00:00.54 | JohnnyC | "Open of mISDN Failed" anyone working with mISDN ? |
00:04.15 | *** join/#asterisk litage (n=nick@203.220.55.70) |
00:11.54 | *** join/#asterisk coppice (n=chatzill@123.192.17.210.dyn.pacific.net.hk) |
00:13.04 | galel | some body know why i can register to a iax-truk ... this is what it shows:XX.XX.XX.XX:4569 master <Unregistered> 60 Rejected |
00:14.21 | galel | i have to change something in manager.conf |
00:14.22 | galel | ? |
00:14.25 | *** part/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca) |
00:14.41 | galel | i changed to accept:0.0.0.0 |
00:14.50 | galel | some one? |
00:15.04 | *** join/#asterisk fugitivo (n=ajf@209.13.243.83) |
00:18.01 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
00:22.27 | halorgium | what computer specs would be bare minimum to support asterisk + a x100p fxo? |
00:22.30 | *** join/#asterisk file (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net) |
00:23.22 | Nugget | a cpu, some ram, and a unix. |
00:23.28 | halorgium | :P |
00:23.31 | Nugget | really |
00:23.33 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
00:23.54 | Nugget | well, actually, s/unix/linux/ for the zaptel card. |
00:24.35 | xheliox | The IAXy is requesting a registration period of 0 seconds, when Asterisk's default is 60 seconds. Is there any way to tell the IAXy how long to register for? And yes, I know the minreg can be set in iax.conf, but that creates a different issue, which isn't really relevant. :) |
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01:04.32 | Icemaann | after a call to Set(GROUP()=TEST), GROUP_COUNT(TEST) still returns 0. Is this the expected behavior? |
01:04.43 | Icemaann | using 1.2beta1 |
01:13.28 | *** join/#asterisk Juxt (n=Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
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01:22.16 | drumkilla | so chatty in here ... |
01:22.21 | Juxt | yep |
01:22.26 | drumkilla | russell-: ?! |
01:22.26 | brookshire | russell.. you're a nubb ;) |
01:22.28 | JerJer | can't keep up |
01:22.30 | drumkilla | THAT'S MY NAME |
01:22.33 | Juxt | anyone have a take on goiax.com |
01:22.42 | brookshire | just kidding |
01:22.43 | brookshire | <3 |
01:22.45 | Juxt | i am seriously confused of how they can afford to do what they do |
01:22.48 | drumkilla | brookshire: it's nub. |
01:22.59 | brookshire | lol |
01:23.04 | brookshire | well.. ok |
01:23.07 | brookshire | i really don't care |
01:23.35 | drumkilla | your mom doesn't care ... |
01:23.37 | JerJer | 404 |
01:23.48 | drumkilla | FOUR OH FOUR! |
01:24.01 | Juxt | www.goiax.com |
01:24.13 | drumkilla | i barely slept that entire week. |
01:24.14 | file | I remember reading that on the mailing list |
01:24.30 | JerJer | i missed it :( |
01:25.06 | *** join/#asterisk dalfry (n=dalfry@ool-435285b1.dyn.optonline.net) |
01:25.55 | Juxt | they do not pass caller id but the quality is very nice |
01:28.17 | bweschke | Juxt: they're a clec... the termination is cheap and it helps them to work on a scalable iax platform for their paying clients |
01:29.02 | Juxt | so they are taking this traffic so they could test scalability of their system? |
01:29.55 | bweschke | yes - that is the goal.. they want to work on their iax platform so it scales horizontally.... they will be making changes to it over time to see how they can get it to perform better to suit their needs. |
01:30.09 | bweschke | so we're guinea pigs essentially, and in exchange, we get the free termination |
01:30.16 | Juxt | how do you know this |
01:30.59 | bweschke | because I'm one of their paying customers |
01:31.13 | Juxt | oh, what's the name of the company? |
01:31.16 | bweschke | and the clec stated his intentions with the platform on the forums a few weeks back |
01:31.25 | bweschke | used to be txlink, now acquired by CommPartners |
01:32.09 | Juxt | gotcha |
01:32.15 | Juxt | they are brave :-) |
01:32.59 | bweschke | they're also smart... how many minutes you think they need to serve up before they equal the cost of an Empirix Hammer tester and even then how much time/labor would you expend to configure the hammer tester to simulate real world load? |
01:33.16 | bweschke | they're getting real world load for only the cost of the term which is sub a penny per min for them |
01:33.27 | Juxt | yeah it makes sense |
01:34.19 | hardwire | blah |
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01:48.02 | lilneon | hey everyone |
01:50.09 | *** join/#asterisk babak (n=root@r-72-244-140-113.nycmny83.covad.net) |
01:50.14 | lilneon | got a quick problem guys, reinstalled rh9 on my asterisk box (got bigger hard drive).. and added a tdm11b card, keep getting a dma_timer_expiry error followed by a too much work on the interrupt abut 8 times on the screen when linux bootss |
01:50.20 | lilneon | any help? |
01:50.44 | lilneon | is my hard drive bad? or is the digium card grabbing all the interrupts leaving none for the network card? |
01:51.28 | lilneon | hello?? n e one home? |
01:51.52 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
01:52.07 | lilneon | where is everyone? |
01:52.08 | babak | i need help: i have an asterisk system with tdm400p with 4 fxs modules. when i dial out to the pstn i almost always get a "please dial 1 before making this call message". |
01:52.24 | babak | can anyone help? |
01:52.41 | mog_home | add wwww before the ${EXTEN} |
01:52.47 | mog_home | im here lilneon |
01:52.55 | lilneon | hi mog_home |
01:53.03 | lilneon | did u see my problem? |
01:53.07 | mog_home | nope |
01:53.28 | babak | hi mog_home. i'll try wwww. |
01:53.28 | lilneon | getting a dma_timer_expiry error after reinstalling rh9 on my asterisk box |
01:53.47 | mog_home | when and where? |
01:54.11 | lilneon | during first boot |
01:54.17 | lilneon | and well every boot up after that |
01:54.39 | mog_home | dmesg says that? |
01:55.04 | lilneon | during boot up |
01:55.18 | lilneon | didn't bother check dmesg |
01:55.19 | brookshire | mog! |
01:55.26 | mog_home | brooks |
01:55.32 | brookshire | gtknubbjab |
01:55.45 | mog_home | word |
01:56.09 | brookshire | gtkastnubbjab |
01:56.09 | mog_home | we need to hook that up |
01:56.10 | brookshire | :D |
01:56.14 | mog_home | lol |
01:56.17 | mog_home | better |
01:56.18 | lilneon | my network card stopped wrking as well after i added a tdm11B card.. says too much work for the interuupt |
01:56.21 | file | 256 area code omg |
01:56.38 | brookshire | gtknubbastjab |
01:56.47 | brookshire | file what what? |
01:57.02 | mog_home | no gtkastnubbjab |
01:57.57 | lilneon | ?? whats with the gtkastnubbjab? |
01:58.01 | Juxt | looks like you need to go into the bios and resuffle the interrupts |
01:58.20 | mog_home | its a top secret project..... |
01:58.24 | lilneon | it doesnt allow me to do that juxt |
01:58.47 | lilneon | just has plug and play OS yes/No.. and then it assigns interrupts to what ever device it likes.. |
01:59.04 | Icemaann | can someone explain to me how GROUP() and GROUP_COUNT work? If I do, Set(GROUP()=TEST), and then, NoOp(${GROUP_COUNT(TEST)}) it returns 0 for the group... |
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02:00.35 | *** part/#asterisk alibby (n=alibby@pcp01412034pcs.phnixv01.pa.comcast.net) |
02:00.38 | babak | hi mog. thank you, thank you, thank you!!!! that did it. |
02:00.55 | babak | thank you!!!!!! |
02:00.56 | mog_home | no prob |
02:01.09 | mog_home | the line was seazing to fast |
02:01.10 | babak | so, i guess wwww add a wait before dialing? |
02:01.19 | mog_home | the www add 300 millaseconds of wait |
02:01.25 | mog_home | w = 100 millasecond wait |
02:01.38 | babak | what does seazing mean. i've seen it here and here... |
02:01.58 | mog_home | it grabs the line and then dials |
02:02.01 | Icemaann | it actually seems to start the count at 0, which is fine, but when doing a transfer it seems it starts the count at 1 lol. I can work around it with variables, but I wanted to see if Im using the functions right. If I am I will file a bug report |
02:02.04 | babak | ah |
02:02.14 | mog_home | now what happens is it grabs the line waits 300 millaseconds then dials |
02:02.24 | babak | this is great. |
02:03.02 | babak | is there a line seaz setting that i can set (on my phone or asterisk) or should i just add "w" |
02:03.42 | mog_home | well you could |
02:03.45 | mog_home | but its lame |
02:03.47 | babak | actually, if i think about it, its not my phone... |
02:03.49 | mog_home | better stick with wwww |
02:03.52 | babak | yeah. |
02:04.01 | babak | thank you very much. |
02:04.18 | babak | bye |
02:04.25 | *** join/#asterisk brad_mssw (n=brad@ip24-170-193-14.ga.at.cox.net) |
02:05.39 | Brijn | lilnoen: Did you look at the noapic kernel option.. Just guessing, but there was something you could feed the kernel at boot time |
02:07.01 | Corydon76-home | Icemaann: file a bug report |
02:07.23 | Icemaann | Corydon76-home: k |
02:10.41 | lilneon | brijn... um.. you know what command line parameter? |
02:12.37 | Brijn | linux noapic at the boot prompt |
02:13.12 | lilneon | brijn:googling.. as i ahve no idea what it is |
02:13.41 | brad_mssw | anyone here use teliax? any idea how to improve DTMF code reliability? seems like sometimes when you dial extension 112, it'll think you dial 111 or similar (seems to mainly mess up from a cell phone) ... |
02:13.48 | _Thor | What port is used by CLI? |
02:15.03 | Brijn | me neither, but i know it has something to do with irq routing |
02:16.43 | lilneon | brad_mssw : i use em, but mainly dial from analog phones.. havent had problems.. some users say they have a hard time entering their pin# though when they attempt a call fro their cellphone.. but i have never checked it out.. try mailing them |
02:18.44 | brad_mssw | lilneon, just signed up today, was experiencing the same issues with vonage ... which is why i'm trying teliax ... i guess i'll research it more |
02:19.10 | *** join/#asterisk ubergoober (n=ubergoob@c-24-16-108-200.hsd1.ca.comcast.net) |
02:22.51 | lilneon | brad_mssw : yeah cool, they are ok.. if u find anything shout me, cant test now cuz my asterisk test box is down.. sigh |
02:27.03 | *** part/#asterisk schuylerdigium (n=Bosco@pcp03052091pcs.huntsv01.al.comcast.net) |
02:27.31 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
02:27.32 | pauldy | does teliax support * |
02:27.36 | *** join/#asterisk SplasPood (n=sp@brooklyn.paravolve.net) |
02:27.51 | pauldy | I noticed you can get vontage working if you don't mind their bs 500 minute softphone limit |
02:27.59 | lilneon | pauldy : um yeah.. u can connect via iax, sip etc |
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02:30.45 | brad_mssw | pauldy, teliax supports both SIP and IAX2 for asterisk .... pauldy, vonage has a business plus offering they don't advertise that they allow direct sip |
02:31.30 | pauldy | yea just read the rates though and in comparison to broadvoice teliax is much higher |
02:31.40 | pauldy | really brad_mssw got any links or rates |
02:33.05 | pauldy | just currious been wadding through provider hell trying to find somewhere to park a client I did an asterisk install for |
02:33.21 | pauldy | wanted to go forward with broadvoice and now they don't have any local dids for their area |
02:34.19 | pauldy | not to mention lnp hell because they want to port over three numbers |
02:34.34 | brad_mssw | pauldy, just do a google search for 'vonage business plus', you'll find a bunch of resellers ... vonage doesn't seem to want to sell direct |
02:34.56 | brad_mssw | pauldy, minimum plan is 4 simultaneous lines, 5k minutes, $150/mo |
02:36.00 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net) |
02:36.17 | pauldy | yea to much for a soho setup |
02:36.41 | pauldy | they are paying just a bit more than that right now for their service |
02:36.49 | brad_mssw | teliax pay as you go is a really nice plan |
02:36.51 | pauldy | with POTS lines |
02:36.58 | brad_mssw | just waiting to see if it's reliable enough |
02:37.14 | pauldy | right now I cna't seem to beat the offer broadvoice has |
02:37.43 | brad_mssw | basically it's only $5/mo + 0.02 per minute ... and unlimited simultaneous calls |
02:38.00 | brad_mssw | don't think broadvoice allows more than 1 simultaneous call |
02:38.05 | pauldy | I can move their other lines to buisness lines and do a tele-branch on the numbers they can't port and still be under 70 bucks a month |
02:38.28 | pauldy | I've had 6 simultanious calls in a meetme conference over bv |
02:38.36 | Icemaann | Corydon76-home: I filed a bug report, 5453, I falsely put it under core functions instead of New Functions, sorry. |
02:38.51 | brad_mssw | pauldy, 6 simultaneous calls on a single broadvoice phone number? |
02:39.02 | pauldy | yup |
02:39.04 | brad_mssw | pauldy, I have broadvoice for home service, didn't think it was possible |
02:39.13 | pauldy | very much so |
02:39.39 | pauldy | I've routinly had more than one the 6 was a special occassion to see if it would kill my broadband |
02:39.41 | pauldy | and it did |
02:39.55 | brad_mssw | hmm, is that only the business plans then ? |
02:39.55 | Icemaann | pauldy: I've heard Telasip is good, havent tried them yet |
02:40.24 | pauldy | brad_mssw, nope mine is residential because I just use it to fool around with |
02:40.47 | pauldy | yea I saw a review for them on bbreports |
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02:43.08 | brad_mssw | pauldy, interesting ... |
02:43.47 | pauldy | you trying it now |
02:43.49 | pauldy | hehe |
02:44.44 | pauldy | yea telasip to expensive for a small home buisness |
02:45.36 | Icemaann | yea their business rates do seem hi. I have emailed their tech support and they (Gene) is very responsive. He may work with you on the pricing. The website leaves a lot to be desired |
02:46.00 | file[laptop] | Gene... where have I heard that name |
02:46.11 | file[laptop] | oh I remember |
02:46.12 | pauldy | Yea they have a premium on some of their hardware to which makes me think they are a third party provider |
02:46.25 | alephcom | Gene is working on doing some nice stuff with the rates as well as the packages I believe. I'm not sure exactly when though. |
02:46.53 | alephcom | They resell for Level3 I believe. I heard that from one of their customers. |
02:47.03 | Icemaann | yea they use level3 |
02:47.13 | alephcom | I really don't know what they have for their own lines.... |
02:47.16 | Icemaann | I will probably setup an acct with them in the next couple of weeks |
02:47.50 | alephcom | He will hopefully have online signup by then. He said he was working on it as I don't think they have that right now. |
02:48.00 | Icemaann | no they done |
02:48.03 | Icemaann | dont* |
02:50.50 | Juxt | telasip does level3? |
02:51.41 | crash3m__ | <PROTECTED> |
02:53.04 | pauldy | anyone here happen to know why number are portable to some providers and not to others |
02:53.37 | JonR800 | i think some providers lie because they don't want to deal with the hassle. :) |
02:54.32 | pauldy | thats what I think to but I figure someone else might know if there is a reason so I cans top being pissed at them for not doing it |
02:57.34 | JonR800 | i spoke with 4 providers who were all able to provide me with a number in my area code/exchange.. 2 of the companies known for their support said porting would be no problem, 1 said no, and the last didn't reply. |
02:58.35 | JonR800 | i was left with the impression that the last two just didn't give a shit. |
03:00.55 | Katty | hi lads. |
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03:11.04 | n3u7 | wow |
03:11.19 | n3u7 | I finally have asterisk installed on SuSE9.3 |
03:11.51 | n3u7 | so now I'm this far: |
03:11.55 | n3u7 | usr/lib/asterisk/modules/chan_modem_i4l.so: undefined symbol: ast_unregister_modem_driver |
03:11.55 | n3u7 | Loading module chan_modem_i4l.so failed! |
03:15.57 | pauldy | vi /etc/asterisk/moduls.conf add the line "noload => chan_modem_i4l.so" to the end of the file |
03:16.12 | *** join/#asterisk twisted[digium] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
03:16.12 | *** mode/#asterisk [+o twisted[digium]] by ChanServ |
03:16.48 | drumkilla | twisted[digium]: nub |
03:17.29 | file[laptop] | O.O |
03:18.22 | pauldy | how do you get info on the modules that start with pbx_ |
03:18.43 | pauldy | like show applications |
03:18.50 | file[laptop] | Katty: nooooooo |
03:20.00 | drumkilla | i hope it's good |
03:20.38 | file[laptop] | drumkilla: how is my favorite Russell doing? |
03:21.45 | drumkilla | good |
03:22.17 | *** join/#asterisk Moc__ (n=mochouin@modemcable181.215-82-70.mc.videotron.ca) |
03:23.10 | jake1932 | has anyone recorded prompts that are faster spoken than the defaults for voicemail? |
03:27.00 | Katty | drumkilla: Russell stover, i do hope |
03:27.00 | *** part/#asterisk mcadory (n=mcadory@208.149.64.28) |
03:28.35 | n3u7 | false allarm SuSE9.3 install failing on account of libnewt |
03:30.53 | *** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
03:31.45 | Qwell | jake1932: Allison can, for a small fee |
03:33.26 | Katty | nitenite. |
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03:41.55 | n3u7 | file this is exausting |
03:42.11 | n3u7 | Apthis makes Ubuntu and SuSE |
03:42.19 | _Thor | Anybody knows what pport is used by the CLI? |
03:42.29 | file[laptop] | the CLI doesn't use a port |
03:42.36 | file[laptop] | it uses a unix socket |
03:42.38 | _Thor | ? |
03:42.50 | _Thor | how come?... the manager uses a port |
03:43.21 | file[laptop] | because it doesn't. |
03:43.42 | _Thor | well file... coming from you, I'll take your word for it :) |
03:44.04 | twisted[digium] | file[laptop], you poked? |
03:44.09 | _Thor | thank you |
03:44.43 | file[laptop] | twisted[digium]: your Powerbook is showing. |
03:45.26 | _Thor | ummm.. I got the answer now!... because it never really sends anything out of the box.... |
03:45.37 | twisted[digium] | file[laptop], oops. |
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03:46.36 | Qwell | hmm |
03:46.36 | meanphil | I keep getting "Polarity reversed (-1 -> 1)" and then the same thing but with the numbers the other way around |
03:46.41 | meanphil | anyone know what this means? |
03:46.56 | Qwell | Are there any programs/daemons that will connect to a socket, and open a port for said socket? |
03:46.58 | Qwell | or something |
03:46.58 | meanphil | (when using a Digium TDM11B) |
03:48.27 | arp2 | qwell, you mean like netcat? |
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03:48.40 | arp2 | what do you mean 'connect to a socket' and 'open a port for said socket'? |
03:48.43 | Qwell | arp2: yeah, probably |
03:48.52 | Qwell | arp2: dunno |
03:49.43 | n3u7 | where can i get the patch to make asterisk compile without libnewt? |
03:50.06 | *** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
03:50.06 | Qwell | n3u7: installing libnewt would be easier...and supported |
03:50.08 | lilneon | hey guys, anyone know what siocsifflags incorrect? my network card stopped wrking |
03:50.27 | lilneon | is there a way to see a list of IRQs and devices they are assigned to in RH9? |
03:50.57 | Qwell | cat /proc/interrupts |
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03:51.31 | n3u7 | edited the make file to include OPTIONS=none |
03:51.34 | n3u7 | it worked |
03:51.36 | n3u7 | but |
03:51.42 | n3u7 | cli.o err! |
03:53.29 | lilneon | thnx Qwell |
03:53.35 | dos000 | anyone know what it takes to mimick a CO interface to a modem that talks v22.bis ? i want to avoid renting lines from the CO |
03:53.59 | dos000 | best being using digium cards |
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03:55.19 | Kyreeth | dos: fxs ports emulate a CO.. You can plug phones or modems into them. |
03:55.40 | n3u7 | OMG...it worked |
03:55.53 | n3u7 | ASETERISK on SuSE9.3 |
03:55.57 | n3u7 | ! |
03:56.00 | n3u7 | ::)\ |
03:57.03 | pauldy | n3u7, now go cross compile it for the tivo |
03:57.22 | Igbothom_III | and the imate JasJar |
03:57.45 | Brijn | Is there a reason to keep the default Zap/go channel? |
03:57.48 | Brijn | g0 |
03:57.57 | pauldy | asterisk at home? |
03:58.07 | Brijn | Manual MAP install |
03:58.10 | Brijn | AMP |
03:58.26 | pauldy | Brijn, if you just using voip then nope |
03:59.00 | pauldy | I think it sets it up as the default trunk |
03:59.04 | Brijn | Kill kill kill :) |
03:59.07 | n3u7 | ps -aux |
03:59.13 | dos000 | Kyreeth, nice ... any idea what modem protocol the digium cards support ? |
04:00.20 | n3u7 | awwwwww |
04:00.25 | n3u7 | false alarm |
04:00.28 | Kyreeth | dos: Ah, you need it to decode the modem protocol? Hm. |
04:00.53 | n3u7 | make |
04:01.19 | n3u7 | gcc: cannot specify -o with -c or -S and multiple compilations |
04:01.19 | n3u7 | make: *** [cli.o] Error 1 |
04:01.25 | n3u7 | :0 |
04:01.28 | pauldy | now whats with the fisher price voip phones http://www.voipsupply.com/popup_image.php?pID=557 |
04:01.30 | dos000 | Kyreeth, mind you i do not have a lot of data |
04:02.29 | *** part/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
04:03.03 | Brijn | I just received my SIP details from Dolphintel(.com), outbound calls are working OK.. But if I dial the number I get extention not available.. I don't see anyhin on the * console. So it's Dolphintel I guess.. What is needed to accept incoming calls? |
04:03.20 | *** join/#asterisk copantl (n=copantl@205.240.200.98) |
04:04.13 | ManxPower | Brijn, do a sib debug then try calling the number |
04:04.28 | Kyreeth | dos: You'll probably need the calls to come in over ISDN, maybe from an FXS channel-bank, and look into this: http://www.voip-info.org/wiki/view/Asterisk+ISDN4Linux |
04:04.50 | Kyreeth | dos: Alas, I'm not an expert, and haven't done what you're wanting to do. |
04:04.53 | copantl | any body knows howto interconnect 2 asterisk with iax trunk? |
04:04.56 | pauldy | Brijn, you may need to setup a DID |
04:05.09 | pauldy | inbound sip are rejected by default |
04:05.47 | pauldy | copantl, there is a tut in the documentation part of sourceforge for AMP |
04:05.59 | Brijn | ManxPower: Nothing logged.. I wonder, I don't see a "registered xxx" for the ip of the provider. I also have a fwd trunk, and i see that IP |
04:06.21 | ManxPower | Brijn, do a "sip show registry" |
04:06.24 | Brijn | pauldy: i added a did (i think), in extentions_additional.conf there is a entry for the DID |
04:06.52 | copantl | i tried pauldy but i got an error: autentication reject ... any idea? |
04:06.52 | Brijn | manxPower: Ahhh, it registered ok |
04:06.54 | pauldy | http://sourceforge.net/docman/display_doc.php?docid=26418&group_id=121515 |
04:08.00 | pauldy | copantl, make sure you have the authentication details matched up correctly ie peer should have the username for the remote connection on each config |
04:08.02 | Brijn | Hmmmm, sip debug doesn't show any activity if I dial the number.. Anything else that I need to check on my side.. Or is it likely the other end forgot something? |
04:08.53 | pauldy | copantl, that was the only trouble I had following those docs + a context needs to be set where you see a c |
04:08.53 | *** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
04:09.27 | copantl | can you show me a example? |
04:09.49 | pauldy | copantl, the eample is on the site I pasted |
04:09.56 | copantl | ok |
04:10.07 | pauldy | Brijn, what did you have for sip show registry |
04:10.27 | nexis | is it possable to run meetme if you have no ZAP interfaces? |
04:10.49 | pauldy | nexis, yes just use the ztdummy |
04:10.56 | Brijn | Host Username Refresh State |
04:10.56 | Brijn | voip.lightspeed.ca:5060 001404640234 105 Registered |
04:11.15 | copantl | there is my registry : 63.245.93.140:4569 master <Unregistered> 60 Rejected |
04:11.40 | nexis | pauldy, yea, i have ztdummy in the kernel, it still complains about channel.c:2206 ast_request: No channel type registered for 'zap', then says invalid confrence, and drops it. |
04:12.06 | Brijn | pauldy: does the "user" sip entry need a secret entry? Or can it be left out? |
04:13.03 | copantl | pauldy: what context do you use? |
04:13.26 | n3u7 | how often ids the cvs updated? |
04:13.30 | pauldy | nexis, wierd working here with that config do you have a trunk setup for zap or something |
04:13.51 | *** part/#asterisk Moc__ (n=mochouin@modemcable181.215-82-70.mc.videotron.ca) |
04:13.52 | n3u7 | could this error be fixed in two nights? |
04:13.56 | n3u7 | gcc: cannot specify -o with -c or -S and multiple compilations |
04:14.12 | twisted[digium] | n3u7, cvs is updated regularly |
04:14.25 | pauldy | Brijn, you need the secret for the peer section |
04:14.29 | twisted[digium] | please check out a new version, make clean, and try again. |
04:14.36 | pauldy | copantl, context=from-internal |
04:14.40 | n3u7 | :)\ |
04:14.50 | copantl | in both sides? |
04:14.57 | pauldy | for testing yes |
04:15.05 | pauldy | jsut to make sure you can dial one from the other |
04:15.25 | pauldy | they have another part that says outbound dial prefix don't worry about that |
04:15.45 | pauldy | instead setup an outbound route to use the trunk with a dialing rule of 7|xxx or something |
04:15.47 | Brijn | pauldy: not for user, because we will decide by defining an did context if we want to accept the call? |
04:15.52 | nexis | odd, nothing is documented that you have to have chan_zap.so loaded to use meetme. |
04:15.53 | copantl | i made the 4 version from voip-info |
04:16.14 | pauldy | BrianR, not for user because you don't want to authenticate the incomming call |
04:16.36 | pauldy | probably best if you just leave the incomming settings blank |
04:16.47 | pauldy | just enter the info in peer details |
04:16.50 | n3u7 | inux:/usr/src/asterisk # cvs checkout asterisk |
04:16.50 | n3u7 | cvs checkout: cannot chdir to asterisk: Not a directory |
04:16.51 | n3u7 | cvs checkout: ignoring module asterisk |
04:17.00 | n3u7 | ? |
04:17.14 | Brijn | [frm-dolphin] |
04:17.14 | Brijn | type=user |
04:17.14 | Brijn | context=from-pstn |
04:17.32 | pauldy | Brijn, probably not a good idea to paste the whole seciton in here |
04:17.41 | pauldy | esp the password etc... |
04:17.45 | Brijn | is what I have nowm even if there would be mistajkes there, i should see the call coming in on the console (verbose 30, sip debug)?? |
04:17.54 | twisted[digium] | n3u7, try cvs update -d |
04:17.59 | twisted[digium] | from within your asterisk source |
04:18.00 | Brijn | I left the password out :) |
04:18.07 | pauldy | BrianR, if you aren't registered properly you might have problems |
04:18.28 | Brijn | pauldy: the sip how registered looked ok? |
04:18.31 | n3u7 | that worked! |
04:18.35 | pauldy | from my experience I use the following register string and it has worked well for me on several installs |
04:19.01 | pauldy | number@host:pass:number@host |
04:19.26 | drumkilla | twisted[digium]: make update, fool |
04:19.33 | Corydon76-home | twisted[digium]: shouldn't that be: make update |
04:19.39 | drumkilla | I win! |
04:19.41 | twisted[digium] | it works either way |
04:19.46 | twisted[digium] | silly rabbits |
04:19.53 | file[laptop] | tricks are for kids? |
04:19.54 | drumkilla | you forgot -P |
04:19.55 | copantl | pauldy : do i need to change something in /etc/asterisk? |
04:19.57 | Brijn | pauldy: i guess one of the "numbers" is user? |
04:19.59 | Corydon76-home | I think 'make update' does something extra |
04:20.01 | drumkilla | that's why you should just use make update |
04:20.05 | n3u7 | :( |
04:20.05 | drumkilla | and there are other reasons, as well ... |
04:20.07 | twisted[digium] | sure. |
04:20.09 | n3u7 | gcc: cannot specify -o with -c or -S and multiple compilations |
04:20.09 | n3u7 | make[1]: *** [localtime.o] Error 1 |
04:20.18 | twisted[digium] | n3u7, did you modify your makefile? |
04:20.22 | pauldy | copantl, if you have amp installed no |
04:20.28 | pauldy | Brijn, yes |
04:20.38 | n3u7 | yes to compile with out libnewt |
04:20.44 | n3u7 | options=none |
04:20.48 | twisted[digium] | drumkilla, i fixed your streamplayer for osx earlier. |
04:20.52 | twisted[digium] | :P |
04:20.56 | drumkilla | hm? |
04:21.00 | copantl | asterisk@home |
04:21.01 | n3u7 | libnewt is broken in SuSE9,3 |
04:21.12 | twisted[digium] | simple fix, but fixed nonetheless |
04:21.25 | drumkilla | i build on osx all the time ... |
04:21.31 | pauldy | copantl, the web based interface for AAH is AMP so no you shouldn't have to edit any of the files by hand just use the wb interface |
04:21.38 | twisted[digium] | drumkilla, it wasn't building |
04:21.45 | copantl | ok |
04:22.14 | Brijn | pauldy: sorry, you wrote number@host:pass:number@host.. I guess it's number@user:pass:number@host or something.. Where should user be in your string? |
04:22.17 | twisted[digium] | two different macs, two different versions of osx |
04:22.37 | drumkilla | i don't see your commit |
04:22.44 | twisted[digium] | it went in |
04:22.47 | file[laptop] | hrm |
04:22.49 | twisted[digium] | it's probably hung in mailmain again |
04:22.52 | twisted[digium] | er mailman |
04:22.57 | drumkilla | what did you change |
04:22.58 | pauldy | Brijn, replace the number string with user host is the remote connection and pass is your sip secret |
04:23.07 | twisted[digium] | drumkilla, in streamplayer.c, line 42 |
04:23.29 | drumkilla | ok |
04:23.33 | drumkilla | annnd |
04:23.36 | twisted[digium] | err not 42 |
04:23.38 | pauldy | copantl, another thing is just make sure that the two machines are talking to eachother by using iax debug |
04:23.47 | twisted[digium] | wtf |
04:23.49 | twisted[digium] | i commited it |
04:24.02 | twisted[digium] | strange. |
04:24.05 | pauldy | Brijn, is your inbound number not the user |
04:24.31 | twisted[digium] | lemme try again |
04:24.52 | drumkilla | happily builds for me. |
04:24.57 | twisted[digium] | because yea, the checkout didn't have the change |
04:24.59 | Brijn | pauldy: Hmmm, one thing, my password contains an @, would that eb a problem?? Do I need to escape that |
04:25.09 | drumkilla | i think you're on crack. |
04:25.10 | twisted[digium] | no |
04:25.19 | twisted[digium] | i was at ABC with brookshire and all |
04:25.27 | Brijn | pauldy: it's a number yes |
04:25.35 | ManxPower | Brijn, Yes, passwords with @ are bad |
04:25.46 | Brijn | I guessed that much :( |
04:25.49 | pauldy | Brijn, or get them to set another pass for you |
04:25.49 | ManxPower | But if sip shows registry shows you registered..... |
04:25.50 | twisted[digium] | it even gave me a rev bump |
04:25.52 | twisted[digium] | *shrugs* |
04:26.03 | twisted[digium] | I saw it not build. Made the change, and it built |
04:26.05 | drumkilla | it probably failed the up to date check |
04:26.15 | JunK-Y | i cant sleep. |
04:26.17 | twisted[digium] | shouldn't it have complained though? |
04:26.19 | drumkilla | I changed something this weekend at some point |
04:26.23 | JunK-Y | sup here? |
04:26.24 | drumkilla | it's easy to miss |
04:26.31 | twisted[digium] | ah... could have been thn |
04:26.33 | twisted[digium] | er then |
04:26.52 | pauldy | I spent 8 hours trying to figure out why my meetme conferences suddenly stopped working only to find someone removed a # from an include line |
04:26.59 | twisted[digium] | i'll recommit as soon as i retest with this fresh tree.. |
04:27.00 | twisted[digium] | grr |
04:27.18 | twisted[digium] | it should make zero functionality change, but allow it to be built on older versions of osx as well as newer ones |
04:27.22 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166095097.nb.aliant.net) |
04:28.09 | drumkilla | what's the change |
04:28.18 | pauldy | hey twisted how well do you think asterisk would run on a g4 433hz with 384MB RAM running 10.3.9 |
04:28.22 | drumkilla | i'm running tiger. |
04:28.34 | twisted[digium] | pauldy, *shrug* |
04:28.37 | drumkilla | pastebin it or something |
04:28.43 | twisted[digium] | drumkilla, me too, but the two boxes i tested were panther and jaguar |
04:28.57 | drumkilla | was it that ifdef line? |
04:28.57 | twisted[digium] | it built fine on tiger all three tries |
04:29.01 | twisted[digium] | yea |
04:29.02 | drumkilla | to include that extra header? |
04:29.08 | Brijn | ManxPower/pauldy: If I escape the @ in the pwd it keeps hanging in Sent auth.. So I guess the @ is ok |
04:29.22 | drumkilla | so you just added, || defined(__Darwin__) |
04:29.23 | drumkilla | ? |
04:29.25 | JunK-Y | wow, http://www.futureshop.ca/catalog/proddetail.asp?sku_id=0665000FS10058254&atab=&spviewed=&newlang=EN&logon=&langid=FR pretty cheap |
04:29.27 | twisted[digium] | *nods* |
04:29.32 | drumkilla | ok, i'll add it ... |
04:29.40 | twisted[digium] | ok |
04:30.06 | ManxPower | Anyone that pays $500 for portable audio is an idiot |
04:30.18 | twisted[digium] | confirmed - no difference in osx 10.4.2 |
04:30.52 | drumkilla | k, it's in |
04:30.52 | pauldy | ManxPower, I fail too see the economic sense in ipods too |
04:30.53 | drumkilla | :) |
04:31.00 | JunK-Y | ManxPower: thats ur opinion :) |
04:31.02 | twisted[digium] | gratsi |
04:31.08 | pauldy | esp when PDAs are cheaper and will perform the same functions |
04:31.12 | twisted[digium] | did you commit from your ipod? :P |
04:31.18 | drumkilla | of course |
04:31.29 | JunK-Y | drumkilla: i wonder if i sould take the new one, with video streaming. |
04:31.31 | *** join/#asterisk fugitivo (n=ajf@209.13.243.217) |
04:31.34 | ManxPower | pauldy, I can see spending $150 for such a thing if you really love music and are not near your computer. |
04:31.36 | drumkilla | my ipod is in my compilation farm |
04:32.13 | ManxPower | that reminds me, I need to re-down.oad my music collection to my laptop. |
04:32.31 | JunK-Y | pauldy: when i do jogging, i prefer having an ipod in my adidas pants instead of a pda :) |
04:32.38 | pauldy | yea my pda is now a sip extension, gps, movie player, ipod lookalike, and day planner |
04:33.06 | ManxPower | I keep thinking about getting a PDA |
04:33.16 | ManxPower | but I need a car and a perm place to live first. |
04:33.21 | pauldy | JunK-Y, to each his own I just can't phathom such a purchase with its limited functionality |
04:33.28 | drumkilla | g'night all |
04:33.37 | JunK-Y | see ya drumkilla |
04:33.46 | fugitivo | i want a nokia 770 |
04:34.01 | ManxPower | pauldy, the thing is, I doubt Junky jogs for more than an 3 hours and most cheap audio players can handle at least that mucn. |
04:34.33 | pauldy | my PDA will play for 6 hours if I tune down the speed from 400Mhz to 100 |
04:34.42 | *** part/#asterisk zedas (i=zedshawc@pizarro.dreamhost.com) |
04:34.48 | fugitivo | pauldy: what pda? |
04:34.59 | pauldy | toshiba e805 |
04:35.06 | fugitivo | how much is that thing? |
04:35.40 | pauldy | still about 500 bucks |
04:35.57 | fugitivo | what OS? |
04:36.01 | pauldy | if you cna find it and they don't make them any more |
04:36.09 | pauldy | winblows ce |
04:36.43 | Brijn | Does exten => 6046289655,3,Goto(ext-local,201,1) mean jump to ext-local context, extention 201 after 1 second? |
04:36.49 | pauldy | spyros been working on a port of linux since early 2004 but its barely more than halfway complet |
04:36.57 | fugitivo | www.nokia.com/770 |
04:37.00 | fugitivo | it runs linux |
04:37.11 | fugitivo | it's not really a pda, it's an internet tablet |
04:37.15 | ManxPower | Brijn, no, itmeans jump to context ext-local, extension 201, priority 1 |
04:37.24 | n3u7 | hem astman.o |
04:37.27 | pauldy | I have an old 9200 that was the last nokia product I ever purchased |
04:37.30 | ManxPower | of course it's not going to work if you don't have priorities 1 and 2 for that extension |
04:37.46 | JunK-Y | Brijn: polux*CLI> show application goto will give u all the answers. |
04:37.53 | n3u7 | trying to get around libnewt so that I can get this going on SuSE |
04:37.54 | pauldy | ManxPower, shouldn't it be prefixed with an _ too |
04:38.36 | ManxPower | pauldy, it's not a pattern so it doesn't need a _ |
04:38.46 | pauldy | oh the other tweak I do to maximize usage is to turn off the PDAs screen |
04:39.32 | n3u7 | what di I do now? |
04:39.54 | pauldy | n3u7, you need a bigger hammer to get that round peg in the square hole |
04:40.08 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
04:40.13 | n3u7 | heh, like editing the make file |
04:40.27 | n3u7 | but I wouldneed lots of helpz0rs |
04:41.14 | pauldy | <PROTECTED> |
04:41.29 | fugitivo | pauldy: i'm waiting the release to get one |
04:41.52 | fugitivo | pauldy: www.maemo.org |
04:41.53 | Brijn | What does the hint part in exten => 201,hint,SIP/201 mean?? |
04:42.01 | ManxPower | Does anyone have recommendations for a podcast client for wn32? |
04:42.03 | pauldy | any idea on what the hardware specs are nokia is rather limited |
04:42.06 | Brijn | I tried shop app hint ;) |
04:42.52 | fugitivo | pauldy: http://europe.nokia.com/nokia/0,,75023,00.html |
04:43.08 | Brijn | Brijn, but it's not a command so that is not surpising |
04:43.31 | fugitivo | pauldy: version 2006 of the os will have voip |
04:43.33 | pauldy | fugitivo, yea no cpu specs etc... |
04:44.15 | fugitivo | pauldy: check www.maemo.org, it's for developers |
04:44.29 | pauldy | found it 220Mhz OMAP 1710 powered by an ARM9 core |
04:44.36 | pauldy | wierd is that not a TI chip? |
04:44.57 | fugitivo | i think i've read it's slow |
04:45.03 | pauldy | It is it says it further down the page |
04:45.17 | pauldy | yea that thing isn't meant to handle a display that large |
04:46.07 | pauldy | but I believe if it is the chip I think it is there is a nice programmable DSP piggy backing it |
04:46.15 | fugitivo | http://maemo.org/maemowiki/Nokia_770_Hardware_Specification |
04:46.24 | pauldy | which would be nice for compression codecs for audio |
04:47.00 | fugitivo | no PIM |
04:47.04 | fugitivo | that's bad |
04:47.13 | copantl | pauldy: i tried the configuration and the aix2 trunk debug say: Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT |
04:47.13 | copantl | <PROTECTED> |
04:47.13 | copantl | <PROTECTED> |
04:47.31 | copantl | i used from-internal |
04:48.17 | pauldy | context=from-internal |
04:48.29 | copantl | yes |
04:48.38 | pauldy | that goes in the user settings not peer |
04:49.20 | copantl | right |
04:49.43 | pauldy | yea fugitivo to bad this device will be pretty much outa date before it ships hopefuly the next revision will sport a faster cpu and built in video acceleration |
04:49.53 | fugitivo | gimp will run on that device |
04:50.00 | pauldy | nice |
04:50.24 | pauldy | copantl, run this cd /etc/asterisk;grep [from-internal] * |
04:50.45 | pauldy | no wait |
04:50.49 | Corydon76-home | Heh |
04:50.57 | Corydon76-home | Doing a character class, eh? |
04:51.12 | pauldy | cd /etc/asterisk;grep "\[from\-internal\]" * |
04:51.25 | Corydon76-home | You need to double the backslashes |
04:51.37 | Corydon76-home | Once for the shell, once for grep |
04:51.51 | *** join/#asterisk NoRemorse (n=axel@202.161.68.6) |
04:51.53 | pauldy | Corydon76-home, I just ran it and it worked fine |
04:52.08 | pauldy | not using egrep |
04:52.10 | NoRemorse | hi all, how do I recompile asterisk with a RF2833 dtmf payload of 99 please? |
04:52.31 | pauldy | it just hit me after I pasted that those brackets will cause issues |
04:52.38 | Icemaann | is markster in here? |
04:52.39 | Corydon76-home | Yeppers |
04:52.54 | Corydon76-home | Mark is probably asleep |
04:53.03 | Icemaann | nah, he is replying to bugs ;-) |
04:53.08 | fugitivo | scummvm will run on the nokia 700 |
04:53.10 | fugitivo | 770 |
04:53.14 | fugitivo | so we can play monkey island |
04:53.25 | pauldy | hehe |
04:53.40 | pauldy | I had mame running on my e805 for a while it was neat |
04:54.10 | pauldy | but since I actually use my pda for productive things it came off as soon as the novelty wore off |
04:54.37 | fugitivo | too bad no PIM is included |
04:54.53 | copantl | pauldy: a lot info :))! |
04:55.09 | pauldy | yea copantl did you try my revised version of the command |
04:55.32 | pauldy | just looking to make sure it finds the line in extensions |
04:55.36 | NoRemorse | hi all, how do I recompile asterisk with a RF2833 dtmf payload of 99 please? |
04:56.19 | pauldy | fugitivo, http://oss.kernelconcepts.de/maemo/ there you go |
04:58.00 | fugitivo | niiiiiice |
04:58.22 | fugitivo | now we need a calendar and it'll be perfect |
04:58.34 | pauldy | the only drawback is the device is seriously underpowered from a cpu perspective |
04:58.57 | pauldy | fugitivo, like this http://www.steinbauer.org/matthias/computer/linux/gpe-calender-770/ |
05:01.10 | NoRemorse | you are a font. not. |
05:01.12 | *** part/#asterisk NoRemorse (n=axel@202.161.68.6) |
05:01.25 | *** join/#asterisk websae (n=websae@CPE-24-167-206-63.wi.res.rr.com) |
05:02.14 | websae | anyone here have experience with new WIFI phones? |
05:02.32 | websae | if so how are they with asterisk--and what seems to be the best one? |
05:02.37 | copantl | zyxel? |
05:03.18 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
05:03.45 | Kyreeth | I've got one of the zyxel wifi phones, but it's having trouble registering with DHCP at work. Seems to work fine at home. |
05:03.48 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
05:04.11 | copantl | i got one too |
05:04.54 | copantl | but myone works... but don't now how tranfer a call in that phone |
05:06.06 | pauldy | have you tryed just pressing pound with it |
05:06.17 | *** join/#asterisk erickj_az (n=erickj_a@wsip-68-98-222-74.ph.ph.cox.net) |
05:06.37 | erickj_az | Is there a channel dedicated to DUNDi |
05:07.23 | pauldy | have you tried #dundi |
05:07.52 | copantl | yes |
05:08.04 | copantl | but hangup the call |
05:08.16 | pauldy | haha well thats not good then |
05:08.27 | copantl | right |
05:08.44 | *** join/#asterisk ag0ny_ (n=ag0ny@ares.sengawa-networks.com) |
05:09.03 | *** join/#asterisk logicalonline (n=Ken@209.242.52.25) |
05:09.14 | copantl | pauldy, i can't found what's wrong with my iax2 trunk setup |
05:10.35 | pauldy | can you do some screenies of the two trunk pages and put them up somewhere |
05:11.29 | erickj_az | Does anyone here know anything about dundi? |
05:11.34 | pauldy | wen can get the config fixed tehn you can change the pass info or whatever |
05:13.00 | *** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net) |
05:16.04 | copantl | ok |
05:20.56 | copantl | pauldy : what do you need?... just tell me? |
05:22.24 | *** part/#asterisk copantl (n=copantl@205.240.200.98) |
05:22.41 | *** join/#asterisk copantl (n=copantl@205.240.200.98) |
05:23.18 | digime | anyone recommend a good SIP softphone |
05:23.22 | digime | that is open source |
05:23.26 | digime | besides SJlabs |
05:24.12 | copantl | is there in digium some one called kenny? |
05:25.20 | wunderkin | no.. they killed kenny! those bastards! |
05:26.29 | copantl | i need support from digium |
05:26.53 | copantl | i bought a te110p |
05:26.54 | wunderkin | and only someone named kenny can do that? |
05:27.23 | copantl | i'm not shure of hes name |
05:27.26 | copantl | but no |
05:27.38 | wunderkin | ok and? |
05:27.55 | wunderkin | <insert problem here> |
05:28.48 | copantl | ok this guy kenny or kenneth or whatever , fix me a PRI/isdn issue with my telco |
05:29.21 | copantl | and now my cdr not work |
05:29.51 | wunderkin | and define by not work , what happens, maybe someone can help you then |
05:30.50 | copantl | i'm using asterisk@home |
05:31.43 | copantl | and when i go to amp for see my cdr reports... they giveme the reports from 2 days ago |
05:31.59 | *** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca) |
05:34.19 | copantl | and i made a cdr status and give me this: cdr status |
05:34.20 | copantl | CDR logging: enabled |
05:34.20 | copantl | CDR mode: simple |
05:34.20 | copantl | CDR registered backend: cdr_manager |
05:34.20 | copantl | CDR registered backend: csv |
05:35.17 | JamesDotCom | and they're meant to be logging to a database? |
05:36.07 | JamesDotCom | oh |
05:36.09 | JamesDotCom | a@h |
05:36.12 | JamesDotCom | meh |
05:36.13 | JamesDotCom | learn asterisk |
05:36.18 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
05:36.28 | copantl | thanxs.... i know |
05:37.40 | copantl | but for a "simple mortal " like me is a good beginning |
05:37.56 | *** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net) |
05:40.32 | copantl | the other problem is the ANI dont pass thought the telco |
05:40.37 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
05:42.01 | copantl | i configured 2 ATA's with 2605600 and 2605601 |
05:42.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:43.00 | copantl | and when i placed a call from the 2605601 in the CID shows 2605600 |
05:43.56 | copantl | sorry for my english is not my first language !! :)) |
05:44.14 | n3u7 | hoe do I compile asterisk without astman? |
05:44.24 | n3u7 | hoe do I compile asterisk withoutlibnewt? |
05:45.24 | *** part/#asterisk logicalonline (n=Ken@209.242.52.25) |
05:45.46 | L|NUX | have any one treid braintel DID here ? |
05:46.04 | JamesDotCom | copantl: a@h adds a whole layer of complexity above asterisk, you need a@h support |
05:46.34 | n3u7 | I've been tyrying to install for 2 weeks |
05:46.42 | n3u7 | two different distros |
05:47.39 | *** join/#asterisk enota (i=dimka@freelsd.net) |
05:48.31 | copantl | do you know the channel? |
05:49.16 | JamesDotCom | what distro now n3u7? why not just install the libs? |
05:49.24 | JamesDotCom | copantl: no idea sorry, might mention something on their website |
05:49.59 | copantl | i tried to install asterisk - debian - amp really painfull!!! |
05:53.13 | *** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
05:58.19 | *** join/#asterisk brookshire (n=matt@esbrooks3.traveller.com) |
05:59.34 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-105-214.38-151.net24.it) |
06:00.39 | X-Rob | n3u7 -why not install newt and astman? |
06:00.54 | X-Rob | usually 20 seconds of isntallation is worth more than 2 weeks of scratching your head. |
06:01.03 | pauldy | damb I bet he never thought of doing that |
06:03.18 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
06:15.11 | *** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
06:15.13 | *** join/#asterisk kram (n=mark@pdpc/sponsor/digium/kram) |
06:15.13 | *** mode/#asterisk [+o kram] by ChanServ |
06:15.32 | *** join/#asterisk twisted[laptop] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
06:15.32 | *** mode/#asterisk [+o twisted[laptop]] by ChanServ |
06:18.40 | *** part/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
06:20.03 | tzafrir_laptop | considering that there are (unofficial) amportal debs for debian, it can't be that painful. |
06:20.44 | wunderkin | wb kram |
06:21.31 | brookshire | twisted's new laptop is hot |
06:21.36 | brookshire | speaking of laptops |
06:21.38 | brookshire | heh |
06:23.17 | kram | thanks wunderkin, but i'm headed to bed |
06:23.24 | wunderkin | bed? wow |
06:23.39 | wunderkin | but its only 02:23 |
06:23.39 | wunderkin | :D |
06:24.29 | wunderkin | - another stupid mistake by me.. whee |
06:29.53 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
06:37.47 | *** join/#asterisk litage (n=nick@203.220.55.70) |
06:40.14 | *** join/#asterisk tuxinator_linuxM (n=spabin@24-53-55-28.ontrca.adelphia.net) |
06:42.59 | argos73 | ick - puppy runny poo at 2:45 am. fun |
06:51.32 | mmmToop | lets keep it clean ; ) |
06:55.26 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
06:55.26 | *** mode/#asterisk [+o twisted] by ChanServ |
06:56.30 | twisted | i am the nat king. |
06:56.53 | twisted | cisco 7960 --SIP--> NAT1 --> NAT2 --> Server |
06:56.59 | twisted | works perfectly, with two way audio |
06:57.51 | *** join/#asterisk af_ (n=af@ip-142-250.sn1.eutelia.it) |
07:00.30 | X-Rob | twisted - re #5150 |
07:00.38 | X-Rob | oej said 'next week' |
07:00.43 | X-Rob | and it was meant to be in 1.2beta |
07:00.45 | twisted | 5150... |
07:01.04 | X-Rob | <PROTECTED> |
07:01.09 | X-Rob | <PROTECTED> |
07:01.11 | X-Rob | wups |
07:01.29 | twisted | you can't submit bug reports on code that is not in the codebase |
07:01.34 | X-Rob | I didn't |
07:01.36 | X-Rob | OEJ did |
07:01.51 | X-Rob | note the 'Hey, I could make a copy!' |
07:02.01 | twisted | why would he set you as the reporter? |
07:02.04 | X-Rob | FIIK |
07:02.14 | twisted | lol |
07:02.14 | X-Rob | because on 4877 I was the reporter |
07:02.17 | X-Rob | ? |
07:02.28 | X-Rob | or maybe I wasn't |
07:02.30 | twisted | 4877 was gst |
07:03.08 | X-Rob | no idea |
07:03.30 | *** part/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca) |
07:03.50 | *** part/#asterisk twisted[laptop] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
07:04.43 | X-Rob | so what should I do - I've already asked OEJ to comment, and there's been no responce. |
07:04.50 | X-Rob | it does, definately, cause * to crash. |
07:05.02 | X-Rob | all AMP distros do NoCDR/ResetCDR |
07:05.21 | twisted | didn't you see markster's notes? |
07:05.44 | X-Rob | yesh |
07:05.50 | X-Rob | and then I commented immediately after it |
07:05.54 | X-Rob | saying 'Olle, any comments?' |
07:06.08 | X-Rob | like 5 minutes after he posted |
07:09.12 | twisted | okay, well, the bug is on code that isn't in the codebase apparently |
07:09.17 | twisted | and we cannot reproduce it |
07:09.40 | twisted | but i'm not going to take any action on it until OEJ responds |
07:09.54 | twisted | since you did the right thing and got ahold of me;) |
07:11.48 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
07:11.48 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 was a great success! Thanks to everyone who made it happen, as well as to everyone who made it out! |
07:12.29 | infinity1 | hmm ..when dialing a gizmo #, i'm getting Got SIP response 488 "Not Acceptable Here" back from 198.65.166.131. |
07:12.35 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
07:13.00 | swm_ | Anyone know why Linux shows 4 Processors on a 2 Processor System under /proc/cpuinfo ? |
07:13.14 | infinity1 | swm_: hyperthread? |
07:13.20 | brookshire | hyperthreading |
07:13.22 | swm_ | Yeah It's hyperthreading |
07:13.48 | swm_ | Linux 8.x didn't show 4, but Linux 10.x shows 4 of them ... amazing |
07:14.08 | brookshire | linux 8.x? |
07:14.14 | brookshire | suse? |
07:14.20 | swm_ | Slackware |
07:14.25 | brookshire | ahh |
07:14.33 | swm_ | *slutware* |
07:14.48 | swm_ | Not I have to find something to use all that power... Hmm |
07:15.11 | brookshire | well.. it's actually still only two processors |
07:15.50 | brookshire | but you can balance tasks on it better |
07:15.52 | brookshire | lol |
07:16.01 | swm_ | balance is nice |
07:16.04 | e-Hernick | I laugh at you and your weak linux 8.0! I am using Turbo Linux2000 X11 XR2006 |
07:16.30 | e-Hernick | Advanced Edition ! |
07:16.32 | brookshire | gentoo! |
07:16.34 | swm_ | Turbo linux? lol |
07:16.41 | brookshire | i don't think it has a version |
07:17.13 | e-Hernick | Well, I'm using the 2005.0 and 2005.1 profiles |
07:17.32 | e-Hernick | I may still have a 2004.3 somewhere but I think they're all upgraded. |
07:18.22 | swm_ | My only problem is I cannot get KDM or XDM to run . I dont care which one is running as long as I can access it via remote |
07:19.04 | e-Hernick | Don't you like freenx and vnc ? |
07:19.25 | swm_ | dont know much about the linux graphical side |
07:21.02 | pauldy | swm_, tried piping the display or running vnc |
07:21.22 | pauldy | or are you trying to get a remote x sesson via XDMC or domething |
07:21.58 | swm_ | oh it got it working no problem, somone generated a shitty client version and it donesnt work. Reflection for Windows connects me fine |
07:22.15 | pauldy | kewl |
07:23.25 | infinity1 | does anyone here have asterisk connected to gizmo? |
07:23.31 | infinity1 | ..and it works? |
07:23.59 | *** join/#asterisk szer (n=Miranda@217.116.36.22) |
07:24.09 | szer | hi all |
07:24.16 | pauldy | sip phone for macintosh? |
07:26.21 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
07:27.33 | pauldy | infinity1, are you talking about the sip phone for the macintosh |
07:29.01 | JerJer | Captain Crunch is working on a sip phone for OS X |
07:29.20 | pauldy | so what you gota whistle to make it work |
07:30.50 | JerJer | 2600hz unlock tone to launch the app |
07:30.56 | *** join/#asterisk LANmower (n=LANmower@ndn-165-153-187.telkomadsl.co.za) |
07:31.04 | LANmower | hi there |
07:31.07 | JerJer | moo |
07:31.33 | JerJer | asterisk is not an operating system |
07:31.41 | pauldy | neat I've only seen asterisk as an app until now |
07:31.43 | infinity1 | pauldy: no. |
07:31.49 | LANmower | thats why I need to make one |
07:32.02 | *** join/#asterisk gvag11 (n=g@84.254.12.52) |
07:32.04 | JerJer | the fact remains asterisk is not an operating system |
07:32.04 | pauldy | infinity1, ok there are several tutorials on getting it to work |
07:32.15 | infinity1 | pauldy: gizmo is some free sip voip service |
07:32.26 | JerJer | Linux is an operating system - and there are already a dozen micro distros out there suitable for Asterisk |
07:32.36 | infinity1 | pauldy: well, not totally free, but if you're talking to another gizmo user it is free |
07:32.57 | infinity1 | pauldy: there is probably a mac client as well. i'm just trying to get it to work with asterisk instead of their client. |
07:33.18 | pauldy | I came across a couple of totorials on getting it setup |
07:33.34 | infinity1 | pauldy: yea. there is a page on voip-info, but i can't get it to work. :/ |
07:33.47 | LANmower | jerejer, I have my reasons |
07:33.53 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
07:33.58 | LANmower | anyway I have 2 qstns |
07:34.00 | LANmower | what ap does asterisk use to mail? and how do you set what server mail forwards to? |
07:34.30 | Akelavlk | Hello guys, I just installed spanDSP + RxFAX and TxFAX applications and it's working properly.. :-) |
07:34.40 | pauldy | your lying |
07:35.04 | pauldy | haha Akelavlk do you have it working over a BRI or SIP |
07:35.23 | Akelavlk | BRi.. I don |
07:35.33 | Akelavlk | I don't have SIP client yet.. |
07:35.37 | Akelavlk | Do you have one? |
07:36.04 | pauldy | yup tried to get it working finally gave up and setup a hylafax server with a hardware modem |
07:36.34 | pauldy | I got one of those damb linspire boxes from frys for 151 so I figured I wasn't out much |
07:36.53 | Akelavlk | What? |
07:37.18 | Akelavlk | You used SIP hardware phones for faxing? |
07:37.30 | pauldy | yea |
07:37.50 | Akelavlk | Hmm, how much does it cost? |
07:38.10 | *** join/#asterisk hansii (n=nn@cpe.atm2-0-1021108.0x50a0f2ba.odnxx6.customer.tele.dk) |
07:38.17 | Akelavlk | I rather test it with software SIP client . It's cheaper. |
07:38.27 | pauldy | total setup was 200 bucks to build a hylafax server and get the HT386 to emulate a phone line |
07:39.03 | LANmower | snom makes a nice windows-based sip client |
07:39.06 | infinity1 | if someone has a sec, could they look at this debug and see whats wrong? i'm at a loss |
07:39.10 | infinity1 | http://pastebin.ca/25817 |
07:39.36 | LANmower | if your target is the technically-apt, sjphone is better though |
07:39.42 | Akelavlk | My setup cost 100 $. One hardware fax and spanDSP. |
07:40.14 | pauldy | wow nice what kind of computer |
07:40.17 | Akelavlk | HylaFAX support also SIP FAX-ing? |
07:40.42 | JerJer | there is no such thing as 'sip faxing' |
07:40.47 | Akelavlk | You mean what PC i used for PBX? |
07:41.29 | pauldy | I"m going to try and get it working now |
07:41.29 | Akelavlk | JerJer, you should check last version of txfax and rxfax application.. |
07:41.39 | JerJer | that is called T.38 |
07:41.43 | JerJer | not 'sip faxing' |
07:41.48 | Akelavlk | Yes exactly.. |
07:42.08 | Akelavlk | As I know, it's FAX extension for SIP. |
07:42.11 | JerJer | then use that term |
07:42.16 | JerJer | sip faxing means nothing |
07:42.27 | Akelavlk | How do you mean that? |
07:42.31 | JerJer | T.38 |
07:42.35 | JerJer | its not 'sip faxing' |
07:42.40 | pauldy | I wsa doing faxes over a sip connection |
07:42.47 | pauldy | T.38 was disabled on the ht386 |
07:43.04 | pauldy | I can't think of what else to call that but sip faxing |
07:43.07 | JerJer | T.38 is a codec encapsulation |
07:43.13 | JerJer | um T.38 |
07:43.53 | Akelavlk | Yes, but T.38 allow you send fax over SIP as I know.. |
07:44.08 | JerJer | but sip has nothing to do with it |
07:44.15 | pauldy | JerJer, read what I did if you still think it is T.38 ok |
07:44.23 | JerJer | you can do T.38 over any signaling transport that supports it |
07:44.33 | Akelavlk | Yes, of course SIP is such as TCP/IP in this case. |
07:45.46 | LANmower | oh well, as usual, no help here |
07:45.55 | LANmower | thx anyway |
07:45.57 | JerJer | LANmower: then leave |
07:47.44 | pauldy | well gizmo is a wash not going to bother with something that tells me my username is taken just because it doesn't have numbers in it |
07:49.00 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
07:49.49 | Akelavlk | How can I create correct TIFF image for faxing.. I use FAX ccitt 3 compresion but it's not working.. What compresion should I use? |
07:50.46 | *** join/#asterisk Willem_ZA (n=willem@wbs-146-146-209.telkomadsl.co.za) |
07:51.03 | JerJer | use a solution that can deal with PDFs |
07:51.47 | pauldy | or postscript |
07:51.52 | Akelavlk | Hmm, at this time rxfax and txfax support just TIFF.... |
07:52.26 | *** join/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
07:52.43 | Akelavlk | HylaFAX has also PDF support? |
07:52.49 | MuppetMaster | Hi |
07:54.19 | MuppetMaster | Anyone know the ETA for Asterisk1.2beta to come out of beta? |
07:55.43 | *** join/#asterisk szer (n=Miranda@217.116.36.22) |
07:55.47 | Ahrimanes | 1.2 Beta1 is out |
07:55.59 | MuppetMaster | Indeed, but when is it due to come out of beta? |
07:56.06 | JerJer | MuppetMaster: when its ready to come out |
07:56.11 | MuppetMaster | ie - Generally Availablle |
07:57.32 | MuppetMaster | I understand that. |
07:57.36 | MuppetMaster | But any estimates? |
07:57.39 | MuppetMaster | What is holding it up? |
07:58.14 | Akelavlk | MuppetMatster, I am not sure if there are some betas versions. You know it's development.. I am using lastest version and basic functions are working properly.. |
07:58.53 | MuppetMaster | Yes, I have been using the 1.2beta. The only problems I have had are some crashes on reload at the CLI, other than that functionality is fine. |
07:59.50 | JerJer | then report the crashes |
07:59.59 | JerJer | or use a more sane method to reload your config |
08:00.04 | Akelavlk | Yes, what kind of problems do you have exactly. |
08:00.47 | MuppetMaster | Correct. I did not report the crashes as there are already appear to be bug reports on the subject. Don't want to go and open dups. |
08:00.59 | MuppetMaster | Akelavik: Not reproduceable and appear to be somewhat random. |
08:01.13 | MuppetMaster | I use Realtime in conjunction with the required text files of course. |
08:01.54 | JerJer | no wonder |
08:02.26 | MuppetMaster | Why all of the ill will towards Realtime? |
08:02.31 | MuppetMaster | I have seen quite a bit. |
08:02.32 | Akelavlk | Aha, so problem happend in any config file? |
08:02.36 | MuppetMaster | Should Digium have not allowed it into the core? |
08:02.47 | MuppetMaster | Akelavik: Not sure, making changes to the likes of sip.conf and iax.conf. |
08:02.54 | JerJer | Akelavlk: yes HylaFAX can deal with PDF |
08:02.54 | MuppetMaster | If I don't do reloads, very stable. |
08:03.13 | *** join/#asterisk terracon (n=tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
08:03.20 | JerJer | how about realtime is simply crap |
08:03.38 | MuppetMaster | JerJer: Then why did Mark Spencer/Digium allow it in the core? |
08:03.48 | MuppetMaster | As opposed to say an add-on? |
08:03.56 | Akelavlk | I had quite same problem with sip and zapata.conf. Then I found problem with config parameters.. |
08:04.01 | MuppetMaster | And why is it crap? |
08:04.09 | MuppetMaster | Honest questions. |
08:04.16 | JerJer | it was disclaimed and mark felt having some sort of database based config was better than nothing |
08:04.27 | JerJer | but has also admitted that it needs serious work |
08:04.37 | MuppetMaster | What are the key issues with it? |
08:04.47 | Ahrimanes | hence still only beta1 |
08:04.52 | JerJer | dependence on an external system |
08:05.05 | JerJer | massive abuse of system resources |
08:05.08 | MuppetMaster | Well, that is the definition of database driven configuration, no? |
08:05.11 | Akelavlk | As I remember, there was some problem with parsing of file.. |
08:05.15 | JerJer | lack of sanity in processing |
08:05.17 | MuppetMaster | Ah, so takes too much CPU... |
08:05.29 | MuppetMaster | So, may work for a small system but does not scale well. |
08:05.53 | MuppetMaster | I do not believe the crashes are related to realtime, but with config file parsing of Asterisk. As nothing changed in extensions.conf with the crashes occured. |
08:05.59 | JerJer | how about 4 SQL queries per extension priority |
08:06.00 | MuppetMaster | Also, if I stopped and started it worked fine. |
08:06.08 | MuppetMaster | JerJer: I see, that is excessive. |
08:06.24 | Akelavlk | May be you should download lastest Asterisk version.. |
08:06.45 | MuppetMaster | I am not too worried about it, this is not a production system. |
08:06.54 | *** join/#asterisk _m_ (n=m@nat-ph3-1.rz.uni-karlsruhe.de) |
08:07.02 | MuppetMaster | Just curious if there is any expectant ETA on the GA for 1.2... |
08:07.59 | Akelavlk | Asterisk is in very early stage, so there are some basic functionality bugs, so I recommend update to lastest CVS version. |
08:08.18 | MuppetMaster | Ah, too often you catch the CVS mid-bug and can't even compile. |
08:08.25 | JerJer | bullshit |
08:08.29 | JerJer | cvs always compiles |
08:08.29 | MuppetMaster | Not bullshit. |
08:08.45 | MuppetMaster | Even sent in a bug report which Mark Spencer immediately fixed and apologized for it. |
08:08.46 | Akelavlk | I had few problems with ZAPATA ports and parsing config files etc etc.. But last version seems to be pretty stable.. |
08:08.58 | MuppetMaster | I was impressed, he did it within 10 minutes. |
08:09.00 | JerJer | then re-check it out and it compiles |
08:09.04 | MuppetMaster | But it was not compiling in those 10 minutes. |
08:09.05 | JerJer | problem solved, as you stated |
08:09.08 | MuppetMaster | After it was fixed. |
08:09.09 | JerJer | MOVE ON |
08:09.13 | MuppetMaster | But not during. |
08:09.18 | JerJer | it was a mistake |
08:09.21 | MuppetMaster | And the point is, it does happen. |
08:09.22 | MuppetMaster | That is fine. |
08:09.24 | JerJer | MOVE THE FUCK ON |
08:09.27 | MuppetMaster | The point is, the CVS HEAD is a moving target. |
08:09.36 | JerJer | and the problem is |
08:09.38 | JerJer | ? |
08:09.47 | Akelavlk | Hmm, this should never happend. But you know shit happend.. |
08:09.49 | MuppetMaster | You were the one who said it was bullshit. Simply correcting your bullshit. |
08:09.51 | MuppetMaster | Now, carry on. |
08:10.45 | JerJer | i'm done |
08:10.48 | MuppetMaster | Once 1.2 is out it has all of the functionality I currently require. |
08:10.58 | MuppetMaster | So, happy to wait for that and suffer some reload issues in the meantime on a lab system. |
08:10.58 | *** part/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer) |
08:11.23 | Akelavlk | What OS do you use? |
08:11.45 | MuppetMaster | I use OpenSuSE v9.3 for my production/lab systems. And also play around on OSX. |
08:12.21 | Akelavlk | I am using Fedora Core 2.. |
08:12.52 | MuppetMaster | Ah, I had problems with FC and various stages. Tried it, didn't like it, so was very happy when SuSE made their Pro version availabe. |
08:13.09 | MuppetMaster | I prefer the green lizard to the red fedora... ;) |
08:13.15 | Akelavlk | :-) |
08:13.38 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
08:13.53 | Akelavlk | I know, FC is not such stable, but I think for Asterisk it's quite good platform. |
08:14.23 | MuppetMaster | Yes, it works. I just had strange issues at one point (this was a while ago with FC2 and a pre-1.0 Asterisk platform) and when I switched to SuSE those problems dissappeared. |
08:14.30 | MuppetMaster | At that time it was SuSE Pro v9.1 |
08:14.32 | pauldy | hrm I cna';t even get this pos to auth |
08:15.09 | Akelavlk | BTW, what system is Mark using for developing? |
08:15.27 | MuppetMaster | Not sure to be honest, I think Redhat though. |
08:15.44 | MuppetMaster | The new O'Reilly Asterisk book is also predicated on installing with Redhat. |
08:15.52 | MuppetMaster | Seems to be prevalent in the community. |
08:15.55 | *** join/#asterisk grumpie (n=vijay@dsl093-139-122.sfo4.dsl.speakeasy.net) |
08:16.12 | grumpie | hi twisted |
08:17.00 | Akelavlk | :-) You know, they use redhat because it's pretty good software.. |
08:17.08 | Akelavlk | I prefer Gentoo anyway.. |
08:17.32 | MuppetMaster | Yes, it is one of the best commericial distros. |
08:17.41 | MuppetMaster | I am telling you, I just think that Red Fedora is one of the ugliest logs around. |
08:17.47 | MuppetMaster | Green lizard much, much better. |
08:17.49 | pauldy | wow who was asking about gettin gizmo to work with asterisk? |
08:17.59 | MuppetMaster | Other than that, they are much of the same. Although I do like YaST config under Suse better. |
08:18.10 | MuppetMaster | pauldy: Why do you ask? |
08:18.22 | MuppetMaster | I use them together, although the Gizmo client does not connect directly to the Asterisk box. |
08:18.31 | MuppetMaster | Also, there is an entry on the wiki that someone kindly put up. |
08:19.00 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-82-213.37-151.net24.it) |
08:19.05 | pauldy | yea but it doesn't tell you that you have to use the 1 |
08:19.18 | MuppetMaster | pauldy: The ? |
08:19.20 | MuppetMaster | 1? |
08:19.38 | pauldy | yea for your number i.e. 1747XXXXXXX |
08:19.52 | MuppetMaster | Ah, I see. |
08:19.56 | MuppetMaster | Well the Wiki may be updated. |
08:19.59 | MuppetMaster | I will do that. |
08:20.34 | pauldy | I had to bust open etherreal to see how the app was registering then customize asterisk to it then everything worked |
08:21.16 | MuppetMaster | Those details are in the Gizmo knowledgebase. |
08:21.30 | MuppetMaster | As well as on the SIPPhone website, which is the proxy you are actually using with Asterisk. |
08:22.38 | MuppetMaster | Added a little blurb under Outbound Calls: http://www.voip-info.org/wiki/view/Asterisk+settings+Gizmo |
08:23.08 | MuppetMaster | Also, you can see the way it parses the example of extensions.conf under Outbound SIP Calls that it is providing the 1. |
08:23.56 | Akelavlk | What is Gizmo project exactly? |
08:24.38 | *** join/#asterisk tessier (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
08:26.03 | pauldy | 99-3 erro neat |
08:26.56 | *** join/#asterisk tobiasWolf (n=konversa@195.162.255.10) |
08:27.26 | *** join/#asterisk damned (n=vpol@prior.lanck.net) |
08:30.05 | MuppetMaster | http://www.gizmoproject.com |
08:30.19 | MuppetMaster | A SIP softphone with XMPP for Presence developed by the folks from http://www.sipphone.com. |
08:30.27 | MuppetMaster | Meant to be a Skype killer, but not really up to par yet |
08:31.32 | Akelavlk | Hmm, I see. It seems to be quite same.. Also prices are same.. |
08:33.30 | MuppetMaster | Except it is open standards based SIP/XMPP |
08:33.37 | MuppetMaster | May dial SIP URIs |
08:33.43 | MuppetMaster | And have your Asterisk registry with the proxy as welll |
08:33.47 | snitt | i like gizmo conference rooms |
08:34.19 | MuppetMaster | Yes, especially since they also integrated with http://www.freeconferencecall.com |
08:34.20 | pauldy | wow I never realized what a difference a codec could make |
08:34.23 | MuppetMaster | It is great for large conferences |
08:34.40 | pauldy | just runnig ilbc with gizmo reduces the latency by a ton vs ulaw |
08:34.51 | many | i know its a meta question, but did *anyone* connect a fax device behind a analogue card like tdm400? |
08:34.56 | MuppetMaster | Yeap, a nice thing. |
08:35.00 | MuppetMaster | Gizmo also likes gsm |
08:35.05 | MuppetMaster | Gotta run. Bye |
08:35.07 | *** part/#asterisk MuppetMaster (n=MuppetMa@169.red-81-184-73.user.auna.net) |
08:35.14 | pauldy | me too bed time |
08:37.05 | *** join/#asterisk ianrid (i=hidden-u@213-131-100-29.onyx.net) |
08:37.36 | *** join/#asterisk FreezeS (n=gido_b@82.208.156.94) |
08:37.46 | FreezeS | hey guys |
08:37.55 | FreezeS | I've got a problem |
08:38.00 | FreezeS | http://lists.digium.com/pipermail/asterisk-users/2004-October/067978.html |
08:38.11 | FreezeS | aparently it happened to this guy last year |
08:38.21 | FreezeS | but nobody answered then |
08:40.03 | ianrid | Hi, Do any of you guys use an E1 card on a UK BT network? |
08:43.59 | *** join/#asterisk apardo (n=w0w0@49.Red-83-41-11.dynamicIP.rima-tde.net) |
08:44.41 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
08:44.44 | puzzled | morning |
08:45.06 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
08:51.42 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
08:55.05 | nfi|ermes | hi all |
08:55.15 | nfi|ermes | i have some problem with nat and xlite |
08:55.31 | nfi|ermes | i cant register with asterisk |
08:56.13 | nfi|ermes | the client send request with NAT=pubblic_address and not his address in the subnet |
09:26.06 | *** join/#asterisk Starmaker (n=magnus@85.8.2.169) |
09:29.27 | *** join/#asterisk szer (n=Miranda@217.116.36.22) |
09:29.31 | szer | hi all |
09:40.55 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
09:42.46 | *** join/#asterisk littleball (n=littleba@bb219-75-114-120.singnet.com.sg) |
09:46.28 | Starmaker | hi, does anyone know if there is a way to build chan_bluetooth without rebuilding asterisk? |
09:47.52 | many | mh. |
09:47.56 | many | i think there is |
09:48.34 | Starmaker | I would guess it is possible by just including the headers for asterisk and some bluetooth library and just compiling it? |
09:48.42 | many | however you need the source |
09:48.50 | Starmaker | of course |
09:49.18 | many | what i did was t take asterisk source, copy blt into channels/ and then toplevel 'make subdirs' |
09:49.34 | nfi|ermes | it's incredible |
09:49.46 | nfi|ermes | xlite send a request to a web server |
09:49.49 | many | and then cp the chan_blt.so manually. |
09:50.02 | Starmaker | nfi|ermes, yeah, I noticed that too |
09:50.10 | Starmaker | I denied that using little snitch :) |
09:50.16 | nfi|ermes | to know your pubblic ip address |
09:50.18 | nfi|ermes | i too |
09:50.39 | nfi|ermes | i closed all request to thAT address in my firewal |
09:50.49 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
09:50.52 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
09:52.19 | cryzeck | Royk :) |
09:52.40 | many | starmaker: you have to beware anyway, not every blt headset works. |
09:53.02 | many | i even crashed one headset to be FUBAR with chan_blt :) |
09:53.37 | Starmaker | many, i was thinking about connecting a cell phone to my asterisk |
09:53.44 | littleball | many, what is the function of chan_blt? |
09:53.50 | many | ah. that should make less problems |
09:54.07 | many | littleball: bluetooth. connect bluetooth headsets/handies to asterisk |
09:54.25 | littleball | what is the special hardware requirement of the asterisk server? |
09:54.27 | many | bt hs obviously is a wireless headset then, bt handy connects asterisk to gsm |
09:54.40 | many | s/handy/mobile/gi |
09:54.50 | Starmaker | since my operator has free phonecalls within it's network, and I have a pre-paid subscription too, i was thinking about how to get cheaper cell phone calls |
09:54.52 | many | germanisms. :) |
09:56.01 | littleball | many, i think a special hardware needed to hook onto the asterisk server to use your bluetooth headset, right? |
09:56.28 | Starmaker | littleball, a bluetooth interface |
09:56.32 | Starmaker | of course |
09:56.43 | *** join/#asterisk DonDonnie (n=don@ip51cd13bc.speed.planet.nl) |
09:57.08 | many | littleball: a BT adapter, my usb/BT dongle works fine |
09:57.08 | littleball | Starmaker, can you recommend a device(which i can purchase) so that i can try it out myself. |
09:57.15 | RoyK | hi |
09:57.22 | littleball | ok. |
09:57.50 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
09:58.25 | Starmaker | yeah, just about any usb bluetooth dongle works |
09:59.04 | many | iam just pondering how many headsets can be talking at the same time over one dongle. |
09:59.18 | Starmaker | depends on the dongle |
09:59.20 | littleball | many, i want to ask the same question :) |
09:59.45 | Starmaker | and of course the implementation |
09:59.46 | many | i was confused since there are only 12 or so RF Channels |
10:00.05 | many | and i thought one rf channel has to be dedicated to one connection |
10:00.17 | Starmaker | but there are dongles that allows like 8 concurrent connections, and there are those that allows only one concurrent connection |
10:00.23 | many | (and afaik even some devices are hardcoded to channels) |
10:01.51 | many | well. :) |
10:02.34 | *** join/#asterisk Willem_ZA (n=willem_Z@wbs-146-146-209.telkomadsl.co.za) |
10:02.43 | many | now if only fax device would be working. |
10:03.34 | Willem_ZA | hi, could anyone help me with a digium tdm400p? |
10:04.04 | many | depends on what yuor problem is |
10:05.11 | Willem_ZA | well, i just like to know, i have a 3x fxs and a 1x fxo config and. |
10:05.44 | Willem_ZA | if i plug a phone into the fxs i do get power on the phone, but no dialtone, |
10:06.17 | Willem_ZA | according to asterisk everything is installed and running properly. |
10:06.21 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
10:07.02 | Willem_ZA | and when i dial the extention, i get a busy tone, but when i call the fxo from my pbx, it just keep ringing.. any suggestions? |
10:07.31 | DonDonnie | @Willem_ZA, do you see anything in the asterisk console? |
10:07.49 | Willem_ZA | yes, hold on, i will give you an example. |
10:08.02 | DonDonnie | k |
10:08.47 | Willem_ZA | Dial("SIP/louis-318b", "Zap/1|20") in new stack |
10:08.47 | Willem_ZA | Oct 18 12:08:21 WARNING[3402]: channel.c:2249 ast_request: No channel type registered for 'Zap' |
10:08.47 | Willem_ZA | Oct 18 12:08:21 NOTICE[3402]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
10:08.47 | Willem_ZA | <PROTECTED> |
10:08.47 | Willem_ZA | <PROTECTED> |
10:08.48 | Willem_ZA | <PROTECTED> |
10:08.50 | Willem_ZA | <PROTECTED> |
10:08.52 | Willem_ZA | <PROTECTED> |
10:08.58 | Willem_ZA | <PROTECTED> |
10:09.00 | Willem_ZA | <PROTECTED> |
10:09.02 | Willem_ZA | <PROTECTED> |
10:09.08 | Willem_ZA | <PROTECTED> |
10:09.10 | Willem_ZA | <PROTECTED> |
10:09.12 | Willem_ZA | <PROTECTED> |
10:09.14 | Willem_ZA | <PROTECTED> |
10:09.16 | Willem_ZA | thefly*CLI> |
10:09.22 | Willem_ZA | the ZAP error i did not get the last time.. |
10:10.08 | DonDonnie | well, obviously there is no Zap/1 channel avaialble |
10:11.09 | Willem_ZA | k, will take a look quickly, thanx |
10:13.55 | *** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net) |
10:14.23 | *** join/#asterisk Tili (i=Tili@202-133-67-23-dialup.sat.net.pk) |
10:14.28 | Willem_ZA | why will this work?: |
10:14.39 | Willem_ZA | or not work? |
10:14.39 | Willem_ZA | dial 1000 |
10:14.40 | Willem_ZA | <PROTECTED> |
10:14.40 | Willem_ZA | <PROTECTED> |
10:14.40 | Willem_ZA | <PROTECTED> |
10:14.40 | Willem_ZA | Use EXIT or QUIT to exit the asterisk console |
10:14.41 | Willem_ZA | <PROTECTED> |
10:14.43 | Willem_ZA | <PROTECTED> |
10:14.45 | Willem_ZA | <PROTECTED> |
10:14.47 | Willem_ZA | <PROTECTED> |
10:14.49 | Willem_ZA | <PROTECTED> |
10:14.51 | Willem_ZA | <PROTECTED> |
10:14.53 | Willem_ZA | <PROTECTED> |
10:15.41 | Willem_ZA | but should this not acctually ring on the extention? |
10:16.16 | DonDonnie | you could try to make a SIP extension, with a softphone and dial Zap/1 with your Sip softphone |
10:16.36 | DonDonnie | like: exten => 100,1,Dial(ZAP/1,20) |
10:16.59 | DonDonnie | you sure Zap is loaded in asterisk? just for the record ;) |
10:17.40 | Willem_ZA | how can i be sure its loaded? |
10:18.06 | X-Rob | 'zap show channels' |
10:18.47 | DonDonnie | from the cli |
10:20.03 | Willem_ZA | ok |
10:20.09 | *** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net) |
10:20.11 | kippi | hey |
10:20.17 | kippi | is there a echo test? |
10:21.45 | X-Rob | Yes. |
10:22.11 | X-Rob | kippi - at the astersk*CLI> prompt, type 'show applications' |
10:22.23 | X-Rob | those are all the toys you that come with asterisk. |
10:23.17 | Willem_ZA | ok, i ran the command, but i got this error: zap show channels |
10:23.18 | Willem_ZA | No such command 'zap' (type 'help' for help) |
10:23.18 | Willem_ZA | , so i tried:show channels |
10:23.18 | Willem_ZA | and got:Channel Location State Application(Data) |
10:23.18 | Willem_ZA | 0 active channels |
10:23.18 | Willem_ZA | 0 active calls |
10:23.47 | X-Rob | no such command 'zap' means that Zap is not loaded. |
10:26.08 | ianrid | Any ISDN30 experts about? |
10:27.09 | DonDonnie | @Willem, what is the output when you run /sbin/ztcfg -vvvv |
10:27.22 | DonDonnie | from your linux shell |
10:28.12 | Willem_ZA | let me check |
10:28.38 | Willem_ZA | Zaptel Configuration |
10:28.39 | Willem_ZA | ====================== |
10:28.39 | Willem_ZA | Channel map: |
10:28.39 | Willem_ZA | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
10:28.39 | Willem_ZA | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
10:28.39 | Willem_ZA | Channel 03: FXO Kewlstart (Default) (Slaves: 03) |
10:28.41 | Willem_ZA | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
10:28.43 | Willem_ZA | 4 channels configured. |
10:29.11 | DonDonnie | ok, did you reloaded asterisk? |
10:29.20 | Willem_ZA | i did yes. |
10:29.22 | DonDonnie | restarted actually |
10:29.25 | Willem_ZA | restart now** |
10:29.27 | DonDonnie | ok |
10:29.38 | ianrid | We've had an ISDN30 box installed and the line tests fine with a loop test althoug the service has yet to be activated. What should the status of our E1 card be when we plug this into the box with a stright through RJ45 cable |
10:30.32 | ianrid | We have red LEDs by the way withan alarm on the BT ISDN box. |
10:30.46 | DonDonnie | @Willem, check this: http://www.digium.com/downloads/hw_article (scroll down a little) it has configuration files for TDM* cards |
10:31.20 | Willem_ZA | ok, thanx |
10:31.40 | DonDonnie | Good luck, I am off for now |
10:31.52 | ianrid | A looped cable into the ISDN box gives green LEDs. A looped cable into the E1 card gives green LEDs. But a straight through between the two gives red LEDs |
10:31.58 | DonDonnie | your cards are installed well btw |
10:32.13 | DonDonnie | as ztcfg shows no errors |
10:32.23 | X-Rob | ianrid - where are you? |
10:32.57 | ianrid | UK |
10:33.50 | X-Rob | ~pb |
10:33.51 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
10:33.58 | X-Rob | your /etc/zaptel.conf |
10:37.04 | ianrid | http://pastebin.ca/25822 |
10:47.51 | *** join/#asterisk dreamler (n=bill@195.28.52.162) |
11:08.59 | *** part/#asterisk dreamler (n=bill@195.28.52.162) |
11:09.43 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
11:27.50 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.169.84) |
11:27.54 | MuppetMaster | Hello |
11:28.00 | MuppetMaster | IAX and DTMF. |
11:28.13 | MuppetMaster | If it is always inband, how does one use a low bandwidth codec? |
11:28.16 | MuppetMaster | Like g729a? |
11:28.34 | *** join/#asterisk rowter (n=SilverDr@201.135.26.195) |
11:28.41 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
11:30.27 | *** join/#asterisk hypomanic (n=foo@FC.wayout.net) |
11:30.43 | MuppetMaster | Anyone have an idea on low band codecs and DTMF on IAX channels? |
11:30.57 | X-Rob | yes. They don't work. use inband. |
11:31.17 | MuppetMaster | I see that, but then how does one use inband with a codec like g729 which destroys them? |
11:31.26 | MuppetMaster | IAX only works reliably with DTMF over alaw/ulaw? |
11:31.31 | MuppetMaster | Seems like a real deficiency. |
11:31.51 | X-Rob | uh |
11:31.56 | MuppetMaster | ? |
11:31.59 | X-Rob | there was a silent 'don't' there |
11:32.02 | X-Rob | 'don't use inband' |
11:32.24 | MuppetMaster | But, according to the Wiki, it only does inband: http://www.voip-info.org/wiki-IAX |
11:32.24 | X-Rob | sorry |
11:32.25 | *** join/#asterisk Talnakh (n=Talnakh@217.22.177.17) |
11:32.27 | MuppetMaster | How to get it out of band? |
11:33.08 | X-Rob | afaik it _doesn't_ use it inband. |
11:33.29 | MuppetMaster | ? |
11:33.54 | X-Rob | it's always oob |
11:34.05 | X-Rob | what's your problem? |
11:34.14 | Talnakh | Hi all. can when i am trying to call to asterisk though PSTN line, if i hang the phone, asterisk still tries to record a voice mail, leaving a voicemeil of approximately 1 minute with hang tones. how can i fix it? |
11:34.32 | *** join/#asterisk skiold (n=userid@84-121-64-126.onocable.ono.com) |
11:34.43 | X-Rob | Talnakh - enable busydetect in zapata.conf. It's all documented |
11:35.07 | Talnakh | X-Rob, i did that :-( |
11:35.32 | Talnakh | ok, although i reloaded astrisk, i ll restart the pbx now |
11:35.49 | Starmaker | crappy phone |
11:35.53 | Starmaker | i have a mt |
11:35.54 | Starmaker | ot |
11:36.01 | Starmaker | motorola a925 |
11:36.08 | MuppetMaster | X-Rob: You have confused me. Does IAX only do inband and therefore can not use codecs like g729. Or is there a way to get it to do out-of-band so that it may use the lower bandwidth codecs? |
11:36.13 | Starmaker | and it won't work with chan_bluetooth |
11:36.33 | *** join/#asterisk kippi (n=chrisfro@untrust-gct.equinoxit.net) |
11:36.33 | X-Rob | IAX does _not_ use in band signalling. |
11:36.41 | MuppetMaster | What does it use? |
11:36.46 | X-Rob | oob |
11:36.48 | X-Rob | out of band |
11:36.49 | X-Rob | as I said |
11:36.51 | X-Rob | what is your problem? |
11:37.13 | MuppetMaster | My DTMF on an IAX channel between two Asterisk boxes is not being transmitted effectively using g729 codecs. |
11:37.23 | MuppetMaster | For example, unable to enter a uname/passwd in the voicemail app. |
11:37.30 | MuppetMaster | But if I use alaw, no problem. |
11:37.39 | MuppetMaster | And then I read the wiki, and it says that IAX does inband ONLY. |
11:37.52 | X-Rob | no |
11:37.54 | X-Rob | it sends it inline |
11:37.57 | X-Rob | not in band. |
11:38.07 | MuppetMaster | What is the difference between inline and inband? |
11:38.23 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
11:38.24 | X-Rob | Wups |
11:38.28 | MuppetMaster | Welcome back. |
11:38.28 | ianrid | X-Rob, Did you spot anything in our zaptel.conf? |
11:38.38 | MuppetMaster | What is the delta between inline/inband? |
11:38.41 | MuppetMaster | As the wiki says inband. |
11:38.46 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
11:38.49 | X-Rob | the wiki says INLINE |
11:38.50 | MuppetMaster | And I do not understand the differentiation. But I can be dense. |
11:39.00 | MuppetMaster | You are right., |
11:39.03 | MuppetMaster | Reading what I wanted to. |
11:39.06 | MuppetMaster | But what is the difference? |
11:39.06 | X-Rob | inband == as an audio stream |
11:39.14 | X-Rob | inline == 'the user pushed 2' |
11:39.45 | X-Rob | it's bad phrasing |
11:39.53 | MuppetMaster | Correct, which can not be used over g729, which destroys DTMF |
11:40.08 | X-Rob | there's another problem. |
11:40.16 | X-Rob | that's not it. |
11:40.39 | X-Rob | iax2 does not, repeat, not, encode DTMF as audio at any stage of the transmission. |
11:40.57 | MuppetMaster | I see, so it is inline, in terms of using the same UDP stream. |
11:41.02 | MuppetMaster | Now I am beginning to get it. |
11:41.12 | MuppetMaster | So why would it be coming through garbled? |
11:41.21 | X-Rob | that's the next question. |
11:41.27 | X-Rob | possibly something else is encoding it. |
11:41.41 | Talnakh | X-Rob, busydetect worked, everything is ok now. |
11:42.02 | MuppetMaster | Hmmmm..... |
11:42.18 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
11:42.31 | X-Rob | ianrid - sorry, what's your pastebin? |
11:42.40 | MuppetMaster | If it is IAX2 between the 2, and I can see both are connecting g729. What else could be encoding? (BTW - Both machines have g729 licenses) |
11:43.06 | *** join/#asterisk Willem_ZA (n=willem_Z@wbs-146-146-209.telkomadsl.co.za) |
11:43.17 | X-Rob | the end device is using g729 and deciding that it wants to send dtmf inband? |
11:43.32 | MuppetMaster | No, IAX2 in Asterisk on both sides, v1.2beta |
11:44.30 | X-Rob | 'the end device' == 'the thing that is pushing '2'' |
11:44.36 | MuppetMaster | But, the endpoint on the Asterisk A intiating the dial is a SIP/Avaya 4602SW IP Phone and then Asterisk A is sending to AsteriskB. |
11:44.49 | MuppetMaster | So, is Asterisk not suppose to translate between SIP and IAX2 seemlessly? |
11:45.07 | MuppetMaster | And AsteriskB is getting the confusing stuff in the voicemail app. |
11:48.26 | MuppetMaster | So IAX2 does not transmit DTMF if the originator is a SIP endpoint on an Asterisk box? |
11:48.35 | MuppetMaster | Transmits, but not properly. |
11:48.37 | X-Rob | No. Something else is wrong. |
11:48.39 | X-Rob | I said that before |
11:48.42 | MuppetMaster | The local voicemail app works on the Asterisk A. |
11:48.43 | X-Rob | you'll have to do some debugging. |
11:48.50 | MuppetMaster | Ah, I see, maybe a bug in the beta. |
11:48.57 | MuppetMaster | Thanks |
11:49.10 | RoyK | ~seen coppice |
11:49.13 | jbot | coppice <n=chatzill@123.192.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 17h 55m 48s ago, saying: 'nortel used to be a really great company. something went horribly wrong'. |
11:49.19 | RoyK | ~seen jesus |
11:49.20 | jbot | jesus <i=jesus@84-122-34-139.onocable.ono.com> was last seen on IRC in channel #debian, 54d 17h 34m 49s ago, saying: 'hi all'. |
11:49.42 | X-Rob | ~seen thelight |
11:49.44 | jbot | X-Rob: i haven't seen 'thelight' |
11:49.52 | X-Rob | MWahahah |
11:49.59 | RoyK | ~seen elvis |
11:50.00 | jbot | elvis <~elvis@9-151.tr.cgocable.ca> was last seen on IRC in channel #kde, 176d 18h 22m 40s ago, saying: 'StevenR, thx a lot bro'. |
11:50.06 | X-Rob | ~seen A good movie recently |
11:50.07 | jbot | X-Rob: i haven't seen 'a good movie recently' |
11:50.15 | Willem_ZA | does anyone know anything about the chan_zap.so module? |
11:50.18 | RoyK | ~seen the light |
11:50.19 | jbot | RoyK: i haven't seen 'the light' |
11:50.27 | X-Rob | THE BAND!!! |
11:50.38 | ianrid | X-Rob, http://pastebin.ca/25822 |
11:51.19 | X-Rob | ianrid - it shuld be 1,1,0... but apart from that it's fine. |
11:51.31 | X-Rob | what card do you have? te110? |
11:51.40 | X-Rob | uh, span=2, means TE400? |
11:52.09 | X-Rob | wct4xx even |
11:52.16 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.169.84) |
11:52.33 | ianrid | The card is a Junghanns Double E1 |
11:53.26 | X-Rob | don't know anything about them, sorry 8-( |
11:56.30 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
11:56.35 | rowter | anyone has an experience similar to http://bugs.digium.com/view.php?id=5406? |
11:59.51 | *** join/#asterisk spiekey (n=spiekey@p549D1B88.dip0.t-ipconnect.de) |
11:59.54 | spiekey | hello! |
12:00.14 | spiekey | can i put Asterisk between my telco and my PBX with two isdn cards? |
12:04.49 | Talnakh | X-Rob, i hear an echo when i use SIP phone with asterisk. could you suggest which part of config should i dig? |
12:05.02 | Talnakh | to get rid of echo |
12:08.31 | *** join/#asterisk FastJack (i=fastjack@2001:8d0:20ff:3:0:0:0:1) |
12:09.26 | *** join/#asterisk victormedrano (n=vmedrano@196.32.128.206) |
12:10.44 | X-Rob | Talnakh - are you using asterisk 1.2 or CVS-HEAD? |
12:11.08 | Talnakh | i use stable |
12:11.12 | X-Rob | no |
12:11.14 | X-Rob | you use 1.0 |
12:11.31 | X-Rob | the name 'stable' was depreciated about 6 months ago. |
12:11.35 | Talnakh | ok, i use 1.0.9 i think |
12:11.44 | Talnakh | :-) |
12:11.58 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
12:12.02 | X-Rob | I suggest upgrading to current CVS-HEAD, which will enable the new KB1 echo cancellation |
12:12.05 | X-Rob | works well |
12:12.24 | Talnakh | ok, thanks. i ll think about it. :-) |
12:12.50 | *** join/#asterisk gonzo- (n=gonzo@web.portaone.com) |
12:13.43 | gonzo- | Hi everybody. Doas anybody know where i can get PCI specs on AVM Fritz! card? |
12:14.02 | Talnakh | their website does not have? |
12:15.34 | *** join/#asterisk ToR\L (i=toril@cpe-24-58-23-240.twcny.res.rr.com) |
12:15.53 | ToR\L | quick stupid question |
12:16.01 | ToR\L | there a way to announce the name before an extension? |
12:17.47 | Dr_Ray | when you call an extension? |
12:19.44 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
12:20.06 | ToR\L | yea |
12:20.24 | Dr_Ray | just put a sound file with their names in your extensions.conf |
12:20.32 | ToR\L | hmm |
12:20.48 | *** join/#asterisk falbala- (i=falbala@bne75-1-81-57-10-154.fbx.proxad.net) |
12:21.44 | *** part/#asterisk falbala- (i=falbala@bne75-1-81-57-10-154.fbx.proxad.net) |
12:25.11 | *** join/#asterisk coppice (n=chatzill@123.192.17.210.dyn.pacific.net.hk) |
12:26.45 | jake1932 | ToR\L: or if you want to get fancy and the user already recorded their name in voicemail, find out where the VM names are and play them directly from there |
12:28.46 | Dr_Ray | mmm... fancy |
12:36.45 | ToR\L | yea thats what I was thinking |
12:37.26 | ToR\L | I'm asking for a friend... I have voip through home (on asterisk) |
12:37.38 | ToR\L | and he setup some an asterisk box for someone |
12:37.49 | ToR\L | so I'm trying to figure out exactly what hes trying to do |
12:38.07 | ToR\L | I remember recording my name through ivr for my vm on my box |
12:39.52 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
12:40.12 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
12:41.34 | jake1932 | ToR\L: if you haven't found it already, the names are in /var/spool/asterisk/voicemail/[vm-context]/[exten]/greet.gsm (or.wav) |
12:44.58 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
12:48.55 | Talnakh | i want to make asterisk server to be online at all times. does anyone know how i can make PC to be always switched on. If lets say there is a powercut, the pc will switch on when electricity appears. |
12:49.15 | jake1932 | Talnakh: use a UPS |
12:49.28 | FreezeS | jake1932: my words exactly :) |
12:50.11 | Talnakh | jake1932, FreezeS, bad idea if there is no power for 4 hours. |
12:50.35 | jake1932 | Talnakh: http://cableorganizer.com/generators/propane-generators-guardian-plus.htm |
12:50.39 | FreezeS | Talnakh: you have 2 solutions. Use a VERY large battery or a generator |
12:51.09 | FreezeS | actually, there is a third sollution: use some extremely low power computer |
12:51.17 | FreezeS | like a laptop or something... |
12:51.24 | Talnakh | but can i make linux to switch on automatically? |
12:51.31 | Dr_Ray | your bios might be setable to power on |
12:52.02 | Talnakh | on old TX cases unless the switch is off, pc will go back to on when power appears. |
12:52.05 | FreezeS | Talnakh, simplest solution: AT power supply :) |
12:52.11 | jake1932 | Talnakh: you don't have to worry about it turning on if it never turns off |
12:52.39 | *** join/#asterisk tdonahue (n=tdonahue@64.201.13.50) |
12:53.01 | tdonahue | good morning all |
12:53.04 | mutilator | usually the bios will let ya do that anymore |
12:53.13 | Talnakh | yes, but motherboard require a different power connector for ATX and TX power supplies |
12:53.32 | Dr_Ray | my motherboard has a bios setting to auto power on |
12:53.34 | FreezeS | of course Talnakh. Just use an AT compatible mobo |
12:54.07 | Talnakh | FreezeS, megalol. it wont be good enough for me now. wit crappy CPU and memory :-) |
12:54.49 | Talnakh | but i think i have an idea how to make POWER supply to be switched on at all times. |
12:55.00 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
12:55.01 | FreezeS | Talnakh: well, in this case buy a VERY large battery for your UPS |
12:55.08 | Dr_Ray | Talnakh - check your bios |
12:55.10 | FreezeS | or link several UPS'es |
12:55.24 | mutilator | ^ wouldn't work |
12:55.40 | FastJack | some bioses have a "power up after power failure" option |
12:55.43 | mutilator | well it would to an extent i guess |
12:55.59 | FreezeS | you can link many UPS'es |
12:56.26 | FastJack | FreezeS: imagine a beowulf cluster of UPSes ;) |
12:56.27 | Talnakh | no, i ll shortcircuit a couple of pins on power supply connector. I ll try it now and i ll let you all know if it works. it might be helpful |
12:56.59 | FreezeS | Talnakh, since you didn't know about UPS or a generator, I guess you're no electronics specialist |
12:57.10 | FreezeS | so I would recommend you DON'T DO THAT ! |
12:57.14 | Talnakh | U assume too much :-) |
12:57.23 | FreezeS | better check your bios options |
12:57.25 | Katty | mew. |
12:57.26 | *** join/#asterisk Lathos42 (n=Lathos42@adsl-69-208-243-229.dsl.lgtpmi.ameritech.net) |
12:57.36 | Katty | Lathos42: :> |
12:57.36 | Dr_Ray | lord forbid you do the easy/correct way first.. |
12:58.00 | Lathos42 | Katty: Good Morning |
12:58.01 | Talnakh | i knew about UPS and generator and all that stuff. i just like electronics haking |
12:58.15 | Talnakh | hacking i mean |
12:58.41 | pif | hi, anyone tried the swissvoice IP10 phone? |
12:58.45 | Katty | Lathos42: mew. |
13:01.07 | FreezeS | what happends Lathos42 ? :) |
13:01.15 | tdonahue | is there any way to use queues to ring people in the same order every time? |
13:01.29 | FreezeS | tdonahue, it's very simple even |
13:01.39 | Lathos42 | FreezeS: I suddenly found myself in #asterisk-unregistered as Lathos42_ :) |
13:01.42 | FreezeS | exten => 1,1,Dial(guy1) |
13:01.47 | FreezeS | exten => 1,2,Dial(guy2) |
13:01.57 | FreezeS | exten => 1,3,Dial(guy3) |
13:02.00 | FreezeS | etc... |
13:02.26 | tdonahue | FreezeS, What happens when guy1 is on the phone? |
13:02.32 | FreezeS | guy2 is called |
13:02.37 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
13:02.57 | tdonahue | ok, i'll try that out. thanks |
13:03.03 | FreezeS | glad to help |
13:03.38 | Katty | the world is coming to an end :< |
13:03.54 | FreezeS | of course Katty |
13:04.22 | FreezeS | actually, everything comes to an end eventually :) |
13:04.56 | *** join/#asterisk docE (n=docE@66.237.242.41.ptr.us.xo.net) |
13:05.09 | docE | Morning! |
13:05.28 | jake1932 | not worried so much about the world - just keep my asterisk box up and running |
13:05.35 | FreezeS | docE: It's 4:04 PM in Romania :D |
13:05.51 | FreezeS | jake1932, what are you using * for ? |
13:05.59 | RoyK | dorstopper..... |
13:06.01 | jake1932 | i have a few |
13:06.06 | RoyK | s/dor/door |
13:06.23 | jake1932 | one is at a client site as an IVR |
13:06.32 | Lathos42 | I use asterisk so I can be hip like all the Cool kids |
13:06.54 | Katty | sad. |
13:07.14 | FreezeS | Lathos42: buy an iPod better |
13:07.17 | jake1932 | i have one here at home so i'm not paying the big money to the bell company |
13:08.18 | Ariel_ | morning folks |
13:08.30 | Ariel_ | Katty, it's not coming to an end yet. |
13:09.05 | Katty | Ariel_: that's not what the JWs are trying to cram down my throat. |
13:10.36 | jake1932 | that's 242 hugs - should be a while |
13:10.49 | docE | ya well.. Ill get around to it. |
13:14.42 | ManxPower | Ugh. I really should start packing. |
13:15.31 | Ariel_ | ManxPower, yes you should. (just got 10 boxes and will be getting another 10 boxes on Thurday to finish up packing). |
13:16.07 | ManxPower | Ariel_, I just need to pack up the laptop and a suitcase. |
13:16.08 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
13:16.38 | Ariel_ | ManxPower, nice. I have a 4 bedroom house to pack. Then get a storage unit to store the stuff then move into a 2 bedroom appartment. |
13:17.12 | ManxPower | Ariel_, Ah. I'm heading to Covington/Gulfport this week. |
13:17.16 | Ariel_ | Sold the house in 3 days with just a sign out front. (Strange I feel I asked way to low). |
13:17.19 | X-Rob | Katty - kick the JW's in the nads. |
13:17.32 | *** join/#asterisk pointer (i=pointer@64.18.103.6) |
13:18.00 | Katty | X-Rob: I shall not. |
13:18.03 | Katty | X-Rob: She is my mother. |
13:18.10 | X-Rob | Bugger. |
13:18.11 | jake1932 | hah |
13:18.12 | X-Rob | you poor thing. |
13:18.22 | Katty | X-Rob: hardly. |
13:18.26 | Katty | X-Rob: I learned to think. |
13:18.43 | Katty | X-Rob: If I hadn't, I'd still be a good little JW girl. |
13:18.59 | jake1932 | does she still come to your door with a booklet? |
13:19.19 | X-Rob | Katty - onya. |
13:19.19 | Katty | jake1932: no, she pesters me everytime I visit. |
13:19.23 | FreezeS | what's JW ? |
13:19.37 | X-Rob | Katty - tell her you've deicded to worship the FSM |
13:19.44 | FreezeS | FSM rule ! |
13:19.44 | Katty | X-Rob: Stopping fixing me. |
13:19.59 | Katty | X-Rob: I didn't ask for help or counsel. |
13:20.00 | FreezeS | oh, Jehova's Whitnesses |
13:20.15 | X-Rob | Katty - sorry, I'm just trying to be humourous. |
13:20.16 | X-Rob | Geez. |
13:20.21 | Katty | k |
13:20.42 | FreezeS | Katty, do you know what FSM is ? |
13:21.05 | Katty | FreezeS: of course. |
13:21.19 | FreezeS | ok, I only found out about 2 weeks ago |
13:23.11 | Katty | Ariel_: that's what google is for. |
13:23.58 | FreezeS | Ariel_, it's the TRUE religion |
13:24.13 | FreezeS | people finally found out the REAL TRUTH |
13:24.14 | X-Rob | FreezeS - that's TRUE(*) religion |
13:24.24 | X-Rob | (*) where true == as true as any |
13:24.38 | Talnakh | hi all, i am back to life |
13:24.39 | Dr_Ray | well, as untrue as the all are |
13:24.41 | Ariel_ | well I got two main ones from google the Flying Spaghetti Monsterism or the Free Speach Movement. |
13:24.56 | Talnakh | i just got a really bad electric shock. |
13:25.16 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
13:25.27 | FreezeS | X-Rob, there is only ONE truth |
13:25.34 | jake1932 | Talnakh: didn't your parents ever tell you not to mess with electricity |
13:25.38 | FreezeS | and finally the people found it |
13:25.51 | Dr_Ray | there is no dog |
13:27.19 | Talnakh | ah, i am just kidding. but remember i said that it is possible to keep ATX motherboard powered on without BIOS settings. well, it actually works. my mobo does not support to switch on using bios. |
13:27.33 | Ariel_ | X-Rob, truth is that they all lie, they all have secrets and you can't belive any of them. |
13:28.23 | FreezeS | Ariel_, have you found the TRUE religion ? |
13:29.28 | Ariel_ | FreezeS, yes I do belive. |
13:30.55 | pointer | this is a silly newbie question, but...when you change the mp3s in your MOH dir for a context, how do you you tell asterisk to restart/notify the mpg123 threads? will a reload work or will it require a restart (when convenient)? |
13:30.56 | FreezeS | one more pastafarian joins the Holy Church |
13:31.35 | FreezeS | pointer: musiconhold.conf |
13:31.36 | iCEBrkr | pointer: I'd just make sure all instances of mpg123 are dead. Then when Asterisk fires it back up, it'll collect the new filenames. |
13:32.11 | pointer | FreezeS: I know what config file to edit...in this case, I shouldn't have to touch it though... |
13:32.31 | FreezeS | oh, I misunderstood the question |
13:32.39 | pointer | iCEBrkr: so it only fires up mpg123 threads when a channel is put on hold? |
13:32.46 | iCEBrkr | pointer: Yeah |
13:33.07 | iCEBrkr | pointer: But sometimes there will be two mpg123 processes out there running even though no one is on hold. |
13:33.14 | iCEBrkr | So just be sure to kill them off. |
13:33.33 | pointer | iCEBrkr: ah, ok....didn't realize that. Can I safely kill running mpg123 processes if I know that nobody is on hold? |
13:33.34 | *** join/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com) |
13:33.44 | iCEBrkr | pointer: Sure |
13:34.09 | pointer | iCEBrkr: thanks, I really didn't have time to test that one and risk the worst |
13:34.15 | iCEBrkr | :) |
13:38.30 | *** part/#asterisk pointer (i=pointer@64.18.103.6) |
13:39.47 | docE | Say anyone know anything about static agents in app_queue or could point me somewhere to find the info. I tried the WIKI and Google with no information |
13:40.23 | RoyK | static agents? |
13:40.31 | RoyK | waddayamean? |
13:40.44 | FreezeS | docE, you could just use lines instead of agents |
13:40.49 | FreezeS | like sip/user |
13:40.49 | iCEBrkr | RoyK: They all have afro's from the static electricity |
13:40.53 | FreezeS | or zap/1 |
13:40.55 | Ahrimanes | lol |
13:41.04 | RoyK | SIP/user |
13:41.06 | RoyK | not sip/user |
13:41.06 | RoyK | :P |
13:41.38 | skyen | what's wrong when debug says "Cannot cretate channel of type SIP", when the call is patched throuh? |
13:41.43 | FreezeS | RoyK: hopefully, he got the idea |
13:41.57 | skyen | everything works as a charm, but that errormsg keeps appearing |
13:42.17 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
13:43.04 | FreezeS | skyen: maybe the problem is on the client part. Are they registered on the server ? |
13:43.15 | *** join/#asterisk gambolputty (n=gambolpu@72.240.241.108) |
13:43.33 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
13:44.32 | skyen | FreezeS: yes |
13:44.57 | FreezeS | can they call ? |
13:45.06 | skyen | everything works. |
13:45.12 | skyen | just complaints in the log |
13:45.21 | FreezeS | when do these messages appear ? |
13:45.28 | skyen | just as the call is patched |
13:45.34 | skyen | when the phone starts ringing |
13:45.55 | docE | Well here is the synario of what I have now.. its working but doesnt work when the system reboots. I have 2 SIP extensions not registered to asterisk so that when a call is in queue and its routed to one of the end points it dials a SIP URI and sends the call of. How do I make it so users dont have to actually login to the queue? |
13:46.47 | FreezeS | docE: you need the extensions to be registered to asterisk |
13:47.09 | FreezeS | what telephone are you using for the agents ? |
13:48.18 | docE | 2 Cisco 3660's |
13:48.20 | docE | :) |
13:49.04 | Ariel_ | I really think that the press is evil. The just want to report doom and gloom. There trying to make everyone think that this Dandemic for the bird flue is here or close to being here in the states. argh |
13:49.21 | *** join/#asterisk nomazda (i=nyyankee@user-0c6tnqf.cable.mindspring.com) |
13:49.44 | docE | The 1 Cisco 3660 accepts SIP and is routed into a NEC PBX via CAS signaling.. The other sits in Melbourne, AUS and routes to a DID on the PSTN |
13:50.07 | jake1932 | damn it |
13:50.13 | Ariel_ | skyen, are you using canreinvite=yes on your system? |
13:50.32 | docE | Damn what? |
13:50.34 | skyen | no, all clients are forced to no on that option |
13:50.54 | jake1932 | i was on a call and the guy i was talking to (SIP to SIP) got another call, he couldn't hear me after the call waiting beep |
13:51.24 | docE | Asterisk call? |
13:51.28 | jake1932 | yep |
13:51.42 | docE | interesting.. |
13:51.54 | docE | What version? I am using head and dont have the issue with call waiting |
13:52.00 | skyen | Ariel_: could that be my problem? |
13:52.18 | jake1932 | HEAD as of 9/15 |
13:52.31 | Ariel_ | skyen, no it should not be. If it works just leave it along in my view. |
13:52.49 | Ariel_ | jake1932, head has had allot of fixes since then. |
13:52.56 | jake1932 | ok - i'm upgrading |
13:53.21 | Ariel_ | jake1932, backup first |
13:53.29 | jake1932 | will do - tnx |
13:53.44 | *** join/#asterisk synthetiq (n=roger@64.201.13.50) |
13:54.01 | synthetiq | what other options than SER do i have for load balancing on cloned machines |
13:54.03 | Ariel_ | why did Mediatix make the 1204.. I know to make my life a living hell. Yes that is it. |
13:54.48 | Ariel_ | cloned machine... hum ... hartbeat for starters. hummm look at the wiki for more info |
13:54.52 | *** join/#asterisk ikey (i=ikey@220.226.13.53) |
13:56.26 | *** join/#asterisk trym (n=trym@cD9088B17.sdsl.catch.no) |
13:57.07 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
13:58.01 | trym | what was the name of that handy app that converted sound files into .gsm again? |
13:58.18 | trym | sox |
13:58.22 | Ahrimanes | trym: sox? |
13:58.25 | trym | indeed |
13:58.29 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
13:59.28 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
13:59.36 | KriS83 | Hi |
13:59.39 | *** join/#asterisk lars-- (n=lars@lars.debian.us) |
13:59.43 | Katty | hi. |
13:59.45 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
14:01.02 | *** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
14:01.07 | wmandra | morning all |
14:01.37 | wmandra | is anyone else having trouble logging into the wiki??? |
14:03.33 | Katty | morning. |
14:05.05 | trym | file.c:492 ast_openstream_full: File velkommen does not exist in any format |
14:05.13 | trym | file.c:804 ast_streamfile: Unable to open velkommen (format ulaw): No such file or directory |
14:05.15 | trym | im very sure it exists |
14:06.57 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
14:07.09 | szer | knows anybody something from bug 5442? |
14:08.28 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
14:08.53 | psk | anyone with problems sending faxes with asterisk and getting T4 timeout? |
14:10.23 | IronHelix | wmandra- try logging out on every machine/browser you are logged in with, then login again |
14:10.34 | paryl | i've got a wierd problem... when a call comes in from an analog line, i have it ring all SIP phones (gxp-2000's)... when one person is on line1, the incoming call rings to their second line, but the other phones ring on line1 AND line2.. i can't figure out why |
14:11.17 | IronHelix | that is very odd |
14:11.25 | IronHelix | and if nobody is on the line, then only line1 rings? |
14:11.26 | jake1932 | does line 1 support call waiting? |
14:12.53 | *** join/#asterisk clennon (n=clennon@dialup144.ts009.bmt.esat.net) |
14:13.15 | KriS83 | I need a hint.. I have just compiled * beta 1.2.0 because I need the mysql cmd feature. Everything seems to work fine. Also compile chan-capi (using a AVM B1 card) set incomingmsn=27 in /etc/capi.conf but when I call my MSN 27 I get the following: Oct 18 13:51:38 NOTICE[15395]: pbx.c:1688 pbx_extension_helper: Cannot find extension context 'capi-in' |
14:13.16 | KriS83 | Oct 18 13:51:38 ERROR[15395]: chan_capi.c:2012 start_pbx_on_match: ISDN1: did not find exten for '27', ignoring call. |
14:13.48 | IronHelix | kris- you have capi setup right, but maybe not extensions.conf |
14:13.53 | Juggie | KriS83, you essentially answered your question |
14:13.57 | Juggie | create a context in extensions.conf |
14:14.02 | IronHelix | called capi-in |
14:14.06 | Juggie | [capi-in] |
14:14.09 | Juggie | inside it |
14:14.14 | Juggie | add your extensions |
14:14.18 | Juggie | eg |
14:14.23 | KriS83 | ok so If I renamed [demo] to [capi-in] it should work right? |
14:14.26 | Juggie | exten=> 27,1,Answer |
14:14.43 | Juggie | possibly... but your not going to learn anything that way |
14:15.02 | Juggie | try |
14:15.06 | Juggie | [capi-in] |
14:15.08 | IronHelix | kris- yup, or you could make a [capi-in], and then put exten =>27,1,Goto(demo,s,1) |
14:15.29 | KriS83 | Right I'll try that |
14:15.30 | Juggie | either way, your missing the context |
14:15.35 | Juggie | which is why its broken |
14:17.52 | KriS83 | works fine ;) |
14:18.32 | joelsolanki | Hello anybody used the http://voip-info.org/wiki/view/Asterisk+CDR+csv+mysql+import to import the text cdr from /var/log/asterisk/** |
14:19.25 | KriS83 | one more thing... When I create a IVR and add exten => 27,4,wait(3.0) for example because I want to have a Pause in the Text, in this pause the caller can not press any keys? is that correct? |
14:19.29 | *** join/#asterisk ikey1 (i=ikey@220.226.23.83) |
14:20.16 | IronHelix | as i recall it depends on what is happening |
14:20.33 | IronHelix | like if you do ringing() then wait() in my experience it doesnt hear dtmf |
14:21.04 | IronHelix | but if you just wait() it might |
14:21.06 | KriS83 | IronHelix, what I'm doing is just playing like for TEchnical support -> wait |
14:21.10 | KriS83 | press 1 |
14:21.16 | KriS83 | wait(3.0) |
14:21.26 | *** join/#asterisk jsiddall (n=jsiddall@206-248-134-222.dsl.teksavvy.com) |
14:21.29 | KriS83 | for sales -> wait(0.3) press 2 |
14:21.30 | tzafrir_laptop | joelsolanki, CSV is CSV, it is data. Filter it to your favorite format |
14:21.32 | IronHelix | 3 seconds is a lot of time on the phone |
14:21.41 | KriS83 | IronHelix, I know was only an example |
14:21.44 | IronHelix | ah |
14:21.52 | KriS83 | but I have like 1.5 |
14:22.10 | KriS83 | and when pressing 1 in this 1.5 second period it is not accepted |
14:22.14 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
14:22.19 | IronHelix | give it a shot, make something like background(hello) wait(30) and see if it will listen to dtmf |
14:22.42 | joelsolanki | tzafir_laptop: not able to understand u. I need to convert csv cdrs in to mysql and put into cdr tables. so that i can be able to do billing. |
14:22.51 | IronHelix | then maybe another way to do it (ugly hack) make a 1.5 second long blank gsm file and background() it |
14:23.02 | joelsolanki | Can u please give some suggestion on this. |
14:23.23 | IronHelix | ahhh |
14:23.25 | KriS83 | IronHelix, ok... |
14:23.27 | IronHelix | you want WaitExten() |
14:23.29 | IronHelix | there you go |
14:23.32 | IronHelix | that will wait and listen |
14:23.39 | KriS83 | Ok thx |
14:23.51 | KriS83 | Put that down on a piece of paper ;) |
14:23.57 | wmandra | ironhelix: closing all browser windows and trying to login again didn't work (couldn't log out, cause I was never logged in) I even tried to create a new account, got the registration email, clicked the link, entered a new password, no joy - back to the homepage without being able to log in. |
14:24.01 | *** join/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro) |
14:24.14 | IronHelix | try clearing all cookies? |
14:24.26 | ManxPower | asterisk-sounds has silent .gsm files of various lengths |
14:24.54 | KriS83 | IronHelix, now the only thing I'll be fighting with again is Passing my Calls from Line 1 (incoming) to Line 2 -> MSN 17 for example (via CAPI) |
14:25.12 | IronHelix | i suggested that cuz its done this a few times, but I use both IE and firefox. sometimes it wont log me in on IE, then i just logout in firefox and it works again |
14:25.27 | IronHelix | you just need to dial() |
14:25.35 | IronHelix | get the right channel and you should be set |
14:26.05 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
14:26.18 | KriS83 | IronHelix, I did that yesterday (with the old 1.0.6 install) and It connected to 17 but so short that it didn't even ring |
14:26.29 | IronHelix | any errors in the log? |
14:26.37 | KriS83 | I only saw it on my display ->Missed calls |
14:26.58 | KriS83 | This morning I found a capiHOLD and capiEXT function |
14:27.24 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
14:27.27 | KriS83 | this worked with one minus point... it always told me the number of the calling person before giving me the call on 17 ;) |
14:27.29 | KriS83 | No no error |
14:27.42 | KriS83 | But I'll try again now with the fresh install |
14:28.05 | Anthro | I would like a box with Asterisk, a place to plug in my home's phone wiring (RJ11, I assume), a place to plug in ethernet (RJ45), and a nice web interface to configure the more common aspects (SIP<->POTS provider information, mapping phone numbers to SIP addresses, etc.). Does such a thing exist? If not, would there be an interest in it once I develop it? |
14:28.10 | IronHelix | maybe then the dial plan isnt doing anything interesting with it? |
14:28.19 | IronHelix | anthro- it does |
14:28.32 | Anthro | IronHelix: URL? |
14:28.36 | IronHelix | you need an old pc, a copy of asterisk@home, and a digium TDMxx board |
14:28.44 | jsiddall | Is there any solution yet for the choppy audio when using ztdummy on the latest FC4 2.6.13 kernel? |
14:28.48 | wmandra | ironhelix: go figure, IE works fine - just can't login with FF |
14:28.57 | IronHelix | http://asteriskathome.sourceforge.net/ |
14:29.54 | IronHelix | http://www.voipsupply.com/index.php?cPath=99_103 FXO ports connect to phone lines, FXS ports connect to phones (like your existing phone handsets) |
14:29.59 | Anthro | IronHelix: I'm thinking of a turnkey solution that people like my parents would be able to install on their own. Something as easy to plug in and start using as a Linksys router. |
14:30.10 | synthetiq | i have an interesting situation...and am lookign for suggestions.... we have 3 cloned servers which we want to load balance.....the issue is, what if i need a two phones to make calls to each other when the two phones are registered on different machines? |
14:30.38 | synthetiq | with out going to out termination provider and back in |
14:30.43 | IronHelix | syn- use realtime? |
14:30.50 | IronHelix | if there is a single database it can store registrations |
14:30.53 | IronHelix | or replicate the db |
14:31.16 | IronHelix | anthro- its hard to make * linksys simple without ripping out most of it |
14:31.25 | synthetiq | we will be using realtime |
14:31.34 | Anthro | IronHelix: Ripping out or just setting up plausible defaults? |
14:31.40 | IronHelix | AAH certainly isnt linksys simple, you need to understand a bunch of stuff |
14:31.50 | IronHelix | both |
14:32.14 | IronHelix | syn then store the registrations in a db that all the servers get access to, so a registration will be valid on any server |
14:32.34 | trym | I cant seem to convert my .wav to .gsm and make it work in asterisk |
14:33.01 | jontow | trym; have you seen the wiki page on that topic? |
14:33.18 | trym | hmm on sox and gsm yes |
14:33.21 | trym | im using the same options |
14:33.54 | *** join/#asterisk yartelecom (n=no-email@62.33.183.215) |
14:34.08 | Anthro | IronHelix: Mm. Well, I intend to learn what I need to know. Hardware-wise, is a middle of the road VIA box with one PCI slot going to be sufficient for the task? I haven't bought any hardware, so I'm aiming for low cost, small size, and low power consumption/heat dissipation. |
14:34.53 | IronHelix | you'll need to both rip out much of the un needed stuff (few usa users will want capi, for example) and set defaults for the rest (voice mail etc) |
14:35.00 | trym | ah lol |
14:35.00 | trym | nm |
14:35.08 | IronHelix | via should be nice but check the incompatibility lists first. |
14:35.25 | IronHelix | 1 pci slot will accomidate up to 4 POTS lines, either fxo or fxs depending on how you load the card |
14:35.33 | *** join/#asterisk ayano_ (n=erik_lee@68-117-160-098.static.chtn.wv.charter.com) |
14:35.34 | IronHelix | check the voipsupply link i posted above |
14:35.40 | Anthro | IronHelix: Which incompatibility lists? |
14:35.47 | IronHelix | you can get a digium card with any combination of fxo or fxs modules |
14:35.47 | *** part/#asterisk ayano_ (n=erik_lee@68-117-160-098.static.chtn.wv.charter.com) |
14:36.17 | *** part/#asterisk case__ (n=case@mailhost.seeft.com) |
14:36.19 | IronHelix | there's a list on voip-info.org |
14:36.26 | IronHelix | mostly old Dell systems |
14:36.34 | IronHelix | as long as the card can get a free IRQ you should be fine |
14:36.47 | synthetiq | ironhelix how do i go about storign registraions realtime |
14:36.52 | IronHelix | digium cards generate tons of irq traffic |
14:37.43 | *** join/#asterisk l1nux (n=moi@lns-bzn-4-82-250-119-242.adsl.proxad.net) |
14:37.49 | l1nux | hi |
14:38.00 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-240-180.dsl.irvnca.pacbell.net) |
14:38.03 | synthetiq | dell machines lvoe to sue the same irqs for ...everything |
14:38.05 | Anthro | IronHelix: I see. What kind of power consumption/heat dissipation should I expect from a Digium card? They aren't like modern video cards that need their own fans or anything, right? |
14:38.05 | synthetiq | use |
14:38.35 | synthetiq | the digium cards will melt your heatsink :-P |
14:38.43 | *** join/#asterisk LostFrog (n=really@69-174-51-210.chvlva.adelphia.net) |
14:39.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:39.09 | Anthro | synthetiq: Eh? That's not so good. So a fanless system is out of the question? |
14:39.20 | l1nux | howto add "busy tone" if "SIP response 404 "Not Found" ? |
14:39.28 | IronHelix | hmmm, syn not sure about realtime registrations |
14:39.31 | IronHelix | i had thought it was possible |
14:39.33 | LostFrog | Is there anything short of the SNOM 4S ITSP that you can put in front of asterisk to support SSIP/SRTP? |
14:40.10 | LostFrog | Like, does the 4S Proxy support SSIP/SRTP? |
14:40.15 | spiekey | can i put Asterisk between my telco and my PBX with two isdn cards? is that possible? |
14:40.20 | IronHelix | sure |
14:40.25 | IronHelix | er |
14:40.34 | IronHelix | i dunno if an isdn card can 'serve' |
14:40.44 | IronHelix | but why would you want to do that? |
14:40.55 | MGSsancho | not enough bw |
14:40.57 | MGSsancho | i think |
14:41.52 | ManxPower | isdn bri or isdn pri? |
14:42.33 | *** join/#asterisk lehel (n=asd@82.79.20.17) |
14:42.57 | IronHelix | hehe |
14:43.05 | IronHelix | i think i was the only US user of ISDN |
14:43.14 | Anthro | IronHelix, synthetiq: Do the Digium cards really run that hot? |
14:43.25 | synthetiq | im joking man |
14:43.36 | iCEBrkr | IronHelix: Are you kidding? ISDN sold pretty good in NE. Ohio-- If you were a telco guy or geek. |
14:43.37 | ManxPower | IronHelix, you are 8-) |
14:43.38 | synthetiq | they dont run hot at all |
14:43.44 | Anthro | synthetiq: Ah, good. |
14:43.48 | MGSsancho | still works with the 8 switches on it |
14:44.00 | iCEBrkr | I used ISDN for about 5yrs. |
14:44.08 | IronHelix | im not anymore, had it for a few years before cable came out |
14:44.14 | MGSsancho | good times |
14:44.21 | IronHelix | back then it was the shit, connect in like 3 seconds and WOW 64 REAL KBPS! |
14:44.25 | lehel | hello |
14:44.30 | IronHelix | hi |
14:44.32 | MGSsancho | lol |
14:44.35 | MGSsancho | hi |
14:45.07 | iCEBrkr | IronHelix: I liked the fact that I could set a threshold on my Cisco ISDN router for more bandwidth and then have it drop the second line when I was done leeching |
14:45.15 | IronHelix | http://www.digium.com/index.php?menu=compatibility digium card compatibility notes |
14:45.21 | iCEBrkr | Or if a call came in, drop the second channel and ring my phone |
14:45.29 | IronHelix | yeah multilink was cool, i never got it to work tho |
14:45.36 | IronHelix | i think my isp didn't support it |
14:45.41 | iCEBrkr | Ahh |
14:45.51 | MGSsancho | AOL doesnt :( i called |
14:45.59 | iCEBrkr | I worked for an ISP at the time. I was kinda a test case. |
14:46.07 | MGSsancho | >_> |
14:46.30 | clennon | hi |
14:46.38 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
14:46.38 | iCEBrkr | My first ISDN terminal adaptor was this Hayes Modem. Sucked. |
14:46.39 | MGSsancho | now if only they mde T1 the same price as cabel or dsl |
14:46.40 | *** join/#asterisk galel (n=galel@63.245.93.138) |
14:46.45 | MGSsancho | or even $40 a month |
14:46.50 | IronHelix | i got a nastygram from my isp once complaining that i was dialed in too much |
14:46.56 | IronHelix | yeah cheap pri would rock |
14:46.57 | *** join/#asterisk galel (n=galel@63.245.93.138) |
14:47.04 | IronHelix | even if it only had 2-3 channels |
14:47.17 | *** join/#asterisk copantl (n=galel@63.245.93.138) |
14:47.24 | *** join/#asterisk copantl (n=galel@63.245.93.138) |
14:47.38 | Anthro | IronHelix: So Asterisk@home is a web administration interface, basically? |
14:47.43 | MGSsancho | i cant find drivers for my old hayes supermodem |
14:47.44 | MGSsancho | yup |
14:47.46 | IronHelix | no |
14:47.50 | iCEBrkr | Drivers? |
14:47.53 | MGSsancho | and it automates a lot of stuff |
14:47.55 | spiekey | ermm..sorry, had to grep something to eat... |
14:47.59 | iCEBrkr | Since when does a serial device require drivers :P |
14:48.00 | spiekey | i use bri ISDN |
14:48.01 | MGSsancho | lol |
14:48.01 | IronHelix | asterisk@home is a distribution of linux which includes a web interface, asterisk, and a bunch of other stuff |
14:48.17 | IronHelix | it includes Asterisk Management Portal (AMP) which is the web interface |
14:48.24 | spiekey | a normal german ISDN line with two "channels" |
14:48.25 | IronHelix | it is (relatively) easy to configure |
14:48.26 | Anthro | IronHelix: Ah, I see. I was confused. |
14:48.30 | iCEBrkr | Windows made modems more difficult to deal with/install |
14:48.40 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
14:48.43 | IronHelix | no worries |
14:48.47 | IronHelix | yeah i had hayes too |
14:48.50 | LostFrog | No help on SSIP/SRTP?? |
14:48.56 | IronHelix | courier i-modem isdn with v.34 |
14:49.01 | IronHelix | lost- its hard |
14:49.03 | MGSsancho | T_T |
14:49.08 | IronHelix | there is no real srtp support for * atm |
14:49.13 | iCEBrkr | Speaking of which, I should get this PRI hooked up to my asterisk box to start configuring it. |
14:49.13 | IronHelix | there are plenty of bounties for it |
14:49.16 | LostFrog | IronHelix, that I know.. |
14:49.28 | LostFrog | I would be willing to put another software packages in front of it. |
14:49.29 | IronHelix | iax2 has some sort of encryption routine but i dont know how well developed or secure it is |
14:49.35 | IronHelix | and thats really only for site to site |
14:49.52 | jake1932 | Anthro: if you're confused already - this is nothing |
14:49.57 | LostFrog | I want to secure my work-from-home users using snom phones. |
14:50.11 | IronHelix | heres another idea |
14:50.12 | LostFrog | Without out using IPsec to secure their whole networks. |
14:50.20 | IronHelix | get them all Linksys wrt54g's |
14:50.28 | IronHelix | load linux on them (sveasoft, openwrt, whatever) |
14:50.31 | MGSsancho | lol |
14:50.46 | IronHelix | set them as pptp clients, and instruct the users to plug only their work computer and phone into them |
14:50.57 | IronHelix | deny phone registrations from the Internet, only allow from behind pptp |
14:51.10 | IronHelix | clumsy, but works and lets them secure other stuff too |
14:51.25 | Anthro | I don't know a whole lot about phone systems (yet). The wiring in my house is designed to work with POTS, of course. Is there a wire somewhere in the house that I can plug into a port on a Digium card to provide phone service to all of the extensions in the house? |
14:51.31 | LostFrog | Hmm.. I thought about that. |
14:51.37 | IronHelix | anthro- anywhere |
14:51.44 | IronHelix | pots phone wiring is a loop |
14:51.57 | IronHelix | that means any port is electrically the same as any other port |
14:51.59 | LostFrog | Anthro, make sure you disconnect the connection from your phone service, first. |
14:52.06 | IronHelix | just make VERY VERY SURE that you unplug your telco |
14:52.18 | LostFrog | 75-90 Volts to a FXS port can suck. |
14:52.18 | IronHelix | because ringing voltage from the telco will possibly fry something expensive |
14:52.26 | iCEBrkr | 120v DC. |
14:52.30 | MGSsancho | 50V usealy |
14:52.34 | MGSsancho | in LA |
14:52.35 | LostFrog | no, ringing is AC. |
14:52.36 | iCEBrkr | 50? |
14:52.41 | Anthro | IronHelix: So I could have my Asterisk box in my office, plug it into the phone extension there, and provide service to the entire house? Sweet! |
14:52.46 | jake1932 | i got a shock from a ring before - not fun |
14:52.49 | MGSsancho | my lines are 48V and 49.2 at home |
14:52.52 | iCEBrkr | Unless they changed it.. It's DC |
14:53.01 | *** join/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
14:53.14 | IronHelix | anthro- yes, sort of |
14:53.14 | Anthro | IronHelix: At the moment I have no landline service. |
14:53.16 | scoates | anyone know of a mirror for app_conference? sourceforge's CVS isn't playing nice. |
14:53.19 | IronHelix | depends on what you are trying to do |
14:53.22 | LostFrog | ok, iCEBrkr, that explains why they have sine generators in FXS channel banks.. :) |
14:53.47 | Anthro | IronHelix: I basically want all calls to route through SIP, but I want the interface to be identical to POTS from my wife's perspective. |
14:53.58 | IronHelix | you dont need asterisk to do that |
14:54.05 | IronHelix | but you can use it |
14:54.13 | Anthro | IronHelix: What would you use, if not Asterisk? |
14:54.39 | IronHelix | the cheapest way- sign up with any voip provider, they will ship you a little box called an ata (analog telephony adapter). it has ethernet on one end and 1-2 fxs ports on the other |
14:54.41 | iCEBrkr | hrrm. |
14:54.50 | *** join/#asterisk litage (n=nick@203.220.55.70) |
14:54.50 | iCEBrkr | Grandpa's memory is going.. |
14:54.51 | iCEBrkr | <== |
14:55.02 | IronHelix | disconnect the telco interface, plug the ata's phone port into the wall |
14:55.02 | LostFrog | Stupid question: I shouldn't have any problems with assisted transfer with snoms and Asterisk, in regards to having control of the call if the remote extension doesn't answer, should I? |
14:55.11 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
14:55.15 | IronHelix | your wife won't know the difference, she will get dialtone, ringing, etc etc |
14:55.17 | nexis | i hate to spit it out in here, but is anyone else seeing problems with goiax today? |
14:55.50 | IronHelix | i suggest get a provider that supports BYOD (bring your own device) so if you decide to upgrade to * later you can |
14:55.59 | Anthro | IronHelix: Hrm, okay. That sounds pretty good. Is there a place with a good list of providers? |
14:56.04 | scoates | <PROTECTED> |
14:56.05 | LostFrog | Mmmm.. broadvoice. :) |
14:56.07 | IronHelix | by putting the SIP settings from the adapter straight into the asterisk box |
14:56.07 | Katty | hmm. |
14:56.12 | nexis | yea, vonage is evil, except they helped me get my hands on a grip of ATAs for about 2 bucks each. |
14:56.15 | kiko69 | Earthlink |
14:56.19 | IronHelix | i recommend broadvoice or quantumvoice if you are in the USA |
14:56.23 | nexis | nufone is my choice. |
14:56.35 | jake1932 | callvantage is good but expensive |
14:56.37 | IronHelix | quantumvoice especially, their webiste looks like ass but they have a good forum and always answer emails within a day |
14:56.41 | iCEBrkr | http://www.textfiles.com/phreak/black3.txt |
14:56.43 | IronHelix | cant say the same for broadvoice |
14:56.44 | synthetiq | i dont think asterisk registers phones via realtime |
14:56.50 | iCEBrkr | THOSE where the days |
14:56.55 | nexis | IronHelix, whats the pricing like on quantum? |
14:56.59 | *** join/#asterisk power1 (n=marktren@rndf-146-36-59.telkomadsl.co.za) |
14:57.05 | Anthro | IronHelix,nexis: I'll look into them, thanks! |
14:57.09 | IronHelix | not too bad, 25/mo unlimited usa as i recall |
14:57.09 | LostFrog | Why can't linksys put the the version numbers of their products on the packaging??? |
14:57.11 | Nugget | jason's "bbs documentary" dvd set is amazing. buy it. :) |
14:57.11 | heison | can anyone recommend a good TTS software that works well with Asterisk and supports Cantonese? |
14:57.41 | nexis | yea, nufone is 2c a min outgoing/incoming 800, and a 25 one time setup for a MI DID |
14:57.45 | IronHelix | lostfrog- i wish they'd stop with version numbers all together |
14:57.55 | nexis | but you get full control over your call with setcidnum ect. |
14:58.02 | kiko69 | Earthlink 14.95 for 500 minutes, or 24.95 unlimited |
14:58.07 | synthetiq | speakign of phreakign i know way to make free voip calls and avoid keepign registration on a server |
14:58.09 | synthetiq | =] |
14:58.16 | IronHelix | hehe |
14:58.25 | nexis | synthetiq, you mean free outgoing or incoming? |
14:58.26 | power1 | Hey all, Ive just got asterisk @ home 1.5 up and running. everything is working correctly except for voicemessages going out as emails, where do i tell asterisk to use my internal mail server as its outgoing smtp? |
14:58.26 | synthetiq | cdrs dont get written |
14:58.30 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) |
14:58.32 | synthetiq | outgoing |
14:59.44 | *** join/#asterisk hypa7ia (i=hypatia@wsip-24-234-241-145.lv.lv.cox.net) |
14:59.46 | nexis | well, i suppose you could put like dial([IAX2|SIP]/username:password@host/exten) if they support that. |
15:00.03 | nexis | and just never register |
15:00.10 | LostFrog | You don't have to register for outbound calls. |
15:00.21 | IronHelix | power1- use the webmin console and configure sendmail to use a smart relay host |
15:00.44 | IronHelix | register is just a 'hey im here send me my calls' for incoming |
15:00.46 | LostFrog | ok.. guess I will flash this brand-new WRT54G V4.0 |
15:00.57 | MGSsancho | lol |
15:01.02 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
15:01.02 | IronHelix | they're up to 4.0 now? |
15:01.03 | IronHelix | gah |
15:01.11 | LostFrog | up to 5.0, I believe. |
15:01.11 | MGSsancho | bad? |
15:01.13 | IronHelix | why cant they stick with ONE design... |
15:01.19 | MGSsancho | lol |
15:01.19 | IronHelix | esp with network cards |
15:01.30 | power1 | IronHelix, thanks....could you point me in the direction of the correct sendmail conf file? |
15:01.31 | LostFrog | USB wireless is the worst. |
15:01.38 | IronHelix | if i know i have a linksys lne100tx i shouldnt have to open my pc and take the card out to figure out which card i have |
15:01.40 | LostFrog | One card will work and then the next wont. |
15:01.53 | IronHelix | power i mean use the webmin thing to configure sendmail |
15:01.54 | Anthro | Okay, so this might be naive, but... IIRC, iChat AV uses SIP. Is there a way to set things up so an iChat AV user can attempt to call me and have an Asterisk server treat it as any other SIP call? |
15:02.03 | IronHelix | it should have a sendmail setup thing as i recall |
15:02.11 | IronHelix | anthro- not naieve at all |
15:02.15 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
15:02.25 | LostFrog | I guess I should download the linksys firmware first.. :) |
15:03.13 | power1 | IronHelix, webmin thing : does a default @ home iso install run webmin on port 10000 ? |
15:03.15 | MGSsancho | ya |
15:03.17 | *** join/#asterisk Meaty-Wrk (n=cp_simbu@office.abi.ca) |
15:03.55 | mmlj4 | power1: sendmail is not trivial to configure... your best bet is to use webmin if at all possible |
15:03.56 | IronHelix | anthro- turns out ichat uses some sip derivative, which is not actual sip and thus not compatible with asterisk |
15:04.09 | Anthro | IronHelix: Oh. How disappointing. |
15:04.27 | IronHelix | yeah :( |
15:05.07 | Anthro | Okay, I'll have more questions (naive or otherwise) later. For now I have to get back to work. Thanks for all your help! |
15:05.13 | *** part/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro) |
15:05.14 | power1 | IronHelix, oh ok,,,thanks..is webmin installed by default on an asterisk @ home distro? |
15:05.20 | IronHelix | dont think so |
15:05.25 | IronHelix | http://www.voip-info.org/wiki/view/Asterisk@home+Handbook+Wiki chapter 6.3 |
15:05.28 | IronHelix | has a command to grab it |
15:05.31 | IronHelix | i had thought it was |
15:05.37 | IronHelix | no problem anthro |
15:06.06 | mmlj4 | power1: you can determine what port webmin runs on by using the netstat command from the console or a login prompt: "netstat -an | grep LISTEN" will tell you what ports are open, and you can deduce from that |
15:06.24 | power1 | mmlj4, thanks..will have a look |
15:09.22 | LostFrog | why not 'netstat -tan'? |
15:09.27 | LostFrog | webmin is TCP |
15:09.38 | *** join/#asterisk CBTCWwW (n=CBTCWwW@165.154.121.241) |
15:09.39 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
15:10.24 | nexis | woot, i now have full control over my X10 from my phone |
15:10.47 | Ahrimanes | bluetooth phone and xten eyebeam? |
15:11.42 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
15:12.12 | nexis | no, X10 home automation |
15:12.26 | Ahrimanes | ah |
15:12.27 | Ahrimanes | crap |
15:12.28 | Ahrimanes | hehe |
15:12.53 | nexis | i can pick up a phone, turn on and off lights ect |
15:13.01 | nexis | has motion and heat detectors |
15:13.31 | Ahrimanes | how much did you spend for that? |
15:13.31 | IronHelix | hehe |
15:13.34 | IronHelix | those are fun |
15:13.49 | IronHelix | did you use the x10 dtmf module or something more exotic? |
15:13.58 | nexis | IronHelix, X10 usb module |
15:14.11 | IronHelix | ahh |
15:14.11 | Katty | what's for lunch? |
15:14.20 | IronHelix | pancakes |
15:14.26 | nexis | i got a lot of it from rat shack on a sale |
15:14.56 | nexis | all in all, my asterisk box, all the modules in the system, the 2 voip phones, and the stack of linksys PAP2s, i have about 300 bucks tied up. |
15:15.31 | Ahrimanes | nice |
15:15.54 | Katty | any /other/ lunch suggestions? |
15:16.12 | trym | When I reach agent login (AgentCallbackLogin), I enter my agent number with #, but nothing happens.. it just hansg up |
15:16.13 | nexis | Katty, chineese sounds good. |
15:16.14 | IronHelix | pancakes |
15:16.19 | IronHelix | :) |
15:16.25 | nexis | ohh, chineese pancakes. |
15:16.28 | Nugget | I've got leftover manicotti. |
15:16.31 | Nugget | come on by. :) |
15:16.41 | Katty | Nugget: it has cheese in it. |
15:16.46 | Nugget | feh |
15:16.53 | nomazda | what's wrong w/ cheese? |
15:16.54 | jovu | anyone got any suggestions as to why my cisco 7960 wouldn`t pass dtmf tones through correctly? |
15:17.15 | IronHelix | trym- your asterisk box doesnt like you. buy it a Digium board to win its good graces back |
15:17.24 | IronHelix | /or/ anything on the console? |
15:17.39 | IronHelix | katty- yeah get scallion pancakes |
15:17.42 | IronHelix | or lo mein |
15:17.43 | trym | ill put it on a pasteshiznit.. gimme a sec |
15:17.44 | IronHelix | those are good |
15:18.13 | Katty | IronHelix: i'm not interested in pancakes or lo mein for lunch. thanks anyway. |
15:18.13 | nexis | jovu, try dtmfmode=rfc2833 |
15:18.23 | trym | http://pastebot.nd.edu/235 |
15:18.27 | IronHelix | oh well |
15:18.30 | IronHelix | was worth a shot |
15:18.35 | Katty | nomazda: i do not eat cheese. |
15:19.02 | znoG | does anyone know how I could find out the admin password to a sipura unit? |
15:19.03 | nexis | jovu, http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
15:20.58 | IronHelix | bbiab |
15:21.08 | nexis | now, if i could only figure out how to build a robot to bring me a beer from the fridge |
15:21.16 | jovu | nexis, i tried rfc2833, didnt work.. is there anything i need to set on the phone too? |
15:21.31 | LostFrog | nexis: get married. |
15:21.45 | mcf3782 | cheaper to just get the beer yourself. |
15:22.12 | nexis | i think i may train a dog to do it. |
15:22.30 | LostFrog | But, then there will be fang holes in it. |
15:22.47 | LostFrog | Not to mention the pool of beer by the fridge. |
15:23.06 | Dr_Ray | why not just install a beer keg at your cpu desk |
15:23.24 | *** join/#asterisk marc324 (n=marc3234@206-248-159-56.dsl.teksavvy.com) |
15:23.29 | LostFrog | Yeah.. and use the cooling unit to cool your cpu/mb too. |
15:23.40 | lunk | Dr_Ray: almost a good idea, except you don't want ice melting everywhere |
15:24.17 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
15:24.18 | LostFrog | You would have to drink a lot of beer to justify a keg. |
15:24.55 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
15:25.58 | Starmaker | i'm having a problem with my timer |
15:25.59 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
15:26.04 | LostFrog | I hate to say it, but I need another windows machine. |
15:26.17 | Starmaker | I don't have a zaptel-card, so I should use ztdummy, right? |
15:26.21 | nexis | well, i could put in a keg-o-rator here at the desk |
15:26.27 | LostFrog | Hmm.. then again.. maybe I can use vmware. |
15:26.37 | *** join/#asterisk brookshire (n=pfffft@gateway.digium.com) |
15:26.49 | nexis | LostFrog, i drink a 30 of long necks a week, at least |
15:27.27 | synthetiq | drink beer to ahve stomach lookign liek a keg |
15:27.27 | nomazda | must have a gut like a trampoline |
15:27.31 | trym | IronHelix: any ideas ? |
15:27.44 | Starmaker | the problem is, with ztdummy it's even worse than without. without it i can't use MP3Player(), because it plays REALLY slow, but with ztdummy, it doesn't fix the MP3Player()-problem, and it screws up my voicemail |
15:27.49 | nexis | i have a bit of a ponch, but not a geer gut. |
15:27.59 | *** join/#asterisk ursuspacificus (n=paul@wsip-24-249-27-197.ri.ri.cox.net) |
15:27.59 | LostFrog | What is the small distribution of windows 98 that you can run from CD called? |
15:28.20 | nexis | im not aware of a livecd windows. |
15:28.27 | jsiddall | <PROTECTED> |
15:28.32 | Starmaker | 2.6.13.4 |
15:28.34 | LostFrog | TNot officially, nexis, but there is. |
15:28.51 | LostFrog | hmm.. 2.6 doesn't even need UHCI for ztdummy, supposedly. |
15:28.53 | trym | http://pastebot.nd.edu/235 <-- trying to login as an agent.. I enter my agent ID after the prompt and hit #, but after a few seconds asterisk just hangs up |
15:28.54 | Starmaker | I'm using a 250 Hz kernel |
15:29.04 | jsiddall | Yup, same here, same problem. Don't know when it appeared, no one seems to have a solution short of buying a digium card :( |
15:29.13 | *** join/#asterisk darwin35 (n=darwin35@208.139.193.178) |
15:29.23 | *** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
15:29.26 | jsiddall | How do you know you have a 250 Hz kernel? |
15:29.27 | RoyK | does zaptel work with 2.6.13 yet? |
15:29.31 | *** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
15:29.33 | nexis | im runnin 2.4.31 on my asterisk box. |
15:29.45 | Starmaker | jsiddall, because I compiled it with that setting? |
15:30.01 | szer | bye al |
15:30.02 | szer | l |
15:30.06 | nexis | RoyK, it only runs on the asterisk box, the rest are running 2.6 or freebsd |
15:30.19 | ursuspacificus | Hello, all... anyone have any experience getting a TDM22B working on WhiteBox Enterprise Linux 4 (Full install, plain vanilla, stock SMP kernel on P4 w/HT)? |
15:30.32 | RoyK | nexis: why 2.4? |
15:30.50 | jsiddall | Hmmm... I'm using RPM'd kernel (yeah, yeah) but I'm betting the ztdummy expects 1024 Hz from the kernel |
15:30.54 | RoyK | why oh why..... |
15:30.59 | Starmaker | jsiddall, 1000 hz |
15:31.02 | Starmaker | not 1024 |
15:31.17 | nexis | RoyK, cas before it was a asterisk box, it was my router, sat in the closet for a while, then came out of retirement for a asterisk box |
15:31.24 | Starmaker | hmm, i could 1. try ztrtc |
15:31.25 | jsiddall | AFAIK it has to be a power of 2, so ztdummy drops a few hz to compensate |
15:31.35 | marc324 | not everything is in power of two. |
15:31.47 | Starmaker | or 2. recompile my kernel |
15:31.52 | RoyK | does zaptel ztdummy from -stable support the 2.6 timer? |
15:31.56 | jsiddall | no, but the kernel RTC does apparently |
15:32.12 | RoyK | jsiddall: was that to me? |
15:32.40 | jsiddall | No, sorry, that was to marc324. I'm using the ztdummy from stable |
15:33.00 | RoyK | jsiddall: with the kernel timer or usb? |
15:33.02 | darwin35 | use rtc on 2.6 not ztdummy |
15:33.24 | nexis | wonder if you can pull timer data off the intel v92 card |
15:33.25 | jsiddall | kernel. I don't think the 2.6 ztdummy uses USB at all |
15:33.36 | RoyK | darwin35: rtc instead of ztdummy??? |
15:33.48 | RoyK | darwin35: you mean ztdummy using rtc..... |
15:33.51 | jsiddall | Where is ztrtc now? |
15:33.56 | RoyK | er |
15:33.58 | RoyK | ztrtc? |
15:33.58 | Starmaker | ztdummy for 2.6 does not use USB |
15:34.42 | jsiddall | there was a ztrtc, but I thought that was like 2.6 ztdummy for 2.4 (ie: used kernel timer instead of usb) |
15:34.42 | SwK[Work] | ztdummy uses RTC on 2.6 |
15:34.43 | *** join/#asterisk facecake (n=facecake@81.29.64.26) |
15:35.01 | jsiddall | Yeah, but if doesn't work in 2.6.13 anyway :( |
15:35.02 | nexis | http://www.voip-info.org/wiki-Asterisk+timer |
15:35.29 | RoyK | jsiddall: use 2.6.12, then |
15:35.48 | *** join/#asterisk hellagony (n=egutierr@irc.americatelnet.com.pe) |
15:35.57 | jsiddall | That might be the path of least resistance. Aside from compiling the kernel, how can you tell what frequency the kernel RTC runs at? |
15:36.00 | facecake | Hi, just wondering if anyones experienced any issues with the te110p's where they refuse to notice that the nte is pluged in however with a loopback cable it gives the green light |
15:36.00 | nexis | Zaprtc will not work on SMP systems |
15:36.09 | RoyK | nexis: que? |
15:36.16 | RoyK | nexis: why? |
15:36.50 | ursuspacificus | The problem I'm running into, after having followed the "quick install guide" is that when I modprobe wctdm, I get a large pile of newlines, then "line 0: Unable to open master device '/dev/zap/ctl'". When I look in /dev, I see zap1, zap2, zap3, zap4, zapchannel, zapctl, zappseudo and zaptimer... but no /dev/zap/... and, as you might expect, nothing in the nonexistent /dev/zap/ |
15:37.05 | nexis | RoyK, from the looks of it, SMP systems are locking the RTC for SMP timing |
15:37.20 | *** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) |
15:37.43 | nexis | ursuspacificus, which distro? |
15:37.58 | *** join/#asterisk ayano_ (n=erik_lee@68-117-160-098.static.chtn.wv.charter.com) |
15:38.13 | ursuspacificus | White Box Enterprise Linux 4 full install, plain vanilla with SMP kernel |
15:38.15 | nexis | cas it seems the files that it expects to see in /dev/zap are being made in /dev/ |
15:38.33 | *** part/#asterisk ayano_ (n=erik_lee@68-117-160-098.static.chtn.wv.charter.com) |
15:38.36 | n3u7 | is there anyway to compile asterisk without libnewt? |
15:38.53 | nexis | also, are you running stable, or cvs head, are you using the same version of zaptel drivers? |
15:39.22 | ursuspacificus | nexis: I did a CVS checkout, as per the "quick install guide" |
15:39.28 | jsiddall | Make sure you read README.udev if your system has udev |
15:39.32 | Ariel_ | ursuspacificus, use udev |
15:40.09 | *** join/#asterisk RussC (n=face@216.157.205.211) |
15:40.13 | jsiddall | Hmmm, tried cat /proc/driver/rtc |
15:40.15 | nexis | yea, udev will screw with ya if its not setup correctly. |
15:40.27 | jsiddall | Noticed periodic_freq : 1024 |
15:40.41 | Ariel_ | ursuspacificus, http://www.voip-info.org/wiki/view/Asterisk+OS+Platforms pick the closes which would be CentOS 4.1 |
15:40.47 | RussC | Hello I am getting an error pbx.c:1331 pbx_extension_helper: Cannot find extension context 'default' when ever I try to connect to a sip on an internal network |
15:40.50 | jsiddall | Unfortunately that should be the right setting |
15:40.57 | RussC | Any help would be great |
15:41.22 | *** join/#asterisk razu_ (n=razu@ip58.cab60.mus.starman.ee) |
15:41.30 | nexis | RussC, you are missing a context. |
15:41.32 | Ariel_ | RussC, it's saying you don't have any context default. Check your settings |
15:41.36 | n3u7 | nexis:I just had that problem |
15:42.17 | n3u7 | used sym links |
15:42.25 | n3u7 | to dev/zapctl |
15:42.48 | RussC | Ariel_: context defaults in wich confs extensions? |
15:43.02 | RussC | or sip those are the only two I have edited |
15:43.05 | nexis | n3u7, thats one way of doing it, untill you restart. |
15:43.09 | Ariel_ | RussC, your settings might not have the proper context setup |
15:43.35 | RussC | Ariel_: thank you |
15:43.45 | iCEBrkr | I'm going to assume the zaptel stuff can't see my TP100 |
15:43.45 | iCEBrkr | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
15:43.47 | iCEBrkr | ?? |
15:43.54 | Ariel_ | RussC, you can see example of context in /usr/src/asterisk/configs/extensions.conf.sample |
15:44.07 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
15:44.39 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
15:44.55 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
15:46.19 | jlewis | anyone ever seen asterisk "slow down" at which point after calls are connected, it's a few secs before you can hear each other...and you can watch dialplan logic execute relatively slowly when normally it would fly by in the CLI? |
15:46.20 | n3u7 | libnewt is not cooperating |
15:46.53 | iCEBrkr | jlewis: check the load on the machine? |
15:47.02 | eKo1 | What could be causing the primary d-channel to go up and down all the time. WTF?! |
15:47.04 | jlewis | we've had this problem on and off (using various CVS stable snapshots...usually only happens after * has been running for a "long" time |
15:47.10 | jlewis | load is negligible |
15:47.29 | nexis | jlewis, also, check your ram. |
15:47.41 | iCEBrkr | free -m |
15:47.48 | jlewis | dual HT 2.8ghz xeon...97+% idle |
15:47.52 | ursuspacificus | Ariel_: Thanks for the link. I got everything compiled and all... It looks like a udev config issue.... I will have to play with it a bit. |
15:48.11 | jlewis | 800mb of buffers+cache |
15:48.16 | nexis | man, thats a lil over kill for a asterisk box |
15:48.50 | jlewis | its got 1gb ram...it's the PBX for around 100 users and has 1 PRI to the PSTN |
15:48.55 | iCEBrkr | jlewis: naaaa, what's it say for mem: free? |
15:49.11 | jlewis | <PROTECTED> |
15:49.11 | jlewis | Mem: 1025392 993272 32120 0 84364 723640 |
15:49.11 | jlewis | -/+ buffers/cache: 185268 840124 |
15:49.11 | jlewis | Swap: 2104496 20184 2084312 |
15:49.57 | iCEBrkr | That's not bad |
15:50.04 | jlewis | when this happens, restarting * always clears it up |
15:50.16 | nexis | a reload fix it? |
15:50.39 | iCEBrkr | use top and sort by mem usage |
15:50.48 | mutilator | w00t |
15:50.55 | jlewis | reload in cli?...I don't remember if we tried that before |
15:50.58 | n3u7 | I'm getting a load of error messages on install to the extent: |
15:51.02 | mutilator | i get a whole week! |
15:51.16 | ursuspacificus | Ariel_: One thing that seems to be eluding me is the names of the devices that wctdm expects to see... I mean, I got /zap/ctl figured out from the modprobe error message... but does the "zap" at the beginning of all the devices I'm seeing correspond to the directory they should be in in /dev and should be stripped off on the actual device names in the directory? |
15:51.30 | *** join/#asterisk LostFrog (n=really@69-174-51-210.chvlva.adelphia.net) |
15:51.35 | mutilator | order the server today and build next week when it comes in, then build asterisk on it, and then hook it to 4 24 port channel banks |
15:51.41 | mutilator | and figure out some dialplan |
15:51.52 | mutilator | then cross my fingers and hope nothing goes wrong |
15:52.06 | mutilator | i love doing stuff at the last minute |
15:52.06 | LostFrog | That sounds like my MO |
15:52.08 | jsiddall | ursuspacificus: README.udev should tell you what to config |
15:52.27 | mutilator | it goes online to about 50 users on the 31st |
15:52.27 | LostFrog | yeah.. magneto-optical. :) |
15:52.33 | mutilator | along with sdsl |
15:52.55 | syle | wish i could get sdsl |
15:53.16 | mutilator | move here |
15:53.17 | mutilator | i'll give ya soe |
15:53.21 | mutilator | some |
15:53.27 | LostFrog | wish I could afford a DS3 in my house. :) |
15:53.32 | LostFrog | Wishing doesn't really work. |
15:53.34 | mutilator | win the lotto |
15:53.44 | mutilator | i'de do it if i did |
15:53.48 | n3u7 | libnet.a(scrollbar.o)(.text+0x237):scrollbar.c indefined reference to 'SLsmg_write_chr' |
15:53.48 | n3u7 | *libnewt |
15:53.49 | n3u7 | hoe do I comppile without libnewt for SuSE9.3 |
15:53.51 | n3u7 | ther is no working patch for SuSE |
15:53.53 | mutilator | then spend a crap load on a few servers |
15:53.58 | mutilator | and just host game servers or some crap |
15:54.02 | nexis | LostFrog, only a DS3? |
15:54.09 | *** join/#asterisk gaspiz (i=gaspi@86.34.6.164) |
15:54.16 | gaspiz | hi there |
15:54.22 | iCEBrkr | What the hell is libnewt? |
15:54.29 | nexis | mutilator, whats your budget like? |
15:54.29 | LostFrog | Only??? |
15:54.30 | iCEBrkr | and what's it gotta do with Asterisk? :P |
15:54.46 | n3u7 | libnewt are libraries |
15:54.48 | mutilator | budget? |
15:54.49 | mutilator | um |
15:54.58 | nexis | LostFrog, if your going for overkill, bust out a OC-786 |
15:55.00 | iCEBrkr | n3u7: ok, I got that much, but what's it for? |
15:55.02 | gaspiz | can someone help me with a realtime question? |
15:55.06 | mutilator | no budjet |
15:55.17 | mutilator | budget* |
15:55.18 | syle | if i won the lotto i think i would play xbox fulltime |
15:55.19 | nexis | call center, or office? |
15:55.21 | syle | :) |
15:55.25 | mutilator | a town |
15:55.30 | ful|work | i got a 100mbits line for a voip provider |
15:55.44 | syle | how much? |
15:55.45 | nexis | eww, so billing and everything. |
15:55.50 | mutilator | oh yea |
15:56.00 | LostFrog | I don't think I could saturate an DS-3, let alone anything higher. |
15:56.01 | nexis | first expernece with asterisk? |
15:56.04 | mutilator | no |
15:56.07 | ursuspacificus | Ariel_: Aaaahhh! (1,000,000,000 candlepower light shines from behind me, through a foggy mist as it all becomes clear to me). Many thanks, Ariel_! |
15:56.34 | syle | 100 megabit line, damn nice :) |
15:56.36 | r0d3nt | libnewt is console graphic libraries for zttool, astman and several other utilities that get compiled with asterisk and zaptel if you have newt libararies available @ compilation, |
15:56.48 | syle | in canada costs us about 2k for a 10 megabit line unmetered |
15:56.56 | mutilator | been using it for sip -> sip -> pstn for a year or so |
15:56.58 | r0d3nt | sorta like ncurses, but extra special crappy from RedHat/Fedora. |
15:57.17 | mutilator | installed a channel bank with a single slot at a local college a few months ago |
15:57.20 | trym | How can I make AgentCallbacklogin use the agent id as the extension to execute ? |
15:57.23 | nexis | have the FXS banks yet? |
15:57.31 | mutilator | no |
15:57.38 | syle | rhino? |
15:57.42 | mutilator | adtran |
15:58.01 | nexis | cas if you have to order the server today, i was gonna sugest find a p3 500 or something, put asterisk on it, and start working on your dialplan now |
15:58.04 | syle | i'd stick with rhino |
15:58.18 | nexis | get a week extra. |
15:58.24 | mutilator | extra week |
15:58.31 | mutilator | we've already got customers signed up |
15:58.37 | mutilator | and ready to go online the 31st |
15:58.45 | syle | SER+asterisk? |
15:58.52 | mutilator | thats our "turn on date" we've been advertising |
15:59.02 | nexis | yea, but you can start working the dial plan now, not when you get the server built |
15:59.08 | LostFrog | He means, you can have the week that you would have spent waiting on the server to work on your dialplan. |
15:59.19 | n3u7 | iCEBrkr: :it's a development library for text user inferaces |
15:59.32 | syle | i spent 3 months solid on my dialplans |
15:59.36 | mutilator | i dun have an extra machine |
15:59.43 | mutilator | i could vmware one i guess |
15:59.46 | LostFrog | Go to walmart.. :) |
15:59.52 | nexis | your telling me you cant even scrounge up a p2 350? |
16:00.02 | mutilator | no |
16:00.09 | nexis | what kind of geek are you? |
16:00.18 | syle | i got a 800 mhz machine just lieing in parts on my floor |
16:00.21 | mutilator | they wouldn't out out and get somethin like that cause i needed it |
16:00.25 | syle | just needs a harddrive |
16:00.38 | mutilator | go out |
16:00.43 | mutilator | * |
16:00.50 | nexis | syle, i have 10 p2 350 HP Vectra's in my closet, unused |
16:00.57 | mutilator | i live in my car right now nexis |
16:01.04 | mutilator | it's full of everythin o own |
16:01.06 | mutilator | i own |
16:01.15 | nexis | ouch |
16:01.17 | syle | girlfriend kick you out? |
16:01.24 | mutilator | yep |
16:01.26 | mutilator | sure did |
16:01.34 | gaspiz | how can I jump to another realtime context from a realtime context? |
16:01.35 | eKo1 | chan_zap.c:1938 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! <--- wtf?! why am I getting this all of a sudden |
16:01.36 | LostFrog | Steal her computer. :) |
16:01.47 | nexis | i take it your working at a telco or a ISP right? |
16:01.56 | mutilator | isp yea |
16:02.09 | nexis | they should have a extra box you can steal. |
16:02.16 | mutilator | nope |
16:02.26 | syle | he needs to rent an apartment before he can do anything lol |
16:02.32 | nexis | yea, that too |
16:02.42 | nexis | time to call CDW and order a fully built server for next day air. |
16:02.46 | mutilator | i don't make enough to rent |
16:02.55 | LostFrog | Geez.. where are you working, mutilator, bumfcuk?? |
16:02.56 | syle | then switch jobs |
16:03.02 | mutilator | i've tried |
16:03.18 | mutilator | only gotten 1 interview in like 2 years of trying |
16:03.26 | mutilator | phone interview at that |
16:03.30 | nexis | man, i make paper, and i have no money problems, you would think that a ISP would pay better then a factory. |
16:03.39 | mutilator | nope |
16:03.46 | syle | you don;t know how to get a job then hehe |
16:03.55 | mutilator | syle: obviously |
16:04.30 | syle | personally if i wanted a job, i just walked into a place i wanted to work, asked for owner, and just had a chat with him |
16:04.43 | syle | sometimes you get lucky |
16:05.27 | iCEBrkr | I make good cash, and I have all sorts of money problems :P |
16:05.56 | syle | rent out a bedroom hehe |
16:05.57 | iCEBrkr | mutilator: Sell yourself like a whore, man.. |
16:06.15 | mutilator | i;'ve tried that too syle |
16:06.27 | mutilator | this is the only place i landed a job doing that |
16:06.37 | mutilator | and i'm like their genius slave labor |
16:06.47 | syle | can you program in php? |
16:06.54 | mutilator | i do like 100x more than i was hired to do |
16:06.55 | mutilator | yea |
16:06.56 | syle | you can get jobs off irc if you can |
16:07.17 | syle | camp in #php on efnet |
16:07.26 | nexis | heh |
16:07.31 | iCEBrkr | mutilator: Monster.com works |
16:07.38 | syle | in past i;ve done about 10k in contract work just from people in that channel |
16:07.41 | syle | also from diff sites |
16:07.43 | mutilator | iCEBrkr: not really |
16:07.45 | nexis | thats soo true syle, people come in there saying ill pay 300 bucks for somone to code this up. |
16:07.53 | mutilator | thats where i got my 1 phone interview |
16:07.54 | iCEBrkr | mutilator: Not really? I've gotten 2 jobs off it. |
16:07.59 | mutilator | out of i think 800 apps i put it on there |
16:08.05 | iCEBrkr | mutilator: and I dunno how many face to face intervies. |
16:08.22 | mutilator | u probly have a degree too |
16:08.29 | iCEBrkr | mutilator: Nope |
16:08.35 | mutilator | well i dunno then |
16:08.37 | syle | well my last job i got from #php to, they offered me fulltime 60k a year position |
16:08.42 | syle | i worked there for 2 years |
16:08.46 | iCEBrkr | mutilator: apt-get install confidence |
16:08.49 | mutilator | i apply for every job from help desk to network engineer |
16:08.57 | nexis | i have no degree, got my high school deploma last year, and i have no problems gettin jobs. |
16:09.00 | LostFrog | That works, iCEBrkr?? |
16:09.07 | mutilator | iCEBrkr eh.. |
16:09.10 | iCEBrkr | LostFrog: It was worth a try :P |
16:09.12 | LostFrog | I've been trying rpm -i confidence.rpm |
16:09.21 | LostFrog | It keeps saying 'file not found' |
16:09.27 | ikey1 | rt |
16:09.30 | nexis | iCEBrkr, i would be happier with apt-get install girlfriend |
16:09.40 | iCEBrkr | mutilator: Seriously, you gotta go in there and tell them what you know and be confident in what you're telling them. |
16:09.44 | eKo1 | hmm...looks like a full reboot did the trick |
16:09.46 | *** join/#asterisk philm (n=Phil@73.236.204.68.cfl.res.rr.com) |
16:09.47 | iCEBrkr | nexis: That'd be sweet! |
16:09.48 | IronHelix | you need to understand that package first, do man woman |
16:09.50 | eKo1 | wtf man |
16:09.59 | iCEBrkr | IronHelix: lol |
16:10.04 | LostFrog | man man should be illegal. :) |
16:10.04 | mutilator | yeh |
16:10.09 | mutilator | if i got the interviews i could be |
16:10.11 | mutilator | but i don't get em |
16:10.37 | nexis | or |
16:10.53 | nexis | apt-get remove --purge dishes |
16:11.01 | iCEBrkr | mutilator: then you need to go over your resume format. |
16:11.06 | syle | btw reason i left my job at 60k a year was dude woudl never give me a raise, so if you own a company don;t forget to do that :) |
16:11.19 | mutilator | iCEBrkr: ya i've been thinking of saving up and blowing cash on the monster.com resume writer ppl |
16:11.44 | nexis | mutilator, no need for that, you have a local community college right? |
16:11.44 | iCEBrkr | mutilator: ehhh, I wouldn't go that far. Just look around and find a nice looking for mat. |
16:11.49 | LostFrog | I don't make shit, but I'm happy.. (except for the bills) |
16:11.57 | mutilator | i make $8.50/hr |
16:11.58 | mutilator | :) |
16:12.08 | LostFrog | I make $27.5k |
16:12.08 | nexis | damn, where in the hell do you live? |
16:12.13 | iCEBrkr | Yeah, really. You could probably get a nice looking resume from a college student for $20 |
16:12.18 | LostFrog | You can make $10 at McDonalds |
16:12.24 | syle | i make 0.00/hr right now, still trying to get my site up |
16:12.26 | mutilator | ya LostFrog |
16:12.33 | mutilator | i could go do a factory job for like $12 |
16:12.37 | mutilator | and get insurance and shit too |
16:12.38 | Ariel_ | 10 dollars at McDonalds??? |
16:12.48 | mutilator | but that isn't what i wanna do |
16:12.51 | LostFrog | Here you can, areski. |
16:12.51 | philm | Does anyone know a site with information on hardware hacking for voip, like connecting a fxs port to a loud speaker or microphone? |
16:12.51 | mutilator | so i stick around here |
16:12.54 | LostFrog | Ariel, even. |
16:12.57 | nexis | well, i was saying, the community colleges usualy offer free classes on how to write a resume |
16:13.09 | syle | 10 at mcdicks? that must be california |
16:13.11 | nexis | philm, why, use a sound card |
16:13.20 | nexis | chan_oss or chan_alsa |
16:13.20 | LostFrog | Northern Virginia |
16:13.21 | mutilator | i live in northern michigan |
16:13.29 | nexis | damn dude |
16:13.35 | syle | isn;t minimum wage there like 10-11 bucks an hour now |
16:13.36 | philm | I want to havee allot of them. |
16:13.46 | nexis | no, min wage is still 5.25 |
16:13.51 | LostFrog | $7.75, I think. |
16:13.55 | mutilator | federal |
16:14.01 | iCEBrkr | It's more than 5.25 anymore.. |
16:14.04 | mutilator | state to state is diff |
16:14.04 | nexis | i havent made min wage sense i was like 16 |
16:14.17 | LostFrog | I never made minimum wage. |
16:14.31 | mutilator | soon as hardwire gets here |
16:14.35 | *** join/#asterisk xunil (i=xunil@66.194.40.30) |
16:14.37 | mutilator | i'm goin to take up his job offer |
16:14.41 | mutilator | see if he's still lookin |
16:14.44 | nexis | ehh, i washed dishes in a kitchen when i was 14 for min wage. |
16:14.49 | syle | i don;t even want to remember jobs i had at 16, picking weeds out of strawberry farms, and picking up horse shit lol |
16:14.52 | *** join/#asterisk gaggaman (n=leo@host-82-135-28-39.customer.m-online.net) |
16:14.59 | mutilator | i'll just be on a frigid island off alaska |
16:15.19 | LostFrog | ewww.. $5.15 |
16:15.21 | nexis | mutilator, dude, alaska is killer |
16:15.32 | nexis | its like .7:1 ratio males to females. |
16:15.35 | iCEBrkr | LostFrog: $5.15 > $.0.00 |
16:15.37 | mutilator | ya |
16:15.43 | mutilator | but the femals are all fat samoans |
16:16.03 | syle | males to females or females to males? |
16:16.21 | LostFrog | I thought samoa was in the south pacific.. |
16:16.29 | mutilator | it is |
16:16.41 | mutilator | but anchorage anyway is filled with em |
16:16.44 | syle | canadian maritimes is where you want to be: 9 to 1 girls to guys |
16:16.48 | mutilator | samoans and asians |
16:17.28 | syle | then there is always the phillipines and tailand hehe |
16:17.38 | LostFrog | Note to self: vmware doesn't like it when you change the MAC for your ethernet card without rebooting. |
16:17.44 | mutilator | hm |
16:17.50 | mutilator | well none of my bosses are around |
16:17.55 | ursuspacificus | Ariel_: well... /dev/zap is now appearing as it should, but when I modprobe wctdm, I still get the same error "line 0: Unable to open master device '/dev/zap/ctl'" |
16:17.59 | mutilator | i'm going to sneak out to town and get food |
16:18.03 | mutilator | bbl |
16:18.06 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-165.nas28.salt-lake-city1.ut.us.da.qwest.net) |
16:18.16 | syle | lostfrog, stealing ip addresses again? |
16:18.18 | mutilator | maybe i'll get boardwalk at mcdonalds! |
16:18.32 | mutilator | ;) |
16:18.32 | LostFrog | no.. fooling a MAC filter. |
16:19.06 | syle | ahh, licensed software |
16:19.15 | iCEBrkr | grrrr. lspci doesn't show my TP100 card |
16:19.31 | LostFrog | It must not exist then.. :) |
16:19.38 | LostFrog | It was all your imagination. |
16:19.43 | iCEBrkr | I guess so |
16:19.56 | LostFrog | Sue Digium for vaporware. :) |
16:19.59 | iCEBrkr | haha |
16:20.06 | iCEBrkr | This is weird. |
16:20.14 | n3u7 | I'm not sure that asterisk can be installed on SuSE9.3 |
16:21.21 | LostFrog | I'm sure it can be installed, running sucessfully is a fish of a different color. |
16:21.46 | gaspiz | how can I jump to another realtime context from a realtime context? |
16:21.56 | LostFrog | If I wasn't lazy, I would download it and help you out. |
16:22.55 | *** join/#asterisk pjz (n=pj@place.org) |
16:24.02 | iCEBrkr | Ya know. It helps if the card is pushed in all the way |
16:24.11 | LostFrog | lol.. yes it does. |
16:25.03 | iCEBrkr | w00t! configured! |
16:25.22 | LostFrog | Grrr.. why does RendezvoisProxy not save it's configuration when it exits?? |
16:25.26 | LostFrog | -i |
16:25.38 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
16:25.41 | *** part/#asterisk pjz (n=pj@place.org) |
16:26.11 | *** join/#asterisk in (n=int@secure.wifihacker.org) |
16:27.13 | iCEBrkr | ok, now how the hell do I Dial() through this thing :P |
16:27.53 | LostFrog | Dial(Zap/<x>,<number>)? |
16:28.21 | iCEBrkr | I'm lost on the <x> part... I figured Zap/ |
16:28.29 | iCEBrkr | Doc's claim g1 |
16:28.32 | *** join/#asterisk smcmahon (n=admin@digitaldatabits.net) |
16:28.48 | *** join/#asterisk damned (n=vpol@damned.vpol.org.ru) |
16:28.59 | LostFrog | Whatever port it is. |
16:29.04 | LostFrog | Starting with 0 |
16:29.04 | iCEBrkr | haha |
16:29.08 | iCEBrkr | I dunno |
16:29.14 | iCEBrkr | I've never done PRI stuff with Asterisk |
16:29.18 | LostFrog | zap show channels? |
16:29.24 | iCEBrkr | oh, duh!! |
16:29.34 | darwin35 | ouch |
16:29.40 | darwin35 | thats my forhead |
16:29.46 | iCEBrkr | Sorry |
16:29.47 | LostFrog | lol.. I knew that was coming.. |
16:30.07 | LostFrog | Better your forehead than your foreskin. |
16:30.11 | iCEBrkr | doh |
16:30.19 | smcmahon | Wow running Xwindows platform over Windows 2000 is pretty cool. Looking for a better X-Client to use tho besides reflection |
16:30.22 | darwin35 | hey keep my foreskin out of this |
16:30.36 | darwin35 | kde |
16:30.39 | iCEBrkr | Is there a zap module? Cuz I don't have the zap command available. |
16:30.41 | darwin35 | xfce |
16:30.48 | LostFrog | chan_zap.so? |
16:30.56 | Ahrimanes | smcmahon: xfree's win32 edition? |
16:30.58 | darwin35 | hey hey hey |
16:31.10 | smcmahon | heh sorry had to darwin35 |
16:31.15 | darwin35 | leave it where it is |
16:31.19 | LostFrog | You know what the best thing about being a rabbi is? |
16:31.24 | smcmahon | but that would be limp right? |
16:31.25 | LostFrog | You get to keep the tip. |
16:31.37 | darwin35 | hahahah |
16:32.44 | nextime | can * detect a pulse tone like a dtmf on a pri zap channel? |
16:33.19 | *** join/#asterisk hugo-v6 (n=hugo@195.140.232.114) |
16:33.31 | hugo-v6 | hiho |
16:33.45 | azzie | nextime, pulse over PRI ? that's funny :) |
16:34.16 | *** join/#asterisk [Airwolf] (n=airwolf@airwolf.xs4all.nl) |
16:34.49 | nextime | azzie : i know, but i have some ivr apps with a large user base to support, and about 1% of those users can use only a pulse phone line |
16:35.04 | iCEBrkr | nextime: Sucks to be them! |
16:35.10 | syle | wonder what would happen if i used a rotory phone and dialed 2 to a PRI line |
16:35.25 | iCEBrkr | nextime: When was the last time you ever heard of a IVR system that supported pulse-dial? |
16:35.40 | azzie | nextime, my guess - you'll be getting flash events for every digit... |
16:35.46 | gaggaman | hi! |
16:35.58 | nextime | iCEBrkr : i've no hear about that, i'm asking if it is possible |
16:36.09 | iCEBrkr | nextime: I don't think it is. |
16:36.16 | gaggaman | could maybe somebody help me with my bristuffed asterisk 1.09 and call Pickup? |
16:36.24 | jarrod | im getting droped voice in the middle of phone calls.. i guess thats the RTP stream and it only happens one way |
16:36.26 | iCEBrkr | chan_zap.so is being stuborn |
16:36.28 | jarrod | anyone ever heard of this |
16:36.42 | gaggaman | when I do a Pickup, i get; |
16:36.44 | iCEBrkr | load_modules: Loading module chan_zap.so failed! |
16:36.47 | gaggaman | <PROTECTED> |
16:36.47 | gaggaman | <PROTECTED> |
16:36.47 | gaggaman | <PROTECTED> |
16:36.50 | iCEBrkr | blah blah blah |
16:37.20 | nextime | iCEBrkr : ok thanks, i think the same. |
16:37.37 | gaggaman | and then, the picked up channel (Zap/1-1) is ZOMBIE |
16:37.47 | ursuspacificus | w00000000000000000000000000000000000000t. |
16:38.07 | azzie | nextime, try here: http://www.voip-info.org/wiki-Dial+Pulse+to+Touchtone+DTMF+Converters |
16:38.23 | ursuspacificus | must modprobe zaptel first, then modprobe wctdm. |
16:38.24 | tainted_ | what is parkandannounce used for? |
16:39.04 | iCEBrkr | weird.. Seems as if I put a noload => chan_oss.so it worked. |
16:39.10 | iCEBrkr | I guess I could test that. |
16:39.22 | gaggaman | nobody? |
16:40.01 | gaggaman | what does "<MASQ>" in Hungup 'SIP/61-a2fe<MASQ>' mean? |
16:40.21 | nextime | azzie : the pulse to dtmf converters are for a phone line attached to a fxs on the * server, not for an external line come in from a remote and unknown user to an E1 line |
16:40.23 | syle | means incorrent symbol for NASDAQ trade |
16:40.45 | iCEBrkr | syle: boooo hissss :) |
16:40.59 | gaggaman | could have guessed this :-) |
16:42.09 | LostFrog | There is no such thing as a pulse-only phone line. |
16:42.27 | LostFrog | Make your users buy dual-mode phones with a switch for pulse/DTMF. |
16:43.12 | LostFrog | That's what I used to do when the phone company wanted to charge me extra for DTMF dialing. |
16:43.19 | nextime | LostFrog : i can't ask to all italian phone users to buy a device for call my service 1 time |
16:43.20 | *** part/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
16:43.49 | nextime | they are about 60 milion of people |
16:44.08 | nextime | and i don't know which of them will call my ivr |
16:44.10 | LostFrog | 1% would only be 600,000. |
16:44.22 | iCEBrkr | nextime: Seriously man, it ain't happening. |
16:44.40 | *** join/#asterisk zedas (i=zedshawc@pizarro.dreamhost.com) |
16:44.43 | *** part/#asterisk darwin35 (n=darwin35@208.139.193.178) |
16:46.46 | iCEBrkr | *CLI> pri show span 1 |
16:46.46 | iCEBrkr | Status: Provisioned, In Alarm, Down, Active |
16:46.52 | iCEBrkr | Errrrrm. |
16:47.25 | nextime | iCEBrkr : my problem is that the ivr that i'm talking about serve a service like a "lottery" over a premium number, so, if a user call and can't play to the game, he pay for the service but he can't win anything... |
16:47.55 | iCEBrkr | Welcome to 2005 |
16:48.01 | nextime | anyway, i will put a "check if your phone is tone compatible" menu on the ivr app |
16:48.03 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc1bi.dialup.mindspring.com) |
16:48.04 | iCEBrkr | That'd be my greeting |
16:48.09 | nextime | this sound like the only one solution |
16:48.54 | iCEBrkr | That's how everyone used to do it |
16:53.43 | mutilator | omg mcdonalds monopoly sucks |
16:53.46 | mutilator | every, and i mean every time i've gone i get B&O Railroad |
16:53.50 | mutilator | i have like 9 sittin in my car right now |
16:54.16 | brookshire | :( |
16:54.23 | brookshire | you need to buy more mcdonalds |
16:54.25 | brookshire | hehe |
16:54.52 | brookshire | i wonder if you can buy stamps on ebay |
16:55.14 | mutilator | most of the time i got get breakfast |
16:55.28 | mutilator | actaully went for lunch today |
16:56.16 | syle | ignorepat => 9 |
16:56.21 | syle | am i missing something |
16:56.26 | syle | why doesn;t this work |
16:57.51 | *** join/#asterisk fugitivo (n=ajf@209.13.241.249) |
16:58.05 | pauldy | anyone know why did doesn't seem to work right and if there are any examples on how to make it work right |
16:58.19 | pauldy | sorry i should add with incomming sip calls |
17:00.23 | *** join/#asterisk nobell (n=jdegraff@67.137.31.58) |
17:02.59 | mutilator | wonder if i can instant win anything |
17:05.22 | *** join/#asterisk AgiNamu (n=AgiNamu@dsl081-096-215.den1.dsl.speakeasy.net) |
17:06.42 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
17:08.43 | AgiNamu | shit, configuring a cisco as5X makes configuring asterisk look easier than anything |
17:22.00 | *** join/#asterisk bkw_ (n=bkw_@adsl-69-148-34-63.dsl.tulsok.swbell.net) |
17:22.18 | *** part/#asterisk bkw_ (n=bkw_@adsl-69-148-34-63.dsl.tulsok.swbell.net) |
17:25.14 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
17:26.51 | *** join/#asterisk Assid (n=assid@203.115.64.62) |
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17:30.34 | *** join/#asterisk nexis (n=nexis@12-219-60-252.client.mchsi.com) |
17:30.39 | nexis | blah, that sucked |
17:32.27 | *** join/#asterisk phb (n=phb@c213-100-46-84.swipnet.se) |
17:34.10 | nextime | detach |
17:36.12 | eKo1 | wtf, this is the second time * crashes on me. It always crashes after "-- Stopped music on hold on ..." |
17:36.39 | *** join/#asterisk lars-ut (n=me@67.137.31.58) |
17:38.25 | gaggaman | could maybe somebody help me with my bristuffed asterisk 1.09 and call Pickup? |
17:38.31 | *** join/#asterisk pnviking (n=pnviking@c83-248-7-150.bredband.comhem.se) |
17:38.33 | gaggaman | when I do a Pickup, i get: |
17:38.46 | gaggaman | <PROTECTED> |
17:38.55 | gaggaman | <PROTECTED> |
17:39.02 | gaggaman | <PROTECTED> |
17:40.05 | *** join/#asterisk phb (n=phb@c213-100-46-84.swipnet.se) |
17:40.14 | snitt | afaik picking up does not work between different technologies |
17:40.26 | snitt | a sip exten can only pickup an other ringing sip exten |
17:40.38 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
17:41.25 | nexis | man, aint cell proviers supposed to have battery backups and crap for when the power dies? |
17:41.36 | *** part/#asterisk frenzy (n=frenzy@193.220.82.108) |
17:41.52 | *** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca) |
17:42.28 | asterisk99 | anyone know if ztfcg is supposed to be run after every reboot?? |
17:42.45 | asterisk99 | (amke the ztcfg) |
17:42.56 | eKo1 | yes |
17:43.18 | asterisk99 | eKol: any reason why it would sudennly stop running? |
17:43.43 | eKo1 | what suddenly stopped running? |
17:44.23 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
17:44.33 | asterisk99 | eKol: it used to work... it would reboot fine... now it doesn't --- I have to manually run ztcfg and start asterisk |
17:45.13 | asterisk99 | eKol: I wonder if I should d/l and install the latest Zaptel driver |
17:45.45 | nexis | asterisk99, sounds like your init script is gone, or messed up |
17:45.53 | asterisk99 | hmmmmm |
17:46.40 | g__ | Strageness: I'm calling a 1-888 number on our PRI, and it times out after 45 seconds, even though the call is still in progress. |
17:47.37 | *** join/#asterisk JohnnyC (n=JoaoCorr@195-23-115-68.net.novis.pt) |
17:48.05 | g__ | Since it's a PRI.. all the signallying should be happening out of bound.. so how could asterisk not know the phone call has been answered? |
17:48.15 | g__ | s/bound/band |
17:48.52 | *** join/#asterisk loick (n=loick@APuteaux-151-1-30-110.w82-124.abo.wanadoo.fr) |
17:49.29 | g__ | Does anyone know what could be going on? |
17:50.03 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
17:50.23 | malverian[work] | If I set a variable in sip.conf for a phone using setvar=FOO=bar, how can I access that from the dialplan? |
17:51.24 | *** join/#asterisk pehrjansson (n=pehr@adsl-68-90-188-15.dsl.austtx.swbell.net) |
17:51.58 | pehrjansson | Dare I ask a newbie question? |
17:52.00 | devel | word, everybody. can i have music-on-hold while ringing? |
17:52.11 | devel | pehrjansson, have at it. |
17:52.25 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
17:52.32 | wunderkin | g__: the other end isnt answering the line, the easiest way i know of to fix it is to answer it on your side, you can set it to just the specific numbers you have problems with.. i think there are other ways but dunno |
17:52.45 | malverian[work] | devel, ? |
17:53.00 | malverian[work] | devel, Yeah... Dial(...,,m[(class)]) |
17:53.17 | JohnnyC | <PROTECTED> |
17:53.18 | JohnnyC | Open of mISDN Failed |
17:53.24 | JohnnyC | Im getting mISDN failed |
17:53.52 | devel | malverian[work], i'm sorry, i meant while ringing via transfer |
17:53.58 | g__ | wunderkin: I'll look into it.. I *just* tripped over a mailing list post that describes my problem. |
17:54.45 | devel | you hear the MOH while they're actually dialing the exten, then goes to dead air until answered.... |
17:55.43 | pehrjansson | I have a couple of Grandstream Budge Tone and cannot get the VM to work. Changed the dtmf in sip_additoinal.conf dot "dtmfmode=Info". Still it will not work to access the vm mailboxes from the telephone set. |
17:56.38 | devel | pehrjansson, i use dtmfmode=rfc2833 all the way around with no problems |
17:57.04 | devel | pehrjansson, both on the grandstream side and the asterisk side (set for each exten in sip.conf) |
17:57.44 | pehrjansson | in the Asterisk hitchhiker's guide it is stated "Info |
17:57.44 | pehrjansson | The Info method sends SIP Info messages from phone to Asterisk with the text of the buttons pressed on the phone. SIP Info tends to be a better choice than Inband because the key-presses are sent textually, independent of the audio codec. Users of Grandstream phones should use Info for the Asterisk voicemail; the other methods won't work. |
17:57.49 | pehrjansson | " |
17:58.32 | devel | bah. |
17:58.48 | pehrjansson | do you use grandstream phones? |
17:58.55 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
17:59.01 | devel | yes. both budgetone and handytone |
17:59.10 | ful|work | haven't problems compiling oh323 with asterisk -> http://pastebin.ca/25865 |
17:59.18 | ful|work | *i'm having |
17:59.21 | pehrjansson | and you have no problem with rfc2833? |
17:59.41 | ful|work | anyone knows how to fix this ? |
17:59.42 | devel | that is affirmative, pehrjansson, so long as it is set the same in both the phone and in sip.conf |
18:00.04 | wunderkin | exten => 1,n,Monitor(raw|/home/dialer/monitor/${leadid}-${UNIQUEID}|b) |
18:00.04 | wunderkin | exten => 1,n,Dial(SIP/2|18) |
18:00.12 | wunderkin | that should be ok right? its only recording 0.040 sec hmm :( |
18:00.23 | pehrjansson | the sip.conf I know how to manipulate but I haven't found dtmf on the web interface for the telephone set |
18:01.19 | devel | pehrjansson, it's labeled "Send DTMF" |
18:01.39 | pehrjansson | i see it now. |
18:01.47 | pehrjansson | thanks. i'll try it. |
18:01.52 | wunderkin | hey mark think i found a bug with the queue monitoring.. it core'd on me when the agent answers.. using 10/15 have to check to see if anything has changed on that yet :D trying to work around the problem now |
18:02.41 | devel | so, nothing about MOH while ringing (via transfer)? |
18:03.24 | *** join/#asterisk junbug (i=junya@adsl-144-134-75.mia.bellsouth.net) |
18:03.33 | devel | heck, even ringing would be acceptable (i just get dead air) |
18:03.50 | wunderkin | devel, maybe the moh is getting stopped early? |
18:04.19 | devel | wunderkin, it acts like as soon as the "transfer" takes place (i.e. first phone releases call) the MOH stops |
18:04.32 | wunderkin | well i meant check the cli :D |
18:04.37 | iCEBrkr | Alrighty, this blows. |
18:04.42 | devel | i am. i watch it stop. |
18:04.51 | devel | so now i need to know how to make it unstop. |
18:05.03 | iCEBrkr | TP100P says my PRI is in alarm, but the same PRI just worked on our PoS Dialogic box |
18:05.44 | wunderkin | devel, you have moh on the dial for who you are dialing? |
18:06.17 | devel | i don't, in fact. |
18:06.24 | pehrjansson | devel - that worked. thanks again. |
18:06.32 | devel | but shouldn't i at least be getting the ringing? |
18:06.36 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
18:06.36 | devel | no problem, pehrjansson |
18:06.51 | *** join/#asterisk Godsey (i=lanny@pdpc/supporter/sustaining/Godsey) |
18:06.59 | syle | i prefer |
18:07.15 | syle | Monitor(wav|/home/dialer/monitor/${leadid}-${UNIQUEID}|bm) |
18:07.17 | syle | in your case |
18:07.23 | wunderkin | devel, ive heard people say sometimes they dont get ringing.. im not sure about all the internals so i dunno just giving hints where to look |
18:07.27 | wunderkin | syle, yeah i know about m :P |
18:07.36 | `Sauron | hm. |
18:07.39 | *** join/#asterisk dasuberdavid (n=david@gateway.digium.com) |
18:07.42 | `Sauron | Monitor() lets you record calls? |
18:07.51 | AgiNamu | yea |
18:07.55 | syle | wav i can;t listen to from my windows computer |
18:07.55 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net) |
18:07.56 | syle | can |
18:07.59 | devel | wunderkin, i'll force the m(class) for that exten, since it's a group. thanks for the pointer. |
18:08.03 | wunderkin | syle, im just going to have a script to mix and convert to mp3 at the end of the day |
18:08.03 | `Sauron | Apparently I should update my * install. :) |
18:08.21 | pehrjansson | next question, which I believe relates to txgain, people we talk to complain about the volume being too muffled. I ran ztmonitor and it did appear that the levels were very low. how do I best increase the output volume. |
18:08.26 | *** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net) |
18:08.36 | wunderkin | syle, my problem is that it didnt record a darn thing though :P |
18:09.25 | syle | strange |
18:09.39 | devel | pehrjansson, that's in /etc/asterisk/zapata.conf (rxgain/txgain) |
18:09.40 | syle | are you sure that dir is writeable by asterisk user? |
18:09.50 | syle | did you check debug log? |
18:10.04 | wunderkin | syle, yeah well the file did get written only 0.040 sec tho |
18:10.25 | AlexCTI | Hi... Some can help me to get a good quality on Music on Hold, I tried raw and mpg123 but the quality is not good. |
18:10.27 | syle | what version of asterisk |
18:10.30 | devel | pehrjansson, i think you have to actually restart asterisk after that change (rather than just reload) |
18:10.34 | wunderkin | syle, head 10/15 |
18:10.51 | g__ | wunderkin:it sounds like I'm having problems related to "Answer Supervision".. I'm just reading up on it now. |
18:10.57 | Renacor | ugh what am I doing wrong with this polycom phone, I can dial out on it, but when I dial it's extension it goes to VM |
18:11.15 | syle | strange, well worked for my zap and sip tests |
18:11.15 | *** join/#asterisk mmmToop (n=chatzill@rrba-146-64-241.telkomadsl.co.za) |
18:11.23 | wunderkin | g__: yeah.. sometimes the other end doesnt send an answer while in the ivr i guess |
18:11.24 | Renacor | it waits but the phone doesn't ring |
18:11.41 | g__ | Yeah.. do you know (off the top of your head) what the fix is? |
18:11.41 | syle | i;m using latest cvs head as well |
18:11.55 | wunderkin | not sure what i changed i thought i had this working already |
18:12.09 | syle | hmmmmm |
18:12.24 | syle | i have a feeling maybe it can;t find sox in the path |
18:12.27 | syle | check |
18:12.29 | syle | which sox |
18:12.30 | wunderkin | im not using m |
18:12.37 | *** join/#asterisk ic (n=ic@staff.rbi.speka.net) |
18:12.40 | ic | hi there |
18:12.55 | syle | try it |
18:13.03 | syle | what you got to loose |
18:13.06 | wunderkin | it shouldnt be running sox until afterwords anyways though, the file should be several seconds |
18:13.07 | g__ | Wunderkin: I presume you have a T1 as well? Would you care to trade zapatel.conf files? |
18:13.10 | syle | fix by trial and error |
18:13.37 | wunderkin | g__: yeah i have a pri, nothing special in the config though, i commented out everything and only added in callerpres |
18:13.59 | wunderkin | i can't call 800 numbers though since its only LD, and i dont call ivrs :D |
18:14.01 | g__ | Thanks, I'll compare mine with the default then. |
18:14.14 | syle | do you own the PRI? |
18:14.23 | wunderkin | yeah its mine |
18:14.24 | devel | wunderkin, that's the stuff. after adding the m(class) to the group extens all works as expected. thanks. |
18:14.26 | ic | has someone here interfaced asterisk (h323) with a eads telecom ipbx with a pt2 card ? |
18:14.29 | g__ | Not personally, but my company does. |
18:14.35 | wunderkin | oh him |
18:14.49 | g__ | why? |
18:14.49 | wunderkin | devel: ok thats what i was wondering, not sure about no ringing though |
18:16.22 | *** join/#asterisk SplasPood (i=jwb@ludicrous.paravolve.net) |
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18:20.02 | *** part/#asterisk T-Squared (n=ted@hidden.serreyn.com) |
18:21.22 | wunderkin | not working w/o b either |
18:21.46 | wunderkin | weird :( |
18:21.58 | wunderkin | it stops monitoring after the dial probably |
18:22.08 | wunderkin | i mean after answer |
18:22.13 | wunderkin | why? |
18:22.50 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
18:25.46 | Renacor | argh why wont this damn phone ring |
18:26.18 | nexis | anyone using goiax? |
18:26.49 | *** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net) |
18:28.37 | ronaldl79 | How are you guys dealing with clients who are Qwest, SBC, BellSouth or Verizon subscribers that require white page listings? After porting a clients number, are you finding these companies still list your clients number and address, or not? |
18:31.16 | Ariel_ | ronaldl79, when I setup any system all my customers keep one or 2 lines with the local phone co. Then we forward the local number to the voip one. |
18:31.32 | Ariel_ | other reason for this is faxes and 911 |
18:33.21 | *** join/#asterisk vader-wrk (n=johndoe@204.183.88.101) |
18:33.27 | vader-wrk | hello |
18:35.39 | vader-wrk | does anyone in here use ADIT 600 channel banks in their setup? |
18:38.06 | ronaldl79 | Hmmm, Ariel. I don't want the telco anywhere in the picture. There are sold on the cost savings of VoIP alone. |
18:38.14 | ronaldl79 | They* |
18:38.46 | Ariel_ | ronaldl79, well don't have any suggestion. Due to all my setups must have one pots. for 911. It's just makes no sence in my view not to have it. |
18:39.09 | g__ | 911 is a very good reason.. |
18:39.19 | g__ | Safety first.. |
18:39.26 | AlexCTI | Ariel_, do you know how can i do to get better quality on MOH? |
18:40.07 | Ariel_ | AlexCTI, I use normal mpg123 in the setups seems to work fine. |
18:41.07 | infinity1 | Ariel_: who do you use for voip gateway usually? |
18:41.52 | Ariel_ | infinity1, depending on the setup I use either a sipura 3000 or tdm400p. Also channel banks like the Adtran. It really depends on the setup. |
18:41.53 | AlexCTI | I did, but the quality is not the best, and in some parts of the song it sounds like underwater.. or noise, i'm using G729 i don't if it affect the quality. |
18:42.42 | Ariel_ | AlexCTI, ahh then convert them to g729 or native (raw) if your using cvs head. I only use stable at present. |
18:42.52 | infinity1 | Ariel_: what about voip to pots service? like teliax and such? |
18:43.32 | Ariel_ | infinity1, they work I use them. I have some customers using teliax, VoicePulse, Nufone asterlink. But I always maintain one pots line. |
18:44.21 | infinity1 | Ariel_: have you found an inexpensive solution for incoming callerid info over voip? |
18:44.23 | Ariel_ | Not everyone needs the same type of service. |
18:44.34 | AlexCTI | Ariel_, How can I know if i'm using cvs head? |
18:44.53 | AlexCTI | i'm new.. sorry :-( |
18:44.54 | ronaldl79 | Speaking of G729, who's using it for themselves or a client for termination? |
18:44.58 | Ariel_ | infinity1, that is a loaded question. Due it's more of what they need then just inexpensive |
18:44.59 | infinity1 | AlexCTI: did you download a tgz to use cvs to download it? |
18:45.09 | Ariel_ | show version |
18:45.52 | AlexCTI | tgz |
18:45.57 | *** part/#asterisk junbug (i=junya@adsl-144-134-75.mia.bellsouth.net) |
18:46.25 | infinity1 | Ariel_: the CNAM issue i guess it is called. thats what i find as a major drawback with teliax |
18:46.42 | Ariel_ | there is a tgz of 1.2beta1 so that is not a valid. AlexCTI in the CLI do show version |
18:46.52 | AlexCTI | ok |
18:47.11 | AlexCTI | Asterisk 1.0.9 built by root@ACS on a i686 running Linux |
18:47.22 | Ariel_ | infinity1, I don't have much problems with them. Since I am only changing the number not the name when we use them for dial out. |
18:47.34 | *** join/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
18:47.39 | Ariel_ | AlexCTI, you have stable. |
18:47.44 | infinity1 | Ariel_: oh . do you use voip for incoming calls? |
18:47.52 | Ariel_ | infinity1, yes I do |
18:48.06 | AlexCTI | Ariel, that's the same like yours..correct? |
18:48.24 | Ariel_ | Yes |
18:48.36 | infinity1 | Ariel_: is there a solution for callerid in incoming calls? |
18:48.49 | scoates | I'm getting what I can only describe as "DTMF doubling" iax->asterisk->SIP-><conferencing bridge> (it detects a single "4" as "44") -- any ideas? I don't mind reading, I just don't know what to look for. |
18:48.49 | Ariel_ | But I have not run into any problems do you have enough cpu power for the transcoding? |
18:49.50 | Ariel_ | infinity1, Yes there is. My home I and work I have an did from connect.voicepulse.com Which gives me full caller ID info. Also allot of others do as well. |
18:50.02 | iCEBrkr | So is there some special cable I'm supposed to use with this T100P card? |
18:50.16 | Ariel_ | iCEBrkr, t1 crossover cable |
18:50.19 | iCEBrkr | The cable I have now seems to work just fine in another box with a Dialogic card in it. |
18:50.35 | infinity1 | Ariel_: guess i just got unlucky with teliax :) |
18:50.37 | iCEBrkr | I'm assuming this little adaptor thing is the crossover. |
18:50.38 | Ariel_ | dialogic card is not a digium card |
18:50.38 | xheliox | The IAXy is requesting a registration period of 0 seconds, when Asterisk's default is 60 seconds. Is there any way to tell the IAXy how long to register for? And yes, I know the minreg can be set in iax.conf, but that creates a different issue, which isn't really relevant. :) |
18:50.51 | iCEBrkr | Ariel_: I understand that.. I'm trying to rid this company of Dialogic cards |
18:50.51 | AlexCTI | Ariel_, yes, the CPU is enough |
18:50.53 | iCEBrkr | :P |
18:51.03 | *** join/#asterisk Inv_arp (i=junya@adsl-144-134-75.mia.bellsouth.net) |
18:51.22 | iCEBrkr | I've had nothing but headaches with Dialogic cards in the 15yrs I've been around phone 'systems' |
18:51.27 | Ariel_ | AlexCTI, are you using mpg123 or mpg321? which version of mpg123? |
18:51.47 | iCEBrkr | not saying I know anything about phone systems-- But I'm an expert that Dialogic cards are nothing but problems :P |
18:52.04 | Ariel_ | iCEBrkr, http://www2.adtran.com/support/technotes/t1ddsadptxvr/ |
18:52.14 | AlexCTI | Right now, I switch it by rawplayer, it sound a little bite better, but i was using mpg123. |
18:52.16 | Inv_arp | Ariel_: sup man |
18:52.23 | iCEBrkr | Adtran? had one of those at the last job... |
18:52.35 | Ariel_ | iCEBrkr, I have about 50 of the 4pci and dialog4 boards. I think i know them also |
18:52.52 | iCEBrkr | :) |
18:53.11 | infinity1 | Ariel_: wow. 11 / month for a voicepulse did. but free incoming minutes! |
18:53.26 | Inv_arp | Ariel_: the client i was talking to you about never got the phones in yet so thats why i havent updated you on anything yet |
18:53.45 | *** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) |
18:53.56 | Ariel_ | Inv_arp, not much just stopping by to answer some questions. No problem |
18:54.14 | AlexCTI | Ariel, how can i get the mpg123 version? |
18:54.26 | Inv_arp | infinity1: inphonex offers 7.95 incoming DID ... SIP/lIbc/g729 support only tho |
18:54.32 | Ariel_ | infinity1, I have used them for over 2 years now and it's been rock solid. There not the cheapest but they have worked. |
18:54.47 | Ariel_ | AlexCTI, at the linux prompt do mpg123 |
18:54.51 | infinity1 | Ariel_: well the fact that they have callerid is huge. |
18:55.08 | ronaldl79 | I'm actually looking for a rock solid provider as well for a client. |
18:55.19 | Ariel_ | infinity1, not all sections or calls will give you caller ID |
18:55.26 | AlexCTI | High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. |
18:55.26 | AlexCTI | Version 0.59r (1999/Jun/15). |
18:55.30 | Ariel_ | some are blocked depends on who is calling. |
18:55.42 | ronaldl79 | I cannot risk these guys with BroadVoice, they need a solid sip/iax provider ... my reptuation is on the line .. and so is VoIP. |
18:55.49 | Ariel_ | AlexCTI, that is the correct one |
18:56.12 | scoates | heh.. it was echo |
18:56.15 | scoates | silly moi |
18:56.44 | AlexCTI | let me give you the config.. hold on |
18:56.48 | Ariel_ | ronaldl79, look at voicepulse, teliax, then. |
18:56.57 | iCEBrkr | Ok, the cable I have is a straight-through... I'm going to have to assume this little blackbox hang'n off the interface here is the crossover |
18:57.05 | Ariel_ | AlexCTI, I have to go back to work I will be back later. |
18:57.12 | AlexCTI | ok.. |
18:57.17 | iCEBrkr | ronaldl79: Good luck |
18:57.21 | Ariel_ | iCEBrkr, never assume |
18:57.27 | iCEBrkr | Ariel_: agreed :P |
18:57.52 | ronaldl79 | hmmmm |
18:58.07 | iCEBrkr | I guess all I need to know is what kinda cable I need for this Digium card. I briefly looked on Digium's site, and didn't find anything.. Tho, I didn't look hard |
18:58.28 | infinity1 | Ariel_: do they let you pick your did #? how much of a selection do they have? |
18:58.29 | g__ | After reading more posts on the Internet, the only solution to calling big companies with IVRs that don't send their a "Answer Supervision" until you're speaking to a real person is either: navigate the IVR menu really fast, or make special dialing rules that Answer() the before Dial(). Is there no other solution? |
18:58.32 | Ariel_ | iCEBrkr, did you not get my link |
18:58.32 | ronaldl79 | I don't see anyone else offering BroadVoice's service plans though ... damn, what a wedge to be in... |
18:58.44 | iCEBrkr | Figures I get all the zaptel/libpri stuff compiled and working and now I'm gonna have issues with hardware :-/ |
18:58.47 | Ariel_ | infinity1, you pick from a list |
18:58.51 | iCEBrkr | Ariel_: Yeah, what am I looking for on this thing? |
18:58.58 | *** join/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro) |
18:59.07 | Ariel_ | I have to go see you all later. |
18:59.08 | AlexCTI | If I use mp3 files wich is the correct format of these files? |
18:59.31 | Ariel_ | ~docs |
18:59.32 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
19:00.31 | iCEBrkr | I'm quite aware of the docs and the wiki |
19:00.53 | AlexCTI | cf-usa-en => custom:/var/lib/asterisk/mohmp3/cf-usa-en,/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s |
19:02.23 | *** join/#asterisk ap0ught (n=ap0ught@208.19.226.94) |
19:04.06 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
19:05.26 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
19:07.06 | wunderkin | looks to me like * is sending the rtp to the right ip and port why aint it workin :( |
19:07.27 | infinity1 | i'm trying to connect * to gizmo. when making a call, i get an error. i haven't been able to figure out a working configuration. error: Got SIP response 488 "Not Acceptable Here" back from 198.65.166.131 |
19:11.13 | pauldy | ahh infinity1 after you left last night I set it up using the config from the voip-info site |
19:11.25 | pauldy | works fine |
19:11.31 | infinity1 | pauldy: no )(#*) way! |
19:11.37 | pauldy | yup |
19:11.52 | pauldy | you running amp? |
19:11.57 | infinity1 | amp? |
19:12.07 | pauldy | asterisk management portal |
19:12.08 | infinity1 | can you post your sip.conf and your extenion to pastebin.ca |
19:12.11 | infinity1 | pauldy: no. |
19:12.29 | pauldy | it is the sme as whats on voip-info |
19:12.42 | infinity1 | pauldy: i've tried that! hrmz. |
19:12.48 | pauldy | http://www.voip-info.org/tiki-index.php?page=Asterisk+settings+Gizmo |
19:12.49 | infinity1 | pauldy: and many variations |
19:13.04 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-219.cpe.netcabo.pt) |
19:13.06 | infinity1 | pauldy: what version of * are you useing? |
19:13.20 | pauldy | 1.0.9 |
19:13.28 | infinity1 | pauldy: i'm using head. |
19:13.31 | pauldy | well no scratch that I updated to head last night |
19:13.43 | infinity1 | ahh ...now i must hurt you :) |
19:14.00 | infinity1 | can you post your sip.conf and extension anyway? |
19:16.02 | infinity1 | it works for me if i use xten client. but not * ...heh |
19:17.12 | pauldy | why not post yours to see what is wrong |
19:17.16 | pauldy | mine has way to much for me to cherry pick out user/pass pairs |
19:17.37 | Renacor | anybody know why a polycom phone will not ring? |
19:17.59 | pauldy | volume turned down, no ones calling it, is it plugged in |
19:18.44 | harryvv | Renacor it wont ring because you dont have it configured and or dont have a ftp server it log into and copy the image files. |
19:20.12 | infinity1 | pauldy: http://pastebin.ca/25871 |
19:20.20 | infinity1 | pauldy: mine is a mess. i've tried everything |
19:20.44 | vader-wrk | does anyone in here use ADIT 600 channel banks in their setup? |
19:20.44 | infinity1 | pauldy: just post your gizmo and your register line |
19:20.46 | Renacor | harryvv it is set up |
19:21.27 | Renacor | harryvv: it never gets to SIP/phonename-xxx is ringing |
19:21.38 | Renacor | but it executes the rest of the dialplan |
19:25.25 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
19:25.46 | *** join/#asterisk Los415 (n=los415@64.201.104.186) |
19:25.58 | vader-wrk | can an FXO card only accept phone calls? |
19:26.38 | syle | analog trunk |
19:27.45 | *** part/#asterisk lars-ut (n=me@67.137.31.58) |
19:27.46 | vader-wrk | say if i have 8 analog trunk lines |
19:28.11 | vader-wrk | do i need (2) quad FXO cards and (2) quad FXS cards? |
19:28.16 | infinity1 | anyone have a gizmo phone # i can try |
19:28.57 | snitt | infinity1: i have one |
19:29.07 | snitt | can you send dtmf inband? |
19:29.28 | snitt | if no, my number would ring at the livingroom |
19:29.39 | infinity1 | hmm ..can i try? |
19:30.34 | snitt | dtmfmode=rfc2833 |
19:30.38 | snitt | well.. |
19:30.48 | snitt | 1-747-622-2653 |
19:30.51 | *** join/#asterisk wzlwzl- (n=wzlwzl@wsip-70-183-60-181.oc.oc.cox.net) |
19:30.54 | wzlwzl- | hey all.. have a * setup using 3 broadvoice lines and Polycom IP300 phones... We have 1.5Mbit up and down via cable. 40ms (ave) pings to the bv proxy and no packetloss. The phones sound like cell phones. The person on the other end complains about it cutting in and out. Any help troubleshooting this plz? |
19:30.59 | snitt | dial 2 when you hear voice |
19:31.49 | snitt | FUCK |
19:31.54 | infinity1 | ? |
19:31.56 | snitt | i told u to dial or hangup |
19:32.03 | snitt | Asterisk Ready. |
19:32.09 | infinity1 | i dialed. didn't hear anytihng |
19:32.19 | *** join/#asterisk mogorman (n=mogorman@gateway.digium.com) [NETSPLIT VICTIM] |
19:32.28 | snitt | no. i dont believe that |
19:32.37 | snitt | it works, maybe not at your side |
19:32.39 | infinity1 | either my side is broke. or your side is :) |
19:35.46 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
19:38.35 | infinity1 | hmm. when calling into mine using gizmo, i don't even get anything on the cli |
19:39.13 | *** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
19:40.29 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) [NETSPLIT VICTIM] |
19:42.42 | infinity1 | i appear to be able to call gizmo users. they just cant call me. |
19:43.42 | *** join/#asterisk juice (n=juice@mo-67-77-176-229.dyn.sprint-hsd.net) |
19:46.30 | *** join/#asterisk YaP (n=YaP@ppp-60-43.27-151.libero.it) |
19:46.38 | YaP | hi |
19:46.39 | YaP | is it possible to authenticate iax phones using their mac address? |
19:47.10 | AgiNamu | well, if you're on a single ethernet network |
19:47.19 | AgiNamu | you could hack asterisk to do that |
19:47.37 | AgiNamu | but mac addresses aren'tr transmitted into TCPIP so... |
19:48.46 | Nugget | well, there's also the challenge that the server doesn't necessarily see the mac address of the phone. |
19:48.51 | Nugget | it would be quite limiting to presume so |
19:50.20 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
19:51.32 | YaP | Nugget: with hack you mean modify asterisk code? |
19:51.56 | YaP | i need to authenticate the phone, do you have any other idea? |
19:53.50 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
19:55.50 | *** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
19:56.19 | Igbothom_III | tzafrir_laptop; art thee here? |
19:57.51 | *** join/#asterisk Nexis (n=nexis@12-207-56-108.client.mchsi.com) |
19:58.01 | Igbothom_III | tzafrir_laptop; art thou here? |
19:58.54 | *** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
20:00.52 | vader-wrk | any of you work with channel banks? |
20:01.12 | harryvv | not yet |
20:01.26 | harryvv | I hear the rhino channel bank is one of the easiest to work with. |
20:01.45 | vader-wrk | id like to do an intermix of channel banks and ip phones/software |
20:01.46 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
20:01.49 | generalhan | whats up everyone ! |
20:02.12 | vader-wrk | ive only found a few 24-48 channel banks but i need more connections |
20:02.16 | generalhan | im having a serious issue with transfering calls right now between my users ... im getting this NOTICE |
20:02.16 | generalhan | Oct 18 13:01:07 NOTICE[3710]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' |
20:02.16 | generalhan | <PROTECTED> |
20:02.18 | vader-wrk | can you hook up more than one channel bank |
20:02.39 | generalhan | anyone know what the deal is ? |
20:03.52 | Nexis | vader-wrk, correct, you can hook up as many banks as you would like |
20:04.04 | Nexis | generalhan, you do have chan_sip.so loaded correct/ |
20:04.57 | *** part/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
20:04.59 | generalhan | lol i dont know, ive had this asterisk setup running nonstop for the past month with no changes and this just started happening today |
20:05.12 | generalhan | should i just reload everything and see if that fixes it ? |
20:05.47 | generalhan | well that didnt help any so scratch that idea ... crap |
20:06.06 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
20:06.09 | Nexis | try a restart |
20:06.16 | generalhan | crap |
20:06.16 | hardwire | never! |
20:06.17 | generalhan | i cant do that |
20:06.19 | hardwire | CRAP! |
20:06.23 | hardwire | NOOOOOO!O!O!! |
20:06.23 | generalhan | i have 20 peopole on the phones right now |
20:06.23 | Nexis | why cant you restart? |
20:06.29 | Nexis | eww |
20:06.33 | generalhan | yea no good |
20:06.38 | generalhan | lemme see if i can tell them all to get off ! lol |
20:06.45 | hardwire | heh |
20:06.47 | Nexis | call center, or can you ask them to logoff for like 3 min? |
20:06.49 | hardwire | you tell them like this |
20:06.51 | vader-wrk | wat do you guys think of the ADIT 600 E1? |
20:06.52 | hardwire | restart now |
20:07.05 | vader-wrk | i was looking at the E1 because you don't need a PRI card supposly just a network card |
20:07.23 | generalhan | when i restart it do i do a "shutdown now" then a "safe |
20:07.27 | generalhan | _asterisk" |
20:07.34 | generalhan | or is there a better way ? |
20:07.42 | Nexis | stop gracefully |
20:07.53 | Nexis | are you running asterisk as a non privlaged user? |
20:07.59 | generalhan | nope |
20:08.01 | generalhan | as root |
20:08.08 | Nexis | wow, you have balls. |
20:08.19 | generalhan | ?? |
20:08.26 | hardwire | heh |
20:08.27 | Nexis | then yea, you would execute safe_asterisk |
20:08.31 | hardwire | restrart when convenient |
20:08.33 | hardwire | or gracefully |
20:08.34 | hardwire | never works |
20:08.38 | hardwire | nor shutdown when ... |
20:08.50 | hardwire | never ever has worked correctly for me |
20:08.53 | hardwire | always drops the channels |
20:09.03 | Nexis | stop when convenent works good |
20:09.03 | paryl | i'm looking for a SIP phone with a nice price/feature point for a rollout of about 40 stations. i tried the gxp2000 in a satellite location, and they just seem a little too flimsy. any suggestions? |
20:09.17 | hardwire | Nexis: CVS? |
20:09.21 | Nexis | yea |
20:09.28 | hardwire | in 1.0.0 1.2.0 and CVS it has never worked correctly |
20:09.41 | hardwire | there will be bridged calls.. then poof.. no calls |
20:09.43 | Nexis | paryl, do you need any special features off a sip phone? |
20:09.45 | mutilator | i run asterisk with daemon-tools |
20:09.47 | mutilator | never dies |
20:09.49 | mutilator | :P |
20:10.14 | Nexis | and are your users going to be abusive to the hardware? |
20:10.37 | paryl | nexis: 2+ lines, standard headset capable, speakerphone |
20:10.53 | paryl | and, no, they won't be terribly abusive |
20:10.57 | Igbothom_III | paryl; polycom 501 |
20:11.03 | *** join/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
20:11.08 | paryl | i'm going to buy some of the 2000's for the warehouse, etc. |
20:11.11 | Igbothom_III | or snom 360 |
20:11.30 | Nexis | paryl, a channel bank would be a choice. |
20:11.53 | paryl | a channel bank? |
20:11.57 | Igbothom_III | but channel banks only allow old analog phones to be used, don't they? |
20:12.03 | paryl | ah |
20:12.10 | Nexis | correct |
20:12.24 | paryl | currently we're on a siemens system, and i'm replacing it with asterisk... so the phones go with it |
20:12.25 | Nexis | but he could plug old phones in, and drop normal phones around |
20:12.27 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
20:12.33 | Nexis | ahh |
20:12.35 | Igbothom_III | actually, do they allow old PABX analog phones to be used? |
20:12.57 | *** join/#asterisk newl (n=newlook@203-59-214-216.dyn.iinet.net.au) |
20:13.10 | jsaunders | In a sip 200 OK after a REGISTER from client, is the "Contact" header required? |
20:13.11 | Nexis | not sure, i havent looked into them much |
20:13.50 | paryl | igbothom: i was looing at the 501... you like it? |
20:14.33 | Nexis | paryl, dont know if you have looked at the stuff on voip-info, but that will give you a idea as to how well phones would work. |
20:14.34 | Nexis | http://www.voip-info.org/wiki/view/Asterisk+phones |
20:15.25 | paryl | yeah... i've done a lot of searching... just wanted to see if anyone could give me real-world views on it |
20:15.31 | harryvv | can anyone recomend a voip phone that has the best tx/rx range? |
20:15.35 | paryl | people swore i'd love the grandstream phone :) |
20:15.45 | harryvv | voip wifi phone with those ranges? |
20:15.55 | Nexis | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg38942.html |
20:16.08 | Nexis | thats a good review there also, real world experence. |
20:17.11 | generalhan | Nexis:i restarted it all and still no go. same error about creating a SIP channel |
20:17.19 | Nexis | odd |
20:17.50 | Nexis | its only when they try to transfer, or? |
20:17.56 | generalhan | i dont understand it ! lol. its been working great for over a month now |
20:18.24 | generalhan | wheather you try and transfer or just 4 digit extension dial it goes directly to their VM Box and i get that error in the console |
20:18.49 | Nexis | and nothing has changed? |
20:18.54 | generalhan | nope |
20:18.55 | *** part/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
20:19.17 | Nexis | cvs head, or 1.0? |
20:19.21 | generalhan | not with asterisk anyway. i just finished installing Festival, but i dont know why that would change anything with asterisk |
20:19.25 | generalhan | 1.0.9 |
20:20.05 | *** join/#asterisk Connor- (n=billy@198-144-174-5.knx.tn.nxs.net) |
20:20.20 | Connor- | Hey guys, how much difference is there with HEAD vs 1.2Beta 1 right now? |
20:20.32 | Connor- | Was 1.2B1 just a snapshot of the HEAD tree? |
20:21.01 | Nexis | can you set verbose in 1.0.9? |
20:21.07 | generalhan | yep |
20:21.11 | generalhan | its at 42 right now ! lol |
20:21.16 | Nexis | ok, set debug 50 |
20:21.21 | generalhan | lol hang on |
20:21.34 | Nexis | should be able to do it from the CLI |
20:21.54 | generalhan | k |
20:22.05 | Nexis | try it |
20:22.06 | generalhan | :Verbosity was 42 and is now 103: |
20:22.07 | Nexis | see what you get |
20:23.19 | *** join/#asterisk JASON-0 (n=jason@jason.unitz.ca) |
20:23.35 | generalhan | Nexis: this is what i get in the console http://generalhan.pastebin.ca/25881 |
20:23.43 | JASON-0 | I am compiling asterisk but it fails at : /usr/bin/ld: cannot find -lssl |
20:24.07 | harryvv | ssl modual not installed? |
20:24.14 | Nexis | JASON-0, install the ssl dev package |
20:24.14 | JASON-0 | i installed OpenSSL |
20:24.18 | JASON-0 | and it still does it |
20:24.25 | Nexis | JASON-0, distro? |
20:24.29 | JASON-0 | Mandrake |
20:24.29 | generalhan | Nexis: see anything out of the ordinary ? |
20:24.31 | Nexis | generalhan, do you have a sip phone on your desk? |
20:24.58 | generalhan | kinda ! |
20:25.02 | generalhan | it is now |
20:25.05 | generalhan | Cisco 7960 |
20:25.24 | Nexis | reboot it |
20:25.44 | *** join/#asterisk kippi (n=chrisfro@cpc3-hatf3-6-0-cust49.lutn.cable.ntl.com) |
20:25.50 | generalhan | ok why ?! |
20:26.11 | JASON-0 | Nexis: I'm using Mandrake |
20:26.26 | Nexis | i have a suspicion |
20:26.32 | generalhan | what am i looking for though ?> |
20:26.33 | Nexis | JASON-0, you need the ssl dev libs |
20:26.40 | JASON-0 | where do I get that? |
20:26.51 | Nexis | generalhan, we are looking to see if the phone is not registered in asterisk |
20:27.03 | Nexis | if you can make calls, but not recieve them, it kinda points to 1 of 2 things |
20:27.07 | Nexis | 1. asterisk needs restarted |
20:27.14 | Nexis | 2. your phone is not registering with asterisk |
20:27.33 | generalhan | i wish i would have rebooted a different phone then |
20:27.51 | generalhan | my cisco takes like 3 minutes to start up. the other 25 Aastra phones i have reboot in like 10 seconds |
20:27.57 | Nexis | ahh |
20:28.31 | JASON-0 | Nexis: Do you know where I get the SSL dev libs? |
20:28.47 | Nexis | JASON-0, no, i dont, i dont use a rpm based distro |
20:28.58 | Nexis | #mandrake or the forums could help im sure |
20:29.05 | JASON-0 | ok thank you :) |
20:29.19 | JASON-0 | #mandrake |
20:29.23 | JASON-0 | oops |
20:30.25 | generalhan | Nexis: i rebooted the phone |
20:30.31 | eKo1 | #mandrake...i thought that distro was dead |
20:30.58 | generalhan | but i had about 20 calls comming in and going out on the console so i couldnt see if it said anything about my phone registering |
20:32.03 | harryvv | General. what astra phones u using? |
20:32.56 | generalhan | 9112i |
20:33.06 | Nexis | generalhan, any luck? |
20:33.16 | generalhan | Nexis: nope, still cant transfer inside |
20:33.23 | *** join/#asterisk damned (n=vpol@prior.lanck.net) |
20:33.37 | generalhan | i cant even call the autoattendant and dial an extension there |
20:34.37 | harryvv | generalhan so you system is not up and running yet. what did that phone cost you? |
20:34.52 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
20:35.18 | generalhan | the phones cost only $120 and were well worth it |
20:35.28 | generalhan | my system is having issues right now thats all |
20:36.01 | generalhan | Nexis: ok new info for you. i can get the transfer to work to 3 of the 30 numbers that i have. so why would it allow me to transfer to 3 people and not the other 27 ? |
20:36.12 | harryvv | general not a bad price. |
20:36.24 | harryvv | Are you seeing this system up for your office? |
20:37.00 | vader-wrk | whats the different between IAX and SIP? |
20:37.27 | harryvv | firewall issues vader |
20:37.31 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
20:38.05 | generalhan | harryvv: no the price is great for what it is. its only a one line phone but it works great. and i have already set my system up, its been running great for over a month now, and just today something went haywire |
20:39.07 | harryvv | generalhan do you always write down the steps you do in asterisk? |
20:39.21 | generalhan | no |
20:39.25 | generalhan | what steps ? |
20:40.14 | generalhan | ok i hae to go figure this out ... this is bad ill be back in a bit guys |
20:41.18 | harryvv | see, most errors are created by the person who configures the software. I dont know in if this is the case. |
20:41.36 | *** join/#asterisk Katty (n=katrina@68-112-15-110.dhcp.cpgr.mo.charter.com) |
20:41.47 | Igbothom_III | meow |
20:42.01 | Katty | mew. |
20:42.10 | Igbothom_III | meowning |
20:42.13 | wzlwzl- | hey all.. have a * setup using 3 broadvoice lines and Polycom IP300 phones... We have 1.5Mbit up and down via cable. 40ms (ave) pings to the bv proxy and no packetloss. Using ulaw. The phones sound like cell phones. The person on the other end complains about it cutting in and out. Any help troubleshooting this plz? |
20:42.27 | generalhan | harryvv: i agree with you, but this time, i didnt do anything to it. its the same today as it has been for the past month and it just messed up |
20:42.33 | wzlwzl- | within the office, ip300->ip300, it sounds great |
20:43.39 | harryvv | Wz, have you done any preliminary bandwith calculations and mearsurment test before implementing the ipphone system? |
20:43.58 | Igbothom_III | wzlwzl-; dedicated 1.5/1.5, or shared with other Internet traffic? |
20:44.25 | harryvv | generalhan look at the error logs on what is is doing. |
20:45.06 | wzlwzl- | Igbothom_III: shared, but its the same when all other traffic is stopped... also running QoS (more details of config here: http://www.ospiron.com/ast.txt ) |
20:45.27 | Igbothom_III | 404 |
20:45.46 | wzlwzl- | you sure? i can get there just fine.. |
20:46.03 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
20:46.12 | Igbothom_III | actually, the main page is 404 as well |
20:46.16 | wmandra | wz: connection refused |
20:46.21 | wzlwzl- | hrm.. 1 sec |
20:46.30 | generalhan | harryvv: the error that is comming up is that it cant creat a channel SIP everyone is busy/congested at this time |
20:46.33 | Igbothom_III | looks like your website is offline to the ret of the world! :( |
20:46.44 | wzlwzl- | dns issue on that domain |
20:46.45 | *** join/#asterisk Caede (n=caede@sentry.zoom.com) |
20:46.49 | wzlwzl- | www.beachrentalsearch.com/ast.txt |
20:47.03 | Igbothom_III | aha - that damn Level3/Whomever bitchfight is rearing its ugly head again |
20:47.22 | Igbothom_III | beachblah works fine |
20:47.30 | wzlwzl- | k |
20:47.37 | Igbothom_III | ~pb |
20:47.39 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
20:47.39 | harryvv | general, busy means that the other line is in use. |
20:47.53 | harryvv | or the line is not connected. |
20:47.54 | generalhan | right, but they arent |
20:47.58 | generalhan | and thats the issue |
20:47.58 | wzlwzl- | jbot: yea.. shoulda used that instead |
20:48.07 | groogs | wzlwzl-: maybe you're not actually getting 1.5mbps |
20:48.36 | Igbothom_III | no chance of 1.5/1.5 on a 1.5/1.5 pipe anyway |
20:48.38 | Igbothom_III | 90% tops |
20:48.38 | Caede | Anyone know what might cause a "T1: Lost our place, resyncing" on a zaptel card? |
20:48.42 | generalhan | those phones can make calls and recieve them if they are called directly, but they cant be the recipiant of transfer or an extension dial |
20:48.44 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfmuo.dialup.mindspring.com) |
20:48.50 | groogs | wzlwzl-: try connecting to another SIP/IAX number |
20:48.54 | Caede | (one that is repeated about 10,000 times an hour) |
20:49.05 | wzlwzl- | 1.1mbit according to various speed tests |
20:49.10 | Igbothom_III | and your overall pipe size needs to be a touch under the maximum speed you get - tried d/ling the latest kernel ands seeing the actual speed? |
20:49.33 | groogs | wzlwzl-: maybe it's just your connection to broadvoice then. check some tracerts against places you know you can get hte bandwidth |
20:49.41 | wmandra | wz, 1) QoS will only work if your ISP supports it. 2) What cable provider are you using?? |
20:49.42 | Igbothom_III | so, your pipe limit set to just under that? |
20:49.42 | groogs | wzlwzl-: and 1.1mbit down, but what about up? |
20:49.50 | wzlwzl- | wmandra: cox |
20:49.52 | wzlwzl- | business |
20:50.03 | groogs | wmandra: no, it will work on your network regardless of your ISP |
20:50.26 | Igbothom_III | exactly - it will only work OUTSIDE of your LAN if your ISP supports it :) |
20:50.30 | groogs | wmandra: ie, if someone on his net is trying to download something, the QoS will place priority on the asterisk traffic and the download will slow down |
20:50.38 | wmandra | groogs: it will work on your network yes, but once the packets leave hisr router the qos tags will be ignored |
20:51.01 | groogs | yes. and in a perfect world that wouldn't matter, since he has 1.5mbit to the isp ... |
20:51.21 | wmandra | true, 3 calls should only be using about 240kbps |
20:51.28 | Igbothom_III | but few ISPs fully implement QoS support |
20:51.40 | wzlwzl- | Your download speed : 1396 kbps or 174.5 KB/sec. |
20:51.40 | wzlwzl- | Your upload speed : 387 kbps or 48.4 KB/sec. |
20:51.45 | Igbothom_III | 3 calls should use a LOT less than that if using G.729 |
20:51.54 | wzlwzl- | guess upstream is getting capped |
20:52.01 | wmandra | he said he was using g.711u |
20:52.09 | Igbothom_III | aha |
20:52.11 | wzlwzl- | but in anycase... even with just 1 call |
20:52.13 | Igbothom_III | furry muff |
20:52.25 | wzlwzl- | it shouldn't sound choppy/cell-phone-like |
20:52.28 | wzlwzl- | no? |
20:52.44 | Igbothom_III | depends on issues between your modem and the end user's handset |
20:53.00 | Igbothom_III | there are MANY things than can cause issues, unfortunately |
20:53.13 | wmandra | wz, to be honest I was using cable for a while here (comcast) and had the same problem, switched to dsl and the problem went away |
20:53.54 | groogs | also they don't really have 1.5 mbit .. ie, if there's 100 subscribers in a given area, they don't have 150mbit dedicated to it .. they have maybe like, 5mbit that has to be shared |
20:54.08 | wzlwzl- | groogs: cox business is dedicated |
20:54.12 | Igbothom_III | issue is possibly that cable is based on the old hub-style architecture - when your neighbor goes at downloads hammer and tongs, you suffer |
20:54.32 | Igbothom_III | wzlwzl-; no, it isn't |
20:54.39 | groogs | wzlwzl-: perhaps |
20:54.40 | Igbothom_III | they sell it to you that way... |
20:54.45 | *** part/#asterisk Caede (n=caede@sentry.zoom.com) |
20:55.03 | wzlwzl- | anyhow, that's a tangent... |
20:55.09 | *** join/#asterisk jbenson (n=jbenson@genpubad.gotadsl.co.uk) |
20:55.09 | groogs | they might also allocate 2 mbit of a 5 mbit for business only .. and call it dedicated |
20:55.09 | wzlwzl- | is there any way to troubleshoot this |
20:55.10 | groogs | who know |
20:55.12 | wmandra | wz: my first suggestion would be to try g729 |
20:55.12 | wzlwzl- | to isolate the problem |
20:55.17 | *** join/#asterisk spiekey (n=spiekey@p549D1B88.dip0.t-ipconnect.de) |
20:55.26 | groogs | wzlwzl-: yes, call another SIP/IAX number/service and see if the same thing happens |
20:55.35 | Igbothom_III | also, try calling someone else on Cox business and see if that's fine |
20:55.35 | groogs | like the digium test number |
20:55.47 | Katty | heh, i read that as digium - take a number. |
20:55.48 | wmandra | wz: or try a call through a different provider |
20:55.53 | Igbothom_III | if that works, then it is the link between Cox Business and the rest of the world where the issue is |
20:56.13 | Katty | twisted[asteria]: or more like, asteria - take a number and get in line for hysteria! |
20:56.19 | groogs | Igbothom_III: or the link between cox and his provider |
20:56.19 | spiekey | zapta.conf is for the hardware configuration only, right? |
20:56.24 | Katty | twisted[asteria]: oops, did i say that outloud? |
20:56.34 | Igbothom_III | groogs; his provider IS cox |
20:56.43 | groogs | sorry, i mean voip provider |
20:56.48 | Igbothom_III | :) |
20:57.03 | Igbothom_III | yeah, thsat is the link between Cox and the outside world, still |
20:57.04 | jbenson | Hi All - I am trying to set up a connection to FWD via IAX, but I am getting an error on the console: - Host 'iax2.fwdnet.net/496460' not found at line 17. Is anyone else having this problem please? |
20:57.50 | groogs | Igbothom_III: or PART of the outside world.. presumably a 'big' provider like cox (big enough that i've heard of them before, though i live in canada) has more than one connection |
20:58.10 | Igbothom_III | yeah, still relates to their connection to the outside world! |
20:58.39 | wmandra | wz: does the problem happen all the time or only during peek hours? |
20:58.48 | wzlwzl- | wmandra: all the time |
20:58.57 | *** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net) |
20:59.03 | wzlwzl- | even when everything else is taken off the network, and i make a single call |
20:59.15 | jbenson | or, does anyone here connect to FWD via IAX, please? |
20:59.19 | Igbothom_III | sure your users aren't playing with you and talking in a choppy, low quality voice? :) |
20:59.29 | wzlwzl- | lol... yea, im sure =) |
20:59.33 | Igbothom_III | hehe |
20:59.40 | wmandra | igbothom: lmao |
20:59.56 | scoates | jbenson: http://www.freeworlddialup.com/community/support/iax.php |
21:00.05 | wzlwzl- | the other problem i have, and i dunno if its related or not... on inbound calls from the bv numbers into the auto attendant... it picks up duplicate DTMF |
21:00.29 | wzlwzl- | but somewhere i read that's just an issue with bv and not much can be done about it |
21:01.01 | wmandra | wz: using bv here with no problems |
21:01.03 | Inv_arp | hmm.. cant get this sometimes asterisk just gets unregistered to my sip provider , sip reload fixes it |
21:01.16 | jbenson | thanks scoates. I have that all set up, but I get an error Host 'iax2.fwdnet.net/496460' not found at line 17. What does that mean exactly? The host does "exist", but is the login being rejected? |
21:03.24 | wmandra | wz: you can try setting qualify=yes under the section for bv in your sip.conf then use sip show peers from the CLI |
21:03.26 | spiekey | zapta.conf is for the hardware configuration only, right? |
21:03.26 | Inv_arp | jbenson: hmm the host is iax2.fwdnet.net not iax2.fwdnet.net/496460 |
21:05.07 | spiekey | ok...zapta.conf is the config file i will have to edit when i first install/configure asterisk. or where woudl you start? |
21:06.07 | Igbothom_III | spiekey; zapata.conf is to configure the Zaptel ATA card |
21:06.27 | Igbothom_III | sip.conf and extensions.conf are two of the main files |
21:07.28 | wmandra | spiekey: http://voip-info.org/wiki/view/Asterisk+config+files |
21:09.39 | Renacor | anybody familiar with polycom phones know how to make the phone re-read the .cfg file when you reboot it? for some reason it is not updating it from the tftp |
21:12.06 | *** join/#asterisk JohnnyC (n=JoaoCorr@195-23-115-68.net.novis.pt) |
21:12.12 | JohnnyC | hellop |
21:16.22 | *** join/#asterisk ohad (n=ohad@19-231-13-72.cosmoweb.net) |
21:16.44 | ohad | how do i add more ringtons to my BT 101? |
21:17.16 | Igbothom_III | BT 101, as it is the cheapest of the cheap, I'd doubt you can |
21:20.40 | *** join/#asterisk dos000 (n=dos000@i216-58-60-251.cybersurf.com) |
21:23.56 | ohad | Igbothom_III, i am sure i can. the question is how:) |
21:27.33 | *** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3685577.sympatico.ca) |
21:28.10 | tehdely | gah |
21:28.11 | tehdely | dfasdfasdf |
21:28.19 | tehdely | digium replaced my two TDM400Ps |
21:28.21 | tehdely | insalled the new ones today |
21:28.23 | tehdely | still crackling |
21:28.24 | tehdely | :( |
21:28.30 | tehdely | zttest gives consistent 99.98 - 100% |
21:29.51 | *** join/#asterisk delink (n=delink@ziegchen.delink.net) |
21:31.17 | ursuspacificus | Renacor: Re Polycom phone config files... if you're using TFTP, you need to change the filenames on the bootserver... I usually just symlink... so... FE if you changed file 'phone1234.cfg' you will need to 'ln -s phone1234.cfg phone1234.cfg.r001' or something like that and update 000412345678.cfg to point to the updated filename. |
21:32.13 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
21:33.10 | steve___ | Is there something other than callerid that a telco can send that isn't blockable? |
21:33.44 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
21:35.05 | ender | Ftp is the way to go for Polycom. not tftp |
21:35.53 | Katty | tftp-- |
21:35.55 | Katty | ^- skeery. |
21:37.04 | ohad | anyone, help with canceling echo on asterisk? |
21:37.16 | ohad | i am using PRI and therefore shouldn't have any ech |
21:37.16 | ohad | o |
21:39.59 | Nexis | generalhan, sorry about runnin off, you figure it out? |
21:41.19 | *** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
21:41.22 | diclophis | hello |
21:41.27 | *** join/#asterisk darkskiez (n=darkskie@host86-138-169-183.range86-138.btcentralplus.com) |
21:41.34 | diclophis | what is the standard way of switching contexts in a call? |
21:41.49 | devel | a question that somebody may be able to address quickly: we have a t1 (not PRI), and no callerid data. the calls come in with a text label of "asterisk", where does that come from? |
21:41.50 | diclophis | i thought it was the Goto command, however I keep getting this error |
21:42.03 | diclophis | WARNING[3170]: pbx.c:2278 __ast_pbx_run: Channel 'SIP/212-62d9' sent into invalid extension '600' in context 'alt', but no invalid handler |
21:42.06 | spiekey | i have asterisk here with a ISDN card. i finished configuring zapata.conf. Which config file should i configure next? |
21:42.14 | diclophis | Also note that I am using asterisk Realtime config... |
21:43.14 | Renacor | anybody familiar with polycom phones know how to make the phone re-read the .cfg file when you reboot it? for some reason it is not updating it from the tftp |
21:43.24 | darkskiez | has anyone used asterisk head on osx in the last couple of weeks? |
21:45.00 | ender | Renacor: tftp the file would have to be renamed. |
21:45.08 | ender | Renacor: honestly, you _really_ should use ftp instead. |
21:45.34 | ender | Renacor: ftp will work off of timestamps and work easier for uploading files/configs/directories/etc... |
21:49.27 | *** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
21:49.42 | asterisk99 | Q: I am trying to d/l STABLE 1.2 from CVS --- I am geting an error "no such tag v1-2" Ideas? Hints? |
21:50.08 | MikeJ[Laptop] | there is only 1.2beta1 |
21:50.13 | MikeJ[Laptop] | and it is not called stable |
21:50.21 | MikeJ[Laptop] | in fact, there is nothing called stable |
21:50.42 | MikeJ[Laptop] | you can snag the tarball for beta1 off the ftp site or grab it by it's tag |
21:52.26 | kippi | did anyone have problems when installing mysql? for AMP? |
21:52.56 | *** join/#asterisk ManxPower (n=ewieling@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
21:55.55 | *** join/#asterisk wunderkin (i=kev@0-1pool199-201.nas26.tempe1.az.us.da.qwest.net) |
21:56.14 | *** join/#asterisk scud (n=scud@12-214-190-139.client.mchsi.com) |
21:56.32 | mmlj4 | hey PacketLoss :-) |
21:57.14 | pauldy | kippi, I had major problems because I decided to clear a table after I installed it |
21:57.34 | pauldy | apparently it needs it default data or it starts acting wierd |
21:59.32 | nobell | any suggestions on a good web site that has perl agi examples for ivr stuff? |
21:59.36 | spiekey | whats app_capiCD.so good for? |
21:59.55 | spiekey | i get a "Loading module app_capiCD.so failed!" but i am not even sure if i need that module |
22:01.32 | kippi | i just keep on getting this damm error Timeout error occurred trying to start MySQL Daemon. and it is stoping my asterisk time :( |
22:03.39 | pauldy | kippi, sounds like you don't even have asterisk running right |
22:03.52 | pauldy | check your my.conf and make sure everythings good |
22:03.55 | pauldy | have to run again |
22:06.28 | spiekey | what does that mean? "loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber" |
22:06.38 | spiekey | undefined symbol?! |
22:07.44 | diclophis | anyone using extension realtime config? |
22:07.47 | *** join/#asterisk Miggidy (n=Miggidy@203-59-9-189.perm.iinet.net.au) |
22:07.59 | Igbothom_III | undefined symbol - the source code is buggy |
22:08.00 | MikeJ[Laptop] | it means it does not know what that is |
22:08.07 | MikeJ[Laptop] | where is that function? |
22:08.26 | MikeJ[Laptop] | somthing you are not including in -l when you are compiling\linking |
22:08.32 | *** join/#asterisk Shaun2222 (n=ndci@ip68-111-70-41.oc.oc.cox.net) |
22:08.35 | spiekey | i am currently using the version which comes with debian sarge |
22:09.06 | spiekey | i guess i will have to switch to the currend tarballed version ;) |
22:16.31 | zedas | spiekey, see if debian has a bunch of other required packages you need to install first. |
22:16.47 | zedas | spiekey, debian is famous for doing lots of little packages without dependencies. |
22:17.36 | hardwire | blah |
22:20.55 | *** join/#asterisk santiago (n=santiago@63.245.87.62) |
22:21.24 | *** join/#asterisk mxmasster (n=mxmasste@pitbull.media.net) |
22:21.26 | mxmasster | hi all |
22:21.47 | mxmasster | i am looking for docs on how to configure DID to people's extensions - can anyone point me to a url? |
22:22.28 | Inv_arp | mxmasster: do you have the DID yet? |
22:22.35 | mxmasster | yes |
22:22.49 | Inv_arp | mxmasster: sip or iax? |
22:22.59 | mxmasster | that is not the problem, given a block of x sip did's what is the easiest way to sent them to a phone |
22:24.30 | *** part/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com) |
22:24.50 | generalhan | mxmasster: exten => YOUR_DID_NUMBER,1,Dial(SIP/User,20,tT) will work just fine |
22:25.22 | mxmasster | generalhan: is their a shortcut for this - i would assume that a larger organization wouldn't define 100+ did's this way |
22:25.43 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
22:26.01 | generalhan | not that i know of ... my company isnt HUGE but i have a block of 50 DIDs and i have programmed 20 of them in this manner |
22:26.35 | pauldy | you can us a macro and pattern |
22:26.55 | generalhan | eww thats right, but youll need to talk to someone else about macros ... i dont touch them ! lol |
22:26.56 | pauldy | _55555512XX,1 |
22:27.43 | *** join/#asterisk cp5 (n=samy@adsl-69-110-135-49.dsl.irvnca.pacbell.net) |
22:27.58 | generalhan | pauldy: but that is 100 possible numbers what if all those 100 numbers he has and needs each to go to a different place ? |
22:28.20 | hardwire | hmm |
22:28.27 | hardwire | channel based volume normalizationw ould rule |
22:29.12 | generalhan | pauldy: not that i know how to do it or anything but i would love to know how for the future. we might be getting another 50 DIDs and i would like to know how to get it done properly |
22:31.02 | mxmasster | pauldy: do you know of a public example configuration that show's how to do this? |
22:31.25 | pauldy | nope |
22:31.43 | cp5 | is CVS zaptel's fxsdump broken? |
22:31.48 | pauldy | just know thats one way to do it if you were one to one mapping extensions to a did |
22:31.50 | cp5 | it's expecting a missing header |
22:33.25 | Renacor | anybody know where I can get the xml example for the polycom setup cfg's? |
22:33.28 | cp5 | coeffs.h |
22:34.16 | pauldy | if your doing 100 dids though a simple script could generate the dial lines for you |
22:35.07 | wzlwzl- | when editing config files like extensions.conf... can i do reload now, or do i need to restart now |
22:37.34 | *** join/#asterisk K-Bear (n=k-bear@h29n2fls32o815.telia.com) |
22:37.36 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
22:39.19 | *** part/#asterisk jsaunders (n=jsaunder@d154-5-198-14.bchsia.telus.net) |
22:42.22 | K-Bear | I'm having a pretty odd problem. I'm setting up an IAXy. I've manage to provision it and I've set up my Asterisk server to have it accepted. It's working to the point that I get a dial tone, I can dial all the extensions on the PBX. But when a call is connected, even when it's local in my own network, I don't get any sound from the device connected to the IAXy. I can hear the person at the extension I dialed, but they can't hear the IAXy. And |
22:42.22 | K-Bear | after 15s Asterisk hangs up. Anyone know what's wrong? |
22:44.13 | syle | it doesn't like you? |
22:44.21 | K-Bear | Heh :-) |
22:45.02 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
22:46.01 | *** join/#asterisk stevek (n=stevek@slim-eth0.horizonlive.net) |
22:49.18 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-165.nas28.salt-lake-city1.ut.us.da.qwest.net) |
22:50.16 | *** join/#asterisk n3u7 (n=neutrin0@CPE000d8802a707-CM0011e6c7edb1.cpe.net.cable.rogers.com) |
22:51.05 | cp5 | anyone familiar with the queue/agent deadlocking bugs? |
22:51.17 | n3u7 | lokking for answers: |
22:51.20 | n3u7 | http://forums.digium.com/viewtopic.php?t=1973&highlight=suse9+3 |
22:56.36 | *** join/#asterisk jbenson (n=jbenson@genpubad.gotadsl.co.uk) |
23:00.50 | *** join/#asterisk _tekati_ (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
23:01.59 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
23:02.06 | *** join/#asterisk MatsK (n=mk@99.80-202-83.nextgentel.com) |
23:05.24 | *** join/#asterisk bob_too (n=chris@rrcs-24-153-179-246.sw.biz.rr.com) |
23:07.08 | l1nux | is possible use modem without sound (without load oss or alsa modules) ? |
23:09.08 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
23:10.46 | Katty | mrow. |
23:11.11 | SarahEmm | katty! |
23:11.12 | SarahEmm | meow! |
23:11.29 | SarahEmm | *purrrr* |
23:12.09 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
23:12.18 | Katty | Ariel_: hihi |
23:12.28 | Ariel_ | Katty, hello |
23:13.13 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
23:13.16 | *** part/#asterisk nobell (n=jdegraff@67.137.31.58) |
23:13.43 | *** join/#asterisk lilneon (n=tj_r3@cuscon12932.tstt.net.tt) |
23:14.13 | lilneon | hey everyone.. my RH9 asterisk installation not detecting my new digium tdm21B card |
23:14.17 | lilneon | any ideas? |
23:14.39 | Ariel_ | just stopping by to say hello. On my way to see mom. |
23:15.03 | Katty | Ariel_: talk to you later (= |
23:15.32 | lilneon | ok so no one else ever ran into this problem? |
23:15.38 | lilneon | is my card bad? |
23:15.59 | Katty | lilneon: perhaps you should call digium. |
23:16.41 | lilneon | emailed em.. no response yet |
23:16.59 | Katty | k |
23:17.04 | *** join/#asterisk K-Bear (n=k-bear@h29n2fls32o815.telia.com) |
23:17.56 | twisted[asteria] | lilneon, what version pci bus? |
23:18.07 | twisted[asteria] | lilneon, and does the card show up in lspci? |
23:18.16 | lilneon | have no idea really.. cant find the manual for this old board |
23:18.18 | twisted[asteria] | also, did you build and install, and the load the zaptel modules? |
23:18.35 | lilneon | yeah modprobe zapata goes by without a hitch |
23:18.42 | twisted[asteria] | it's not zapata |
23:19.04 | lilneon | i mean zaptel |
23:19.04 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-16-41.cybersurf.com) |
23:19.05 | lilneon | modprobe zaptel |
23:19.08 | twisted[asteria] | you still need to actually load the driver for the card |
23:19.10 | twisted[asteria] | not just zaptel |
23:19.12 | *** join/#asterisk wmandra (n=me@pcp04943183pcs.verona01.nj.comcast.net) |
23:19.20 | twisted[asteria] | in this case, wctdm |
23:19.30 | lilneon | that's when i get errors |
23:19.43 | lilneon | insmod failed... blah blah |
23:19.58 | lilneon | so i've been trying to tweak but nothing |
23:20.11 | twisted[asteria] | does it show up in lspci? |
23:20.16 | twisted[asteria] | do you have power connected? |
23:20.18 | lilneon | no it doesnt |
23:20.24 | lilneon | yes power is connected to it |
23:20.25 | twisted[asteria] | ahhh |
23:20.29 | lilneon | even switched the pci slots |
23:20.31 | twisted[asteria] | it doesn't even show up in lspci? |
23:20.36 | lilneon | nope |
23:20.37 | Katty | lilneon: are they on the same iqr? |
23:20.39 | Katty | i mean irq |
23:20.41 | Katty | if i could type. |
23:20.43 | twisted[asteria] | chances are then that you have an older pci rev |
23:20.49 | twisted[asteria] | and it won't support the card |
23:20.55 | SarahEmm | it sounds like it's not the right PCI rev, and you can't run that board then |
23:21.05 | twisted[asteria] | SarahEmm, :P |
23:21.11 | lilneon | damn it.... |
23:21.15 | wwalker | Does anyone use asterisk on Windows servers? I'm a Linux/*nix/*BSD biggot and need someone else's viewpoint. |
23:21.40 | lilneon | got another pc but that is only for windows.. and running linux in vmware on the box i heard asterisk wont wrk |
23:21.46 | X-Rob | wwalker - it doesn't work on windows. |
23:21.47 | twisted[asteria] | that's correct |
23:21.57 | twisted[asteria] | linux in vmware will not work since you do not have direct access to the hardware |
23:21.58 | twisted[asteria] | well |
23:22.00 | twisted[asteria] | let me rephrase |
23:22.05 | lilneon | so basically i am screwed |
23:22.06 | twisted[asteria] | zaptel under linux in vmware |
23:22.17 | Katty | food network is telling me i must have two different types of peelers in the house at all times. |
23:22.19 | lilneon | oh ok.. |
23:22.27 | wwalker | X-Rob SCORE! |
23:22.36 | wwalker | X-Rob, Thank you!!!!!!! |
23:22.49 | twisted[asteria] | X-Rob, well, actually..... |
23:22.54 | twisted[asteria] | X-Rob, it does |
23:22.58 | twisted[asteria] | X-Rob, but not natively |
23:23.11 | lilneon | does digium have a list of mobos their cards work on? |
23:23.26 | lilneon | dont want to go get another mb and run into the same problem |
23:23.28 | twisted[asteria] | lilneon, no, but they have a list of systems it won't |
23:23.42 | twisted[asteria] | lilneon, make sure you have pci rev 2.2 or higher |
23:24.05 | *** join/#asterisk E|nyPRI_ (n=les@205-200-14-92.static.mts.net) |
23:24.08 | lilneon | twisted:.. how would i tell the difference? |
23:24.10 | wwalker | twisted, what do you mean? |
23:24.11 | E|nyPRI_ | hi |
23:24.19 | twisted[asteria] | lilneon, it'll be in your documentation, and possibly on your bios/post screen |
23:24.34 | lilneon | k |
23:24.40 | twisted[asteria] | wwalker, astwind |
23:24.47 | twisted[asteria] | wwalker, er, astwin |
23:24.54 | twisted[asteria] | google for it |
23:24.57 | twisted[asteria] | asterisk on windos |
23:24.59 | twisted[asteria] | windows, too. |
23:25.08 | lilneon | astwin doesnt allow u to add any hardware though.. |
23:25.14 | twisted[asteria] | correct |
23:25.15 | lilneon | right twisted? |
23:25.31 | twisted[asteria] | YEAH TOAST! |
23:25.36 | lilneon | this sucks sooo bad |
23:25.51 | E|nyPRI_ | any canadians here? |
23:25.54 | twisted[asteria] | lilneon, next time, read the manual :P |
23:26.06 | twisted[asteria] | lilneon, most newer mobo's have pci 2.2/2.3 |
23:26.25 | lilneon | well like i said this was an old mb.. |
23:26.31 | lilneon | was wrking fine with a quicknet phonejack |
23:26.37 | twisted[asteria] | lilneon, does it have ISA? |
23:26.37 | Katty | twisted[asteria]: bob and tom are in cape, kthx. |
23:26.40 | lilneon | decided to try the digium cads |
23:26.44 | twisted[asteria] | Katty, *nods* |
23:26.46 | lilneon | yeah it does |
23:26.55 | twisted[asteria] | ahh, yeah, probably 2.1 or 2.0 |
23:27.05 | lilneon | k\ |
23:27.24 | twisted[asteria] | best way to actually check the card though, is to download a live cd, and then boot a newer box to the livecd. |
23:27.36 | twisted[asteria] | if it shows up in lspci from the livecd, you'll know the card is good |
23:27.45 | twisted[asteria] | well, at least, that it is detectable |
23:28.03 | *** join/#asterisk marc324 (n=marc3234@206-248-135-84.dsl.teksavvy.com) |
23:28.06 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
23:28.14 | twisted[asteria] | because there is the very rare possibility that the card is just hosed, but it's doubtful based on what you've stated here |
23:28.39 | *** join/#asterisk Clearwire (n=a@206.173.47.7) |
23:29.31 | lilneon | ok |
23:29.40 | lilneon | wait.. can i run live cd from inside windows? |
23:29.42 | lilneon | ?? |
23:30.00 | X-Rob | no |
23:30.01 | syle | hmm strange thought |
23:30.07 | syle | anyone met a programmer over 40? |
23:30.08 | X-Rob | you don't run anything 'inside' windows. |
23:30.16 | lilneon | syle : yeah |
23:30.21 | Ariel_ | syle, yes |
23:30.24 | syle | seems people have stopped programming alot by that age to me |
23:30.35 | X-Rob | you boot from a CD that contains a different operating system and doesn't write to your hard drive |
23:30.43 | Clearwire | Hello, can anyone tell me why the 'Read' application command stops execution in the dial plan if it returns 'User Disconnected'? |
23:30.52 | *** join/#asterisk _DAW (n=bob@adsl-222-51-199.msy.bellsouth.net) |
23:31.27 | dos000 | syle, by that time they already moved to management or got nixed |
23:32.23 | syle | true enough |
23:32.36 | syle | they becomes teachers or owners usually |
23:33.30 | syle | idk i;ll see where i am in 10 years and get back to you on it lol |
23:33.40 | dos000 | syle, i have seen one tho. until he was shown the door for trying to stick to old teknology ! |
23:34.09 | *** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
23:34.25 | syle | i think at a certain age you want the learning to stop |
23:34.36 | syle | this is wrong field for that |
23:34.47 | E|nyPRI_ | at a certain age you realise its all bullshit |
23:35.12 | E|nyPRI_ | is this #philosophy? or #asterisk :P |
23:35.29 | syle | i;ll get back to you philosophy when i have smoke a joint or 2 |
23:35.33 | rayvd | Mars. |
23:36.39 | wwalker | syle, I'm over 40, only been programming since I was 13...(29 years) |
23:36.56 | syle | no shit |
23:36.57 | marc324 | programming dulls the mind. |
23:37.09 | syle | i bet you can program in any language then |
23:37.53 | wwalker | syle, you are correct. We get paid less than plumbers but work 40 to 80 hours a week and have to "learn/keep up with new tech" between 10 and 30 hours a week. |
23:38.16 | lilneon | you said it wwalker |
23:38.45 | syle | yeah the keeping up sucks |
23:38.54 | syle | rather watch tv for those 10-30 hours somedays |
23:38.58 | lilneon | plus with all the opensource and sites like hotscripts etc.. my 'boss' usually thinks its just a matter of googling and copying and pasting the code |
23:39.02 | wwalker | A few. 6502, x86 (barely), Pascal, Perl, C, C++, Java, *sh, but spend 98% of my programming in perl |
23:39.17 | lilneon | so he expects alot to be done in one day |
23:39.32 | wwalker | We are the lowest paid workers on the market with any actual skill |
23:39.57 | syle | have you considered working for company making games? |
23:40.14 | syle | thats where the money is |
23:40.29 | wwalker | Hahahahahahahahahahahahahahahahahahahahahahahahahahahahahahaha |
23:40.37 | wwalker | syle, THAT's funny! |
23:40.46 | syle | c++ and some grade 11 trig seems to be |
23:40.46 | Clearwire | Only if he writes/publishes/markets it himself |
23:40.59 | syle | friend makes about 150k a year is toronto c programming games |
23:41.05 | lilneon | here... only really got web application opportunities with a few console / network apps popping up here and there |
23:41.20 | lilneon | no games.. besides simple flash and or java crap for me.. |
23:41.42 | lilneon | but wwalker.. u take the cake man.. been programming 29 yrs? i am not even 29 yrs old |
23:41.49 | wwalker | If you worked for Id and all the money is shared between 8 guys, BIG money. But EA and the rest of the market make game developers into slave labor. $40K US for 80-100 work weeks, and fire them the day the game ships |
23:42.18 | syle | ouch |
23:42.20 | lilneon | ouch |
23:42.24 | syle | well thats more like contract |
23:42.29 | wwalker | syle, Your friend must be VERY good to be able to get any serious money out of a gaming dev company. That is NOT the norm |
23:42.39 | lilneon | wait but doesnt salary bracket increase with experience wwalker? |
23:42.44 | syle | yeah he was in MIT at 13 |
23:42.53 | wwalker | senior game devs get $48K |
23:42.57 | lilneon | syle-wat? |
23:42.57 | *** join/#asterisk danowoz (n=danowoz@wsip-24-249-162-89.ph.ph.cox.net) |
23:43.00 | E|nyPRI_ | an MIT grad only making 150k ? |
23:43.21 | lilneon | my life is soo unfulfilled now.. sigh |
23:43.27 | ender | wwalker: that really depends on the dev house. |
23:43.41 | syle | i didn;t say he still does it lol |
23:43.42 | ender | wwalker: we're a game dev house but our devs and artists and even IT guys (me) are very well paid. |
23:43.52 | syle | now he usually day trades on the stock market |
23:43.56 | X-Rob | "GET /gxp2000.bin HTTP/1.0" 200 12695 "-" "Grandstream GXP2000 1.0.1.13" |
23:43.59 | X-Rob | WEeeeee! |
23:44.03 | wwalker | lilneon, That's OK, I listened to Robert Frost - "took the road less traveled by" I am still a developer, not a lead, not a manager |
23:44.34 | *** join/#asterisk deezed (i=none@adsl-065-006-189-182.sip.bct.bellsouth.net) |
23:44.46 | danowoz | Hi, I have asterisk box set up with two extensions. both extensions are registered and I am able to make outbound calls via an outbound trunk. but whne I recieve incomming calls from an external service and when I dial extension to extension my calls are sent staight to voicemail. can anyone help? |
23:44.53 | wwalker | ender - sweet. I know some places do that. My President ran two game companies and paid his people well, but it's Not the industry norm :( |
23:44.55 | lilneon | well i like programming.. |
23:45.17 | X-Rob | danowoz - sounds like you're using asterisk@home. |
23:45.22 | lilneon | feel like i actually did somethingafter the project done and i go thru it |
23:45.33 | danowoz | :/ |
23:45.40 | wwalker | I still program because I enjoy it. The hours suck, but I do half of them on the couch with Buffy, SG1 or the like as video wallpaper. |
23:45.40 | ender | wwalker: right. We're Casual games too, so it's a different market. |
23:45.50 | danowoz | yes, how can I fix this problem? |
23:45.52 | ender | wwalker: still very good incom. |
23:46.12 | syle | idea of a programmer to begin with is you learn the trade to create a company of programmers down the road to build some nice software |
23:46.17 | X-Rob | danowoz - you _are_ using asterisk@home? |
23:46.29 | danowoz | X-Rob - yes |
23:46.49 | lilneon | syle : kinda what most programmers end up doing |
23:46.53 | wwalker | ender, glad someone has a good gaming desgin gig. But I got sick of running purify and valgrind type tools long ago. As I aged I got lazy. Perl, backend work, 1/10 the lines, 1/5 the debugging. |
23:47.03 | lilneon | but there are some who just cant stop coding |
23:47.04 | X-Rob | danowoz - try #amportal |
23:47.10 | X-Rob | I'm on there but I Can't help you. Someone elsemight. |
23:47.39 | syle | unfortunately that means building some nice income on the side these days while working at your day job to do that |
23:48.06 | danowoz | X-Rob - is this a problem with @home? or amp in particular, cause I am not agains editing config files by hand. |
23:48.18 | lilneon | syle: kinda burns you out after a while though |
23:48.35 | wwalker | if I won the lottery tomorrow, I'd still be in the office Th and Fr (except for meeting with an accountan t and a lawyer). Then I'd invest enough in my current gig to control dev. Then I'd sit back down and start typing again. |
23:48.44 | lilneon | it will be like u have three jobs.. each demanding 40-80hrs per week |
23:49.09 | syle | i think that is how the american dream is built ....good paying job, develop hobby site on the side, site or products grows over the years, don't need dayjob anymore--quit!, hire people, incorporate |
23:49.15 | wwalker | "running your own business" == "you can work whichever 100 hours a week you wish" |
23:49.21 | X-Rob | danowoz - it's probably a problem with dialparties.agi |
23:49.36 | X-Rob | but still, get someone to help you on #amportal |
23:50.16 | danowoz | X-Rob - thanks |
23:50.44 | syle | wwalker is that any different from working 8 hours a day programming, and how many hours keeping up on tech? |
23:50.53 | lilneon | wwalker: yeah was hoping asterisk adn voip would have been a nice side business.. to grow... spending more tiem on it than i do @wrk |
23:53.23 | znoG | argh, if only my provider didn't use https to download its config |
23:53.29 | znoG | i could have the sipura config in my hands right nowwwww |
23:54.36 | ender | wwalker: hehe, yeah, thats why I do IT and tools development, rather than being a full time developer. |
23:54.57 | lilneon | ender:IT and tools??? explain |
23:55.18 | file[laptop] | I have mail, who wants to delete it for me? |
23:55.29 | ender | lilneon: Information Technology, the name that gets slapped on system admins. Tools are jsut that. Tools that a sysadmin uses to complete tasks. |
23:55.36 | ender | lilneon: I manage servers. |
23:55.38 | X-Rob | file, whats your pop3 passworD? 8) |
23:55.42 | X-Rob | I'll sort it out for you! |
23:55.45 | file[laptop] | pop3? pfft, I use imap |
23:55.53 | X-Rob | 1 LOGIN file@is.hawt.com |
23:56.00 | X-Rob | 1 AUTH s3xx0r |
23:56.03 | lilneon | ender: oh.. net Admin?? |
23:56.22 | ender | lilneon: net, phone, hardware, etc... |
23:56.37 | syle | i think we know how to read an rfc but thx for update hehe |
23:56.41 | twisted[asteria] | oh no |
23:56.43 | ender | X-Rob: I'd like to see you type it out in Imaps (: |
23:56.45 | twisted[asteria] | not more file hittingupon |
23:57.02 | file[laptop] | my poor inodes |
23:57.21 | file[laptop] | what |
23:57.23 | file[laptop] | did it really? |
23:57.26 | twisted[asteria] | haha yeah |
23:57.30 | file[laptop] | you bastard |
23:57.32 | file[laptop] | you killed a Powerbook! |
23:57.35 | twisted[asteria] | it went to sleep |
23:57.37 | twisted[asteria] | and didnt' wake up |
23:57.47 | twisted[asteria] | i didn't kill it |
23:57.57 | syle | idk if imap is that popular though...i seen qmail+vpopmail+pop3 but not imap |
23:57.58 | twisted[asteria] | it was sitting here just fine, and then fell asleep, and that was the end |
23:58.00 | file[laptop] | killall -9 powerbook |
23:58.11 | file[laptop] | syle: I like imap... I never used to |
23:58.16 | twisted[asteria] | file, i'm getting a brand new one |
23:58.23 | file[laptop] | but I'm at three computers usually and they all stay in sync |
23:58.24 | twisted[asteria] | apple is announcing the new powerbook line tomorrow |
23:58.25 | n3u7 | i'm working on installing Asterisk on SuSE9.3 |
23:58.35 | twisted[asteria] | and the store is going to replace mine with one of the newly announced ones |
23:58.36 | n3u7 | :) |
23:58.41 | syle | anyone have any idea what digium and intel are up to? |
23:58.41 | file[laptop] | twisted[asteria]: yayyyyy |
23:58.45 | twisted[asteria] | file[laptop], yea :) |
23:59.04 | twisted[asteria] | might have to wait a few days to get it |
23:59.08 | twisted[asteria] | but it'll be worth it in the long run |
23:59.23 | twisted[asteria] | 1900x1200 native resolution :) |
23:59.29 | file[laptop] | mmm |
23:59.57 | twisted[asteria] | thats a lot of pixels |