| 00:00.41 | JT | the question is more, why would it? |
| 00:01.32 | Zaw | because you'll get good quality code from wannabe programmers who seek to brag about doing work for google and open source software |
| 00:01.35 | *** join/#asterisk tg_ (i=tg@2001:618:1a23:0:0:0:0:1) |
| 00:02.12 | JT | just not sure how easily the code could be integrated with asterisk |
| 00:02.34 | *** join/#asterisk jcool (n=zoro@124.106.204.57) |
| 00:03.32 | jcool | hi! good day guys, has anyone can claim is it plausible to have a 700+ local extension with re-invite enabled on a single server? |
| 00:03.34 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 00:03.52 | genz | Anyone know how to get the zaptel 1.4.1 build? |
| 00:03.54 | __dante__ | so why other great projects are in soc? like apache, pgsql... integrate the new code would not be the problem... |
| 00:04.44 | jcool | genz: it should be straightforward make && make install could you paste the error if any? |
| 00:05.18 | genz | jcool: Yes, but I can't find the 1.4.1. Only 1.4. I'm looking for the zaptel with the hpec directory. |
| 00:07.26 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 00:07.35 | jcool | genz: oopps, i'm sorry man, if not familiar with hpec, i'm afraid i can't help you to sort this is w8 let me check |
| 00:08.27 | genz | jcool: Reading their README - http://ftp.digium.com/pub/telephony/hpec/README - it looks like 1.4.1 should be available. But I can't find it in SVN. Then again, I can't even find the 1.2 HPEC subdirectory in SVN |
| 00:08.53 | Shaun2222 | Bobthehunter: what about L3 resellers? |
| 00:08.56 | jcool | genz: this is the hardware high performance echo cancellation stuff, w8 |
| 00:09.20 | jcool | let me check this :) |
| 00:10.24 | JT | jcool: counds like it might be possible with the right hardware seeing as reinvite is on |
| 00:10.35 | JT | probably not advisable to only use one machine |
| 00:10.39 | JT | i'd never do it, anyway |
| 00:11.21 | BigIceCream_Read | what is the diference between "Outbound Proxy" and "Proxy" in SPA-3102? |
| 00:12.55 | jcool | JT: yes, i really do consider clustering but one of the customer is asking how large it really was on a / server |
| 00:13.15 | jcool | genz: this is the close i get http://svn.digium.com/svn/zaptel/betas/1.2-hpec/ |
| 00:13.54 | JT | " / server" ? |
| 00:14.00 | jcool | on a per server |
| 00:14.04 | *** join/#asterisk JoseBravo (n=JoseDavi@190.9.74.206) |
| 00:14.17 | JoseBravo | How can I set time out of a time of in extensions.conf? |
| 00:14.21 | JT | look at the asterisk dimensioning page on the wiki |
| 00:14.39 | JoseBravo | t,1 and t,2 ? |
| 00:14.40 | __dante__ | im interested in improve cdr_radius support in asterisk 1.4, i have ported the implementation to 1.2 and modified to work properly with network failures |
| 00:14.44 | jcool | JT: i already did, but i can't seem to find the right answer for this large extension :) thanks |
| 00:14.54 | jcool | JoseBravo: could you please elaborate more |
| 00:15.10 | JT | jcool: the right answer is "who knows, if in doubt, get more servers" |
| 00:15.25 | genz | jcool: I just can't believe Digium sold me something that I can't even implement yet. |
| 00:15.46 | JT | genz: what is it? |
| 00:16.15 | jcool | genz: did you already called them for support? |
| 00:16.41 | genz | they stop answering at 6 |
| 00:16.42 | jcool | genz: if in 1.2 is in beta stage, what more in 1.4? |
| 00:16.55 | genz | jcool: and i bought it at 5:50 |
| 00:17.19 | jcool | genz: nice 1 :) |
| 00:18.03 | genz | jcool: the tech who told me to buy it "you'll have no problem installing it on your system" |
| 00:18.35 | jcool | genz: hehehe, why don't you try it first on 1.2 ? |
| 00:18.51 | JT | genz: what did you buy? |
| 00:18.57 | jcool | genz: since the driver is available at your own disposal even thou it's in beta stage |
| 00:19.02 | genz | jt: hpec license |
| 00:19.18 | JT | is it software only |
| 00:19.24 | genz | jt: yes |
| 00:19.29 | JT | ah ok |
| 00:20.15 | jcool | genz: this is really a new stuff :) can you point me on some reading material please about this? |
| 00:21.08 | genz | jcool: will zaptel 1.2 work ok with asterisk 1.4? |
| 00:21.14 | jcool | genz: definetly not |
| 00:21.21 | genz | that's what i thought |
| 00:21.48 | JT | do you need asterisk 1.4? |
| 00:22.11 | *** join/#asterisk jjshoe (n=jjshoe@72.54.121.98) |
| 00:22.16 | jcool | genz: yes that's exactly the question, do you need some feature for 1.4 that it doesn't have in 1.2? |
| 00:22.19 | genz | JT: I've changed enough of my configs to work with 1.4 that I'd rather not go back. |
| 00:22.37 | jjshoe | is there a way to get a beep on attended transfer complete when using the hard transfer buttons on phones and asterisk 1.2.14 ? |
| 00:23.53 | JT | genz: hrm, well that sucks, what sparked the migration? |
| 00:24.10 | genz | JT: Echo problems on our T1 PRI |
| 00:24.17 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 00:24.35 | genz | JT: 1.4 fixed it a lot, but didn't destroy them. Supposedly the HPEC on a TE210P is as good as the TE212P |
| 00:24.35 | JT | i don't think 1.4 would've done much for echo problems? |
| 00:24.36 | *** join/#asterisk bok (n=bok@proct.odynia.org) |
| 00:24.48 | genz | JT: The 1.4 zaptel drivers did. |
| 00:24.50 | JT | genz: you should just get hardware echo cancellation |
| 00:24.53 | JT | to get rid of it |
| 00:25.22 | bok | hey guys |
| 00:25.47 | bok | anyone seen a situation in 1.4.0 where the RealTime() application works fine, but the REALTIME() func fails with the same options? |
| 00:25.52 | jjshoe | genz get a sangoma t1 card with hw echo cancellation. |
| 00:25.58 | jcool | genz: did you do this ? hpec-8.20-i686.tar.gz |
| 00:26.16 | jcool | genz: did you try to download the file then put in on 1.4 zaptel then compile? |
| 00:26.18 | genz | jcool: Yes. But it requires base files in the build to include the .so |
| 00:26.43 | genz | jcool: Namely these - http://svn.digium.com/view/zaptel/branches/1.2/hpec/ |
| 00:27.28 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:27.38 | file | genz: yes, 1.4 does not have the echo cancellation changes needed for HPEC yet - but it's being worked on... I'm also talking to support to make sure they realize this |
| 00:28.01 | genz | jjshoe: And what exactly do you propose I do with my current TE210P? |
| 00:28.07 | jjshoe | genz throw it away |
| 00:28.25 | genz | (throws jjshoe away) |
| 00:28.39 | jcool | jjshoe: it's hard to throw it away on something that you already paid it for ? :) |
| 00:28.40 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 00:28.42 | jjshoe | ebay it, maybe someone else will want a t1 card with no echo can :P |
| 00:28.55 | file | if he waits HPEC will be available on 1.4 soon |
| 00:29.51 | file | genz: would you mind if I messaged you? |
| 00:29.57 | genz | file: If I port the 1.2 code, would Digium want it? Go right ahead |
| 00:32.57 | *** join/#asterisk tg (i=tg@x-net.hu) |
| 00:39.44 | *** join/#asterisk T-1 (n=T1@unaffiliated/t-1) |
| 00:39.49 | jjshoe | no thoughts on getting a beep in when someone does an attended transfer using a phone's transfer button? (and sip) |
| 00:40.26 | ManxPower | jjshoe: what brand of phone |
| 00:43.05 | *** join/#asterisk [Tesser] (n=Tesser@unaffiliated/tesser/x-000001) |
| 00:44.45 | bok | odd |
| 00:45.02 | bok | select statement from REALTIME doesnt even seem to reach the db |
| 00:45.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:46.33 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 00:48.28 | *** join/#asterisk Mike800 (n=mike800@cpe-76-167-156-224.socal.res.rr.com) |
| 00:49.13 | Mike800 | I'm having a problem with Asterisk and ${EPOCH}. For some reason, it thinks its the wrong time. Anyone around to help? |
| 00:50.09 | bok | timezone set wrong? |
| 00:50.14 | Mike800 | in voicemail.conf? |
| 00:51.58 | jjshoe | ManxPower any brand. |
| 00:51.59 | ManxPower | and "date" in a shell shows the correct time? |
| 00:52.15 | ManxPower | jjshoe: you want a beep or you are getting a beep you don't want? |
| 00:52.34 | Mike800 | ManxPower: in shell, 'date' is correct |
| 00:52.40 | jjshoe | ManxPower I would like a beep to signle to the party that they have completed the transfer |
| 00:52.45 | Mike800 | which is why I'm so confused :-) |
| 00:52.47 | *** part/#asterisk genz (n=chatzill@im.jobdig.com) |
| 00:52.48 | jjshoe | signal |
| 00:52.48 | jjshoe | wow |
| 00:52.49 | ManxPower | Mike800: then check the timezone stuff in voicemail.conf |
| 00:52.58 | ManxPower | jjshoe: the calling or called party? |
| 00:53.14 | jjshoe | called party |
| 00:53.31 | jjshoe | a calls b, a attends to c, c gets a beep just before b is bridged |
| 00:54.10 | ManxPower | no idea |
| 00:54.22 | ManxPower | I doubt it can be done without massive kluges |
| 00:54.23 | Mike800 | ManxPower: under [zonemessages] in voicemail.conf, I have it set to san-diego=America/Tijuana|'vm-received' Q 'digits/at' IMP |
| 00:54.46 | ManxPower | Mike800: good, now do you have a tz=san-diego on each mailbox line? |
| 00:55.13 | ManxPower | all your setting does is say what the timezone named san-diego means, it does not apply it to anything |
| 00:55.18 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 00:55.19 | *** join/#asterisk oQPa (n=uawename@233.Red-81-44-147.dynamicIP.rima-tde.net) |
| 00:55.28 | *** join/#asterisk Waverly360 (n=mirc@209.149.58.214) |
| 00:55.37 | Mike800 | ya |
| 00:55.49 | jjshoe | ManxPower :\ |
| 00:55.51 | Mike800 | so still, if i type ${EPOCH} it doesnt give the right time |
| 00:55.53 | jjshoe | seems fairly basic to want this |
| 00:56.00 | jjshoe | you can't tell when someone has completed the transfer without it |
| 00:56.33 | `Sauron | --> <-- that much |
| 00:56.42 | JT | asterisk uses UTC by default |
| 00:56.47 | `Sauron | bah |
| 00:56.47 | Waverly360 | Can anyone tell me why my Polycom phones can call each other no problem, but if I try to dial any other number, I get a fast busy? I never see the fast busy calls in asterisk. |
| 00:57.00 | JT | if you want a different timezone, you'll need to change it yourself i think |
| 00:57.32 | Mike800 | JT: how do I change it to PST? |
| 00:57.34 | jjshoe | Waverly360 not registered |
| 00:57.37 | jjshoe | Waverly360 type 'sip show peers' |
| 00:57.57 | *** part/#asterisk T-1 (n=T1@unaffiliated/t-1) |
| 00:58.03 | Waverly360 | jjshoe: The two phones I have connected are registered |
| 00:58.08 | Waverly360 | they just can't dial any other numbers |
| 00:58.16 | jjshoe | Waverly360 and nothing shows in asterisk? |
| 00:58.20 | Waverly360 | nope |
| 00:58.20 | JT | Mike800: perform an operation on the output of epoch to put the hour offset in |
| 00:58.23 | jjshoe | Waverly360 do you have a high enough debug level set? |
| 00:58.48 | cervi | Waverly360: set verbose set? |
| 00:59.00 | Mike800 | JT: hmm...ok...the weird thing is that it just changed today. The time was perfect and there was no problem. |
| 00:59.01 | jjshoe | I'm not buying what you're telling me :) |
| 00:59.04 | jjshoe | but if it's truly the case |
| 00:59.08 | Waverly360 | jjshoe, cervi: Just set verbosity and debug to 99..still get nothing |
| 00:59.10 | jjshoe | then you need to use something like ngrep to watch sip traffic |
| 00:59.14 | cervi | Waverly360: maybe you are in the wrong context or dont have permission |
| 00:59.15 | JT | Mike800: maybe it's something else then |
| 00:59.27 | Waverly360 | it's like the phones are shortcircuiting and just assuming nothing exists |
| 01:00.38 | Mike800 | Jt: its exactly 8 hours ahead of PST (5:00PM PST, 1:00AM asterisk time) |
| 01:00.47 | cervi | Waverly360: Do you see something with "sip debug" ? |
| 01:00.57 | *** part/#asterisk [Tesser] (n=Tesser@unaffiliated/tesser/x-000001) |
| 01:01.37 | ManxPower | *sigh |
| 01:02.05 | ManxPower | *sigh* Waverly360: 3724 => 1234,Jeanne Taravella,,,|tz=central |
| 01:02.20 | ManxPower | in your case it would be tz=san-diego |
| 01:02.45 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 01:02.50 | Qwell | san diego has it's own timezone now? |
| 01:02.56 | *** join/#asterisk gerphimum (n=trekkie@207.190.58.85) |
| 01:02.57 | jjshoe | helo this qwell |
| 01:03.19 | [hC] | is there a way to make format_mp3 not launch the mp3 for music on hold at the beginning of the file every damn time? |
| 01:03.21 | [hC] | thats rather annoying |
| 01:03.25 | Carp1 | Is there any kind of volume control on MOH? It says its playing but I can't hear anything on the other end. |
| 01:03.25 | JT | Mike800: what is the PST UTC offset? |
| 01:03.33 | Mike800 | My problem isn't with the voicemail. Its with all of asterisk |
| 01:03.36 | Mike800 | not sure |
| 01:03.41 | ManxPower | JT: -8 |
| 01:03.59 | JT | there you go Mike800 |
| 01:04.15 | JT | lol how can you live in a timezone and play with the time without knowing the utc offset :P |
| 01:04.23 | JT | important in unix systems and the Internet |
| 01:04.37 | Carp1 | hmm |
| 01:04.41 | Mike800 | oh..haha...i didnt know thats what you were asking for |
| 01:05.15 | Mike800 | so how do I set all of asterisk within PST? |
| 01:06.04 | Carp1 | I just got a Polycom 501 today and I press the transfer button and transfer to 700 (parking......*700 doesnt work for some reason ??) and it says 701 and MOH starts playing, but its playing on the phone where I transfered from, not the other one...also when I hang up the phone I transfered from, it hangs p the channel the other call is on |
| 01:06.20 | ManxPower | Mike800: most of asterisk uses the system timezone, but since voicemail users could be in different timezones |
| 01:06.38 | Mike800 | my problem isnt with voicemail |
| 01:07.24 | Carp1 | Any idea's on that? |
| 01:08.05 | Mike800 | basically, heres the problem is that I am monitoring all the phone calls. The file name of the recording contains the date and time (${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)-${CALLERID(num)}}.WAV). The file names are all wrong because ${EPOCH} is wrong. |
| 01:08.24 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 01:09.03 | JT | dud you change the way the system stores the time? |
| 01:09.08 | JT | in linux |
| 01:10.03 | Mike800 | when i type "date" in the shell, it gives me the proper date/time |
| 01:10.24 | ManxPower | Mike800: the epoch is the number of seconds since midnight jan 1 1970, I doubt it is wrong |
| 01:10.25 | JT | using that time function has always given me utc |
| 01:10.33 | Mike800 | but im pretty sure when I installed linux, I checked off that box that says "system clock uses utc" |
| 01:11.25 | ManxPower | <PROTECTED> |
| 01:11.25 | ManxPower | <PROTECTED> |
| 01:11.26 | ManxPower | <PROTECTED> |
| 01:11.47 | Mike800 | ? |
| 01:12.08 | ManxPower | or %+ The date and time in date(1) format. (TZ) |
| 01:12.27 | ManxPower | Mike800: the STRFTIME function is based on the C strftime function. See "man strftime" |
| 01:13.01 | Mike800 | ahh...ok |
| 01:15.54 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 01:17.28 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:18.35 | *** join/#asterisk oQPa (n=uawename@233.Red-81-44-147.dynamicIP.rima-tde.net) |
| 01:19.36 | *** part/#asterisk oQPa (n=uawename@233.Red-81-44-147.dynamicIP.rima-tde.net) |
| 01:21.13 | Carp1 | voicemail asks for mailbox and password... |
| 01:21.19 | Carp1 | says login is incorrect |
| 01:21.22 | Carp1 | but I know its not. |
| 01:23.38 | ManxPower | Carp1: What device are you calling from? |
| 01:23.58 | Carp1 | Its working now, but I dont know why |
| 01:24.05 | Carp1 | I tried like 10 times and it was wrong info each time |
| 01:24.11 | Carp1 | then I just tried again and it works |
| 01:24.28 | Carp1 | I really am wondering about the call parking, its weird. |
| 01:25.34 | *** join/#asterisk HellHound (n=hellhoun@d54C5F86E.access.telenet.be) |
| 01:28.25 | ManxPower | Carp1: try ending the mailbox number and password with a # |
| 01:31.35 | Carp1 | I got it working. |
| 01:31.49 | Carp1 | I have another question, what priority is timeout? |
| 01:31.59 | Carp1 | like after 10 seconds, i want to send to voicemail....102? |
| 01:38.31 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 01:42.42 | *** join/#asterisk shwa (n=shwa@ip-62-235-203-59.dsl.scarlet.be) |
| 01:42.47 | shwa | salut |
| 01:43.29 | JT | Carp1: what did you need to do to get it working? |
| 01:46.03 | *** join/#asterisk D3V|L (n=d3v@200.118.122.44) |
| 01:46.12 | D3V|L | good night |
| 01:46.32 | JT | already? |
| 01:46.47 | D3V|L | I'm searching for a click to talk solution |
| 01:47.59 | *** part/#asterisk notoriousrab (n=robert_m@207.47.34.74.static.nextweb.net) |
| 01:49.04 | *** part/#asterisk D3V|L (n=d3v@200.118.122.44) |
| 01:49.18 | J4k3 | hah |
| 01:49.26 | J4k3 | that was confusing |
| 01:52.08 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 01:57.37 | *** join/#asterisk mercestes (n=merceste@inet.hou.devry.net) |
| 01:59.19 | *** part/#asterisk mercestes (n=merceste@inet.hou.devry.net) |
| 02:01.23 | ManxPower | Mozilla put the confirmation message from the FTC USA Do Not Call List into my Junk folder. |
| 02:05.08 | Carp1 | I just got a Polycom 501 today and I press the transfer button and transfer to 700 (parking......*700 doesnt work for some reason ??) and it says 701 and MOH starts playing, but its playing on the phone where I transfered from, not the other one...also when I hang up the phone I transfered from, it hangs p the channel the other call is on |
| 02:06.46 | ManxPower | Carp1: you need to press the transfer button a 2nd time after you hear the number |
| 02:07.19 | ManxPower | You also need to get a manual for the phone, it should have come on a CD with the phone |
| 02:08.10 | Carp1 | That didnt work. |
| 02:08.15 | Carp1 | It didnt come woth one |
| 02:08.20 | Carp1 | Everything looked new though. |
| 02:08.30 | *** join/#asterisk X-Rob_ (n=Rob@CPE-58-169-113-104.vic.bigpond.net.au) |
| 02:08.45 | Carp1 | After I hit transfer, I get dial tone, I dial 700, and then the send button |
| 02:08.53 | Carp1 | THen it starts music on hold |
| 02:09.00 | Carp1 | but only on the phone I tried to park the call from. |
| 02:09.17 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 02:09.27 | Carp1 | Then when I hangup, it it says I got tired of being parked and hangs everything up |
| 02:11.18 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 02:11.19 | elriah | Hey guys, in a situation where it's polycom_phone->nat_gateway->public_internet->public_asterisk_box, when one polycom phone calls another inside that source network, after the call is established will they be talking to each other? I know this is how it works when the asterisk server is on the lan, but when using nat=yes in this scenerio, what happes? Is everything bridged through asterisk? |
| 02:11.28 | elriah | And what about conference calls? bridged through asterisk? |
| 02:12.19 | *** join/#asterisk SECGOD (i=SECGOD@71.57.36.106) |
| 02:12.38 | ManxPower | Carp1: once you press transfer the 2nd time the parking is complete |
| 02:13.01 | ManxPower | elriah: it should work |
| 02:13.10 | Carp1 | ManxPower, I tried. |
| 02:13.21 | ManxPower | Carp1: then something else is wrong. |
| 02:13.28 | ManxPower | I have over 100 polycoms and you can park just gine. |
| 02:13.31 | elriah | If I have nat=yes, the Polycom phones should talk to each other instead of going through asterisk, inside the private lan? |
| 02:13.51 | elriah | What happens in the case of conferecing a third party? |
| 02:13.57 | ManxPower | elriah: That wasn't your question 8-) with canreinvite=yes they should talk directly |
| 02:14.04 | Carp1 | When I press a second time it says "Notify answer on owned channel?" |
| 02:14.09 | JerJer | its not nat=yes, its the value of your canreinvite |
| 02:14.13 | ManxPower | Carp1: using 1.4? |
| 02:14.19 | elriah | Oh! Got ya. Great, thanks! |
| 02:14.31 | Carp1 | How do I check the version? |
| 02:14.36 | Carp1 | I installed new source 23 days ago. |
| 02:14.44 | ManxPower | "show version" in the asterisk CLI |
| 02:14.50 | elriah | When would you ever NOT want to use canreinvite=yes? |
| 02:14.59 | Carp1 | No such command |
| 02:15.36 | ManxPower | Carp1: then you are using 1.4. you either need to use 1.2 or upgrade to the Asterisk SVN. 1.4.0 is so full of bugs as to not be usable in many situations like yours |
| 02:16.18 | *** join/#asterisk MaartenB_ (n=Maarten@84-105-196-31.cable.quicknet.nl) |
| 02:16.40 | Carp1 | Ok, I think there is a small tutorial on the asterisk site on how to update with SVN? |
| 02:16.49 | ManxPower | elriah: when you have two phones or sip devices behind DIFFERNET nats |
| 02:17.12 | ManxPower | Carp1: I assume so. I run production systems and don't use SVN as I like to keep my job |
| 02:17.29 | elriah | Wouldn't it just not succeed on the reinvite and bridge anyway? Or would canreinvite=yes just simply break that call in your scenerio? |
| 02:17.37 | Carp1 | Lol, compile from source? |
| 02:17.42 | ManxPower | elriah: it would break the audio |
| 02:19.45 | elriah | sum beotch! We have like 80 or so phones with canreinvite=no in about 11 different locations all going to a public asterisk server. |
| 02:19.53 | elriah | ManxPower: is there a canreinvite=auto? lol |
| 02:20.11 | ManxPower | nope |
| 02:20.57 | elriah | So in this scenerio, if they are calling extensions in other locations, canreinvite=no, if not, canreinvite=yes is absolutely the best scenerio, right? |
| 02:21.20 | JerJer | other locations would need to be ran thru its own type=peer entry |
| 02:21.24 | JerJer | with canreinvite=no |
| 02:21.45 | JerJer | then leave canreinvite on for each local device |
| 02:22.23 | elriah | So is the canreinvite=yes setting in sip.conf or is there a phone setting as well? (thanks for the help on this guys) |
| 02:22.42 | JerJer | canreinvite defaults to ye s |
| 02:22.43 | JerJer | yes |
| 02:22.47 | elriah | Ah! |
| 02:22.48 | elriah | Cool. |
| 02:24.18 | elriah | In asterisk, in the case of Polycom phones, if I change from canreinvite=no to yes, will it pick up the setting on the next registration or does it need to be restarted? |
| 02:24.20 | elriah | (phone) |
| 02:24.38 | *** part/#asterisk j0anna (n=joanna@222.126.13.68) |
| 02:26.34 | Carp1 | How do I uninstall 1.4? |
| 02:27.01 | elriah | sudo rm -fR / |
| 02:27.08 | ManxPower | elriah: "sip reload" |
| 02:27.10 | elriah | no, not really |
| 02:27.14 | *** kick/#asterisk [elriah!n=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 02:27.18 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 02:27.22 | elriah | Oh come on I was joking. |
| 02:27.31 | Qwell | next time somebody does that, I will include the ban |
| 02:27.44 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 02:27.48 | *** part/#asterisk SECGOD (i=SECGOD@71.57.36.106) |
| 02:27.49 | elriah | I appologize, it was wrong, my bad. |
| 02:27.52 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:27.53 | brettnem | hi all |
| 02:28.14 | brettnem | hey, is there any way to pass arbitrary data in an IAX call, like how you can with sip headers? |
| 02:28.14 | *** join/#asterisk joaovianna (i=joaovian@ool-4354d1a8.dyn.optonline.net) |
| 02:28.51 | ManxPower | brettnem: I believe IAX will pass __ prefixed channel variables. |
| 02:28.56 | JerJer | brettnem: anything with __ |
| 02:29.01 | JerJer | damn't you are fast |
| 02:29.06 | brettnem | oh neat.. even in 1.2? |
| 02:29.58 | brettnem | so on the receiving side, it's just seen as a chan var eh? |
| 02:30.37 | wunderkin | no... that was something special added in 1.4 somewhere |
| 02:30.37 | Qwell | I thought it was just a patch on mantis? |
| 02:30.39 | Qwell | marked post 2.0 |
| 02:30.47 | joaovianna | Hi Gurus ! I'm receiving "Unknown RTP codec 126" when I try to make a call from a Grandstream video 3000 using asterisk... Any clue ? |
| 02:30.48 | ManxPower | brettnem: if it is supported it would show up as a normal channel variable on the far end |
| 02:30.53 | wunderkin | maybe... i thought it as added... i dunno.. :D |
| 02:31.20 | ManxPower | Carp1: try "make uninstall" |
| 02:31.39 | brettnem | yaeh, I thought I saw it marked post 2.0 |
| 02:31.48 | ManxPower | BTW, does anyone know if DUNDi would work for SIP in addition to IAX? |
| 02:32.09 | n|cotine | ManxPower: voip-info has a sample config using SIP, I believe. |
| 02:32.17 | joaovianna | Anyone using video in * ? |
| 02:32.22 | ManxPower | n|cotine: Oh? Cool. |
| 02:32.35 | JerJer | i think my asterisk box still has a video card |
| 02:32.45 | Qwell | pringles minis...wtf |
| 02:32.55 | n|cotine | 2nd the wtf |
| 02:32.55 | Qwell | wtf is the point? |
| 02:33.10 | JT | when does the Pringles Nanos or Pringles Shuffle come out? |
| 02:33.16 | ManxPower | to make more money |
| 02:33.22 | Qwell | JT: I'm holding out for the pringles video |
| 02:33.28 | brettnem | does anyone know if you can auth a SIP call with RSA keypair? |
| 02:33.49 | JT | shilouettes of people stuffing their faces with pringles? |
| 02:34.44 | ManxPower | n|cotine: "Note: This configuration is currently non-functional, as chan_sip does not support "dbsecret" at this time." |
| 02:34.49 | ManxPower | I'll have to check |
| 02:36.13 | ManxPower | last modified 2 years ago. Hmm |
| 02:36.46 | n|cotine | ManxPower: no dbsecret for sip in branch 1.4 |
| 02:36.56 | JerJer | one can query using DUNDi but there is no authentication |
| 02:37.00 | ManxPower | *grumble* |
| 02:37.19 | perd | crocodile dundi is better, it comes with a whip and a snazzy hat |
| 02:37.19 | ManxPower | No auth should be OK, as actual calls would still be authed. |
| 02:37.38 | n|cotine | And at this stage, a walker |
| 02:38.10 | n|cotine | ManxPower: Like it says, you could try it with the auth details in the mappings themselves. |
| 02:38.43 | ManxPower | as long as the calls are authed, I'm not terribly worried about exposing my dialplan |
| 02:39.35 | ManxPower | most of my servers are behind firewalls with no port forwarding and the 1 that isn't I can use packet filtering |
| 02:41.23 | ManxPower | heck, my dialplan is exposed using the current system of ENUM |
| 02:43.42 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 02:46.10 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 02:46.44 | DocHolliday | anyone have a default configurating file for the Cisco 7941 / 61? |
| 02:46.50 | DocHolliday | *configuration |
| 02:47.28 | elriah | DocHolliday: Man, am I gonna save you some time and hassle. |
| 02:47.43 | DocHolliday | elriah, you are my hero. |
| 02:48.00 | elriah | DocHolliday: First, half the stuff you read on voip-info is for the 79x0's and the 79x1's have different firmware. |
| 02:48.11 | DocHolliday | yeah i know i'm having a bitch of a time |
| 02:48.14 | DocHolliday | care to PM me? |
| 02:48.20 | elriah | DocHolliday: Did you compile asterisk from source? |
| 02:48.29 | DocHolliday | oh its an existing asterisk isntall |
| 02:48.38 | DocHolliday | i have Cisco 7940's configured and working |
| 02:49.17 | elriah | DocHolliday: Ok, first things first, you'll have to modify channel_sip.c and take out a string that says (0/0) and recompile. Without doing this, the Cisco MWI will never work on the 79x1's, regardless of firmware revisions. |
| 02:49.36 | elriah | What exact version of asterisk do you have? My compiled chan_sip.c may work for you as a drop in. |
| 02:50.19 | elriah | Secondly, forget about NAT, just period, lol |
| 02:50.21 | DocHolliday | Connected to Asterisk 1.2.11 |
| 02:50.31 | DocHolliday | its a local asterisk server? |
| 02:50.49 | Carp1 | Ok, I downgraded to 1.2 |
| 02:51.07 | DocHolliday | 1.2 is soo stable, i really dont want to upgrade |
| 02:51.09 | elriah | Ok, I have 1.2.13, not sure if chan_sip.c is different, do you have your sources? |
| 02:51.25 | Carp1 | Now I'm getting no music...it started and stopped music like 6 times in a row on CLI |
| 02:51.51 | elriah | DocHolliday: PM me your email address, I'll send you some files that will get you started (i.e., the CORRECT cisco firmware for the 79x1 phones) |
| 02:52.02 | DocHolliday | /usr/src/asterisk-1.2.10 |
| 02:52.15 | elriah | hrm... But you say it's running 1.2.11? |
| 02:52.22 | *** join/#asterisk umop3plsdn (n=da@cpe-76-179-77-186.maine.res.rr.com) |
| 02:52.32 | Carp1 | now when I hit the transfer button for hte second time, it doesnt tell me which extension its parked on |
| 02:52.35 | DocHolliday | yeah, is it necessary to upgrade? |
| 02:52.36 | Carp1 | it tells the other caller |
| 02:52.54 | elriah | DocHolliday: Are yours SIP or Skinny firmware? |
| 02:54.12 | DocHolliday | elriah, SCCP but i want to switch over to SIP |
| 02:54.17 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 02:54.33 | elriah | You'll need the SIP firmware. I have 8.2.1, works great. |
| 02:55.30 | Carp1 | Anyone here have SellVoIP service? |
| 02:55.42 | *** join/#asterisk Avochelm (n=damien__@gw-morphett.koalatelecom.com.au) |
| 02:55.48 | DocHolliday | thank you very much |
| 02:56.17 | DocHolliday | any idea why Asterisk doesn't support it out of the box? |
| 02:57.13 | elriah | DocHolliday: Like I said before, the MWI (message waiting light) won't work without a modification to chan_sip.c, removing the string "(0/0)" and recompiling. You don't have to re-install asterisk, just dropping the new chan_sip.o (i think it's .o) over top of the old one and restart as long as the versions match. |
| 02:57.21 | elriah | DocHolliday: Support what? |
| 02:57.32 | Qwell | broken implementations of SIP? no |
| 02:57.35 | Qwell | not all of them |
| 02:57.37 | JT | .so |
| 02:58.11 | DocHolliday | elriah, gotcha (the cisco 7941) |
| 02:58.14 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 02:58.52 | DocHolliday | should i upgrade to the version you've got (of asterisk) |
| 02:58.55 | elriah | The files I sent you, just drop them in your TFTP server home directory. Modify the SEP<mac>.cnf.xml to suite your needs, it's pretty easy to follow. Of course, replace the mac address with the mac address of your phone. |
| 02:59.17 | DocHolliday | elriah, perfect! and in conjunction with the Firmware it will load the SIP stuff? |
| 02:59.22 | elriah | DocHolliday: It's probablly an easy upgrade from 1.2.11 to 1.2.15 (latest I think), so I say go for it and make this change before you compile. |
| 02:59.29 | elriah | DocHolliday: Right. |
| 02:59.37 | n|cotine | elriah: I might end up with a 79x1 here in the near future, would it be possible to get a copy of your little care package? :) |
| 02:59.44 | Qwell | 07 |
| 02:59.49 | Qwell | erm |
| 03:00.02 | elriah | If you have problems with modifying chan_sip.c, send it to me and I'll modify it and send it back. |
| 03:00.12 | Qwell | xchat != mythfrontend |
| 03:01.05 | elriah | Mine is for 1.2.13 and to play it safe I would definitely use the one that comes with your revision of asterisk. |
| 03:01.14 | JT | Qwell: only just noticed? ;) |
| 03:01.31 | Qwell | JT: I thought it was that new text drama show |
| 03:01.42 | JT | ah right |
| 03:02.58 | ez` | <PROTECTED> |
| 03:03.07 | ez` | is it normal ? |
| 03:03.11 | DocHolliday | elriah, okay i have copied and pasted the ZIP file into my TFTP root |
| 03:03.12 | DocHolliday | did you by chance send me a default SEP file? |
| 03:03.40 | elriah | Oops! Nope, just a sec. (it's small) |
| 03:03.45 | DocHolliday | sure :) |
| 03:03.49 | DocHolliday | by the way you are a life saver |
| 03:04.21 | JT | ez`: zomething in your dialplan is obviously querying the astdb for PARK/79 |
| 03:04.22 | [TK]D-Fender | ez`: Its your dialplan... where do you THINK you're setting it? |
| 03:04.39 | ez` | k |
| 03:06.11 | elriah | DocHolliday: Hey, your email box is over quota and my last email got rejected... Let me know when to resend... |
| 03:06.23 | DocHolliday | LOL |
| 03:06.30 | DocHolliday | you are taking up all my space! |
| 03:06.53 | DocHolliday | elriah, please try again :) |
| 03:07.47 | Carp1 | maybe my call parking problem is becuase I don't have a t or a T anywhere? |
| 03:07.57 | elriah | DocHolliday: Resent. |
| 03:08.09 | elriah | n|cotine: You want the firmware/configs as well? PM me your email please. |
| 03:08.27 | DocHolliday | elriah, i dont' know if its advisable to change the OS79XX file for my entire fleet of phones? |
| 03:08.29 | [TK]D-Fender | Carp1 :Quite possibly... |
| 03:08.51 | Carp1 | external extensions should have tT? |
| 03:08.54 | n|cotine | Don't the newer versions of the SIP firmware not check that file anymore? |
| 03:08.59 | DocHolliday | is there anyway i can set this phone to a different directory? |
| 03:09.10 | Carp1 | beucase it doesnt matter who called who internally...so they should both be able to transfer? |
| 03:09.59 | elriah | DocHolliday: You mean in your TFTP dir? I haven't found a way... |
| 03:10.06 | DocHolliday | ah okauy |
| 03:10.42 | [TK]D-Fender | Carp1 : No, I don't think INBOUND callers should have the right to transfer themselves or park other people... |
| 03:11.53 | Carp1 | Ok. |
| 03:11.56 | Carp1 | So just lower case? |
| 03:12.33 | elriah | n|cotine: Which file? |
| 03:12.41 | elriah | n|cotine: And I still need your email address |
| 03:12.43 | n|cotine | OSXX |
| 03:12.48 | elriah | n|cotine: No, they do not. |
| 03:13.18 | n|cotine | elriah: PM'd. |
| 03:13.24 | elriah | n|cotine: Also, you'll need to modify chan_sip.c and remove the text "(0/0)" for MWI to work on Cisco firmware 8.3 or better. This change doesn't affect other phones as far as I can tell. |
| 03:14.02 | elriah | I think the (0/0) is a placeholder for old/new messages. |
| 03:14.35 | DocHolliday | elriah, lets do one step at a time (bare with me), i changed the IP in the SEP file, and changed the name of said file to my MAC ID |
| 03:14.46 | elriah | SEP<mac>.cnf.xml |
| 03:14.48 | DocHolliday | i have also temporarily modified my TFTP root to reflect a temporary dir for the 7941 |
| 03:14.49 | joaovianna | anyone using video / h264 in * ? |
| 03:14.54 | DocHolliday | elriah, exactly, done that |
| 03:15.00 | elriah | Make sure its SEP and not SIP, a change from the 79x0's to 79x1's. |
| 03:15.12 | DocHolliday | sure! |
| 03:15.15 | *** join/#asterisk xsquared (n=scead@202.10.84.172) |
| 03:15.37 | DocHolliday | Okay, i have *not* yet provisioned a SIP registration on Asterisk but that can wait? |
| 03:15.38 | elriah | And if you're using DHCP, give it an option 66 or option 150 and point it to your TFTP server. |
| 03:15.53 | elriah | And did I mention forget about NAT? |
| 03:15.54 | DocHolliday | yeah i cant do that but i used 'alternative TFTP' |
| 03:16.13 | DocHolliday | yes you did, if the phone and asterisk are on the same subnet i'm fine? |
| 03:16.16 | elriah | DocHolliday: Are you using a windows combo tftpd/dhcpd like solarwinds? |
| 03:16.32 | DocHolliday | nope, DHCP is running on my Router, TFTPD is running using SolarWinds |
| 03:16.44 | xsquared | hi, i don't know whether this software is what I want or not. I'd like to use conventional telephone lines with a piece of software that can handle the incoming calls and play music if they're on hold etc.. Am I in the right place? |
| 03:16.46 | elriah | Got ya. |
| 03:16.54 | DocHolliday | elriah, can i safely restart the phone? |
| 03:17.02 | elriah | Yea, it works great. |
| 03:17.23 | DocHolliday | *crosses fingers* |
| 03:17.24 | elriah | I love our 7941's, I HATE the fact that NAT is broke. |
| 03:17.49 | [TK]D-Fender | xsquared : Yes. * can be used as a full-service PBX, managing multiple kinds of phones & lines and being able to process calls from any source in many popular ways |
| 03:18.00 | elriah | DocHolliday: it's really easy to replace ringtones and background images on the phone. If it was $50 cheaper, worked with NAT, and used FTP for firmware, it would be my #1 choice. |
| 03:18.16 | elriah | Are you going to modify chan_sip.c? |
| 03:18.18 | xsquared | [TK]D-Fender, ok, great, i wasn't sure if this was voip only or not |
| 03:18.24 | DocHolliday | okay, it said mk-sccp.jar file not found |
| 03:18.24 | [TK]D-Fender | elriah : Oh... you mean like Polycom? ;) |
| 03:18.42 | Qwell | polycom sucks! |
| 03:18.45 | Qwell | yeah, I said it |
| 03:18.46 | elriah | [TK]D-Fender: I applaud you, sir, for waiting as long as you did to RUB IT IN (you bastard) lol |
| 03:18.47 | [TK]D-Fender | xsquared :It can be all-VoIP, or all TDM, or anywhere in-between |
| 03:18.48 | Qwell | what now? |
| 03:19.28 | Carp1 | On a Polycom 501, does anyone know how to program the " |
| 03:19.34 | DocHolliday | elriah, apparently i'm missing files :( |
| 03:19.35 | Carp1 | messages" button |
| 03:19.41 | Carp1 | to dial the voicemail ext |
| 03:19.52 | xsquared | great. My mum is opening a new business and she was looking into a commercial serivce that provided telephone switching and extensions |
| 03:19.54 | elriah | DocHolliday: No, that's normal. Which file extension did it report missing? |
| 03:19.56 | [TK]D-Fender | Carp1 : Its all nice & layout out in the admin guide |
| 03:20.01 | xsquared | thats why i stopped her and looked around first |
| 03:20.02 | DocHolliday | it took the SEP file, but it wants a .tlv, the mk-sccp.jar and g3-tones.xml |
| 03:20.13 | elriah | [TK]D-Fender: Hey, I did get that 650HD phone in today. It looks way cool, but haven't configured it yet. |
| 03:20.20 | elriah | Yea, ignore those. |
| 03:20.39 | DocHolliday | okay, but the firmware didn't switch :P |
| 03:20.46 | [TK]D-Fender | xsquared : * is often much cheaper, and if not, at least much more flexible & featureful. |
| 03:21.03 | elriah | Oh, you need to power off and then back on. A soft-reset won't do it with the 79x1's, which apparently is also different from the 79x0's. |
| 03:21.07 | JT | xsquared: hrm, how many lines you looking at? |
| 03:21.10 | elriah | Actually yank the power. |
| 03:21.33 | elriah | Oh, and hold down # on boot. |
| 03:21.33 | DocHolliday | elriah, i unplugged it then plugged it back in.. |
| 03:21.33 | DocHolliday | oh :P |
| 03:21.34 | [TK]D-Fender | Qwell : And yeah... we heard you SAY it... we also know better than to believe you MEAN it ;) |
| 03:21.38 | elriah | When the lights start flashing, enter 123456789*0# |
| 03:21.39 | Qwell | :P |
| 03:21.48 | elriah | This will reset to factory and upgrade the firmware. |
| 03:21.49 | xsquared | well, only 1. 2 lines in total (phone and fax & direct deposit machine) |
| 03:21.59 | JT | oh ok |
| 03:22.02 | [TK]D-Fender | elriah : 8-6-7-5-3-0-9? |
| 03:22.03 | elriah | By the way, to soft reset, hit the option button and then **#** |
| 03:22.03 | *** join/#asterisk kuto (n=h57ye@58.69.158.114) |
| 03:22.12 | xsquared | unless * can handle faxs too :P |
| 03:22.17 | elriah | lol |
| 03:22.18 | *** join/#asterisk mxyoung (n=mxyoung@mail.netlogic.net) |
| 03:22.24 | JT | best to leave fax and eftpos seperate to asterisk |
| 03:22.34 | [TK]D-Fender | xsquared : I wouldn't if I were you. I'd leave fax & ATM on their own line. |
| 03:22.42 | xsquared | thats what im planning to do |
| 03:22.46 | DocHolliday | yp, yello lights blinking nothing happening |
| 03:22.47 | kuto | hi people, im trying to register but could not locate the registration page..any idea? |
| 03:22.51 | xsquared | i just wanted something to handle the phone calls |
| 03:22.56 | [TK]D-Fender | xsquared : For verything else, there's Asterisk :) |
| 03:23.02 | kuto | hi people, im trying to register to www.asterisk.org but could not locate the registration page..any idea? |
| 03:23.02 | xsquared | :) |
| 03:23.05 | JT | xsquared: you sure 1 line is enough for customer enquiries? |
| 03:23.16 | DocHolliday | elriah, can i pm you again? |
| 03:23.20 | xsquared | for now i'll have to do |
| 03:23.23 | elriah | DocHolliday: You'll notice that these procedures aren't anywhere out there and a bit different than the 79x0's. I've been meaning to go build on the wiki but haven't had time. |
| 03:23.31 | elriah | Sure. |
| 03:23.42 | JT | xsquared: check if you're in an optus exchange coverage area, it's much cheaper than telstra |
| 03:23.46 | JT | $20/mo/line |
| 03:24.11 | xsquared | we are in an optus exchange coverage area, but she's already gone telstra |
| 03:24.17 | xsquared | she did some deal |
| 03:24.28 | JT | optus will change you over for free |
| 03:24.41 | JT | i doubt telstra do lines for less than $20/mo :P |
| 03:24.41 | xsquared | i'll look into it, thanks |
| 03:24.51 | JT | only problem is if she's in a contract |
| 03:26.03 | xsquared | hmm, okay |
| 03:27.31 | mxyoung | Anybody seen this message before: WARNING[3139]: translate.c:163 framein: no samples for lintoulaw |
| 03:27.48 | mxyoung | Ast 1.4 server running meetme keeps crashing, that is the only thing at the end of the log |
| 03:27.50 | xsquared | i have another few questions, just so i understand the basic concept... 1. how would it actually work if i wanted to hook it up to the conventional phone line? Does the computer need some special pci card? |
| 03:28.09 | JT | mxyoung: do you have zap hardware or ztdummy? |
| 03:28.13 | *** join/#asterisk Fr0zen_ (i=Fr0zen_@67.175.92.171) |
| 03:28.20 | mxyoung | JT: Sangoma A101 |
| 03:28.30 | Fr0zen_ | anyone here use a Cisco 7970g with asterisk? |
| 03:28.32 | JT | xsquared: yes or external hardware |
| 03:28.43 | xsquared | I have a conexant card here with 2 telephone sockets in it |
| 03:28.46 | xsquared | will that do? |
| 03:28.51 | JT | no. |
| 03:29.14 | xsquared | what would i need then? |
| 03:29.25 | JT | a tdm400p or similar pci card |
| 03:29.40 | JT | or a sipura spa-3102 or similar external ATA |
| 03:30.38 | xsquared | eeek. will most computer hardware shops have this card? |
| 03:30.45 | JT | no |
| 03:30.50 | DocHolliday | elriah, it restarted for the second time |
| 03:30.53 | JT | telephony is not meant to be done on the cheap |
| 03:31.00 | JT | ask a pabx company how much they want |
| 03:31.01 | DocHolliday | but it appears to be doing nothing |
| 03:31.13 | JT | because you will realise it's not cheap |
| 03:31.49 | elriah | DocHolliday: Just let it go... |
| 03:31.57 | elriah | DocHolliday: It's working, watch your tftpd log... |
| 03:32.06 | DocHolliday | hrmm, for how long ~? |
| 03:32.17 | DocHolliday | right now nothing is happening in TFTP |
| 03:32.22 | elriah | DocHolliday: It first installs a universal bootloader, reboots, installs something else, etc. |
| 03:32.31 | elriah | Formats its filesystem, bla bla |
| 03:32.38 | elriah | I think it took about 7-10 minutes on my phones. |
| 03:32.42 | DocHolliday | i understand, just wondering why there is no TFTP activity |
| 03:32.45 | DocHolliday | ah! |
| 03:32.55 | elriah | DocHolliday: patients, padiwan learner... |
| 03:32.58 | elriah | lol |
| 03:33.32 | DocHolliday | lol :( its just the 40s are a bit more instant so to speak |
| 03:33.33 | xsquared | whats the difference between FXO and FXS? |
| 03:33.39 | JT | ~fxofxs |
| 03:33.41 | jbot | extra, extra, read all about it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 03:34.02 | elriah | FXO = plugs into the wall, FXS = plugs into a phone |
| 03:34.07 | DocHolliday | elriah, 4th restart.. weird? |
| 03:34.23 | elriah | DocHolliday: Nope. |
| 03:35.23 | elriah | WooHoo! We have E911 now! |
| 03:35.31 | xsquared | oh, so what do I need? :S |
| 03:35.51 | Bobthehunter | think ofthis FXORIGINATED FXSUPPLIED |
| 03:36.07 | elriah | DocHolliday: Gotta run, email me if you have problems or find a NAT solution. |
| 03:36.17 | DocHolliday | elriah, i am just wondering why its restarted 5 times with no TFTP activity |
| 03:36.21 | kuto | i got some question, does source installation config files are the same with asterisk 1.1.15 config files, unlike trixbox that have its own config files? |
| 03:36.38 | Bobthehunter | 1.1.15 ? or 1.2.15 |
| 03:36.39 | Bobthehunter | ;) |
| 03:36.44 | DocHolliday | hope he didn't brick my phone |
| 03:36.57 | kuto | eer 1.2.15 |
| 03:37.03 | Carp1 | This transfer problem is driving me nuts....I cant figure out what it wrong. |
| 03:37.24 | JT | Bobthehunter: easier to thinking of it as Office and Station :P |
| 03:37.24 | xsquared | Bobthehunter, i don't really know what that means either |
| 03:37.34 | mxyoung | DocHolliday: When my phone reboots over and over it means it can't find the tftp server.... what kind of phone do you have? |
| 03:37.54 | DocHolliday | mxyoung, Cisco 7941 |
| 03:37.55 | xsquared | im new to telephony |
| 03:37.56 | JT | xsquared: you need a TDM400P or similar with 1FXS module and 1FXO module |
| 03:38.04 | JT | xsquared: or a sipura spa-3102 |
| 03:38.08 | DocHolliday | so its not defaulting to the TFTP server iset? |
| 03:38.21 | *** join/#asterisk sharp (n=sharp@2001:470:1f01:ffff:0:0:0:1c23) |
| 03:38.22 | JT | xsquared: you in sydney? |
| 03:38.26 | xsquared | brisbane |
| 03:38.27 | Bobthehunter | Well the signal /supply is FX Originated by other end.. or FX Supplied to other end |
| 03:38.37 | mxyoung | DocHolliday: Yeah, I switched to Polycoms a few years ago b/c I hated dealing w\Cisco. Hang on, let me find my old docs about it |
| 03:38.40 | xsquared | why do i need both modules? |
| 03:38.41 | JT | ah ok |
| 03:38.50 | Bobthehunter | so FXO = POTS / FXS = VOIP more or less |
| 03:39.00 | JT | Bobthehunter: wtf that's bullshit |
| 03:39.01 | DocHolliday | mxyoung, i set the TFTP server in the configuration file but has it defaulted to the one on my DHCP server? |
| 03:39.11 | Bobthehunter | its just somethignto remember easier |
| 03:39.16 | JT | xsquared: 1 connects to a phone line, 1 connects to a fine |
| 03:39.19 | mxyoung | DocHolliday: Did you get an IP via DHCP? |
| 03:39.26 | JT | Bobthehunter: no, that makes no sense |
| 03:39.31 | DocHolliday | yes, but i set an alternate TFTP server |
| 03:39.33 | JT | s/fine/line/ |
| 03:39.47 | Bobthehunter | yes but 48 volt is supplied by other end on FXO 's and supplied by yourself on FXS's |
| 03:39.50 | mxyoung | DocHolliday: might get overwritten. Can you set the tftp server in the DHCP settings? |
| 03:39.54 | joaovianna | elriah: I need 911 for my clients... Can you sujest one company where I can pay by did ? |
| 03:40.04 | JT | Bobthehunter: i know that, has NOTHING to do with VoIP |
| 03:40.08 | DocHolliday | naw, its just a regular router? |
| 03:40.31 | Bobthehunter | yes i know just that most voip phones need FXS.. |
| 03:40.32 | Bobthehunter | ;) |
| 03:40.51 | JT | xsquared: alternatively you could just use 1 FXO port for the line, and a SIP hardware phone (a voip phone) |
| 03:40.54 | JT | Bobthehunter: wtf.... |
| 03:41.00 | flenders | Bobthehunter: voip phones don't need fxs |
| 03:41.16 | flenders | Bobthehunter: analog phones need fxs modules |
| 03:41.19 | mxyoung | DocHolliday: Did elriah go over the whole mac address file layout with you? |
| 03:41.33 | flenders | Bobthehunter: analog lines need fxo modules |
| 03:41.33 | DocHolliday | yes of course |
| 03:41.47 | mxyoung | DocHolliday: ok, I wasn't paying attention... |
| 03:42.13 | DocHolliday | but the fact is now that i have done a hard reset i cant go back and change anything? |
| 03:42.54 | mxyoung | DocHolliday: where did you set the alternate TFTP server? |
| 03:43.06 | DocHolliday | on the phone |
| 03:43.19 | mxyoung | DocHolliday: and then it rebooted... right? |
| 03:44.19 | DocHolliday | yep |
| 03:44.31 | DocHolliday | but i was told to do 123456789*0# or what not |
| 03:44.39 | DocHolliday | which clears the thing :( |
| 03:44.45 | mxyoung | DocHolliday: I think you're going to have to tell it via DHCP where the tftp server is. What kind of router? |
| 03:44.59 | DocHolliday | Firebox X5 |
| 03:45.37 | kuto | i got some question, does source installation config files are the same with asterisk 1.2.15 config files, unlike trixbox that have its own config files? |
| 03:45.38 | xsquared | JT, well, mum wants 2 wireless phones. doesn't really matter if they are conventional or voip does it? |
| 03:45.54 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 03:46.03 | xsquared | if they are conventional, it needs an FXO module too right? |
| 03:46.09 | JT | xsquared: they will have to be conventional then |
| 03:46.10 | mxyoung | DocHolliday: don't know that one... don't think I can help more:-( but I think that is the key |
| 03:46.14 | JT | nah, FXS for phones |
| 03:46.23 | JT | xsquared: there are wifi voip phones, but they are all shit |
| 03:46.51 | xsquared | okay |
| 03:47.12 | JT | xsquared: you might want to take a look at the book for a good into |
| 03:47.21 | JT | and the wiki for reference (as well as the book) |
| 03:47.23 | JT | ~thebook |
| 03:47.24 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:47.25 | xsquared | what book? |
| 03:47.26 | JT | ~thewiki |
| 03:47.27 | jbot | somebody said thewiki was at http://www.voip-info.org/wiki-Asterisk |
| 03:47.29 | xsquared | thanks |
| 03:47.39 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 03:49.35 | mxyoung | JT: any idea on the "translate.c:163 framein: no samples for lintoulaw"? |
| 03:51.05 | JT | no idea man, might be a bug in 1.4? |
| 03:51.11 | JT | check the bug tracker |
| 03:51.29 | mxyoung | I did... nothing. |
| 03:51.52 | *** join/#asterisk bmg505 (n=leon@196.209.249.141) |
| 03:51.57 | mxyoung | I hate to open a bug report without better documentation.... but I don't get anything out of it other than this error. |
| 03:52.07 | Bobthehunter | ~thecodec |
| 03:52.50 | xsquared | JT, so what about the SPA3102? does that have FXO and FXS? |
| 03:52.56 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 03:53.10 | xsquared | i wish it actually says something in the details |
| 03:53.31 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 03:53.39 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 03:53.46 | JT | xsquared: yes |
| 03:53.53 | JT | xsquared: it usually does |
| 03:54.40 | xsquared | this is so confusing :( |
| 03:55.01 | JT | i can make it less confusing if you pay me to do it :P |
| 03:55.23 | xsquared | haha |
| 03:55.47 | JT | i'm in australia, you know, local knowledge and all ;) |
| 03:56.06 | xsquared | if i can't figure it out, i might |
| 03:56.44 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 03:57.42 | xsquared | so how does the SPA3102 work? all i see is 2 phone connectors in it. how does it connect with asterisk? |
| 03:58.13 | flenders | xsquared: you should als see a netwrk prt |
| 03:58.28 | JT | ok, it's like a little router thing, but basically it converts them to SIP voip protocol |
| 03:58.33 | JT | and you can connect with ethernet |
| 03:58.51 | xsquared | isn't it great that they show you the front of the device and not the whole thing? |
| 03:58.55 | flenders | wow, I thought my 'o' died |
| 03:59.01 | xsquared | http://www.clearnet.com.au/xcart/product.php?productid=16709 |
| 03:59.11 | xsquared | stupid promo pictures :( |
| 03:59.31 | flenders | was just on the phone with jerry from clearnet |
| 03:59.37 | JT | heh, ebay usually has better pictures |
| 03:59.41 | xsquared | haha :P |
| 03:59.51 | xsquared | do you know where they are based? |
| 03:59.56 | flenders | canberra |
| 04:00.12 | flenders | but their shipping is pretty good |
| 04:00.27 | flenders | takes a day or 2 max to syd |
| 04:00.31 | xsquared | i want it in the next 2 days |
| 04:00.42 | flenders | send him an e-mail |
| 04:00.58 | xsquared | i might |
| 04:01.03 | flenders | he might be able to post it to you today. you might get it on monday |
| 04:01.04 | JT | the next 2 days are the weekend |
| 04:01.04 | xsquared | first i have to find out what i need ;) |
| 04:01.09 | JT | people tend not to ship then |
| 04:01.29 | xsquared | i mean business days |
| 04:01.33 | JT | heh |
| 04:02.26 | flenders | jerry's got pretty good deals on SPA921s as well |
| 04:02.33 | flenders | and I think 941s too |
| 04:03.08 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:07.59 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:08.38 | hegars | how do you activate colour on the rasterisk cli? |
| 04:10.25 | *** join/#asterisk weazahl (n=weazahl@adsl-66-136-148-13.dsl.kscymo.swbell.net) |
| 04:11.25 | weazahl | can anyone recommend a GOOD low cost single port FXO card/modem. i've been finding only crap as of late |
| 04:12.19 | weazahl | i just tried "infomagics" card. WORTHLESS! |
| 04:13.38 | JT | err |
| 04:13.41 | JT | for use in asterisk? |
| 04:13.44 | JT | for voice? |
| 04:13.52 | weazahl | yes |
| 04:13.59 | weazahl | that is correct |
| 04:14.07 | weazahl | to interface my landline at home |
| 04:14.22 | JT | ... |
| 04:14.24 | weazahl | want to play with dundi |
| 04:14.24 | J4k3 | get a nice ATA |
| 04:14.33 | JT | it must be SUPPORTED by asterisk |
| 04:14.40 | JT | you can't just grab random modems |
| 04:15.20 | weazahl | i know that. this one braged of great compatability. but it was SO noisey |
| 04:15.47 | JT | yeah they're shit, those x100p clones |
| 04:15.53 | JT | get an ATA like a sipura |
| 04:15.57 | JT | cheapest option |
| 04:16.03 | JT | otherwise a TDM400P |
| 04:16.10 | weazahl | good hybrids on them? |
| 04:16.29 | weazahl | i cant afford a TDM400P for home |
| 04:16.49 | JT | i've never heard of complaints with regards to the hybrids |
| 04:19.30 | flenders | weazahl: a TDM400P with a single FXO channel is not that expensive |
| 04:19.51 | The_DoC^ | $140 average |
| 04:23.08 | weazahl | mine was horrid. i had to drop the RX to -30DB then i could not hear far end at all |
| 04:23.55 | weazahl | i saw someone else is making TDM clones that are $80 w/ 1 FXO. brand new product |
| 04:24.09 | JT | openvox? |
| 04:24.20 | weazahl | i wish digium still made the 100's |
| 04:24.42 | JT | they can't |
| 04:25.31 | JT | intel chipset was discontinued |
| 04:25.55 | weazahl | ahhh. no it wasnt openvox. but they have $40 single FXOs |
| 04:26.05 | weazahl | any good? |
| 04:26.18 | JT | shrug, i wouldn't go for the x100p clone |
| 04:26.23 | JT | i don't trust any of them |
| 04:26.56 | J4k3 | I trust them to get 26.4k on any one of my POTS lines ;) |
| 04:27.26 | weazahl | ok... got ya on that... |
| 04:29.14 | weazahl | on a different question. these people in this hotel cannot decide how many lines they want (incoming) i think at least 8. what would be the most cost effective way to get them 8 with options? just go with 8 ports and add TDM400Ps if they need more? or go with a 16 port |
| 04:31.49 | JT | no |
| 04:31.55 | JT | digital ISDN PRI |
| 04:32.03 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 04:32.10 | JT | at about 8 lines, it definately becomes the better option |
| 04:32.14 | JT | for most areas |
| 04:32.25 | weazahl | class D ratecenter |
| 04:32.39 | ManxPower | weazahl: I'm sorry to hear that. |
| 04:32.47 | weazahl | yeah so was i |
| 04:32.49 | ManxPower | What is this a hotel on Mt Everist? |
| 04:32.53 | JT | due to pricing, additional capabilities over analogue, and ease of use |
| 04:33.00 | JT | i have no idea what class D ratecenter means |
| 04:33.14 | weazahl | boonville. mo. middle missouri. |
| 04:33.22 | J4k3 | class Z |
| 04:33.26 | weazahl | it means all PRIs are per minute |
| 04:33.27 | ManxPower | JT: It means "you are in the sticks so we are going to screw you on rates" |
| 04:33.37 | JT | hrm |
| 04:34.20 | JT | TDM2400P or similar |
| 04:34.20 | flenders | I regret not getting ISDN here |
| 04:34.24 | weazahl | im wondering how ATT will treat us if we use unlimited LD to forward to a PRI |
| 04:34.33 | JT | i think the 2400 base board is not much more than the 800 |
| 04:34.35 | flenders | hopefully we will swap over soon. |
| 04:35.22 | weazahl | you think they will freak if we have 8 calls forwarded LD with the unlimited LD? |
| 04:35.28 | JT | the tdm2400p has the option of hardware EC |
| 04:36.27 | weazahl | i did it in my shop with 2 lines. but i got luck enough to 'sneak' in a VPRI at $8/month with no per minute |
| 04:36.45 | JT | vpri? |
| 04:36.51 | weazahl | Virtual |
| 04:36.58 | JT | i see |
| 04:37.02 | JT | voip? |
| 04:37.40 | weazahl | yeah, i forward my local number to a VPRI and let someone else handle the AD conversion |
| 04:37.46 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:38.12 | JT | vpri is a misnomer isn't it? |
| 04:38.16 | JT | it's just voip |
| 04:38.30 | flenders | so you need a pretty decent bandwidth |
| 04:38.37 | weazahl | i used to have to forward it to TX before i snuck this one in. |
| 04:39.12 | weazahl | i got shitloads of bandwidth... odd, we have a huge optical line here but are class D |
| 04:39.28 | JT | where does the line go? |
| 04:39.54 | ManxPower | weazahl: why not just take the incoming calls somewhere else and transport them to the final location? |
| 04:40.21 | weazahl | the local DID number is a PRI that runs 120 miles to kansas city. it is on the same OC3 that aol is here |
| 04:40.54 | JT | sorry i'm confused, is this the hotel or something else? |
| 04:41.41 | ManxPower | weazahl: no CELCs provide service? |
| 04:41.50 | weazahl | that is my system... hotel is in the same town but it would be .0149 to get same thing here now |
| 04:42.10 | JT | does the hotel have fibre? |
| 04:43.28 | weazahl | yeah, they dont want to do the T1 though. again, Class D |
| 04:43.57 | weazahl | $1400/month |
| 04:44.13 | J4k3 | wtfbbq |
| 04:44.14 | weazahl | analouge is $31/month/line |
| 04:44.18 | JT | do they have badwidth? |
| 04:44.20 | J4k3 | I can get a T1 to my house for about half that |
| 04:44.42 | J4k3 | and I'm 110+ miles to the nearest 'big router' :P |
| 04:44.47 | weazahl | 120 miles from the nearest backbone |
| 04:45.20 | J4k3 | I'm like 119 from AT&T's router for the connection I'm talking on |
| 04:45.25 | J4k3 | its $850ish/mo after taxes |
| 04:45.29 | J4k3 | 1 year contract |
| 04:45.38 | *** join/#asterisk chasecen (n=chase@70-57-161-205.spkn.qwest.net) |
| 04:45.39 | J4k3 | $0 install, $500 deposit which was refunded after 1 year. |
| 04:45.42 | weazahl | we have 2x 6mb down and 768kb up dsl |
| 04:45.50 | JT | weazahl: the hotel? |
| 04:46.02 | J4k3 | ADSL really isn't terribly optimal for voice application |
| 04:46.04 | weazahl | yeah, i only have one at home |
| 04:46.25 | weazahl | low latency |
| 04:46.27 | *** join/#asterisk angom_h (n=Angel@www.aysco.com.mx) |
| 04:46.57 | J4k3 | that ADSL rocked until I got too many PRIs into the building. The signal quality went to poop. |
| 04:47.07 | J4k3 | and SBC was dragging heels about bringing us fiber |
| 04:48.12 | weazahl | yeah, when the up goes up, the latency does too. QoS! |
| 04:48.56 | weazahl | ok, im gonna see if i can hunt down cheaper T1. do i want data only then? or do i want voice? |
| 04:49.08 | J4k3 | thats worth the extra money alone |
| 04:49.09 | JT | weazahl: voice |
| 04:49.23 | *** join/#asterisk nortex (n=nortex@70.43.65.194.nw.nuvox.net) |
| 04:49.58 | weazahl | i talked these people into a $4000 layer 3 switch, and they dont wanna pony up for T1. silly aint it |
| 04:50.13 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 04:50.13 | ManxPower | weazahl: voice PRI is easier to deal with. |
| 04:50.19 | J4k3 | weazahl: most unlimited LD accounts have pretty strict logging and nasty TOS. |
| 04:50.21 | JT | not really, switch doesn't have the recurring costs :P |
| 04:50.32 | J4k3 | if you're cought, sometimes you're backbilled and its within the parameters of the original use agreement |
| 04:50.34 | J4k3 | if so, that blooooooows. |
| 04:50.52 | weazahl | ahhhh |
| 04:51.03 | J4k3 | I hear multiple concurrent calls will flag |
| 04:51.15 | ManxPower | If it must be a data t-1, get a point to point data t-1 to somewhere there is a CLEC that can provide you with a PRI |
| 04:51.27 | J4k3 | yeah |
| 04:51.34 | weazahl | i never got nailed... but my volume is 120 minutes a day |
| 04:51.42 | J4k3 | or access from a provider that they have extremely good peering/connectivity with. |
| 04:51.53 | J4k3 | A lot of networks have rather good internal performance |
| 04:52.01 | J4k3 | plenty good for cross-country voice applications |
| 04:58.31 | weazahl | $1000 plus .03/min for voice |
| 05:05.09 | FuriousGeorge | so i just noticed i wasnt registered with my iax2 DID provider. ive noticed asterisk takes a while to reregister if the connection is broken for some reason, and its after hours so i just restarted asterisk to make sure that will be right before i go to bed |
| 05:05.48 | FuriousGeorge | not only did it not fix the problem, but the entry no longer shows up with an "iax2 show registry" |
| 05:05.51 | weazahl | what would you do if you had a hotel with 50 rooms and 5 retail spaces, analouge lines are $32 and T1 with reasonable rates are $1400 |
| 05:06.30 | FuriousGeorge | 32x26 = 832 right |
| 05:06.44 | FuriousGeorge | but i think 2channels are used for signalling |
| 05:07.06 | FuriousGeorge | as you can see i have no t1 experience |
| 05:08.29 | JT | clearly :P |
| 05:08.32 | JT | 1 D channel |
| 05:08.44 | weazahl | analouge also makes for free local too (depending on incomming load) |
| 05:08.55 | JT | you can 23 lines in T1 pri |
| 05:08.59 | JT | 23 B channels + 1D |
| 05:09.57 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-155-134.ny325.east.verizon.net) |
| 05:12.22 | JT | FuriousGeorge: E1 is 32 * 64kbit/s |
| 05:12.49 | JT | in pri mode, 1 is used for a D channel, and 1 is uses for framing, leaving 30 for voice |
| 05:14.00 | FuriousGeorge | i was only off by 1 :) |
| 05:14.10 | JT | not for T1 ;) |
| 05:14.31 | FuriousGeorge | i said 26-2 = 24 but it was 24 -1=23 |
| 05:14.51 | JT | ah heh |
| 05:14.53 | FuriousGeorge | im a little more concerned about my iax2 provider being "not found" |
| 05:15.05 | JT | that's a worry i guess |
| 05:15.12 | FuriousGeorge | mfer |
| 05:15.33 | FuriousGeorge | verizon's infrastructure must literally be that original copper from bell labs |
| 05:15.51 | FuriousGeorge | my traceroutes look like packets running a gauntlet. |
| 05:16.21 | FuriousGeorge | a second ago asterisk couldnt even resolve switch-1.myprovider.com |
| 05:16.23 | FuriousGeorge | now its just fixed |
| 05:16.50 | J4k3 | FIOS!! OMG!! |
| 05:17.19 | J4k3 | heh. I'm amused by FIOS and those that think its something amazingly wonderful |
| 05:17.31 | FuriousGeorge | ill be the first to switch if it doesnt suck, but im forced to use dsl till i go there and switch to cable in abvout 1 hour at 1:am |
| 05:18.46 | J4k3 | I hate verizon with a passion |
| 05:18.50 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 05:19.01 | J4k3 | I only use them for cellular because I'm forced to (the competition is GSM, and GSM is worthless in rural locations) |
| 05:19.07 | FuriousGeorge | they are unuseable right now |
| 05:19.27 | J4k3 | otherwise you could not force me to do business with them |
| 05:19.53 | FuriousGeorge | at least at this location, i was just there, they could barely use the web, much less make a phone call |
| 05:20.09 | J4k3 | geez |
| 05:20.13 | J4k3 | :| |
| 05:22.05 | *** join/#asterisk ManxPower (n=manxpowe@117.sub-75-200-33.myvzw.com) |
| 05:22.49 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 05:24.14 | Fr0zen_ | anyone here use a Cisco 7970g with asterisk? |
| 05:31.55 | weazahl | so the SPA-3000 will convert a PSTN call and allow my * box to route it as a X100 would but with quality. if i understand it right? |
| 05:33.12 | JT | right, it will convert it to sip and it has a network connection |
| 05:33.20 | JT | and allows a handset to be connected to it |
| 05:34.26 | weazahl | so it would be a DID/Trunk and a FXS ATA. plus has PSTN failover |
| 05:35.44 | JT | i guess so |
| 05:35.50 | JT | it's a FXO and an FXS |
| 05:38.11 | weazahl | danm linksys owns em now |
| 05:38.29 | JT | that's true |
| 05:40.09 | weazahl | i hate my linksys rep. complete moron |
| 05:40.25 | JT | then don't buy through a rep? |
| 05:40.30 | weazahl | my netgear rep is awesome. |
| 05:40.46 | JT | i hate netgear products |
| 05:41.38 | JT | anyway the linksys ATAs have sipura heritage |
| 05:42.03 | weazahl | i found them to have the best the 48p PoE switch for the best money, layer 3. so perfect for a mixed data/voip network |
| 05:42.35 | JT | heh |
| 05:42.54 | JT | all their lower end stuff seems like rubbish, so i just refuse to use them |
| 05:43.09 | JT | i hate the way they seem to treat their customers like idiots |
| 05:43.21 | JT | their product data sheets have no data in them |
| 05:43.27 | weazahl | the lowend stuff is rubbish. but the pro-safe APs beat the shit out of linksys |
| 05:43.29 | JT | netgear i'm talking |
| 05:43.42 | JT | these are all consumer rubbish brands really :) |
| 05:44.44 | weazahl | FSM752PS check it out. i got one. freggin awesome. 20Gb fabric |
| 05:44.58 | JT | nah it's cool |
| 05:45.12 | JT | i run HP ProCurves at the moment and they work fine |
| 05:45.21 | weazahl | 3 large wholesale. |
| 05:45.21 | JT | unlimited lifetime warranty |
| 05:45.42 | weazahl | yep. advanced replacement |
| 05:46.41 | JT | i don't need PoE right now |
| 05:46.45 | JT | but modules for the procurves are cheap |
| 05:46.53 | JT | and these switch chassis do up to 80 ports |
| 05:46.58 | weazahl | i needed the PoE/QoS and vlan for the phones. |
| 05:47.12 | JT | sure if i was buying now i'd get something that does PoE if i needed it |
| 05:47.16 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 05:47.22 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 05:47.43 | JT | i'd probably go HP or Foundry or Cisco |
| 05:48.07 | weazahl | cisco was crazy expensive |
| 05:48.08 | DocHolliday | hello i have a cisco 7941 that i did a hardreboot on, however i was under the impression it would poll the TFTP server i set in the SCCP config, however instead its directing to my DHCP server.. what are my options? |
| 05:48.26 | JT | yeah probably not cisco due to them being arseholes with support too :) |
| 05:49.28 | DocHolliday | any cisco people that can perhaps help me out? |
| 05:49.40 | weazahl | the NG is BSD based each port is a net interface on the kernel. real nice flexability |
| 05:50.09 | JT | meh i'd prefer it be a dedicated switch firmware really :P |
| 05:50.40 | weazahl | DocHolliday: not a cisco channel. weare jusr waxing poeticly. |
| 05:51.49 | DocHolliday | weazahl, i need help regardless, stemming from the ineptitude of another user in here |
| 05:51.49 | weazahl | well it is heavily customized. everything runsBSD now. copiers, TVs, cash registers, etc |
| 05:52.18 | JT | sure, but it's still a netgear |
| 05:52.25 | DocHolliday | nobody in here that can help me? :( |
| 05:55.29 | weazahl | i do have a nortel baystack 450-24 for home though |
| 05:55.39 | weazahl | WAY overkill |
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| 05:56.22 | JT | i have a HP ProCurve at home :) |
| 05:56.29 | weazahl | my TV has the highest priority on QoS queue |
| 05:57.02 | JT | my servers all have redundant power supplies and multiple stage UPSes, people say it's not necessary for home, screw those people :P |
| 05:57.05 | weazahl | yeah, overkill also. i get the baystacks for $5 with cascade modules included |
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| 05:57.27 | J4k3 | all one needs for their home |
| 05:57.36 | J4k3 | is 24 hours of battery backup and 30 days of propane onsite |
| 05:57.56 | JT | haha |
| 05:58.05 | JT | my power hardly fails though |
| 05:58.13 | weazahl | mine aint that overkill. but i do have 12 charged batteries and a 1kW inverter for those exteneded power failures |
| 05:58.27 | J4k3 | I have a 2KW inverter in a dodge truck. |
| 05:58.33 | JT | i considered a diesel genset, but found it unnecessary considering my median power downtime is not more than 1 hour a year |
| 05:58.41 | J4k3 | for our backup power requirements beyond the battery. |
| 05:58.50 | weazahl | mine blinks alot and i have plenty of UPS power where needed |
| 05:59.31 | weazahl | have my viewsonic TV on protected power |
| 06:00.08 | JT | i'm looking at multihoming my home Internet connections |
| 06:00.19 | JT | there are no particularly elegant ways of doing this |
| 06:00.26 | JT | connectioN, i mean, at the moment |
| 06:00.45 | weazahl | wanted to pickup a salvage 5K symetra but i couldnt lift it on my own. |
| 06:01.14 | JT | heh |
| 06:01.24 | weazahl | there were 4 of em. i could have run all my house in a low power mode that way |
| 06:01.38 | JT | it wasn't 3 phase? |
| 06:02.02 | weazahl | nope 220 |
| 06:02.27 | JT | must be a small model |
| 06:02.35 | weazahl | 5k, yeah |
| 06:02.43 | JT | 5kva, that's usually apc matrix teritory |
| 06:03.07 | weazahl | maybe it was, that was long ago |
| 06:03.46 | weazahl | still wish i had em, add a couple sub panels and i would be set for any storm |
| 06:04.07 | JT | heh yeah |
| 06:04.54 | weazahl | small portable gen to recharge, could run for days |
| 06:05.22 | JT | yep |
| 06:06.37 | weazahl | well, i gotta roll 2 netshelters 2 blocks down the roadto the hotel tomorrow. i should get to bed. fuck loading em up on a truck. 2 blocks and the servers arent in yet. |
| 06:06.56 | fetcher | APCs can be picky about generator power (AC frequency needs to be spot-on 50Hz/60Hz, for one thing) |
| 06:07.09 | DocHolliday | anyone here that can help me with cisco phones? |
| 06:07.36 | JT | fetcher: if you have dodgy input power you need a double conversion online UPS, not a line interactive model |
| 06:08.02 | weazahl | the caprenters are draging ass on my one closet so if i show up with the racks, i should get results as the developer wont want 10 large in the hall during construction |
| 06:10.06 | fetcher | JT: yeah, I run important stuff at home either directly from DC, or from inverters off the battery bank. So only rectifiers/chargers are plugged into utility power |
| 06:10.20 | weazahl | have 120 lines to terminate, then i gotta get them to decide on a route for the DIDs so i can order the servers/cards and have the phones ready for 20 users on april fools day |
| 06:10.54 | fetcher | weazahl: heh, turnup on April Fools'? :) |
| 06:11.02 | weazahl | nice huh? |
| 06:12.01 | JT | fetcher: cool |
| 06:12.12 | JT | weazahl: this the hotel? |
| 06:12.31 | weazahl | time is tight, we just got papers signed last thursday. have all the backhaul together except the firewall (monday) already. they are way behind on thier IT planning. |
| 06:12.39 | weazahl | JT: yeah |
| 06:12.55 | JT | they are getting 120 phone lines? |
| 06:13.05 | JT | i thought you said they were getting maybe 8 |
| 06:13.25 | weazahl | 8 inbound. 50-60 users |
| 06:13.40 | weazahl | plus wifi and hard data ports |
| 06:13.47 | JT | 120 extensions? |
| 06:14.06 | weazahl | 60 extensions. |
| 06:14.24 | weazahl | then additional data infrastructur |
| 06:14.27 | JT | not sure where the 120 comes from? |
| 06:14.30 | JT | hmm ok |
| 06:14.43 | JT | analogue or sip extensions? |
| 06:14.51 | weazahl | sip |
| 06:14.54 | JT | cool |
| 06:14.58 | JT | what phone? |
| 06:15.45 | weazahl | aastras for rooms i think and pc 650 with 3 exp's for desk |
| 06:15.56 | weazahl | simple one button transfers |
| 06:16.16 | JT | ah ok |
| 06:16.53 | weazahl | though the 480 CT is also a good option for desk, as is the 57i with modules |
| 06:17.17 | weazahl | too bad the 57i dont have a CT. would be perfect |
| 06:17.37 | JT | hmm |
| 06:18.02 | weazahl | 2 6mb dsl lines. one data, one voice |
| 06:18.27 | JT | so you're considering whether to get POTS or ISDN on top of that? |
| 06:18.53 | weazahl | i think POTS is the only option. |
| 06:19.03 | JT | maybe |
| 06:19.15 | JT | not too bad if you have voip as well |
| 06:19.25 | JT | as long as you're not running IVRs on it |
| 06:19.31 | weazahl | $1300 plus .04 LD and .09 Local |
| 06:19.33 | JT | that can tend to suck |
| 06:20.09 | weazahl | nah, most calls would be human answered, just failover to IVR |
| 06:20.21 | JT | hmm |
| 06:20.30 | JT | inbound be voip or pots or both? |
| 06:21.09 | weazahl | so if i go pots at $30 a line for inbound, i can get .008 for voip out |
| 06:21.22 | weazahl | 800 voip inbound |
| 06:21.44 | weazahl | toll would be rollover analougue |
| 06:21.52 | JT | can the voip failovr to pots |
| 06:22.00 | weazahl | sure could. |
| 06:22.34 | weazahl | just cut the number of incoming cals you could handle. really makes for a robust system |
| 06:22.48 | weazahl | adds a 9 or two to uptime |
| 06:23.08 | weazahl | then heartbeat w/ ip takeover to add aonther 9 |
| 06:24.33 | weazahl | i think openbox's 12 port cards will do well. 12 should handle all it needs. if they ever need more, could use dundi on the spare server. |
| 06:25.12 | weazahl | then just drop to one server should one fail and fall back to 12 lines. |
| 06:25.41 | weazahl | but, the retail storesare going to fuckup usage on the incoming lines. |
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| 06:26.38 | weazahl | they should have been looking at this a year ago. instead up 8 weeks before going live |
| 06:27.00 | J4k3 | haha, like you can get people to think ahead. |
| 06:27.07 | J4k3 | its hard enough to get them to think at all |
| 06:27.50 | weazahl | hey, its nicefor us though. they are pretty much locked into paying my salary for years to come |
| 06:28.13 | weazahl | try finding someone to maintain asterisk in BFE missouri |
| 06:28.14 | xsquared | i have another question. How does Zapateller work? "Generates special information tone to block telemarketers from calling you." |
| 06:29.05 | weazahl | xsquared: it plays the tone that makes em think the line is dead and tricks the PD into thinking it got a wrong number |
| 06:29.36 | xsquared | ah okay cool. But any normal person will still hang on the line right? |
| 06:29.36 | weazahl | as long as noone mentions remote admin... im set |
| 06:29.58 | weazahl | nope. it is about 250ms |
| 06:30.13 | weazahl | its annoying to the calling party. that is all |
| 06:30.46 | xsquared | ok great |
| 06:30.48 | xsquared | thanks |
| 06:30.52 | weazahl | i hate calling a zaptel though |
| 06:31.19 | weazahl | alright im out folks. |
| 06:31.35 | weazahl | time to wax the weasel and go to sleep |
| 06:32.19 | weazahl | *poof* |
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| 06:38.12 | JT | xsquared: it's a tone that makes some autodiallers disconnect |
| 06:38.24 | JT | and you can make asterisk use it only if callerid is blocked |
| 06:39.13 | xsquared | ok :) |
| 06:39.19 | xsquared | do you use it? |
| 06:39.21 | JT | no |
| 06:39.40 | JT | i don't think telemarketers are a big problem in australia |
| 06:39.45 | JT | i rarely get them |
| 06:40.20 | xsquared | ok |
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| 07:16.49 | _MDC_ | Hi all, is there a possibility to get the same info in chanisavail in CLI? |
| 07:19.56 | tzafrir_laptop | _MDC_, to get where? |
| 07:21.49 | _MDC_ | What I want is the to get the chanisavail status in a web page, so via I thought that via cli i could get that info.. |
| 07:23.21 | _MDC_ | tzafrir_laptop, i just want to see if a sip user is on the phone or not.. |
| 07:23.54 | tzafrir_laptop | have you looked at manager interface functions? |
| 07:24.26 | tzafrir_laptop | sorry: manager interface command. This is something that is easier to use in web interfaces |
| 07:24.49 | _MDC_ | tzafrir_laptop, I've read somewhere that the manager interface is a little bit unstable and could cause asterisk to die.. |
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| 07:28.56 | Corydon76-home | _MDC_: where'd you hear that? |
| 07:30.34 | _MDC_ | Corydon76-home, thought it was on the wiki, but not sure... something if the client dies then asterisk cannot send the message to the client and end up in a deadlock or something.. but if this isn't the case its great! |
| 07:31.26 | Corydon76-home | _MDC_: I've never heard such a claim |
| 07:32.06 | Corydon76-home | _MDC_: and the proper place to report such a claim would be the bugtracker. I think you got stuck with some FUD |
| 07:32.12 | tzafrir_laptop | Each manager socket runs in its own thread, or is there one thread for the whole manager interface? |
| 07:32.43 | _MDC_ | found the page; http://www.voip-info.org/wiki/view/Asterisk+manager+experience |
| 07:33.12 | Fr0zen_ | anyone here use a Cisco 7970g with asterisk? |
| 07:33.32 | Corydon76-home | tzafrir_laptop: it creates a thread for each connection |
| 07:34.22 | JT | _MDC_: it seems to indicate there was at least a partial bugfix |
| 07:34.24 | Corydon76-home | _MDC_: note that that was fixed back in 2004 |
| 07:34.32 | _MDC_ | then the wiki might be outdated, sorry - but i need to get to work now, thanks anyway, will try the manager interface |
| 07:34.35 | JT | without checking it out more, i can't say if it's a full fix or not |
| 07:34.41 | JT | as it isn't that clear |
| 07:34.49 | tzafrir_laptop | _MDC_, anyway, parsing the output of the CLI is not nicer, as you may suddenly have some verbose messages. Not to mention the overhead for rnning asterisk for every "query" |
| 07:34.56 | Corydon76-home | _MDC_: so something that was fixed 3 years ago is certainly not a current bug |
| 07:34.59 | JT | _MDC_: you failed to read: "Disconnecting the connection between a remote connected terminal and the Asterisk box will often cause a deadlock (http://bugs.digium.com/bug_view_page.php?bug_id=0000861) <<<--This bug has been reported fixed on 2-07-2004" |
| 07:35.04 | JT | ? |
| 07:35.48 | _MDC_ | JT, sorry missed that part, is hard with the wiki to see how old a text is, it could be written yesterday or three years ago, hard to tell |
| 07:35.56 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:36.05 | JT | i thought that bit stood out :P |
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| 07:36.40 | _MDC_ | JT, then i'm the problem ;-) sorry to bother you guys.. |
| 07:37.10 | JT | well check it out man, sounds like manager is the way you need to go |
| 07:37.55 | _MDC_ | yes indeed, something fun to look at over the weekend, now i really get to be at work... bye |
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| 07:38.59 | JT | cya |
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| 07:53.20 | sbingner | life sucks and then you die... happy bday me... later |
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| 08:00.41 | kippi | hey |
| 08:01.07 | kippi | how can I setup asterisk so that I can bridge calls using my speed dial keys? |
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| 08:04.45 | DocHolliday | woohoo phone fixed |
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| 08:46.45 | jserve | Good morning |
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| 09:03.49 | sudhir492 | Hi all |
| 09:03.54 | sudhir492 | is anyone there? |
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| 09:08.21 | kezza491 | How do i fix this error? [Feb 23 19:58:27] NOTICE[2391] chan_iax2.c: Registration of 'kieran491' rejected: 'Registration Refused' from: '192.246.69.186' i am using AsteriskNOW |
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| 09:49.31 | vlt | Hello. I was registered to a SIP server, commented the "register =>" line in sip.conf and reloaded. Now `sip show registry` doesn't show the peer anymore but calls still get through to me. register => 117071:mmil2049@pbx-network.de/pbx |
| 09:49.46 | vlt | sorry. wrong paste |
| 09:50.56 | vlt | ... How can I clear a sip registry? |
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| 10:38.07 | active_si | is it possible to get a status back from asterisk/app_txfax when the task has been completed (if it was successful or not)? |
| 10:39.45 | mendol | anybody knows how to connect 70 analog phones to voip? using smth else then using 35xpap2t? |
| 10:40.45 | vlt | Does anyone know how to destroy a sip registry? |
| 10:41.37 | Ahrimanes | mendol: channel bank |
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| 10:42.04 | Ahrimanes | active_si: that's usually done with the script that emails the fax |
| 10:43.00 | mendol | like Rhino CB24FXS etc/ |
| 10:43.01 | mendol | ? |
| 10:43.37 | Ahrimanes | mendol: yes or http://www.voipsupply.com/index.php?cPath=94_286_122 |
| 10:44.31 | mendol | ouch expensive heh |
| 10:44.40 | Ahrimanes | hehe yeah |
| 10:44.47 | Ahrimanes | how much is the Rhino ? |
| 10:45.16 | active_si | Ahrimanes: the script just returns the email that the fax was accepted for delivery, but there is not status of delivery |
| 10:45.30 | mendol | Rhino Asterisk Channel Bank; 24 FXS Analog Channels - 1200 euro |
| 10:45.38 | mendol | so i would have to buy 3 ;-) |
| 10:45.40 | Ahrimanes | active_si: hm the sample scripts i found on voip-info.org would send email with the status |
| 10:45.49 | mendol | whats patch panel? |
| 10:46.17 | Ahrimanes | mendol: yeah, pap2t's are the cheap solution, but also a paint to maintain compared to the others |
| 10:46.53 | mendol | i know, but you know in this country ppl want the best but cheapest solutions |
| 10:47.24 | mendol | hard to make good and cheap project heh |
| 10:47.31 | mendol | but thanks a lot mate :-) |
| 10:48.13 | Ahrimanes | mendol: also http://www.digium.com/en/products/hardware/tdm2400p.php |
| 10:48.15 | Ahrimanes | mendol: np |
| 10:49.51 | Ahrimanes | mendol: http://www.sangoma.com/datasheets/p_a400-specs |
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| 10:55.46 | JT | mendol: why are people so cheap? |
| 10:56.39 | Ahrimanes | JT: well the software is free, so anything on top is expensive i guess |
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| 10:59.42 | JT | Ahrimanes: hah |
| 10:59.58 | mendol | o |
| 11:01.08 | mendol | hm how much for this sangoma? |
| 11:01.21 | Ahrimanes | JT: logic would have it that if you pay $100.000 for a pbx solution, $6000 is CHEAP for addons.. but if you pay zero, then it's a different story |
| 11:01.41 | JT | heh |
| 11:02.05 | Ahrimanes | mendol: estimated prices: http://www.voipsupply.com/index.php?cPath=99_420 |
| 11:06.38 | kippi | Anyone know what this error means? Feb 23 11:06:16 NOTICE[8665]: chan_local.c:498 local_alloc: No such extension/context 1153@default creating local channel |
| 11:10.05 | *** join/#asterisk phpboy (n=shane@196.211.17.202) |
| 11:10.08 | phpboy | hey all |
| 11:10.35 | phpboy | how would I go about using a QUAD ISDN card and a 4 PORT digium analogue card at the same time? |
| 11:10.51 | phpboy | i need to configure this in zaptel.conf and I can't seem to figure out how ;/ |
| 11:11.33 | JT | which quad isdn? |
| 11:11.48 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 11:11.48 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:13.52 | mendol | thanks again :-) |
| 11:14.01 | phpboy | JT: the junghanns quad ISDN card |
| 11:15.20 | Ahrimanes | mendol: :) |
| 11:15.26 | mendol | hehe well |
| 11:15.35 | mendol | looks like we will have to focus on those 35xpap2t |
| 11:15.36 | mendol | ;-) |
| 11:16.14 | phpboy | JT: what do you think? |
| 11:16.47 | JT | phpboy: running bristuff? |
| 11:16.58 | phpboy | I am, yes |
| 11:17.03 | JT | cool |
| 11:17.09 | Ahrimanes | mendol: hehe, tight budget? |
| 11:17.25 | JT | make sure you run 0.3.0 pre1w minimum |
| 11:17.33 | JT | updates to qozap |
| 11:18.12 | mendol | not mine |
| 11:18.32 | Ahrimanes | ok |
| 11:18.36 | mendol | im part of big isp company |
| 11:18.39 | mendol | which sells voip |
| 11:18.50 | mendol | and i have to find a solution for a client with 70 analog phones |
| 11:18.54 | mendol | and wants to use voip |
| 11:19.08 | mendol | and cant spend more then 2000 euro |
| 11:19.09 | mendol | heh |
| 11:19.15 | Ahrimanes | mendol: ok |
| 11:19.22 | Ahrimanes | mendol: that IS a tight budget |
| 11:19.27 | JT | tell them they're crazy, and move on |
| 11:19.30 | JT | not worth it at all |
| 11:19.46 | *** join/#asterisk bobbytux (n=bobbytux@LNeuilly-152-21-159-81.w193-253.abo.wanadoo.fr) |
| 11:19.53 | Ahrimanes | JT: if they do a lot of international calling, it can be worth it |
| 11:20.03 | bobbytux | lo |
| 11:20.08 | JT | then they will benefit from investing the proper amount |
| 11:20.13 | mendol | its more complicated |
| 11:20.16 | JT | what they want is stupid |
| 11:20.21 | mendol | first they sell internet |
| 11:20.24 | JT | unless you lease it to them |
| 11:20.24 | mendol | then voip |
| 11:20.41 | mendol | well have to work with what i have ;-) |
| 11:21.05 | JT | phpboy: have to go, but i can chat about it another time, good luck with isdn setup and let me know how it goes |
| 11:21.42 | phpboy | ok, thanks ;/ |
| 11:21.48 | mendol | hm ok now i need E1 card |
| 11:22.31 | Ahrimanes | mendol: why? |
| 11:22.56 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 11:23.13 | nextime | hi, is bristuff working with asterisk 1.4? |
| 11:23.26 | mendol | coz first they will provide internet using radio |
| 11:23.40 | mendol | then using e1 or smth |
| 11:24.08 | Ahrimanes | ah ok |
| 11:24.19 | Ahrimanes | get a 1-port sangoma with wanpipe |
| 11:25.01 | nextime | and more, comparing visdn, misdn and bristuff, which are the "best" (read as most stable and reliable) drivers to have an * server with both pri and bri interfaces? |
| 11:25.21 | mendol | but will one e1 be able to handle all those calls? |
| 11:26.45 | kippi | can someone help me with this? http://www.pastebin.ca/368849 the extenstion is there, can't work it out! |
| 11:29.09 | Ahrimanes | mendol: well if you do g729 sure |
| 11:30.15 | *** join/#asterisk js_ (i=js@ua-83-227-232-164.cust.bredbandsbolaget.se) |
| 11:31.19 | stoffell | nextime, bristuff only supports 1.2.14 as of now .. |
| 11:31.58 | stoffell | nextime, I hear very good rumours on mISDN.. I'd prefer mISDN or visdn above bristuff |
| 11:32.21 | stoffell | or buy a digium bri-card... even better, you can use the digium drivers then.. |
| 11:42.14 | nextime | stoffell : personally i don't have any BRI, only PRI. But i'm packaging asterisk 1.4 for a debian derived distro, so i must do something that is usable even to non digium bri cards |
| 11:44.58 | phpboy | could somebody please help me configure zaptel to run my ISDN card and my analogue car |
| 11:45.00 | phpboy | card even |
| 11:45.03 | mafkees | use visdn |
| 11:45.27 | mafkees | nextime: debian has visdn branch in subversion. maybe that's an option |
| 11:45.30 | phpboy | I'm using bristuff |
| 11:45.41 | phpboy | I don't really have a choice on freebsd |
| 11:46.24 | stoffell | phpboy, what card are you using? (nr of ports) |
| 11:47.02 | phpboy | junghannes Quad ISDN card and digium 400 quad analogue card(4 FXO modules) |
| 11:47.45 | *** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com) |
| 11:52.59 | phpboy | stoffell: any ideas? |
| 11:56.23 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:56.46 | *** join/#asterisk knathraak (n=zach@151.196.142.242) |
| 11:58.12 | stoffell | phpboy, what do you need? a sample zaptel.conf ? |
| 11:58.35 | phpboy | Well, I know how to configure the cards to run alone |
| 11:58.42 | phpboy | but I can not get them to run together |
| 11:58.43 | phpboy | :/ |
| 11:59.26 | mendol | yea Ahrimanes but they want quality as well ;-) |
| 11:59.41 | stoffell | phpboy, should be not so hard? |
| 11:59.42 | Ahrimanes | mendol: g729 is quite good quality |
| 11:59.42 | smace | My asterisk is working but calls have very bad quality. But latency seems low =/ |
| 11:59.54 | Ahrimanes | mendol: but it does require a license in most cases |
| 12:01.04 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 12:01.49 | smace | if latency is not my problem. what could be my problem? |
| 12:02.03 | penguinFunk | bandwidth ? |
| 12:02.06 | penguinFunk | jitter ? |
| 12:02.18 | smace | jitter? |
| 12:02.26 | penguinFunk | latency = delay |
| 12:02.32 | penguinFunk | jitter = variation in delay |
| 12:02.40 | smace | oh |
| 12:02.41 | *** join/#asterisk vanumo (n=david@62.99.154.2) |
| 12:02.46 | vanumo | hi :-) |
| 12:02.47 | penguinFunk | do you get a contstant latency ? |
| 12:02.51 | smace | sometimes I get low latencis but just a few packets. |
| 12:02.53 | penguinFunk | like 20ms all the time |
| 12:02.58 | *** join/#asterisk coppice (n=chatzill@13.168.17.210.dyn.pacific.net.hk) |
| 12:02.59 | penguinFunk | or is it 20, 40, 80 |
| 12:03.01 | penguinFunk | etc |
| 12:03.12 | vanumo | how can i install chan_cellphone ? |
| 12:03.29 | phpboy | vanumo: what does chan_cellphone do? |
| 12:03.32 | smace | penguinFunk, after 10 pings I get one of 100ms. |
| 12:03.42 | vanumo | i'am to stupid to do install it (-) |
| 12:03.55 | penguinFunk | well anything +/-20ms isnt good |
| 12:04.00 | penguinFunk | not a very stable latency |
| 12:04.12 | smace | should it be lower than 20ms? |
| 12:04.14 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 12:04.25 | vanumo | phpboy with chan_cellphone you can connect you cellphone to asterisk an run it as gateway in to gsm network |
| 12:04.29 | penguinFunk | ideally yes |
| 12:05.10 | phpboy | vanumo: but you'd need a cell phone physically connected to the asterisk server |
| 12:05.14 | phpboy | obviously, yes? |
| 12:05.24 | smace | penguinFunk, http://200.149.32.178/ping.txt |
| 12:05.43 | penguinFunk | hmm |
| 12:05.53 | penguinFunk | very bad jitter |
| 12:06.02 | vanumo | phpboy: you can use bluetooth to connect the cellphone to it, you can use more phone in the same time |
| 12:06.16 | mendol | ahm |
| 12:06.19 | penguinFunk | id prefer a contstant 10ms than get 4ms sometimes for it to jump up to 57ms |
| 12:06.22 | phpboy | vanumo : nice |
| 12:06.45 | vanumo | phpboy, yes i think so but i can install it, iam to stupid for it |
| 12:06.46 | penguinFunk | whats causing that bad jitter ? |
| 12:06.52 | penguinFunk | what type of line is it ? |
| 12:07.07 | phpboy | hectic |
| 12:07.29 | smace | penguinFunk, it is one wireless network. |
| 12:07.33 | phpboy | stoffell: i need to first somehow establish that my BRISTUFF is actually working :/ |
| 12:07.41 | penguinFunk | ahhh |
| 12:07.46 | penguinFunk | that explains it |
| 12:07.59 | smace | but it is one stable wireless network. |
| 12:08.09 | smace | I am surprised it is not good for voip. |
| 12:08.26 | penguinFunk | well it depends on the surroundings |
| 12:08.29 | smace | We use Skype/msn thought it most time. |
| 12:08.37 | vanumo | can anbody help me to install chan_cellphone ? i don't know what i should do ? |
| 12:08.45 | penguinFunk | all those bad pings might be related to anything ? |
| 12:08.51 | penguinFunk | someones farts |
| 12:08.53 | penguinFunk | etc |
| 12:09.04 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 12:09.18 | penguinFunk | bad jitter could quite well be your problem |
| 12:09.21 | smace | penguinFunk, I dont have bandwidthd problem. If I change to g729 Am I going to have better results? |
| 12:09.37 | penguinFunk | best way to test is to use a cable, check your jitter, if its improved.. has your problem got any better? |
| 12:09.57 | penguinFunk | what codec are you using now ? |
| 12:10.09 | penguinFunk | i find alaw is better sound quality than 729 |
| 12:10.18 | smace | g711a |
| 12:10.18 | penguinFunk | you can just tell the difference straight away |
| 12:10.37 | penguinFunk | well if bandwidth aint a problem id say stick with alaw |
| 12:10.54 | penguinFunk | but if you are happy with 729's sound quality use that i suppose |
| 12:11.02 | phpboy | stoffell: I can only configure one of the cards on port(s) 1-4 |
| 12:11.05 | phpboy | not both |
| 12:11.14 | penguinFunk | what problems are you experiencing exactly smace? |
| 12:11.15 | smace | penguinFunk, the problem is that i will have to buy licenses |
| 12:11.16 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:11.19 | penguinFunk | i see |
| 12:11.22 | smace | low quality calls. |
| 12:11.37 | coppice | if you can't tell G.729 from G.711 straight away, consult a doctor |
| 12:11.48 | penguinFunk | well since, 729 is compressed audio, it will be LOWER sound quality than 711a (alaw) |
| 12:11.48 | phpboy | so it's either fxsks=1-4 for analogue only or bchan=1-4 for ISDN only |
| 12:11.55 | penguinFunk | rofl coppice |
| 12:12.11 | coppice | alaw is also compressed, but not very agressively |
| 12:12.15 | penguinFunk | ah ok |
| 12:12.31 | penguinFunk | alaw = top quality its what phone companies use |
| 12:12.57 | coppice | yes. alaw and ulaw are what you hear on any PSTN call to another PSTN phone |
| 12:13.22 | coppice | its nasty, but its the reference standard :-) |
| 12:13.40 | mendol | heh |
| 12:13.40 | penguinFunk | whats your favourite codec then coppice? |
| 12:14.15 | coppice | alaw is narrowband. almost any wideband codec will sound better |
| 12:14.39 | smace | coppice, whats your favourite codec then? |
| 12:15.09 | coppice | 192K samples/second 24 bit PCM, of course |
| 12:16.05 | penguinFunk | how much bandwidth does one call require ? |
| 12:16.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:16.33 | penguinFunk | and surely if the whole world's PSTN systems use some form of 711 it cant be that bad? |
| 12:16.48 | penguinFunk | dont need cd quality phone calls |
| 12:16.51 | penguinFunk | lol |
| 12:17.09 | e-ddie | i do |
| 12:17.12 | e-ddie | :D |
| 12:17.33 | coppice | one of the goals of ISDN In the 80's was to get rid of G.711. it never happened. Rather sad, really |
| 12:17.37 | Ahrimanes | e-ddie: well, you're swedish.. |
| 12:18.07 | e-ddie | Ahrimanes: no, i'm not |
| 12:19.59 | Ahrimanes | e-ddie: yes... |
| 12:20.11 | e-ddie | i'm norwegian |
| 12:20.29 | Ahrimanes | ah |
| 12:20.41 | e-ddie | big difference |
| 12:20.51 | Ahrimanes | explains the hair ;) |
| 12:21.01 | e-ddie | heheh |
| 12:21.11 | e-ddie | it's a cold world |
| 12:21.16 | vanumo | works here anybody with chan_cellphone ? |
| 12:23.51 | phearless | hi guys ! |
| 12:23.59 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:24.18 | phearless | is it possible to set "Withheld number" as a callerID when the incoming number is withheld ? |
| 12:31.54 | vanumo | how can i install chan_cellphone |
| 12:31.56 | vanumo | ? |
| 12:35.48 | *** join/#asterisk shinux__ (n=shinux@196.207.1.30) |
| 12:37.27 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 12:37.55 | phpboy | vanumo: google prolly knows how :/ |
| 12:38.25 | mendol | Ahrimanes: still here? :-) |
| 12:38.41 | mafkees | phpboy: yeah |
| 12:39.00 | vanumo | phpboy it works :-) |
| 12:39.06 | mafkees | phearless: yeah, that's possible |
| 12:39.46 | vanumo | you must make checkout asterisk trunk and patch in the trunk directory |
| 12:39.46 | mafkees | show function CALLERID |
| 12:48.40 | vanumo | how about the gui ? is the gui nice to work with it ? |
| 12:49.53 | nibbler_de | phpboy: tried to prefix #31#? |
| 12:50.02 | nibbler_de | sorry |
| 12:50.09 | nibbler_de | phearless, not phpboy :) |
| 12:52.15 | *** join/#asterisk hal2 (n=hal5@host86-149-56-223.range86-149.btcentralplus.com) |
| 12:52.34 | phearless | nibbler_de: incoming |
| 12:52.37 | phearless | not outgoing |
| 12:52.50 | phearless | I want to display "Withheld number" on my phone |
| 12:52.50 | *** join/#asterisk shinux__ (n=shinux@196.207.1.30) |
| 12:53.16 | hal2 | hello - could someone help me, please? I have set up musiconhold, but the music only plays once for a call rather than looping. Is there a way to get it to repeat indefinitely? |
| 12:55.31 | nibbler_de | phearless: then you can just rewrite the callerid when the string ocurrs that you get when numbers are withheld -> show function CALLERID as mafkees already said :) sorry, was a bit confused here |
| 12:55.52 | phearless | ok ! |
| 13:00.33 | *** join/#asterisk friedrich| (n=friedric@e177243129.adsl.alicedsl.de) |
| 13:01.41 | *** join/#asterisk hal3 (n=hal5@host86-149-56-223.range86-149.btcentralplus.com) |
| 13:04.22 | *** join/#asterisk dj015 (n=damjan@dsl-243-130-53.telkomadsl.co.za) |
| 13:06.11 | hal3 | hello - could someone help me, please? I have set up musiconhold, but the music only plays once for a call rather than looping. Is there a way to get it to repeat indefinitely? |
| 13:07.41 | hal3 | actually, I think I have enabled continuous mode by using the mpg123 -Z switch, however, now I get this if I so a ps -ef | grep mpg123 |
| 13:07.52 | hal3 | root 23156 23154 0 13:00 pts/1 00:00:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 rockford2.mp3 |
| 13:08.16 | hal3 | hold on -that's correct!! I think it has worked for some reason |
| 13:08.18 | hal3 | brb |
| 13:09.07 | hal3 | no - I get this: root 27982 27931 0 13:08 pts/1 00:00:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 /usr/local/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -Z rockford2.mp3 |
| 13:09.16 | dj015 | any developers here? what is the difference betweeen ast_channel's nativeformats and ast_channel_tech's capabilities? |
| 13:09.33 | hal3 | do you see that the mpg123 is listed in the output twice? |
| 13:10.24 | DrukenLPY | hal2: do you only have a single mp3 file? |
| 13:10.45 | hal3 | this is my musiconhold.conf: [default] / mode=mp3 / application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 -Z |
| 13:10.58 | hal3 | yes, I do |
| 13:11.03 | hal3 | is that a problem? |
| 13:11.08 | DrukenLPY | uhmm,.. yeah |
| 13:11.26 | DrukenLPY | you need at least 2 |
| 13:12.09 | hal3 | oh, I didn't realise that |
| 13:12.14 | hal3 | let me try - brb |
| 13:13.00 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
| 13:13.21 | *** part/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 13:13.51 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 13:16.05 | hal3 | that didn't work, druken |
| 13:16.23 | DrukenLPY | did you restart? |
| 13:16.54 | hal3 | also, I am confused why I am getting the mpg123 command repeated when I do a ps -ef |
| 13:17.20 | DrukenLPY | diffrent threads... |
| 13:18.10 | hal3 | no, I don't mean under different processes - it's actually on the same process |
| 13:18.45 | hal3 | root 32259 32206 0 13:16 pts/2 00:00:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 /usr/local/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 rockford3.mp3 rockford2.mp3 |
| 13:20.52 | DrukenLPY | looks normal to me |
| 13:20.56 | hal3 | yes, I did druken |
| 13:21.00 | hal3 | I did restart |
| 13:21.22 | hal3 | ok, I have managed to get it to repeat, by using the -Z argument for mpg123 |
| 13:22.04 | hal3 | it is working fine, but I am concerned about the the ps -ef output having repeated mpg123 commands under one process |
| 13:22.38 | DrukenLPY | if it's working corectly, then it's working... :) |
| 13:22.52 | hal3 | by the way, with the -Z argument, I have found there is no need to have more than one mp3 file in the moh directory |
| 13:24.13 | DrukenLPY | must be one hell of a song if that's the only one you want... :) |
| 13:24.57 | hal3 | yes, it is coool! It's the theme tune for the rockford files! ;-) |
| 13:25.55 | DrukenLPY | are you serious?, i figured you had a custom thing done with advertisments in it or something... |
| 13:26.19 | hal3 | In the show, at the end of the theme it always has rockford's answerphone, with some random person chasing him for money or saying that his cheque has bounced. Shame I can't have that too! ;-) |
| 13:26.42 | hal3 | this is a test machine - only a template for production |
| 13:26.58 | hal3 | adverts would be an excellent idea though |
| 13:29.16 | hal3 | thank you for your help, druken - the "continuous" only actually plays it a few times, but it will do |
| 13:30.52 | hal3 | is there a way to get the phone to beep after a period of time to remind the operator that the call is still waiting? |
| 13:31.25 | jojo^ | What possible causes could a "Dropping incompatible voice frame"-problem have? |
| 13:32.04 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 13:33.31 | hal3 | codec mismatch, jojo? |
| 13:36.20 | anonymouz666 | why zttranscode loads even if I don't have this card ? |
| 13:36.46 | hal3 | Druken, thank you for your help before |
| 13:38.19 | jojo^ | hal3, Yeah, must be in some way. I'm using software IAX-phones, and then a single SIP-provder to which all outgoing calls get routed. The strange thing is that most calls works.. But this morning all calls acted strange, until after a while when it just started to work again. |
| 13:39.36 | hal3 | I am sorry, jojo, I can't think of a reason why that may be. Maybe your provider is having problems? |
| 13:41.04 | jojo^ | hal3, I have spoken to them and it doesnt look like that.. They have a huge customer base so it should be well tested.. I'm going to send some debug-logs to them any way though, just in case. |
| 13:41.27 | hal3 | good idea - I can't think of anything else |
| 13:41.29 | hal3 | sorry |
| 13:42.02 | jojo^ | hal3, Do you think forcing all clients and pathes to just one codec would do the trick? |
| 13:42.51 | hal3 | I am sorry - maybe someone else has some input they can give? |
| 13:45.33 | xboxoslo | Hello I got some help here yesterday and now I can call out but I cant get any incomming cals can someone help me please? |
| 13:49.57 | *** join/#asterisk SLiNK (n=slink@c-68-63-34-189.hsd1.fl.comcast.net) |
| 13:52.00 | *** part/#asterisk shwa (n=shwa@ip-62-235-203-59.dsl.scarlet.be) |
| 13:59.43 | *** join/#asterisk ghatak_g (n=ghatak@84-93-217-81.plus.net) |
| 14:03.21 | *** join/#asterisk KnowWhat (n=KnowWhat@host210-2-165-136.isb.dancom.net.pk) |
| 14:04.54 | ghatak_g | Hi, What does error, "407 Proxy Authentication Required" mean ? |
| 14:04.54 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 14:05.30 | KnowWhat | it means the proxy you are using is not for un unauthorized access |
| 14:05.58 | webman | does anyone know a working site to download spandsp/txfax/rxfax for asterisk 1.4 ?? |
| 14:06.23 | KnowWhat | webman i think you should try google.com |
| 14:06.52 | webman | knowwhat: I have been, for the last 20 minutes... with no success yet :( |
| 14:06.54 | *** join/#asterisk Supermathie (n=michael@justman.NetDirect.CA) |
| 14:06.54 | ghatak_g | KnowWhat: But i have the correct paassword or credentials. Do u mean that it is normally generated when Username/password is incorrect ? |
| 14:07.11 | KnowWhat | ghatak_g, yeah |
| 14:07.29 | KnowWhat | ghatak_g: If problem persists, Please contact your network administrator :D |
| 14:07.40 | Supermathie | Morning all (at least here it is) |
| 14:07.42 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:07.42 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:07.46 | KnowWhat | hmm |
| 14:08.01 | KnowWhat | you wanna use fax with asterisk webman |
| 14:08.15 | ghatak_g | KnowWhat: i am the network administrator :P |
| 14:08.40 | KnowWhat | ghatak_g: then why you asking such questions, i suspect though if you are :P |
| 14:08.51 | Supermathie | On Asterisk/Zaptel, is it possible to have Asterisk determine whether an outgoing line is working (i.e. has a dialtone) before dialing? |
| 14:08.53 | webman | knowwhat: I have been for the past 2 years or more, but I upgraded to 1.4 tonight, and need to update spandsp and friends as well |
| 14:09.18 | webman | supermathie: no |
| 14:09.46 | ghatak_g | ghatak_g: hehe i am not, just kidding, trying to learn more..... that is all |
| 14:10.13 | KnowWhat | webman: why dont you use asterfax? |
| 14:10.34 | Supermathie | Any other hardware that will allow that? |
| 14:10.37 | Gido-E | asterfax |
| 14:10.44 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:11.07 | webman | knowwhat: what is asterfax ?? I thought that was just the script which took the tiff file from rxfax and converted it to pdf and sent it by email ?? |
| 14:11.49 | KnowWhat | The most comprehensive faxing solution for asterisk is AsterFax. AsterFax provides an email to fax gateway which support a large set of file types including MS-Word and OpenOffice. AsterFax supports all Digium hardware as well as a large variety of Fax boards. |
| 14:11.50 | webman | supermathie: any of the digital telephony interface cards.... hmmm, possibly the openswitch cards + drivers might do it |
| 14:12.44 | KnowWhat | webman: http://www.voip-info.org/wiki-Asterisk+fax <-- for you |
| 14:12.48 | Supermathie | So I'd need a PRI at least? |
| 14:13.10 | dj015 | please guys, what is the difference betweeen ast_channel's nativeformats and ast_channel_tech's capabilities? |
| 14:13.34 | KnowWhat | dj015: really dont know much about that |
| 14:13.38 | webman | knowwhat: AsterFax requires trixbox or Asterisk with the spandsp (txfax, rxfax) extensions. |
| 14:13.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:13.44 | webman | supermathie: or BRI |
| 14:14.23 | KnowWhat | there is a link there for tx and rxfax i think |
| 14:15.20 | dj015 | KnowWhat, who would know? |
| 14:15.38 | KnowWhat | may be lots of ppl here who dont wanna tell us :P |
| 14:16.40 | KnowWhat | webman: http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4. |
| 14:16.44 | webman | knwowhat: nope, no links to it, only a link to the voip-info wiki page, which links to www.soft-switch.org which doesn't seem to be online at the moment |
| 14:17.15 | webman | knowwhat: I have that URL, but I can't access the site... seems to be down now..... |
| 14:17.20 | KnowWhat | oh |
| 14:17.25 | KnowWhat | then there would be a problem |
| 14:18.00 | *** join/#asterisk _deg_ (n=deg@200.195.161.164) |
| 14:18.15 | KnowWhat | may be someone have it here |
| 14:18.19 | KnowWhat | but i dont have that |
| 14:18.22 | webman | yeah... :( that is why I was hoping someone might either know the new address (if there is one) or of another place where the files are.... eg, maybe someone downloaded them for themselves and can send them to me :) |
| 14:18.26 | Supermathie | webman: thanks. Darn. :) |
| 14:18.27 | KnowWhat | right now i am stuck with the vicidial stuff |
| 14:18.27 | _deg_ | Someone could help me on building a stress test scenario using sipp + asterisk? |
| 14:18.54 | webman | BTW, I am looking for spandsp + tx_fax + rx_fax for asterisk 1.4 if anybody has them please :) |
| 14:19.24 | webman | knowwhat: hehe, actually, that will be my task for tomorrow :) |
| 14:19.51 | KnowWhat | lol |
| 14:19.56 | KnowWhat | webman: good luck then |
| 14:20.06 | KnowWhat | the problem is that i am only using softphones in here |
| 14:20.09 | KnowWhat | and no PRI here |
| 14:20.12 | webman | supermathie: in my experience, digital interfaces are significantly more reliable, and lead to very few problems!! |
| 14:20.22 | KnowWhat | plus, i am a newbie to linux stuff, still learning |
| 14:20.23 | Supermathie | And also more $$$ :) |
| 14:20.29 | KnowWhat | dont know much about AGI callss |
| 14:21.27 | webman | knowwhat: aren't the instructions pretty straight forward?? (I haven't really looked at the install docs yet, so they might not be) |
| 14:21.49 | webman | I would have thought most of the programming side would be done... just a matter of using it? |
| 14:22.35 | KnowWhat | webman: well if you are familiar with linux, i think then there will be no problem for you |
| 14:23.03 | KnowWhat | and if using the same equipment and same things as mentioned in scratch_install document, then you are ok with it |
| 14:23.10 | webman | knowwhat: well, AFAIK, AGI is almost identical to CGI, if you have ever used that? |
| 14:23.25 | *** join/#asterisk TypMic1004 (n=chrislro@outland.cmf.nrl.navy.mil) |
| 14:23.28 | dj015 | webman, i have them them, but you really want to use openpbx |
| 14:23.29 | KnowWhat | but if you are using softphones like me, then i dont know |
| 14:24.17 | KnowWhat | webman: yeah i got that somehow, but whats actually the relation of screen with AGI? |
| 14:24.22 | webman | dj015: why would I want to do that? |
| 14:25.18 | dj015 | uhm, because steve underwood develops openpbx and not asterisk, because it support T.38, because it has the latest version of everything fax-related, and because it generally works better |
| 14:25.21 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 14:25.23 | webman | knowwhat: well, with CGI stdin comes from the web server and stdout goes to the web browser... in AGI, both stdin/stdout are commannds to/from asterisk.... there isn't really a screen at all |
| 14:25.47 | TypMic1004 | I just built and iinstalled 1.4.0, including the samples and when I run "asterisk -vvvvvc" it dumps the execution info then exits asterisk am I missing something |
| 14:26.22 | *** join/#asterisk anthm (n=anthm@64.241.37.140) |
| 14:26.22 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:26.54 | webman | dj015: yeah, I know those things, but in this case asterisk is a better solution for me.. I don't need T.38/etc... could you please send me the files, or let me know where I could get them from please? |
| 14:26.56 | mafkees | heya all |
| 14:27.22 | *** join/#asterisk vanumo (n=david@62.99.154.2) |
| 14:28.04 | webman | TypMic1004: you will need to provide more information than that if anyone is going to be able to help you. Perhaps you would be better off using trixbox or similar |
| 14:28.22 | dj015 | webman, give me you email address |
| 14:31.06 | KnowWhat | well |
| 14:31.32 | KnowWhat | webman: if i call agi scripts without the screen command, it always says failed to execute script |
| 14:31.55 | webman | knowwhat: what do you mean call them?? you mean from the command line ? |
| 14:32.03 | KnowWhat | if i do screen -L -d -m -S asterisk /usr/sbin/asterisk -vvvvvvvgc |
| 14:32.05 | webman | or from asterisk extensions ? |
| 14:32.10 | TypMic1004 | yes I havent bit level checked the "samples" but one would assume that it would function out of the box, therefore I was wondering what tweaking might the be required to the samples file. The last time I worked with asterisk, about 2 years ago, it worked out of the box |
| 14:32.15 | KnowWhat | webman: from asterisk extension of course |
| 14:33.29 | webman | so if you start asterisk from screen, then the agi works, and if you start asterisk any other method, then the agi doesn't work ?? Is that what you are saying? |
| 14:33.53 | KnowWhat | yup |
| 14:34.23 | webman | knowwhat: so which agi are you using?? does a 'simple' AGI work?? |
| 14:34.30 | vanumo | use here anybody chan_bluetooth ? |
| 14:34.39 | vanumo | sorry chan_cellphone ? |
| 14:35.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:38.18 | KnowWhat | webman: simple AGI?? |
| 14:38.25 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 14:38.38 | KnowWhat | webman: dont mind but i am just a newbie |
| 14:39.05 | webman | knowwhat: do you have php installed on your asterisk box? |
| 14:39.12 | KnowWhat | yeah |
| 14:39.17 | KnowWhat | thats why vicidial is working |
| 14:39.28 | KnowWhat | the vicidial scripts are working |
| 14:39.50 | webman | so what is the problem ?? whats not working? |
| 14:40.21 | KnowWhat | webman: i am notusing a PRI or any device |
| 14:40.26 | KnowWhat | so i am using ztdummy |
| 14:40.38 | KnowWhat | with ztdummy, the rest of configuration should work |
| 14:40.49 | KnowWhat | i dont think so what do you say? |
| 14:41.49 | KnowWhat | and my AGI calls are working like this AGI(AGI(agi://127.0.0.1/call_log) |
| 14:41.55 | webman | the only reason you need zaptel is so that meetme will work. ztdummy will allow meetme to work fine, as long as the box isn't too loaded, and I think a 2.6.x kernel will work better |
| 14:42.11 | KnowWhat | but when i only use AGI(agi://127.0.0.1/call_log) |
| 14:42.22 | KnowWhat | webman: i am on 2.6.xx kernel of course |
| 14:42.46 | KnowWhat | but when i only use AGI(agi://127.0.0.1/call_log) <-- it doenst work |
| 14:43.00 | webman | knowwhat: but your AGI should be like this: exten => 694,n,AGI(lucy.agi) |
| 14:43.49 | KnowWhat | well its not working like that, i dont know why |
| 14:43.57 | KnowWhat | and there is AGI server is running too i think |
| 14:44.56 | webman | knowwhat: the format you are using looks like the type where the agi is running on another machine... I can't remember the name for it but ???AGI |
| 14:45.48 | KnowWhat | FastAGI? |
| 14:46.30 | webman | yeah, that might be it... (I noload'ed it so it didn't show up in my list) |
| 14:47.39 | KnowWhat | i dont know but the scracth install says you need to install some perol components to work AGI scripts |
| 14:47.47 | KnowWhat | i think its Net::Server |
| 14:48.03 | KnowWhat | perl components* |
| 14:48.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:49.14 | KnowWhat | webman: i am using DSL, i wanna start simple, what i wanna know is to make the vicidial working on 10 systems with xlite softphone |
| 14:49.37 | webman | knowwhat: sounds like they are using perl AGI scripts, and they might use the fastagi to be faster/better |
| 14:49.38 | KnowWhat | thats only what i want to achieve, may be you can help me out in this |
| 14:49.53 | webman | knowwhat: I haven't actually installed it yet, but I need to by monday.... |
| 14:51.03 | KnowWhat | oh ok |
| 14:51.24 | KnowWhat | so can i have your any of messenger id, to ask you furthur after you install it |
| 14:57.25 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:58.06 | *** join/#asterisk santibiotico (n=santi@108.Red-88-14-252.dynamicIP.rima-tde.net) |
| 14:58.07 | santibiotico | hi |
| 14:58.14 | KnowWhat | hello |
| 14:58.39 | santibiotico | is there any app i can use in extensions.conf to hide the caller id? |
| 14:59.52 | cpatry | santibiotico: just overwrites it with "" ? |
| 14:59.57 | webman | santibiotico: yes |
| 15:00.25 | webman | SetCallerPres(prohib) |
| 15:00.49 | webman | probably is new syntax in 1.4 or newer 1.2 possibly... I use a pretty old 1.2.x on that system |
| 15:01.51 | KnowWhat | webman: are you going to install vici on 1.4 |
| 15:02.08 | webman | knowwhat: that is the plan... why ?? |
| 15:02.26 | KnowWhat | hmm |
| 15:02.27 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqkq.cable.mindspring.com) |
| 15:02.37 | KnowWhat | do let me know if it works with that, i also wanna upgrade |
| 15:02.50 | KnowWhat | i simply followed the scratch document |
| 15:03.05 | webman | what version do you currently use? |
| 15:03.40 | KnowWhat | 1.2 |
| 15:03.45 | KnowWhat | i have 1.4 installed on the other system |
| 15:03.57 | KnowWhat | but with vicidial i have 1.2 |
| 15:03.58 | webman | which version of 1.2 ?? |
| 15:04.47 | *** join/#asterisk qdk (n=qdk@90.184.3.249) |
| 15:05.02 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:09.01 | KnowWhat | yup |
| 15:10.31 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 15:11.12 | ChicagoBud | is there a way to catch a hangup in a context to perform some wrap up processing? (i.e. after recording a message, I want to call an agi app) |
| 15:11.16 | *** join/#asterisk ManxPower (n=manxpowe@15.sub-70-212-233.myvzw.com) |
| 15:11.45 | webman | ChicagoBud: use the h exten, and call DeadAGI(someagi) |
| 15:11.57 | ChicagoBud | webman, thanks |
| 15:14.12 | ChicagoBud | if I explictly call hangup(), will the h exten still get trigered? |
| 15:14.50 | webman | ChicagoBud: try it and see.... I would guess yes.... |
| 15:15.06 | ChicagoBud | ok |
| 15:16.52 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 15:17.32 | cpatry | ChicagoBud: yes it will triger h. |
| 15:17.46 | mercestes | ~seen anthonynl |
| 15:17.55 | jbot | i haven't seen 'anthonynl', mercestes |
| 15:17.55 | *** join/#asterisk Rhizome (n=Rhizome@c9193BF51.dhcp.bluecom.no) |
| 15:17.57 | ChicagoBud | cpatry, thanks. |
| 15:18.31 | Rhizome | Anyone gotten hints working correctly with 1.4 and snom? |
| 15:18.33 | Rhizome | :D |
| 15:18.59 | mercestes | Gah, who was the guy trying to sell servers in here? Anthonyl? |
| 15:19.35 | phearless | is there anything like snapanumber.com for Linux ? |
| 15:19.53 | phearless | on linux I do not see any plugin for thunderbird for asterisk |
| 15:19.54 | phearless | :( |
| 15:21.12 | *** join/#asterisk eald (n=eald@189.157.105.134) |
| 15:21.33 | mercestes | ~seen anthonyl |
| 15:21.40 | jbot | anthonyl <n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net> was last seen on IRC in channel #asterisk, 1d 20h 43m 36s ago, saying: 'how mang g729 liceneses do i need'. |
| 15:21.48 | *** join/#asterisk AlfaScorpii (n=alfascor@64-12-16-190.fibertel.com.ar) |
| 15:21.54 | AlfaScorpii | morning people |
| 15:22.01 | mercestes | morning Alfascorpii |
| 15:22.10 | AlfaScorpii | need to know if im correct |
| 15:22.18 | AlfaScorpii | mercestes: hello my friend |
| 15:23.19 | AlfaScorpii | if im triying to make voIP the best way is Asterisk + Digium? |
| 15:23.38 | ryant | yep |
| 15:24.04 | AlfaScorpii | ok |
| 15:24.12 | AlfaScorpii | tks ryant |
| 15:24.49 | AlfaScorpii | now... what caind of Digium hard i need to replace this pice of sh. called Micronet SP5050? |
| 15:25.31 | ryant | what's a micronet SP5050, what kind of specs does it have |
| 15:25.38 | ryant | is it for FXO, FXS, PRI? |
| 15:26.48 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:26.54 | coppice | the SP5050 is a nice box |
| 15:27.38 | ryant | hmm yeah it just has six FXO ports |
| 15:27.48 | AlfaScorpii | ryant: gateway voip 6 pstn FXO |
| 15:27.52 | ryant | seems like you could just use that WITH asterisk |
| 15:27.59 | ryant | what's your problems with it? |
| 15:28.04 | AlfaScorpii | coppice: i cant maket work with Asterisk!!!! |
| 15:28.36 | AlfaScorpii | ryant: 2 mounth of problems and i only can make outgoing calls |
| 15:28.42 | ryant | ok |
| 15:28.49 | AlfaScorpii | ryant: the problem is incoming calls |
| 15:29.13 | ryant | well you're best bet is to get a couple digium TDM cards with FXO modules. It'll cost some money, but save you time in the long run. Plus you'd get as much support as you want from Digium |
| 15:30.01 | AlfaScorpii | ryant: now im loosing my job |
| 15:30.30 | mercestes | AlfaScorpii: You were loosing your job two weeks ago |
| 15:30.40 | mercestes | just moon the boss and get it over with |
| 15:31.00 | ryant | AlfaScorpii, looks like a very propietary device, so yeah probably won't speak with asterisk correctly |
| 15:31.05 | AlfaScorpii | mercestes: yes... and the 1st of next mounth i finally lose my job if this dont work |
| 15:31.14 | ryant | get the digium slap then in a server and call it a day |
| 15:31.26 | ryant | damn, get digium then |
| 15:31.28 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 15:31.36 | ryant | if you job literally is dependent on it |
| 15:31.53 | ManxPower | You can do it right or you can waste your life trying to do it wrong. |
| 15:32.14 | AlfaScorpii | ManxPower: ? |
| 15:32.23 | ryant | he's saying to do what I say :) |
| 15:32.46 | mercestes | I second that emotion. |
| 15:32.51 | AlfaScorpii | ryant: tks again... |
| 15:33.04 | ryant | do you have a manula for that device |
| 15:33.14 | mercestes | ofcourse, I daresay if your going to get fired if you can't get it working.....they'll likely fire you immediately after you get it working unless it depends on your presence for some reason. |
| 15:33.15 | AlfaScorpii | how much that it cost? i meen the Digium cards for replace the Micronet |
| 15:33.21 | Defend | any one know if sipura will ever include ilbc? or know of any decent atas that use ilbc? |
| 15:33.24 | ryant | I'd like to see teh SIP stuff for it but the company's website isn't letting me download it |
| 15:33.30 | mercestes | so make certain to write a cronjob that kills asterisk everynight at 11p. |
| 15:33.47 | Defend | my parents isp sucks! and i need a more tolerent codec i think |
| 15:33.56 | ryant | use asteriskNOW but don't tell them about the gui, then they'll have to keep you around. :) |
| 15:34.09 | mercestes | and kill it with a cronjob. |
| 15:34.33 | aydiosmio | in an AGI I'm doing a MixMonitor and a Dial, after hangup the script needs to write to a file but the script seems to terminate on hangup instead of continuing... what's the issue? |
| 15:35.07 | AlfaScorpii | ryant: how much cost that digium card that i need? |
| 15:35.10 | *** join/#asterisk shinux__ (n=shinux@196.207.1.30) |
| 15:35.22 | ManxPower | Defend: You do know that Asterisk before 1.4 does NOT have a jitter buffer for RTP (SIP Audio) right? ANY jitter will cause significant call quality problems. |
| 15:35.29 | Defend | or maybe ya guys have any ideas ... parents ata -> asterisk -> my voip phone sounds good but parents ata -> asterisk -> sip provider sounds jittery |
| 15:35.46 | [TK]D-Fender | AlfaScorpii: I'd expect a decent 6-port solution to cost around 700$ USD |
| 15:36.33 | Defend | so manxpower you saying that i should swap to 1.4 and it will alow me to do some jitter buffering or something? |
| 15:36.36 | ryant | http://www.voipsupply.com/manufacturers/Digium.html look here for acard that does 4 FXO and another card with 2, |
| 15:36.36 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 15:36.39 | AlfaScorpii | [TK]D-Fender: 2.100 pesos argentinos tks |
| 15:37.14 | ryant | http://www.voipsupply.com/product_info.php?products_id=294 this one has four and it's 378.00 |
| 15:37.23 | JoNate | Telepathy works pretty well... |
| 15:37.28 | aydiosmio | ah, guess I need ot use deadagi instead |
| 15:37.29 | webman | ryant: wouldn't the 8port card be better ? |
| 15:37.29 | mercestes | and it's cheaper. |
| 15:37.38 | JoNate | much MUCH cheaper |
| 15:37.49 | ryant | ah, didn't know they had the eight :) |
| 15:37.53 | ryant | been a while since I bought mine |
| 15:37.54 | [TK]D-Fender | AlfaScorpii: Rather than do that it might be worth it to have a consultant look at your entire setup and attempt to get functional "as-is". |
| 15:38.16 | ryant | actually one of the 8 ports are a grand |
| 15:38.36 | AlfaScorpii | [TK]D-Fender: please other words my english is a disaster |
| 15:38.42 | webman | ryant: yeah, but I think they include the echo canceller or something... I haven't looked at them too much |
| 15:38.43 | coppice | an SP5050 should work fine |
| 15:38.58 | [TK]D-Fender | AlfaScorpii: Maybe youshould pay a consultant to make your MicroNet unit WORK. |
| 15:39.06 | mercestes | AlfaScorpii: esta pay dinero hombre trabajo con yo. |
| 15:39.23 | ryant | http://www.voipsupply.com/product_info.php?products_id=1108 this 8FXO is just over a grand |
| 15:39.27 | [TK]D-Fender | AlfaScorpii: And coppice has faith that it'll do the job and likely only needs to be configured right and tested. |
| 15:39.38 | ManxPower | If you need 8 ports why not just go with a PRI |
| 15:39.43 | [TK]D-Fender | AlfaScorpii: At which point it'd be far cheaper to pay someone to make it work. |
| 15:40.46 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 15:40.50 | Rhizome | hm, so the snom lamp won't turn on unless the calllimit is reached? |
| 15:41.00 | [[blah]asfd | I am getting an error... seems asterisk is not happy. http://pastebin.ca/369083 |
| 15:41.07 | [[blah]asfd | can anyone help me to identify the issue? |
| 15:41.19 | *** join/#asterisk apardo (n=apardo@87.217.145.9) |
| 15:41.23 | ryant | http://www.voipsupply.com/product_info.php?products_id=1107 actually this one is only 837 with 8FXO ports but it doesn't ahve teh echo cancellation built in |
| 15:41.36 | AlfaScorpii | coppice: dou you have experience with micronet sp5050 |
| 15:41.37 | AlfaScorpii | ? |
| 15:42.02 | ryant | blah, I'm not sure why you get that |
| 15:42.08 | webman | well, I now have asterisk 1.4.0 installed, seems to work with incoming fax/queues/etc and even my sip hints work too.... now for some sleep :) |
| 15:42.19 | [TK]D-Fender | ryant: http://www.voipsupply.com/product_info.php?products_id=1108 |
| 15:42.21 | kFuQ | [[blah]asfd: bindaddr= |
| 15:42.22 | ManxPower | AlfaScorpii: we spent a year working with Asterisk before we even considered installing it as a production system. |
| 15:42.26 | AlfaScorpii | This is my Micrronet config http://www.pastebin.ca/367732 |
| 15:42.32 | [TK]D-Fender | ryant: You looked REALLY hard didn't you? ;) |
| 15:42.38 | [[blah]asfd | kFuQ: where is that at? |
| 15:42.42 | kFuQ | sip.conf |
| 15:42.49 | [[blah]asfd | ahh.. hell. thats right. |
| 15:42.54 | kFuQ | lol |
| 15:43.03 | AlfaScorpii | ManxPower: yes i now that is no siple work |
| 15:43.04 | kFuQ | always the stupid shit isn't it ........ :-D |
| 15:43.10 | AlfaScorpii | http://www.pastebin.ca/367732 |
| 15:44.28 | [[blah]asfd | kFuQ: its set to 0.0.0.0. that should be ok, right? |
| 15:44.29 | ryant | Fender I already pasted that one a long time ago, and over a grand isnt' too cheap though he hasn't mentioned his limit |
| 15:44.35 | [[blah]asfd | it has worked before... |
| 15:44.53 | [[blah]asfd | i just installed vicidial on it and rebooted, now it doesnt work. |
| 15:45.23 | AlfaScorpii | anybody can take a look http://www.pastebin.ca/367732 please |
| 15:46.03 | kFuQ | [[blah]asfd: should be fine.. |
| 15:46.29 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:46.38 | ryant | Alfascorpii, is your SIP config set to peer-to-peer or proxy? |
| 15:47.11 | AlfaScorpii | ryant: proxy i think |
| 15:47.22 | AlfaScorpii | ryant: how can i checkit? |
| 15:47.28 | [[blah]asfd | y |
| 15:47.34 | [[blah]asfd | err. wrong window... sorry |
| 15:47.38 | ryant | in your web config |
| 15:47.43 | ryant | I'm looking at it in the manual right now |
| 15:48.51 | AlfaScorpii | ryant: in the micronet? yes! is seted like Proy mode |
| 15:49.01 | *** join/#asterisk shinux__ (n=shinux@196.207.1.30) |
| 15:49.10 | ryant | try peer to peer |
| 15:49.16 | [TK]D-Fender | ryant: a fair bit cheaper : http://www.voipsupply.com/product_info.php?products_id=1341 |
| 15:49.28 | *** join/#asterisk intralanman (n=lanman@pool-71-253-242-197.nrflva.east.verizon.net) |
| 15:49.29 | [TK]D-Fender | ryant: or : http://www.voipsupply.com/product_info.php?products_id=1341 |
| 15:50.04 | AlfaScorpii | ryant: but like peer2peer i have to change Asterisk config? |
| 15:50.51 | ryant | AlfaScorpii, you are trying to forward ALL of the FXO ports in the device to asterisk, so asterisk can handle the calls correct? |
| 15:51.14 | AlfaScorpii | ryant: yes |
| 15:51.19 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 15:52.08 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 15:52.40 | ryant | so there devices are pretty much side by side on the same subnet? |
| 15:52.55 | ryant | and you have all the passwords set, etc for the SIP on each line in the device? |
| 15:53.14 | AlfaScorpii | ryant: yes |
| 15:54.01 | ryant | seems like asterisk would have to act as an SIP client and login to EACH of these lines in order to register right? |
| 15:54.10 | ryant | and you probably did this as you can call outbound right? |
| 15:55.03 | AlfaScorpii | ryant: yes |
| 15:55.48 | ryant | so teh outbound calls route correctly, but the device isn't forwarding calls correctly? Did you enable debug to see if anything comes up in asterisk with errors, etc |
| 15:55.48 | ryant | ? |
| 15:56.04 | *** join/#asterisk mtgll (n=mtg@static-71-125-10-2.nycmny.fios.verizon.net) |
| 15:56.25 | AlfaScorpii | ryant: yes, nothig shows on asterisk debug |
| 15:56.44 | mtgll | greetings need some help with modules |
| 15:56.53 | AlfaScorpii | ryant: when i call to the lines from outside nothing shows in debug |
| 15:57.29 | mtgll | updated the system to the most recent branch now asterisk will not start getting WARNING[13702] loader.c: /usr/lib/asterisk/modules/res_convert.so: undefined symbol: ast_module_unregister |
| 15:57.37 | ryant | do you have a route setup in the device to forward calls to each of the registered SIP clients, ie the asterisk box? |
| 15:57.43 | mtgll | any thoughts ? |
| 15:58.13 | *** join/#asterisk Ryushin (n=Ryushin@windwalker.openinnovations.com) |
| 15:58.37 | ryant | sorry mtgll I don't know THAT much :) |
| 15:58.55 | mtgll | did a search but didnt find much thanks |
| 15:59.18 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:59.22 | *** part/#asterisk TypMic1004 (n=chrislro@outland.cmf.nrl.navy.mil) |
| 15:59.23 | AlfaScorpii | ryant: yes |
| 16:00.02 | ryant | seems like this device should work in theory and perhaps just a simple config issue on the device. |
| 16:00.16 | ryant | is there any logging for the device to see if it recognizes the incoming call to it? |
| 16:00.41 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 16:00.55 | AlfaScorpii | ryant: yes micronet have a sip debug |
| 16:01.07 | ryant | do you see anything on that debug? |
| 16:01.12 | *** part/#asterisk mtgll (n=mtg@static-71-125-10-2.nycmny.fios.verizon.net) |
| 16:01.23 | AlfaScorpii | ryant: yes but cant understand it |
| 16:01.30 | AlfaScorpii | ryant: wait i show you |
| 16:01.31 | ryant | pastebin it |
| 16:01.38 | *** join/#asterisk thinwires (n=thinwire@24-49-196-96.kntnny.adelphia.net) |
| 16:01.52 | ryant | also seems like you could debug the analog side for the FXO ports somehow |
| 16:02.22 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 16:02.45 | *** join/#asterisk murdmath (n=vircuser@mail.kimballequipment.com) |
| 16:02.48 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:02.49 | murdmath | Howdy all. |
| 16:03.46 | murdmath | Is it possible to get caller id information to show on a phone that has picked up a parked call? |
| 16:04.39 | cpatry | just to see the callerid from the parked person? |
| 16:05.12 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 16:05.37 | murdmath | Right now If I pickup a parked call I only see the parking spot number on my caller id. |
| 16:05.45 | *** join/#asterisk intralanman (n=lanman@pool-71-253-242-197.nrflva.east.verizon.net) |
| 16:06.33 | ManxPower | murdmath: you actually don't see any callerid, you see the number you dialed |
| 16:06.42 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:07.21 | [[blah]asfd | ok, so i did a make samples to clear out everything i have done in asterisk configs... and I got rid of most of my errors, but this one still remains: Feb 23 08:09:37 WARNING[5472]: chan_sip.c:12947 reload_config: Unable to get own IP address, SIP disabled. Can anyone help me fix this? |
| 16:07.22 | murdmath | ManxPower: That is correct. |
| 16:09.06 | *** join/#asterisk ModocNet (n=d82e036a@69-12-147-8.dsl.static.sonic.net) |
| 16:09.55 | ManxPower | [[blah]asfd: your /etc/hosts file is wrong |
| 16:10.00 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:10.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:10.09 | murdmath | ManxPower: Is that normal? Is it something that can be fixed? |
| 16:10.11 | [[blah]asfd | ok, thanks |
| 16:10.22 | *** join/#asterisk intralanman (n=lanman@pool-71-253-242-197.nrflva.east.verizon.net) |
| 16:10.23 | ManxPower | murdmath: there really isn't support for what you want |
| 16:10.49 | [[blah]asfd | ManxPower: i just have the 127.0.0.1 localhsot.localdomain in there... that should work, shouldnt it? |
| 16:10.52 | ModocNet | running CentOS 4.4 - TE205 (dual span PRI) - EVERYTIME after rebooting I have to modprobe zaptel & wxt2xxp - any ideas? |
| 16:11.22 | murdmath | ManxPower: Is it because the call is "pulled" to the phone vs "pushed" like a transfer? |
| 16:11.36 | ManxPower | [[blah]asfd: does the machine have only 1 IP address? |
| 16:11.38 | [[blah]asfd | ModocNet: from zaptel install directory run make config |
| 16:11.50 | [[blah]asfd | yes |
| 16:11.53 | ManxPower | murdmath: yes |
| 16:12.05 | ModocNet | ahhhhh... just like in the * dir....thanks....it's always the lilttle things...lol |
| 16:12.17 | ManxPower | [[blah]asfd: then don't worry about it. If your machine has only the IP address 127.0.0.1 you can't do any IP anyway |
| 16:12.41 | aydiosmio | does MixMonitor write directly to disk or flush from RAM after hangup? |
| 16:12.59 | [[blah]asfd | aydiosmio: directly to disk |
| 16:13.09 | aydiosmio | 15KPRM disk it is then |
| 16:13.16 | Nugget | it writes as the call proceeds, you can actually listen as the call progresses. |
| 16:13.24 | aydiosmio | oh neat |
| 16:13.41 | [[blah]asfd | i recommend using ramdisk and doing a simlink from your monitor directory to ramdisk |
| 16:14.39 | aydiosmio | hm. |
| 16:14.47 | murdmath | ManxPower: Does Asterisk still retain the caller id info on that call anyway, so maybe I could sms that info to the phone? |
| 16:14.49 | aydiosmio | that's an interesting way to go |
| 16:15.13 | aydiosmio | I'm trying to spec a PC so it can record in the dozens of simultaneous calls |
| 16:15.59 | [[blah]asfd | aydiosmio: I have about 80 concurrent calls being recorded on one of my systems. I record to ramdisk and then a cron moves them to my archive server where they are then comressed to gsm. |
| 16:16.07 | [[blah]asfd | also i do not like mixmonitor |
| 16:16.08 | *** join/#asterisk queztor (i=questor@office.endoria.net) |
| 16:16.09 | aydiosmio | oh great! |
| 16:16.12 | [[blah]asfd | i have had too many issues. |
| 16:16.14 | queztor | hi all :) |
| 16:16.17 | aydiosmio | yeah I hear it's had some issues |
| 16:16.24 | aydiosmio | [[blah]asfd: what are the PC specs? |
| 16:16.29 | [[blah]asfd | just use monitor with the m option. |
| 16:16.34 | ManxPower | murdmath: yes the callerid info is still associated with the call, no I don't know how to do what you want. |
| 16:16.44 | [[blah]asfd | i am using the hp dl380 with 4gb of ram |
| 16:17.25 | aydiosmio | [[blah]asfd: is the ram pretty critical in terms of capacity? |
| 16:17.30 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 16:17.33 | [[blah]asfd | yes |
| 16:17.50 | Qwell[] | [[blah]asfd: That's a pretty decent idea |
| 16:17.52 | [[blah]asfd | because the calls can take some time to complete, you need enough room to store them |
| 16:17.59 | Qwell[] | [[blah]asfd: the ramdisk for recording.. |
| 16:18.16 | [[blah]asfd | yeah... it saved our bacon when we started getting more than 20 concurrent calls |
| 16:18.33 | *** join/#asterisk blinx (n=blinx@unixboard/user/blinx) |
| 16:18.37 | blinx | hi |
| 16:18.47 | blinx | I want to create my own sip id on my server |
| 16:18.52 | aydiosmio | [[blah]asfd: what processor option in the hp? |
| 16:18.53 | blinx | I have a vserver and full root access |
| 16:19.01 | blinx | is there any howto? |
| 16:19.11 | Qwell[] | blinx: if it's a virtual server, you don't have "full root access" |
| 16:19.22 | blinx | but on my instance ;-) |
| 16:19.30 | blinx | xen |
| 16:19.40 | [[blah]asfd | aydiosmio: hang on, let me look |
| 16:19.44 | aydiosmio | thx |
| 16:20.01 | aydiosmio | the difference between 1 xeon and 2 quad cores is pretty big:) |
| 16:20.01 | blinx | is this possible? |
| 16:20.05 | [[blah]asfd | Intel(R) Xeon(TM) CPU 3.00GHz |
| 16:20.13 | [[blah]asfd | dont do the quad core |
| 16:20.19 | [[blah]asfd | just do a single proc |
| 16:20.20 | aydiosmio | yeah I'm sure |
| 16:20.31 | aydiosmio | yeah I'm not goign to venture into SMP for asterisk |
| 16:20.39 | [[blah]asfd | i have 80 concurrent calls right now and I am only using 7% cpu |
| 16:20.42 | queztor | lads, I have a little problem with musiconhold in 1.4.0, anyone time? |
| 16:20.45 | aydiosmio | awesome |
| 16:20.50 | [[blah]asfd | spm for asterisk does not work in my experience. |
| 16:20.55 | [[blah]asfd | smp |
| 16:21.01 | Qwell[] | [[blah]asfd: umm |
| 16:21.14 | [[blah]asfd | i am sure it does... but i have not had a good experience with it |
| 16:21.19 | Qwell[] | [[blah]asfd: asterisk is heavily multithreaded |
| 16:21.49 | [[blah]asfd | hyperthreading i meant... sorry |
| 16:22.09 | Qwell[] | multicore CPUs don't have hyperthreading |
| 16:22.13 | [[blah]asfd | right. |
| 16:22.29 | Qwell[] | ht was a big joke anyhow |
| 16:22.40 | [[blah]asfd | not focusing on the conversation... working on my other little issue and trying to type at the same time :-D |
| 16:23.10 | queztor | hehe |
| 16:23.36 | queztor | Qwell[]: I am at a loss with MusicOnHold.. does it play default to the ringing party if you put them on hold or not? |
| 16:24.41 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 16:24.56 | *** join/#asterisk MarkWD (n=Mark@rrcs-67-78-88-186.sw.biz.rr.com) |
| 16:24.57 | ManxPower | queztor: only with the m option to dial |
| 16:25.01 | aydiosmio | [[blah]asfd: what file format are you writing to the ramdisk? and how big is the average file size in the ramdisk? |
| 16:25.24 | queztor | ManxPower: ok :) thx.. I'll gonna fiddle with that |
| 16:26.17 | [[blah]asfd | aydiosmio: -rw-r--r-- 1 root root 683404 Feb 23 09:23 1172247806.239924.wav |
| 16:26.43 | aydiosmio | thanks |
| 16:26.49 | aydiosmio | I owe you a beer |
| 16:26.56 | [[blah]asfd | come on over |
| 16:27.15 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 16:27.22 | pigpen | with * 1.2, to to intercom's to polycom phones I did: exten => _2XX*,1,SetVar(_ALERT_INFO="Ring_Ans") |
| 16:27.43 | pigpen | I moved to 1.4, changing to: exten => _2XX*,1,Set(_ALERT_INFO="Ring_Ans") |
| 16:27.51 | pigpen | ...but no dice...just rings. |
| 16:28.03 | pigpen | any ideas? |
| 16:28.30 | KnowWhat | any good tutorial about dialplan? |
| 16:29.29 | queztor | ManxPower: that didn't do the job... when it rings, the calling party hears the music, but when I pick up, put them on hold.. they hear nothing |
| 16:29.37 | [[blah]asfd | KnowWhat: http://www.voip-info.org/wiki-Asterisk |
| 16:29.47 | fetcher | pigpen: maybe try removing the leading underscore? Although the need for that is probably a device issue, rather than * version.. |
| 16:30.26 | ManxPower | queztor: they should hear MoH when put on hold regardless |
| 16:30.27 | pigpen | fetcher, I have around 1000 phones working fine as stated.... |
| 16:30.48 | pigpen | the catch is moving to asterisk 1.4, setvar is depreciated in favor of set |
| 16:30.53 | queztor | ManxPower: yes, but that doesn't happen.. any thoughts? |
| 16:31.13 | queztor | it's an mISDN channel btw |
| 16:31.29 | queztor | with the SIP channel it does work |
| 16:31.54 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 16:32.27 | aydiosmio | [[blah]asfd: that's the standard asterisk 8khz 16bit mono wav format right? |
| 16:32.37 | [[blah]asfd | yep |
| 16:32.37 | ez` | tell me if i am wrong : build_route: Contact hop: <sip:480@10.10.10.20:5066> . this mean its try to reach extension 480 on server 10.10.10.20 ??? |
| 16:32.45 | aydiosmio | trying to figure out how many hours will fit on the ramdisk |
| 16:33.16 | [[blah]asfd | how many concurrent calls do you plan on taking |
| 16:34.01 | aydiosmio | well, I'm getting hardware to test capacity, lookign in the 100s |
| 16:34.06 | aydiosmio | so a few servers will be involved |
| 16:34.19 | aydiosmio | 80 per sounds about right though |
| 16:34.39 | [[blah]asfd | you will fill 4 gig pretty quickly if you dont move them off. |
| 16:34.44 | aydiosmio | think you could get more out of your current hardware? |
| 16:34.47 | [[blah]asfd | we pull them off every 5 minutes |
| 16:35.32 | [[blah]asfd | i could get much more out of the hardware, but ram is about 50% used. I dont want it to go much more than that, incase my cron fails, or i loose connection to my archive server |
| 16:35.35 | [TK]D-Fender | pigpen: You need to ensure that you have created appropriate alertinfo modes in your provisioning and associate the right ring-type with them. Also you should be using the new "SIPAddHeader application for this without quotes. |
| 16:36.21 | aydiosmio | [[blah]asfd: how do you know if a recording is complete so you don't move it during a call? |
| 16:36.58 | [[blah]asfd | the incomplete calls are labeled for each part. just exclude those that are -in.wav and -out.wav |
| 16:37.15 | aydiosmio | ah -m option combines them after the call |
| 16:37.23 | pigpen | [TK]D-Fender, yeah..the phones are setup fine....I will look into the new "SIPAddHedader" app. |
| 16:37.29 | pigpen | tks. |
| 16:37.49 | aydiosmio | heh, I guess you could technically increase capacity using MixMonitor |
| 16:38.07 | [[blah]asfd | mixmonitor will be your worst nightmare |
| 16:38.13 | aydiosmio | yeah, I'll avoid it |
| 16:38.19 | aydiosmio | too bad |
| 16:38.20 | [TK]D-Fender | pigpen: np. Typically like SIPAddHeader(Alert-Info: Ring Answer) |
| 16:38.41 | [[blah]asfd | its fine for small office use, but doing large offices like what you want to do, my experience is that it will crash asterisk |
| 16:39.17 | pigpen | [TK]D-Fender, tks again...just drudging through the little things that changed. |
| 16:39.36 | pigpen | I got realtime voicemail working natively with postgresql. |
| 16:40.35 | Nugget | yay postgresql. |
| 16:40.38 | [TK]D-Fender | pigpen: Glad to hear... how difficult this time around? |
| 16:40.59 | [TK]D-Fender | Nugget: Is that sarcastic, or do you actually like/respect it? |
| 16:41.03 | *** join/#asterisk queuetue (n=scott@70.54.254.134) |
| 16:41.06 | pigpen | well, after I got the right info...easy. |
| 16:41.30 | pigpen | just the res_pgsql.conf was not poplulated...nor examples were next to impossible to find..... |
| 16:41.33 | *** join/#asterisk smace (n=smace@200.222.2.228) |
| 16:41.49 | pigpen | but google proved worth it's weight with the search "asterisk res_pgsql.conf" |
| 16:41.55 | tzanger | anyone here used objectworld's microsoft voip solution? just looking for things to be wary of, things they don';t tell you... I'm pitching an asterisk solution and I'm up against them |
| 16:41.58 | giasai68 | hello, can i transfer a caming call on asterisk to ip of e gatekeeper? |
| 16:42.09 | queuetue | Hi. Has anyone noticed VoicePulse quality going downhill? Over the last week, we've got 50-60% of our calls get too choppy to continue. mtr reports right around zero packet loss between our server and theirs... |
| 16:42.20 | aydiosmio | [[blah]asfd: you doing Monitor in extensions or AGI? |
| 16:42.21 | [TK]D-Fender | giasai68: "show application transfer" |
| 16:42.27 | giasai68 | thank you |
| 16:42.31 | [[blah]asfd | aydiosmio: extensions |
| 16:42.50 | [[blah]asfd | exten => s,4,Monitor(wav,/dev/shm/${CALLFILENAME},m) |
| 16:42.54 | smace | my problem with bad voice quality was transcoding from g711a to g711u |
| 16:42.54 | aydiosmio | we're probably going to end up with an AGI for each call |
| 16:43.05 | [[blah]asfd | why an agi? |
| 16:43.13 | [[blah]asfd | just curious |
| 16:43.21 | aydiosmio | CDR stuff |
| 16:43.46 | giasai68 | where can i set the ip in which file? |
| 16:43.50 | aydiosmio | client wants some technical things done with call information |
| 16:44.04 | [TK]D-Fender | giasai68: In the line in extensions.conf where you call Transfer |
| 16:44.13 | pigpen | [TK]D-Fender, worked like a champ....thanks .. yet again... |
| 16:44.14 | [TK]D-Fender | giasai68: Read its instructions and go try it. |
| 16:44.20 | giasai68 | ok thank you |
| 16:44.21 | [TK]D-Fender | pigpen: Quite welcome. |
| 16:45.37 | pigpen | next: allpage |
| 16:46.48 | *** join/#asterisk sharp (n=sharp@2001:470:1f01:ffff:0:0:0:1c23) |
| 16:47.56 | [[blah]asfd | aydiosmio: we do run an agi as well to capture the cdr data, but monitor i left out. all calls get cdr, not all calls get recorded. |
| 16:48.14 | aydiosmio | ah |
| 16:48.22 | *** join/#asterisk shinux__ (n=shinux@196.207.1.30) |
| 16:48.45 | aydiosmio | so you're probably doing 80 concurrent AGI applications as well? |
| 16:48.51 | [[blah]asfd | that is correct |
| 16:48.53 | Qwell[] | eww |
| 16:49.14 | aydiosmio | hey, with his low cpu usage, it's beautiful thing |
| 16:49.15 | [[blah]asfd | as well as some php integration |
| 16:49.20 | smace | [TK]D-Fender, asterisk is working here :-) |
| 16:50.00 | tzanger | nobody here has any experiences with objectworld? |
| 16:50.07 | tzanger | or microsoft unified comms server? |
| 16:50.15 | KnowWhat | what is dialplan number? |
| 16:51.16 | [[blah]asfd | hehe |
| 16:51.35 | *** join/#asterisk aap_ (n=aap_@user-5442f3a4.lns3-c8.dsl.pol.co.uk) |
| 16:55.17 | [TK]D-Fender | aydiosmio: Fake drives, on fake machines..... feel the irony |
| 16:56.02 | [TK]D-Fender | pigpen: AllPage is fairly simple if you just exten the logic of the sample ont he WIKI. |
| 16:56.20 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 16:56.37 | thinwires | has anyone had any FC6 + Asterisk experiances? |
| 16:56.56 | aydiosmio | these numbers are scary, wav 1MB/min x 2 files x 80 concurr = 160/MB per minute to disk |
| 16:57.07 | pigpen | [TK]D-Fender, yeah..I've had it running for some time on 1.2.x...however on 1.4 it broke. |
| 16:58.04 | [TK]D-Fender | pigpen: I here all sorts of little things about 1.4 & SIP, but never mentally assembled a finished picture... |
| 16:58.04 | pigpen | it seems that the alert info is not getting passed. |
| 16:58.25 | pigpen | here is a 4 line snipplet of the agi: |
| 16:58.28 | aydiosmio | [[blah]asfd: think it's feasible to create a script to monitor the directory and remove files as they are created instead of on schedule? |
| 16:58.37 | pigpen | <PROTECTED> |
| 16:58.37 | pigpen | <PROTECTED> |
| 16:58.37 | pigpen | <PROTECTED> |
| 16:58.38 | pigpen | <PROTECTED> |
| 16:58.51 | wunderkin | sipaddheader is an app, not a variable |
| 16:59.14 | pigpen | yeah..I was kinda wondering why it was even there... |
| 16:59.19 | pigpen | I grabbed this off the wiki. |
| 16:59.24 | pigpen | about a year ago. |
| 16:59.56 | pigpen | $callinfo = 'Call-Info:<sip:domain>:answer-after=0'; |
| 17:00.01 | JoNate | the wiki? |
| 17:00.07 | pigpen | $alertinfo = 'Ring_Ans'; |
| 17:00.24 | mercestes | one of these days I will find "The Wiki." |
| 17:00.30 | mercestes | not *a* wiki, *the* wiki. |
| 17:00.32 | pigpen | the alert info is correct. |
| 17:00.35 | mercestes | and it will know all. |
| 17:00.46 | pigpen | but I am unsure of the changes in 1.4 and the call the agi is making. |
| 17:00.50 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 17:00.51 | JoNate | One of these days you won't need the wiki.. |
| 17:00.52 | wunderkin | tiki wiki |
| 17:00.59 | JoNate | you can just use Telepathy to gain all your information |
| 17:01.28 | pigpen | ah..the callinfo one is for snom...I can ditch tat. |
| 17:01.31 | pigpen | s/tat/that |
| 17:01.35 | active_si | I have one technical question, if I have a T1/E1 (PRI) card in my computer in NT mode, what do I need to connect XX (up to 30) users to this interface? |
| 17:02.03 | *** join/#asterisk Ebola (n=Ebola@host86-142-178-37.range86-142.btcentralplus.com) |
| 17:02.21 | ManxPower | active_si: there is no such mode for PRI |
| 17:02.34 | mercestes | JoNate: yes! Collective conciousness is the future!....but, it will eliminate IT as we know it. |
| 17:03.05 | ManxPower | active_si: PRI is mainly for connectng to the telco. BRI has NT/TE mode |
| 17:03.10 | JoNate | it will eliminate many things young padawan... |
| 17:03.38 | JoNate | But fear not the path to enlightenment, for it is filled with entemans chocolate covered donuts and ice cold milk |
| 17:04.33 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:04.34 | active_si | ManxPower: so if I'd want to use old ISDN phone equipment with * I'd have to use BRI cards? |
| 17:04.53 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:05.34 | ManxPower | active_si: ISDN phones use BRI, so yes. |
| 17:05.45 | ManxPower | BRI and PRI are TOTALLY different |
| 17:07.23 | active_si | ManxPower: so only if I'd have a ISDN PBX that supports PRI then I'd have the option to use PRI cards to (for example) enable recording of conversations on channels? |
| 17:07.51 | ManxPower | active_si: active_si: correct. |
| 17:08.44 | active_si | ManxPower: thanks for the explanation |
| 17:09.17 | *** join/#asterisk deb_user (n=deb_user@albuquerque.agroinnovations.com) |
| 17:09.49 | deb_user | so I'm trying to include an * in my dialplan, not as a wildcard but as an extension, like when someone pushes *86 they get voicemail... |
| 17:09.57 | deb_user | how do I do this? |
| 17:10.35 | ManxPower | exten => *86,1,Voicemailmain |
| 17:10.49 | ManxPower | make sure your SIP phone is not eating the * or the *86 |
| 17:10.58 | deb_user | manxpower, I tried that |
| 17:11.15 | ManxPower | deb_user: then your sip phone is eating the digits or not allowing the digits. |
| 17:11.29 | ManxPower | * is not a wildcard in Asterisk |
| 17:11.37 | ManxPower | XNZ. are wildcards in Asterisk |
| 17:11.52 | deb_user | manxpower: its an fxs port via a wildcard tdm |
| 17:12.12 | deb_user | * just gives me a busy signal, same as I would get if i dialed an invalid extension |
| 17:12.23 | ManxPower | deb_user: then you have some other problem |
| 17:12.28 | deb_user | though I don't see any output in the command line, even with verbose set at 3 |
| 17:12.31 | ManxPower | perhaps a context issue. |
| 17:12.38 | cpatry | deb_user: have you reload ur server after adding that extension? |
| 17:13.34 | aydiosmio | [[blah]asfd: haha, 24h of 80 concurrent comes out to about 115GB:P |
| 17:13.35 | deb_user | cpatry: yes...of course |
| 17:13.54 | cpatry | run ethereal and see what you are sending to *. |
| 17:13.58 | cpatry | (asterisk) |
| 17:14.29 | ManxPower | cpatry: he's on FXS |
| 17:14.31 | deb_user | ethereal? |
| 17:14.34 | ManxPower | so not network traffic |
| 17:14.35 | deb_user | i've never used that... |
| 17:14.41 | deb_user | lemme see |
| 17:15.11 | cpatry | and if u calling at **86 ? |
| 17:15.27 | ManxPower | deb_user: ethereal is a network sniffer, if you are using an analog card in the server, ethereal will not help you. |
| 17:16.36 | cpatry | ha, if its analog, see the digit map of ur gateway. |
| 17:16.42 | ManxPower | deb_user: add ,debug to your console => line in /etc/asterisk/logging.conf |
| 17:16.52 | deb_user | ok |
| 17:16.56 | ManxPower | cpatry: HE IS USING A DIGIUM ANALOG CARD |
| 17:17.01 | ManxPower | then do a logger reload |
| 17:17.06 | ManxPower | then do a set debug 9 |
| 17:17.13 | ManxPower | then dial and see what the debugs messages say |
| 17:17.25 | deb_user | ok |
| 17:17.46 | cpatry | ManxPower: ha, sorry didnt read since 2 hrs ago, relax. |
| 17:18.05 | *** join/#asterisk Tako-san (n=sysadmin@24.108.162.254) |
| 17:18.38 | deb_user | manxpower: i've never used logging.conf |
| 17:18.42 | deb_user | what does the syntax look like? |
| 17:18.58 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 17:19.03 | ManxPower | deb_user: the syntax is exactly like the /path/to/src/asterisk/configs/logger.conf.sample |
| 17:19.10 | deb_user | oh |
| 17:19.11 | deb_user | ha |
| 17:19.14 | deb_user | you said logging.conf |
| 17:19.17 | deb_user | but its logger.conf |
| 17:19.22 | ManxPower | sorry. |
| 17:19.25 | deb_user | now I see it |
| 17:19.31 | deb_user | should be some sample config in there |
| 17:19.32 | deb_user | one sec |
| 17:20.59 | deb_user | manxpower: done, done, and done |
| 17:21.03 | Bobthehunter | any idea on why if user using fromuser=username i get "CALLERNAME" <username> in the call ? |
| 17:21.06 | deb_user | now i see the debug output |
| 17:21.14 | js_ | do i need a special kind of addon card in my machine in order to serve sip stuff? or is a network connection sufficient? |
| 17:21.15 | deb_user | but still seems like no info gets passed to * when I dial *86 |
| 17:21.23 | Bobthehunter | and if he doesn it tries to use my inbound context |
| 17:22.01 | deb_user | there's no output at all |
| 17:22.42 | deb_user | i push * |
| 17:22.50 | deb_user | then as soon as I push 8 i get a busy signal |
| 17:23.14 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 17:23.34 | Bobthehunter | <PROTECTED> |
| 17:23.38 | ManxPower | deb_user: do you have * as a DTMF in features.conf? |
| 17:24.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:25.16 | genz | I have random 800 numbers that give me a "all circuits are busy" message. Any ideas why? |
| 17:28.03 | *** join/#asterisk lwh (n=lwh192@rdsl-0270.tor.pathcom.com) |
| 17:29.11 | [TK]D-Fender | genz: Usually that can happen if you've mangled your callerID. 800#'s don't like when you do that. This usually happens over PRI |
| 17:29.13 | genz | I've tried changing the values of pridialplan in zapata.conf to no avail |
| 17:29.35 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:29.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:29.50 | deb_user | features.conf says the default for "pickupexten" is *8 |
| 17:29.53 | genz | D-Fender: I guess that makes sense. Any suggestions on what/where to resolv? |
| 17:29.59 | [TK]D-Fender | genz: Typically * will send the CID of that originating channel (usually a 3-4 digit # associeated with an internal phone for instance)( |
| 17:30.11 | [TK]D-Fender | genz: On your dial-out, FORCE the callerID. |
| 17:30.18 | deb_user | although its commented out |
| 17:30.21 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 17:30.38 | deb_user | i suppose its *8 by default |
| 17:31.26 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:31.44 | ManxPower | genz: the value of HANGUPCAUSE will tell you the reason the telco thinks the call could not be completed |
| 17:32.42 | genz | ManxPower: Is that something I'd see in the pri debug span area? |
| 17:32.51 | deb_user | manxpower: that was it, the default in features.conf for pickupexten was *8 |
| 17:32.55 | *** join/#asterisk froguz (n=alvaro@pc-69-217-46-190.cm.vtr.net) |
| 17:32.58 | deb_user | i changed it, and now its all good |
| 17:33.01 | deb_user | thanks for your help |
| 17:33.11 | ManxPower | genz: yes, but its easier to do a Noop(HANGUPCAUSE is ${HANGUPCAUSE}) as the priority after the Dial |
| 17:34.07 | froguz | somebody know where can i buy a 2n boiceblue enterprise (VoIP GSM Gateway) in the US? |
| 17:35.19 | ManxPower | froguz: does that box even support the GSM friequencies for the USA? |
| 17:36.32 | ManxPower | remember in the USA carriers do not provide trunking features into their GSM network |
| 17:36.44 | mafkees | .nl neither |
| 17:39.26 | genz | ManxPower: HANGUPCAUSE is 4 |
| 17:39.34 | genz | ManxPower: HANGUPCAUSE is 41 |
| 17:39.43 | genz | ManxPower: (stupid copy and paste) |
| 17:40.38 | froguz | ManxPower, it's triband : 900/1800/1900 MHz |
| 17:40.46 | ManxPower | genz: See http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
| 17:40.55 | froguz | 850/1800/1900 MHz also available |
| 17:40.59 | ManxPower | the asterisk codes are in decimal |
| 17:41.15 | ManxPower | genz: what country are you in? |
| 17:41.36 | genz | MaxPower: US |
| 17:41.36 | *** join/#asterisk ruied (n=ruied@bl7-218-228.dsl.telepac.pt) |
| 17:41.55 | ManxPower | genz: then you do not want any pridialplan settings |
| 17:42.02 | froguz | also, i need it here in Chile. but i want to ship it from the US |
| 17:42.06 | ManxPower | remove them and do a unload chan_zap.so and a load chan_zap.so |
| 17:42.50 | froguz | it's less expensive than shiping it from Europe |
| 17:43.21 | Bobthehunter | exten => s,1,Set(CALLERID(all)="Client Name <16131111111>") |
| 17:43.26 | genz | MaxPower: I didn't realize that the dialplan required a reload of the modules. Then I take back saying that they were ever implemented. |
| 17:43.29 | Bobthehunter | is not working.. its hsowing user names |
| 17:43.32 | ManxPower | Bobthehunter: DO NOT USE QUOTES |
| 17:43.45 | genz | MaxPower: I just added them today trying to resolve this |
| 17:43.53 | *** join/#asterisk shinux__ (n=shinux@196.207.1.30) |
| 17:43.56 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:44.01 | Bobthehunter | trying |
| 17:44.30 | queztor | ManxPower: it's really driving me nuts :( |
| 17:45.11 | paolob | Hi guys! How can I get messages in cli not been truncated at x column? Is there a way to set a column width. (E.g.: I want the output of "show channels" not been truncated) |
| 17:45.18 | paolob | ? |
| 17:46.04 | ManxPower | paolob: not that I am aware of. |
| 17:46.18 | ManxPower | if you need that information for an application try using the Manager Interface |
| 17:46.38 | wunderkin | show channels concise or verbose |
| 17:46.54 | ManxPower | wunderkin: is that a 1.4 option? |
| 17:46.59 | wunderkin | 1.2 |
| 17:47.05 | ManxPower | nifty |
| 17:47.06 | paolob | ManxPower, ok, thnks |
| 17:47.17 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 17:47.18 | genz | MaxPower: Could it be the callerid not being forced? |
| 17:47.26 | Bobthehunter | hmm |
| 17:47.57 | ManxPower | genz: no idea. |
| 17:47.59 | fetcher | wunderkin: cool, thanks for the tip. I'd been looking for an easy way to see duration for all calls, rather than one by one |
| 17:48.22 | *** join/#asterisk saftsack (n=oliver@pD9E06B43.dip.t-dialin.net) |
| 17:48.57 | genz | [TK]D-Fender: Any suggestions for how to force the callerid? |
| 17:49.23 | queztor | ManxPower: any suggestions on the problem which I described earlier, or? |
| 17:50.09 | ManxPower | queztor: no |
| 17:52.51 | Bobthehunter | anyone doing virtual PRIS ? |
| 17:53.26 | mafkees | virtual pri ? |
| 17:53.32 | Bobthehunter | yeah |
| 17:53.47 | mafkees | "this is a pri line. it's not real, we emulate it" |
| 17:54.04 | Bobthehunter | yes ;) |
| 17:55.14 | *** join/#asterisk musashibe (n=mus@office.besite.be) |
| 17:55.14 | mafkees | what's the use of that ? |
| 17:55.14 | musashibe | hey |
| 17:55.29 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 17:55.34 | Dr-Linux | hi all |
| 17:55.48 | ryant | hey |
| 17:55.49 | Dr-Linux | anybody is using advance Meetme conference? |
| 17:56.01 | musashibe | anyone experienced with hangup probs in asterisk ? |
| 17:56.06 | ryant | what's "advance" about? |
| 17:56.13 | ryant | musashibe, not me |
| 17:56.24 | mafkees | musashibe: hangup on what channel type ? |
| 17:56.29 | musashibe | pri |
| 17:56.32 | mafkees | what version of asterisk ? |
| 17:56.35 | musashibe | 1.4 |
| 17:56.49 | mafkees | no idea there |
| 17:56.49 | Dr-Linux | ryant: ofcos there are a number of option for meetme, but i need some good idea an example to setup conference dialplan |
| 17:56.50 | musashibe | but 1.2 also |
| 17:56.56 | mafkees | I dont run 1.4 in production yet |
| 17:57.13 | ManxPower | Bobthehunter: it is either a PRI or is is not a PRI |
| 17:57.21 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:57.42 | mafkees | Bobthehunter: or do you mean 'a pri that is not connected to a telco' |
| 17:57.55 | mafkees | that you can do with a 2port pri card and a crossover cable |
| 17:57.59 | Bobthehunter | like getting a flat rate inbound channel... for anywhere national |
| 17:58.03 | Bobthehunter | from |
| 17:58.14 | musashibe | mafkees: for some reason, after we receive a DISCONNECT with error 41 from the operator, the call keeps on running in asterisk |
| 17:58.48 | *** join/#asterisk tonycarstens (n=tony@206.135.21.162) |
| 18:00.08 | tonycarstens | can anyone tell me why when calling from an outside line and connecting through sip to a phone it wont ring? |
| 18:00.51 | mafkees | musashibe: never seen that error |
| 18:01.16 | musashibe | Hangup cause = 41, temporary network failure |
| 18:01.35 | tonycarstens | if i pick up the phone it does answer and communicate |
| 18:01.36 | musashibe | but apparently that happens a lot with this operator for certail cellulars |
| 18:01.41 | *** join/#asterisk Bananaskin (n=Bananask@81-86-102-88.dsl.pipex.com) |
| 18:01.46 | tonycarstens | just doesn't ring? |
| 18:01.53 | musashibe | tonycarstens: what kind of phone |
| 18:02.15 | musashibe | tonycarstens: im using ciscos 7960 and they ring fine |
| 18:02.45 | tonycarstens | grandstream bt-101 unfortuneatly |
| 18:02.49 | *** join/#asterisk irq (n=dan@wsip-70-167-112-5.sd.sd.cox.net) |
| 18:02.57 | irq | hey |
| 18:03.00 | irq | guy i used to talk to |
| 18:03.03 | irq | are you here? |
| 18:03.21 | tonycarstens | i've heard they are garbage but it's the company's choice |
| 18:03.40 | mafkees | I trashed all the BT-101 phones |
| 18:03.40 | musashibe | tonycarstens: soft sip rings ok ? |
| 18:04.00 | tonycarstens | haven't tried that |
| 18:04.02 | anonymouz666 | res_odbc works better than app_mysql for ast 1.2? |
| 18:04.03 | tonycarstens | i will |
| 18:04.24 | musashibe | tonycarstens: gives you an indication if its purely bt101 related |
| 18:04.33 | tonycarstens | yeah, thanks |
| 18:04.44 | mafkees | the BT-101 does ring fine here. |
| 18:04.55 | mafkees | that's the only thing it does fine ;) |
| 18:05.00 | tonycarstens | haha |
| 18:05.14 | tonycarstens | yeah they suck at registering |
| 18:05.23 | mafkees | and call quality |
| 18:05.42 | mafkees | and they look ugly |
| 18:06.10 | tonycarstens | and cheap |
| 18:06.19 | mafkees | indeed |
| 18:06.46 | tonycarstens | maybe we should join #bt101sucksatlife |
| 18:06.51 | tonycarstens | :) |
| 18:06.55 | mafkees | lol |
| 18:07.05 | mafkees | #weliketoburnalotofbt101s |
| 18:07.10 | anonymouz666 | pap2 has some serious static noise, that sucks |
| 18:07.21 | anonymouz666 | another part can't hear you only noise |
| 18:13.23 | [TK]D-Fender | genz: "show function CALLERID" |
| 18:14.07 | ManxPower | anonymouz666: set your rtp packet size in the SIPura to .2 instead of the default .3 |
| 18:14.13 | ManxPower | that will fix the audio problems |
| 18:14.26 | genz | [TK]D-Fender: So I should make sure I'm sending "all"? |
| 18:14.47 | [TK]D-Fender | genz: Set them individually |
| 18:15.10 | [TK]D-Fender | ~gs |
| 18:15.15 | jbot | well, gs is South Georgia and the South Sandwich islands, or ghostscript |
| 18:16.15 | *** join/#asterisk ggilbert (n=ggilbert@tinman.treke.net) |
| 18:16.26 | [TK]D-Fender | jbot: gs is also GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 18:16.28 | jbot | [TK]D-Fender: okay |
| 18:16.42 | [TK]D-Fender | ~gs |
| 18:16.43 | jbot | methinks gs is South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 18:16.49 | [TK]D-Fender | :D |
| 18:17.50 | ruied | My sip phones doesn't work with voicemail but I can access to voicemail with my analog phone... could it be an inband dtmf problem? |
| 18:18.09 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 18:18.11 | coppice | i haven't encountered grandstream stuff very often, but when I do and people are complaining, it usually seems to be the other end doing the wrong thing |
| 18:18.16 | ruied | I0m using g729a |
| 18:18.27 | [TK]D-Fender | ruied: Could be. You can't do inband on G.729 |
| 18:18.29 | ManxPower | ruied: inband only works with ulaw and alaw |
| 18:18.39 | [TK]D-Fender | ruied: Perhaps you could describe the problem a bit more... |
| 18:19.28 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 18:19.28 | coppice | of course you can do inband with G.729. it even works about 90% of the time. of course with a typical number being 10 digits, that's not too useful :-) |
| 18:21.03 | [TK]D-Fender | coppice: "Please enter your account # now. I'm sorry, that is not a valid #, please try again. Please enter your account # now. I'm sorry, that is not a valid #, please try again. Please enter your account # now. I'm sorry, that is not a valid #, please try again. Please enter your account # now. I'm sorry, that is not a valid #, please try again. Please enter your account # now. ... |
| 18:21.05 | [TK]D-Fender | ...I'm sorry, that is not a valid #, please try again. Please enter your account # now. I'm sorry, that is not a valid #, please try again. " |
| 18:21.47 | coppice | in band will work solidly with G.726 as well as G.711 |
| 18:21.51 | *** part/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:21.55 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:21.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:22.39 | coppice | I don't think it works with *, though |
| 18:22.55 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 18:23.23 | clyrrad | Does anyone here use Telantek for DID's? |
| 18:27.33 | CunningPike | Has anyone tried IMAP voicemail storage on MS Exchange? |
| 18:29.41 | *** join/#asterisk angom (n=angom@red-corp-201.143.88.126.telnor.net) |
| 18:32.45 | *** join/#asterisk averren (n=m@CPE000fb5e41c67-CM00e06f14ca46.cpe.net.cable.rogers.com) |
| 18:33.37 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-67-79-2-146.sw.biz.rr.com) |
| 18:34.44 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 18:34.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:36.20 | JoNate | any idea why * would be super slow to respond? |
| 18:37.01 | mafkees | JoNate: voip channels ? |
| 18:37.34 | JoNate | are you asking about them or are you posing an answer in the form of a question |
| 18:38.04 | genz | ruied: still there? |
| 18:38.06 | mafkees | answer in the form of a question ;) |
| 18:38.10 | JoNate | ahhhh |
| 18:38.11 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:38.28 | ruied | here |
| 18:38.35 | mafkees | is it slow in responding to voip channels or PRI channels |
| 18:38.38 | genz | ruied: using Grandstrom GXP-2000s? |
| 18:38.42 | ruied | yes |
| 18:38.43 | JoNate | so sip channels could cause the system to become slowish... |
| 18:38.53 | JoNate | its just slow in everything |
| 18:39.02 | JoNate | loading the first time, reloading... |
| 18:39.03 | ruied | genz, GXP200 and GXP2000 |
| 18:39.14 | JoNate | now problems with linux though |
| 18:39.17 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 18:39.22 | mafkees | JoNate: is your resolving setup correctly ? |
| 18:39.46 | ruied | genz, there is a message key... is that for voicemail? |
| 18:39.51 | JoNate | probably not |
| 18:40.02 | JoNate | I;m an idiot so... |
| 18:40.03 | genz | ruied: in the admin, under the account page, there's a setting |
| 18:40.12 | mafkees | JoNate: check your /etc/resolv.conf |
| 18:40.19 | genz | ruied: it says "Send DTMF", change that to "via SIP INFO" |
| 18:40.36 | JoNate | ahhhhhh |
| 18:40.41 | JoNate | you beautiful bastard you... |
| 18:40.53 | genz | ruied: the MSG key isn't for that, you have to do the *97 or define a key for it |
| 18:40.58 | JoNate | that was meant in a complimentary way... |
| 18:41.18 | mafkees | lol JoNate |
| 18:41.25 | ruied | genz, so what is the 'message' key for? |
| 18:42.18 | genz | ruied: i'm not entirely sure, one sec |
| 18:42.24 | ruied | ok |
| 18:42.24 | JoNate | i'm the perpetual noob at whatever I do... |
| 18:42.47 | mafkees | JoNate: gheh. we all have days like that |
| 18:43.03 | mercestes | ....I want a perpetual virgin. |
| 18:43.03 | *** join/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia) |
| 18:43.03 | JoNate | no no no...you misunderstand...every day is like this... |
| 18:43.13 | mercestes | that would be badass |
| 18:43.20 | genz | ruied: ah, so it is. what firmware are you using? |
| 18:43.21 | Burgwork | line over/hunting, does this work on a single PSTN line? |
| 18:43.39 | clyrrad | Does anyone here use Telantek for DID's? |
| 18:43.41 | JoNate | well...if you had a perpetual virgin, what would you do with it? |
| 18:43.42 | [TK]D-Fender | Burgwork: What is there to hunt for on a SINGLE line? |
| 18:43.59 | mercestes | [TK]D-Fender: A dialtone |
| 18:44.05 | [TK]D-Fender | JoNate: Exfoliate ;) |
| 18:44.23 | mercestes | JoNate: Perpetually |
| 18:44.25 | mafkees | whehehehehe |
| 18:44.37 | JoNate | well i mean...in order for said virgin to stay perpetual...you'd have to lock said virgin up forever... |
| 18:44.43 | mercestes | nononono |
| 18:44.48 | mercestes | your missing the point. |
| 18:44.59 | Burgwork | mercestes: I assume that comment was in reference to mine? |
| 18:45.11 | genz | ruied: so are things better now? sans the msg key? |
| 18:45.19 | *** join/#asterisk chowmeined (n=will_@c-71-231-166-10.hsd1.or.comcast.net) |
| 18:45.23 | mercestes | Burgwork: Of course not, my comment was in reference to D-Fenders which was in reference to yours |
| 18:45.32 | Burgwork | mercestes: right |
| 18:45.37 | Burgwork | and does that sort of thing work? |
| 18:45.51 | mercestes | Burgwork: What sort of thing? |
| 18:46.02 | ruied | genz, not shure, about the version (I'm not finding it) |
| 18:46.02 | chowmeined | Do any of you know how to get mitel 5220s to authenticate with a radius server? |
| 18:46.04 | mafkees | I'm off |
| 18:46.05 | mafkees | latero |
| 18:46.12 | mercestes | chowmeined: Of course, all asterisk ppl know how to do that. |
| 18:46.35 | mercestes | chowmeined: What version of radius are you using? |
| 18:46.57 | chowmeined | freeradius |
| 18:47.05 | mercestes | chowmeined: what version? |
| 18:47.53 | chowmeined | freeradius 1.0.1 |
| 18:48.17 | mercestes | oh dear. Mitel won't authenticate to that version, you have to upgrade atleast two revisions before they are compatible. |
| 18:48.21 | ruied | genz, right in front of my eyes!!! hehe version 1.1.1.14 |
| 18:49.05 | genz | ruied: alright, wasn't sure if you were using some weird beta or something. under Account 1, there's a Voicemail User ID, set that to your voicemail extension (*97) and then it'll work |
| 18:49.22 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 18:50.02 | *** join/#asterisk deb_user (n=Hypnotis@albuquerque.agroinnovations.com) |
| 18:50.29 | *** join/#asterisk Keithdizzle (n=temp@74.93.105.81) |
| 18:50.42 | Keithdizzle | hey, can anyone tell me how i can modify the EXTEN variable? |
| 18:50.57 | chowmeined | mercestes: oh and about me asking here.. I know its not quite appropriate I am just kind of desperate because of mitel's lack of documentation and support. I wish the management had looked around a bit more before spending thousands and thousands of dollars |
| 18:51.09 | ManxPower | Keithdizzle: use Goto(newextension,1) |
| 18:51.44 | Keithdizzle | ok, thanks |
| 18:51.45 | ManxPower | Keithdizzle: Other than that you can't. EXTEN is read only |
| 18:52.25 | Keithdizzle | so wait |
| 18:52.27 | ruied | genz, I was using g729 for epygi PBX (Grandstream-Line1) for asterisk I was using gsm (Grandstream-Line2) |
| 18:52.51 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 18:53.01 | ManxPower | EXTEN contains the currently executing extensions.con extension number |
| 18:53.22 | Keithdizzle | i want it so that i can have an extension where people can dial 1-800-XXX-XXX, but if they dial 800-XXX-XXXX, it adds a 1 to the front and then jumps to the 1-800-XXX-XXXX exten. what do i need to do? |
| 18:53.53 | ManxPower | exten => _800NXXXXXX,1,Goto(1${EXTEN},1) |
| 18:54.05 | Keithdizzle | ok, that's what i needed, thanks |
| 18:54.31 | genz | ruied: i don't know if that means they work or not |
| 18:55.34 | *** join/#asterisk pdtwork (n=ptinsley@209.12.249.243) |
| 18:55.41 | ruied | genz, the MSG key is working, but I can't access to the mailbox... |
| 18:55.54 | pdtwork | i need to call parking from AGI what would be the proper way to call ParkAndAnnounce |
| 18:56.08 | *** join/#asterisk TypMic1004 (n=chrislro@outland.cmf.nrl.navy.mil) |
| 18:56.09 | genz | ruied: and you're connecting to a local asterisk server? |
| 18:56.28 | *** part/#asterisk TypMic1004 (n=chrislro@outland.cmf.nrl.navy.mil) |
| 18:56.41 | ruied | genz, ah, I've configured it for line one again, going to try the line2 |
| 18:58.06 | aydiosmio | [[blah]asfd: thanks again, the math on this project is looking really promising |
| 18:58.12 | ruied | genz, not working... |
| 18:58.31 | [[blah]asfd | aydiosmio: no problem |
| 18:58.51 | deb_user | could somebody please take a look at this and tell me what I'm doing wrong...I couldn't imagine a more simple dialplan, but its not rolling over to voicemail on busy |
| 18:58.53 | deb_user | http://paste.linux-vserver.org/1199 |
| 18:59.11 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:59.12 | CunningPike | Entire industries are founded on the principle that management spends thousands of dollars without looking around |
| 18:59.36 | aydiosmio | call me Consultant Pete |
| 18:59.50 | mercestes | deb_user: try 102, oh, and set priorityjumping=1 in [globals] |
| 19:00.09 | ruied | genz, going to try in rpt mode... |
| 19:00.37 | mercestes | and send your love via paypal to....just kidding..;) |
| 19:00.56 | *** join/#asterisk irq (n=dan@adsl-75-36-60-245.dsl.sndg02.sbcglobal.net) |
| 19:01.04 | genz | ruied: what're we working on now, getting the MSG button to work or getting *97 |
| 19:01.47 | *** join/#asterisk apardo (n=apardo@87.217.145.9) |
| 19:01.59 | *** join/#asterisk irq (n=dan@wsip-70-168-52-194.sd.sd.cox.net) |
| 19:02.16 | deb_user | mercestes: how come 102? |
| 19:02.32 | Keithdizzle | thanks for your help Manxpower |
| 19:02.33 | mercestes | deb_user: did you try it? |
| 19:02.35 | *** part/#asterisk Keithdizzle (n=temp@74.93.105.81) |
| 19:02.44 | ruied | genz, MSG is working (with 500 mail box extension). I connect to the mailbox, but it reports 'login incorrect' |
| 19:03.24 | genz | ruied: i get that until i switch to the SIP Info, you're using a local asterisk server, right? |
| 19:03.36 | ruied | yes |
| 19:03.37 | deb_user | i'm trying now...I'm just wondering what the reason would be so I could understand better |
| 19:03.47 | mercestes | deb_user: Ask me again if it doesn't work. |
| 19:03.51 | deb_user | ok |
| 19:04.08 | *** join/#asterisk seva (i=seva@66.90.103.12) |
| 19:04.18 | seva | how do i set a variable to a space, other than: |
| 19:04.23 | seva | FOO=1 1 |
| 19:04.23 | seva | SEPARATOR=${FOO:1:1} |
| 19:04.38 | genz | ruied: i really think your choice 1 encode should be the G.723.1 |
| 19:04.38 | tonycarstens | would it be RTP that is the problem if i can dial a phone but cannot hear what it is being said |
| 19:04.42 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-191-36.nycmny.east.verizon.net) |
| 19:04.42 | mercestes | seva: set(foo=" ") doesn't work? |
| 19:05.06 | tonycarstens | through SIP |
| 19:05.10 | seva | mercestes: works |
| 19:05.17 | mercestes | tonycarstens: firewall or Nat. Or the person on the other end doesn't like you and refuses to speak to you. |
| 19:05.19 | deb_user | mercestes: tried it, no luck |
| 19:05.20 | mercestes | seva: your welcome. |
| 19:05.20 | deb_user | :( |
| 19:05.21 | seva | i've been defining things as FOO=... |
| 19:05.40 | mercestes | deb_user: try 103 again then. |
| 19:05.44 | deb_user | ok |
| 19:05.52 | mercestes | deb_user: With priorityjumping=1 of course, right? |
| 19:05.58 | ruied | genz, going to try with g723.1 |
| 19:05.59 | seva | actually.. heh SEPARATOR=" " works too |
| 19:06.03 | tonycarstens | merc: how would i open up those ports? |
| 19:06.04 | seva | i've tried SEPARATOR=' ' |
| 19:06.08 | seva | which doesn't ;) |
| 19:06.45 | deb_user | mercestes: yeah, priorityjumping=1...but still nothing |
| 19:06.46 | mercestes | tonycarstens: oh, googling something like asterisk rtp ports or asterisk firewall or asterisk one way voice path or something along those lines. |
| 19:06.55 | deb_user | mercestes: just times out |
| 19:06.57 | tonycarstens | thanks |
| 19:07.12 | ruied | genz, they are negotiating with gsm, maybe I need to set the codec in sip.conf user...? |
| 19:07.23 | mercestes | deb_user: Why are you callin gan answer() and then a dial(blah,20,r) ? btw? |
| 19:07.40 | ruied | genz, already made g729.1 in GXP2000 |
| 19:07.50 | deb_user | mercestes: why not? |
| 19:08.03 | deb_user | it dials a zap interface on an incoming call... |
| 19:08.06 | mercestes | deb_user: you don't want my answer to that question because it might hurt your feelings. |
| 19:08.06 | aydiosmio | [[blah]asfd: what format are your recorded calls in? SIP-Codec/Zap? |
| 19:08.11 | deb_user | haha |
| 19:08.14 | deb_user | just tell me |
| 19:08.25 | mercestes | deb_user: Because it's retarded? lol |
| 19:08.34 | mercestes | deb_user: I just wasn't sure why an "answer" needed to happen before the dial. |
| 19:08.55 | deb_user | mercestes: references I've read recommend using Answer() first... |
| 19:08.58 | ruied | genz, do I need to set it in sip.conf user? |
| 19:08.58 | mercestes | s/wasn't/still not/ |
| 19:09.09 | genz | ruied: using trixbox? |
| 19:09.17 | deb_user | specifically, the O'Reilly * book |
| 19:09.20 | deb_user | recommends it |
| 19:09.23 | deb_user | so I do it |
| 19:09.34 | ruied | nop, asterisk |
| 19:09.41 | mercestes | deb_user: I'd have to see what ${IN4} is set too and what's calling [incomming]. |
| 19:09.43 | genz | ruied: do other call features work from the gs? like call forwarding or ivr |
| 19:09.48 | *** join/#asterisk J4k3 (i=J4k3@dhcp-12-197-128-58.intrastar.net) |
| 19:10.17 | deb_user | mercestes: IN4=Zap/6 |
| 19:10.18 | ruied | genz, forwarding, I think so, going to check... |
| 19:10.28 | genz | ruied: Do your settings look close to this? http://www.inphonex.com/support/grandstream-gxp2000-configuration.php |
| 19:10.43 | mercestes | deb_user: in short, I need more of your dialplan, but I would just s,1,Dial(sip/blah,20) s,2,Voicemail(bexten) s,102,Viocemail(uexten) |
| 19:11.46 | queuetue | If voip quality is great normally, but periodically degrades, it's essentially either network lag, loss, or machine loading, right? We've eliminated network lag as a cause (mtr looks great) , and our machine sits just above idle...so I'm concerned it's a case of the provider having an overloaded server. |
| 19:12.04 | mercestes | deb_user: yea, i tcould be going to priority 3 instead of 103 too now that I think about it. add a s,3,Voicemail(bexten) or whatever. |
| 19:12.04 | *** join/#asterisk Blackhold (n=laura@105.10.223.82.arsystel.com) |
| 19:12.14 | n|cotine | Are there any SIP phones out there that offer configurable soft feature keys? |
| 19:12.23 | Blackhold | hello |
| 19:12.33 | Blackhold | I just installed asterisk 1.4 |
| 19:12.52 | bkruse | awesome. |
| 19:12.59 | Blackhold | and I don't know how to make a call |
| 19:13.05 | Blackhold | 'cause I used dial 500 |
| 19:13.13 | Blackhold | to test that asterisk works |
| 19:13.22 | bkruse | im PRETTY sure asterisk works. |
| 19:13.31 | Blackhold | I'm very new at asterisk |
| 19:14.04 | queuetue | Blackhold: Are you trying to make calls with a voip provider, or a pots line? |
| 19:14.57 | queuetue | (If you don't know, *something* has to carry those calls from your box to the real world. :) ) |
| 19:16.39 | ruied | genz, I've changed the sip registration to yes (that was the only different thing..). Unfortunally I have to leave now, I'll be back later... Thanks alot for the help |
| 19:16.56 | *** part/#asterisk seva (i=seva@66.90.103.12) |
| 19:17.05 | deb_user | mercestes...then it goes to priority 3 instead of n+101 |
| 19:17.17 | mercestes | deb_user: I think so |
| 19:17.31 | deb_user | mercestes: no, it does I'm sure |
| 19:17.33 | mercestes | deb_user: n+101 is an error, and I don't think "busy" is an error, I think "unavailable" is an error. |
| 19:17.49 | averren | does anyone have beginning to end docuementation for creating queues in v1.4? |
| 19:18.04 | mercestes | deb_user: so you should have s,1,dial s,2,VM(Busy) s,102,VM(error) |
| 19:18.12 | mercestes | deb_user: oh, then I'm sure it does too. :) |
| 19:18.58 | deb_user | mercestes: but I want it to go to n+101 if s,2, is busy |
| 19:18.59 | [TK]D-Fender | n|cotine: Aastra 480i is the most configurable phone as far as soft-keys goes. |
| 19:19.12 | mercestes | averren: If you mean copy and paste instructions I suggest a consultant. otherwise google asterisk wiki queues is going to be your best bet. |
| 19:19.14 | deb_user | mercestes: that's what's pissing me off...it should do that but it won't |
| 19:20.42 | mercestes | deb_user: then use ${DIALSTATUS} instead of priority jumping which was removed specifically do to that stupidity. :) |
| 19:20.42 | *** join/#asterisk stives (n=afar@87-196-186-163.net.novis.pt) |
| 19:20.42 | averren | mercestes: not copy and paste, but better instructions than google has provided so far |
| 19:20.42 | [TK]D-Fender | deb_user: Pastebin the whole thing |
| 19:20.42 | [TK]D-Fender | ~pb |
| 19:20.47 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 19:20.47 | deb_user | mercestes: ok |
| 19:20.47 | [TK]D-Fender | deb_user: "busy" is a nearly defunct concept as far as modern phones go. |
| 19:21.05 | mercestes | deb_user: Then you can handle busy, unavailable, congestion, answered, etc. in any way you wish. |
| 19:21.24 | deb_user | mercestes: sounds like a good idea |
| 19:21.28 | [TK]D-Fender | deb_user: For which you probably DON'T really want to differentiate once you sit and think about it.... |
| 19:21.43 | deb_user | mercestes: i'll just write up some simple macros to handle it |
| 19:22.53 | [TK]D-Fender | deb_user: "answered" is irrelvent as your channel usually hangs up. "busy" won't happen on multi-line phone where subsequent calls just CW beep in on you and you typically actively IGNORE them anyways. Therefor "busy" never really happens unless your phone is INCAPABLE of accepting another call (who cares?). |
| 19:23.15 | deb_user | fender: cw is disabled on my system |
| 19:23.24 | tonycarstens | can anyone point me in the direction of finding some info on config * to make outbound calls through SIP |
| 19:23.28 | deb_user | fender: what i need is a rollover to another zap channel on busy |
| 19:23.36 | deb_user | fender: which i was trying to do with n+101 |
| 19:23.36 | [TK]D-Fender | deb_user: Common reality then conculdes that anytime you intend on hitting VM is because you're just "not available". Any sense of WHY is pointless. |
| 19:24.13 | [TK]D-Fender | deb_user: Oh... well then jsut shove your dials back-to-back without any checking. The first one to succeed is the last one it'll try. |
| 19:24.16 | deb_user | fender: I DON'T intend to hit vm in this case, actually, i just set up the simplest dialplan I could to see if you guys could help me figure out why n+101 wasn't working |
| 19:24.19 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 19:24.33 | [TK]D-Fender | deb_user: Priority jumping is DEAD BTW. |
| 19:24.50 | [TK]D-Fender | deb_user: And you shouldn't have to do ANY checks in all likelyhood. |
| 19:25.03 | deb_user | fender: so in 1.4 priority jumping is deprecated? |
| 19:25.07 | [TK]D-Fender | tonycarstens: Calls to what? |
| 19:25.17 | [TK]D-Fender | deb_user: It was deprecated in 1.2...... |
| 19:25.35 | deb_user | sheesh... |
| 19:25.41 | deb_user | talk about being a little behind the times :( |
| 19:25.45 | [TK]D-Fender | deb_user: welcome to yester-year |
| 19:25.50 | mercestes | deb_user: Yea. D-Fender is right..just dump a string of dials with no jumping/checking. |
| 19:26.32 | tonycarstens | [TK] make a call to another phone outside of the network, I have it configured so I can recieve outside calls but cannot dial outside lines |
| 19:26.53 | [TK]D-Fender | mercestes: Similar to the answer to "whats the fastest way to a man's heart?". - Through the chest with a horizontally aligned and very sharp knife. |
| 19:27.16 | deb_user | mercestes: but I need different behaviors depending if the line is busy or if it timesout |
| 19:27.21 | [TK]D-Fender | tonycarstens: You mean through a SIP peer associated with an ITSP you have configured? |
| 19:27.43 | deb_user | i think using ${DIALSTATUS} is a great idea, actually |
| 19:27.46 | deb_user | I'll try it |
| 19:27.51 | tonycarstens | i haven't started configuring it i just wanted to know a good resource to look at to configure it |
| 19:27.52 | mercestes | ... |
| 19:28.11 | [TK]D-Fender | deb_user: Any reason against "ram-dialing" the call? Just gives more chances of success.... |
| 19:28.29 | [TK]D-Fender | deb_user: or you could add a single abort of "noanswer" and then roll-on |
| 19:28.42 | deb_user | fender: because if the fxs is busy, i want it to roll over to a different fxs port |
| 19:28.42 | mercestes | [TK]D-Fender: Shh..I wanna see his dialplan with about 80 Goto(s-${DIALSTATUS})'s |
| 19:28.57 | [TK]D-Fender | deb_user: Depends on how insistant you are on the call going through. |
| 19:28.58 | deb_user | mercestes: you guys are funny |
| 19:29.14 | mercestes | deb_user: mostly me, but I'm like, half engineer, half troll |
| 19:29.20 | [TK]D-Fender | deb_user: How many dialing options are you planning on cycling through? |
| 19:29.22 | deb_user | and if the fxs is not busy, but it rings for 20 seconds, I want it to go to voicemail |
| 19:29.34 | deb_user | fender: not that many |
| 19:29.38 | deb_user | 4 or 5 at the most |
| 19:30.01 | [TK]D-Fender | deb_user: Wait.. you said "line", and now "FXS". I at least ONE of us is confused.... |
| 19:30.48 | tonycarstens | [TK]: i was wondering where i should start to configure this functionality |
| 19:30.57 | [TK]D-Fender | tonycarstens: A question that general will have me pointing you back towards THE BOOK |
| 19:30.59 | [TK]D-Fender | ~book |
| 19:31.01 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:31.24 | deb_user | allright guys, I'm gonna see if this works or not |
| 19:31.28 | deb_user | talk to you later |
| 19:31.44 | *** join/#asterisk komradebob (n=komradeb@164.55.254.106) |
| 19:31.51 | mercestes | good luck. |
| 19:32.55 | Bobthehunter | e aware that setting fromuser= in sip.conf will overide SetCallerID! |
| 19:32.58 | Bobthehunter | that my problem |
| 19:33.10 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 19:33.15 | Bobthehunter | so if i dont use form user then it takes the user of the phone |
| 19:41.10 | deb_user | haha, it works! |
| 19:41.27 | deb_user | and you thought I'd have to write 80 Goto's |
| 19:42.19 | deb_user | now i've got to get rid of all my n+101's in my dialplan... |
| 19:43.06 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 19:46.16 | *** join/#asterisk lumovan2 (n=lumovan@80.122.72.250) |
| 19:46.19 | lumovan2 | hi :-) |
| 19:52.57 | knathraak | anybody have a howto or sample configs for gr303\ |
| 19:53.29 | knathraak | connecting pair of TE110Ps, single span covering both cards, emulating 5ess |
| 19:53.36 | JerJer | knathraak: I might have some - need to clear it first |
| 19:53.48 | knathraak | jerjer, cool thanks! |
| 19:56.06 | Ryushin | What was the program that provided a web page that showed who was currently on a call? |
| 19:56.31 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-da2bea0e28630211) |
| 19:56.57 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 19:57.17 | aydiosmio | I have numbers that say you can have 300 users in conference on a P4 3ghz, I guess this assumes all Zap channels? |
| 19:57.38 | funxion | I'm getting asterisk: relocation error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_copy_string Im recompiled even recompiled after redownloading new sources anyone have a clue? |
| 19:57.50 | anonymouz666 | JerJer: what software did you use to test (simulate the calls) that issue with libpri |
| 19:58.06 | JerJer | asterisk |
| 19:58.17 | deb_user | allright...here's a brain teaser |
| 19:58.24 | anonymouz666 | heh |
| 19:58.56 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-239-113-101.dsl.irvnca.pacbell.net) |
| 19:58.59 | deb_user | exten => s,2,Dial(${IN2},20,t), works fine, when i hit # for transfer, * says "Transfer" |
| 19:59.22 | deb_user | but it transfers the line that hit #, not the line that called in! |
| 19:59.35 | deb_user | why would I want to transfer myself somewhere?? |
| 19:59.39 | JerJer | maybe T ? |
| 20:00.01 | deb_user | jerjer: then it just makes the button noise, without any transfer at all |
| 20:00.31 | deb_user | really weird |
| 20:00.35 | *** join/#asterisk Grnd-Wire (n=groundwi@71-217-104-226.tukw.qwest.net) |
| 20:00.37 | [TK]D-Fender | aydiosmio: And what kind of card will give you 300 Zap channels in 1 box? :) |
| 20:00.50 | aydiosmio | excellent point! |
| 20:00.51 | Grnd-Wire | Good morning [afternoon] guys.. |
| 20:01.17 | Grnd-Wire | [TK]D-Fender: ooh.. umm.. Several quad or octal span E1 boards? :P |
| 20:01.59 | [TK]D-Fender | Grnd-Wire: Which is of course highly recommended and supported by all manufacturers of said cards ;) |
| 20:01.59 | Supermathie | deb_user: do you have goto_on_blindxfr set? |
| 20:02.09 | Grnd-Wire | I am having the weirdest issue with a TDM400P. I'm not sure if it's completely functioning like it's supposed to.. |
| 20:02.18 | aydiosmio | 2 octals would work! |
| 20:02.28 | Grnd-Wire | [TK]D-Fender: Are you familiar enough with those cards to advise? |
| 20:02.35 | BSDTech | is it asterisk or ass trix |
| 20:02.37 | BSDTech | lol |
| 20:02.41 | Grnd-Wire | aydiosmio: dude - That's some serious horsepower.. |
| 20:02.54 | [TK]D-Fender | Grnd-Wire: Enough to say that Digium Disavows all systems with more than 2 cards :) |
| 20:02.59 | Ryushin | I want to roll out that web tool to the employees so they can see who is currently online. I think it was a flash based tool. |
| 20:03.09 | aydiosmio | I'm just wondering how many conference users I could get using voip channels |
| 20:03.25 | Grnd-Wire | BSDTech: I recompiled the newest version of Zaptel and Asterisk 1.2.15 last night, so this isn't a FreePBX issue.. |
| 20:03.32 | [TK]D-Fender | Grnd-Wire: And Sangoma... I'm unsure how * alone will survive... but at least I know the CPU/bus/irq load will be smaller and the port density is double. |
| 20:03.40 | JerJer | aydiosmio: all depends on the details |
| 20:03.44 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 20:03.47 | Grnd-Wire | [TK]D-Fender: yeah you know - I'm going to buy a Sangoma T1 board!! |
| 20:03.55 | [TK]D-Fender | Grnd-Wire: Exactly how many you can get away with is another matter... and I really couldn't say... |
| 20:04.36 | Grnd-Wire | BSDTech: How is the call progress detection supposed to work? I installed a system yesterday, and the damn card wouldn't even recognize that the call was ringing, or had been asnwered.. |
| 20:04.43 | BSDTech | ? |
| 20:04.51 | BSDTech | how did I get pulled into this |
| 20:04.54 | aydiosmio | JerJer: that's what everone says, but I'm sure someone here has done 50 concurrent voip conference users, right? |
| 20:04.59 | JerJer | Grnd-Wire: is it analog? |
| 20:05.02 | aydiosmio | anyone? anyone?:) |
| 20:05.13 | Grnd-Wire | JerJer: yeah, a TDM400p.. |
| 20:05.14 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-239-113-101.dsl.irvnca.pacbell.net) |
| 20:05.37 | JerJer | analog doesnt provide reliabe progress |
| 20:06.01 | [TK]D-Fender | Grnd-Wire: "callprogress=yes" comes bundled with "randomlyhangupcalls=yes", 'generalflakeyness=yes" |
| 20:06.24 | Grnd-Wire | [TK]D-Fender: I understand that.. and it was turned OFF.. |
| 20:06.39 | [TK]D-Fender | Grnd-Wire: Hence your getting NONE. |
| 20:06.40 | *** join/#asterisk python_ (n=chatzill@66-191-97-162.static.eucl.wi.charter.com) |
| 20:06.44 | Grnd-Wire | [TK]D-Fender: But isn't answeronpolarityswitch=yes supposed to work? |
| 20:06.45 | python_ | hello |
| 20:07.36 | [TK]D-Fender | Grnd-Wire: Couldn't say...... I've avoided such things on analog..... |
| 20:07.43 | Grnd-Wire | hmm, ok.. |
| 20:08.11 | Bobthehunter | anyone ser + callerid working ? |
| 20:09.00 | funxion | I'm getting asterisk: relocation error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_copy_string Im recompiled even recompiled after redownloading new sources anyone have a clue? |
| 20:10.31 | tonycarstens | do i need a provider to make outside calls with *? |
| 20:10.42 | tonycarstens | other than a live PTSN line? |
| 20:11.01 | alrs | tonycarstens: I use Gafachi, as it is cheap |
| 20:11.09 | alrs | tonycarstens: though I read that Voxee is cheaper |
| 20:11.16 | tonycarstens | so thats a yes? |
| 20:12.07 | alrs | tonycarstens: If "outside calls" means calls to the standard telephone network |
| 20:12.10 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 20:12.17 | tonycarstens | yeah |
| 20:12.39 | alrs | tonycarstens: you can register with freeworlddialup and make calls to 800 numbers |
| 20:12.48 | tonycarstens | ok |
| 20:13.04 | alrs | tonycarstens: I think they even can do IAX |
| 20:13.23 | tonycarstens | alrs: how come you can recieve incoming calls from standard telephone lines but cannot place them in * |
| 20:13.27 | elriah | Hi all, I'm having a problem with several aastra 480i's. Firmware 1.4. When I have canreinvite=no, audio isn't being sent from the phone, when canreinvite=yes, audio can't be heard. It's a phone->nat->public_asterisk_box config. nat=yes, qualify=3000. Registers fine. sip.conf is exactly like polycoms. ANy suggestions? |
| 20:13.56 | mercestes | elriah: canreinvite=maybe |
| 20:14.10 | elriah | mercestes: Is that really an option? |
| 20:14.13 | tonycarstens | merc: you are a riot let me tell you |
| 20:14.16 | alrs | tonycarstens: ? I don't see how you can accept calls without being hooked up to PSTN |
| 20:14.17 | mercestes | elriah: ... no. |
| 20:14.24 | mercestes | elriah: Check out externip and localnet settings. |
| 20:14.25 | tonycarstens | alrs: i am |
| 20:14.47 | elriah | The asterisk box is public ip. Should the phone have a externip setting? I don't know much about aastras. |
| 20:14.50 | tonycarstens | alrs: i was asking if it is possible to make calls through PSTN with * |
| 20:15.00 | mercestes | elriah: oh....then .....no. |
| 20:15.00 | alrs | tonycarstens: then you don't need a voip provider, you can make calls through * |
| 20:15.11 | mercestes | elriah: try nat=always |
| 20:15.49 | tonycarstens | alrs: i cannot find anything that is helping me set this up do you know a good place that explains how to config it, i look in the * TFOT |
| 20:15.51 | alrs | tonycarstens: you said "other than a live pstn line" |
| 20:16.02 | *** join/#asterisk tklettke (n=tklettke@ip67-155-33-163.z33-155-67.customer.algx.net) |
| 20:16.05 | elriah | mercestes: Is that a real option? |
| 20:16.06 | mercestes | elriah: Yea, that really is an option. ;) |
| 20:16.07 | tonycarstens | alrs: yeah i confuse myself sometimes |
| 20:16.09 | tonycarstens | sorry |
| 20:16.13 | elriah | lol, thanks, I'll try it. |
| 20:16.43 | *** join/#asterisk LostFrog (n=qscript@wsip-68-225-90-115.dc.dc.cox.net) |
| 20:16.53 | *** join/#asterisk Mike800 (n=mike800@cpe-76-167-156-224.socal.res.rr.com) |
| 20:16.54 | LostFrog | Has anyone run pbxsnip and asterisk on the same machine? |
| 20:19.09 | elriah | nat=always didn't seem to do anything except keep it from registering... hrm... |
| 20:19.22 | mercestes | ?? |
| 20:19.52 | mercestes | well, then nat=yes. canreinvite=yes, and...examine the route between these phones and your box. |
| 20:19.53 | elriah | eh? |
| 20:19.59 | elriah | Ok.. |
| 20:20.00 | elriah | Thanks.. |
| 20:20.09 | mercestes | is there a firewall between phones and router?? |
| 20:20.26 | elriah | Yep. Smoothwall. No clue about its config, it was here when we showed up. |
| 20:20.29 | Mike800 | I'm having a problem in Asterisk 1.2.14 with MusicOnHold. When people are placed on hold, and taken off hold, and placed back onto hold, the music doesn't continue from where it left off. Often times, it begins playing an entirely new song. I spoke to Mark and he mentioned this was a bug and to report it. I've never reported a bug, and was trying at bugs.digium.com, but I don't want to post it incorrectly. Can anyone help? |
| 20:20.47 | elriah | what's funny is dtmf tones are working, i.e., voicemail passwords. |
| 20:20.49 | mercestes | Firewall is likely blocking your RTP ports then. see if you can try it withou tyoru firewall. |
| 20:20.51 | elriah | so it's an rtp issue. |
| 20:21.06 | mercestes | Oh yea. |
| 20:21.17 | file | Mike800: it's fixed in 1.2.15 |
| 20:21.27 | Mike800 | yay! |
| 20:21.29 | Mike800 | thanks |
| 20:21.32 | mercestes | your best bet is to have yoru CPUs on their own subnet, and set theri default gateway to the firewall on the same subnet, and have the firewall (gateway) route traffic out of itself filtered. |
| 20:21.57 | mercestes | and then have the phones on a seperate subnet with a default gateway either on an open routing box or the router interface itself. |
| 20:22.00 | *** join/#asterisk NirS (n=Nir@host-84-201-160-138.xdsl.lixxus.net) |
| 20:22.16 | mercestes | that way the phones get to be DMZ and the cmoputers are fully filtered, and you don't have to run seperate lines to everything. |
| 20:22.52 | NirS | Anybody here from London ? |
| 20:23.14 | NirS | I'm staying the night in London, at Heathro, and I'm dying of bordem |
| 20:23.43 | NirS | I'm actually so bored, that I just finished a patch for Asterisk to say numbers in proper hebrew |
| 20:23.46 | NirS | ;-) |
| 20:23.46 | *** join/#asterisk KnowWhat (n=KnowWhat@125.209.64.75) |
| 20:24.18 | Mike800 | i love london |
| 20:24.21 | Mike800 | go to Pickadilly |
| 20:24.28 | Mike800 | :-) |
| 20:24.40 | anonymouz666 | go to carnival |
| 20:24.42 | anonymouz666 | lol |
| 20:24.54 | tonycarstens | nirs: if you are really bored maybe you could tell me how to place calls through PSTN with * |
| 20:25.09 | NirS | tony, that's easy |
| 20:25.22 | NirS | Mike, I live in london for about 6 months |
| 20:25.26 | NirS | I know london very well |
| 20:25.37 | NirS | problem is, all my friends who used to live here now moved |
| 20:25.47 | Mike800 | :p i was there for 2 weeks...and i loved it |
| 20:25.50 | NirS | so I'm literraly stuck alone next to Heathro airport |
| 20:25.57 | *** join/#asterisk dasenjo (n=dasenjo@190.24.179.139) |
| 20:26.02 | anonymouz666 | NirS: drink beer and make new friends |
| 20:26.10 | NirS | :) |
| 20:26.17 | *** join/#asterisk uSuRa (i=usura@d137149.upc-d.chello.nl) |
| 20:26.21 | NirS | tony, what seems to be the problem you are having ? |
| 20:26.41 | tonycarstens | i cannot find out how to do it |
| 20:26.48 | tonycarstens | i've read the * TFOT |
| 20:26.53 | NirS | ok |
| 20:26.53 | KnowWhat | what is the best distro to install asterisk on :P |
| 20:26.57 | NirS | lets start from the begining |
| 20:26.58 | tonycarstens | all it discusses is using FWD |
| 20:27.08 | russellb | KnowWhat: are you trying to start a flame war? :-p |
| 20:27.17 | NirS | do you have any type of connection defined on your box? a ZAP? a SIP trunk? an IAX trunk? |
| 20:27.28 | *** join/#asterisk Aces1Up (n=rich@wsip-24-234-88-23.lv.lv.cox.net) |
| 20:27.32 | tonycarstens | zap |
| 20:27.37 | NirS | KW: I use CentOS and FedoraCore and i'm very happy |
| 20:27.40 | KnowWhat | russellb no man, just want to know |
| 20:27.42 | tonycarstens | i can recieve PTSN calls |
| 20:27.44 | sweeper | KnowWhat: I suggest XENIX |
| 20:27.45 | NirS | tony, ok, what is your dial string ? |
| 20:27.48 | Aces1Up | hey all i have been trying to find some good documentation on a2billing, can anyone point me in the right direction? |
| 20:27.53 | xboxoslo | Hi I wounder if someone can help me with asteriskNOW incomming rules ? |
| 20:27.53 | mercestes | KnowWhat: gentoo |
| 20:27.54 | russellb | KnowWhat: the best answer is whatever you are the most comfortable with |
| 20:27.58 | KnowWhat | NirS: did you install through sources |
| 20:28.01 | tonycarstens | nirs: what is a dialstring |
| 20:28.04 | russellb | KnowWhat: don't listen to anyone else :-p |
| 20:28.13 | KnowWhat | mercestes: you mean emerge asterisk :P |
| 20:28.26 | KnowWhat | NirS: or used rpms |
| 20:28.29 | mercestes | KnowWhat: Precisely |
| 20:28.31 | NirS | KW: my company has its own SRPM packages that we use, so everything is optimized to our products |
| 20:28.50 | KnowWhat | NirS: wow thats kool |
| 20:28.55 | russellb | "optimized to our products" ? |
| 20:29.06 | NirS | tony, are you trying to call from a SIP phone or IAX phone ? |
| 20:29.10 | KnowWhat | NirS: but what if one want to use some predicive dialer with it |
| 20:29.13 | NirS | yes russell |
| 20:29.17 | russellb | how so |
| 20:29.42 | sweeper | FUCK YOU |
| 20:29.45 | tonycarstens | nirs: SIP |
| 20:29.55 | mercestes | oh my. |
| 20:29.56 | NirS | well, my company makes Asterisk based Centrex type products |
| 20:29.59 | KnowWhat | sweeper: its not a sex channel any way |
| 20:29.59 | mercestes | My virgin eyes |
| 20:30.14 | russellb | NirS: that's cool, just curious what you "optimized" |
| 20:30.20 | NirS | our products require specific kernel optimizations to make the OS robust enough for Asterisk and our product at the same time |
| 20:30.38 | NirS | so, we made our own SRPM packages, which are generated nightly from the SVN at Digium |
| 20:30.59 | NirS | it is basically a stock SVN asterisk, but we include various patches |
| 20:31.18 | NirS | tony, did you configure a context for your SIP phone? in sip.conf ? |
| 20:31.40 | NirS | russell, you can check our current website at http://www.atelis.net |
| 20:32.09 | tonycarstens | for incoming and sip calls |
| 20:32.56 | NirS | in sip conf, in your sip phone context, you need to put a context=something line, what does it say ? |
| 20:34.08 | tonycarstens | sip |
| 20:34.21 | xboxoslo | is there anyone that can help with asteriskNOW |
| 20:34.37 | tonycarstens | xboxoslo: read the topic |
| 20:35.04 | xboxoslo | ok sorry |
| 20:35.04 | tonycarstens | asteriskNOW=trixbox |
| 20:35.13 | xboxoslo | ?? |
| 20:35.15 | NirS | tony, you mean that in sip.conf you have a line that says: "context=sip" ? |
| 20:35.16 | tonycarstens | no prob, i'm a newbie too |
| 20:35.21 | ManxPower | NO! AsteriskNOW is NOT Trixbox |
| 20:35.25 | KnowWhat | not equal to any way, but kinda |
| 20:35.25 | tonycarstens | yeah |
| 20:35.25 | NirS | tony, asteriskNOW != TrixBox |
| 20:35.42 | tonycarstens | ok |
| 20:35.45 | tonycarstens | im wrong |
| 20:35.49 | NirS | tony, do you have a [sip] context in extensions.conf ? |
| 20:35.52 | tonycarstens | yeah |
| 20:36.00 | NirS | ok, what does that context contain ? |
| 20:36.13 | NirS | in order to dial out, you should have something like this |
| 20:36.21 | tonycarstens | the extensions for the ip phones i have setup |
| 20:36.23 | NirS | exten => _X.,1,Dial(Zap/1/${EXTEN},120,r) |
| 20:36.44 | NirS | ok, in that case, put the line I just typed at the end of the [sip] context and try dialing out |
| 20:37.01 | NirS | I assume you are using FXO interfaces, and that your FXO interface is located at Zap/1 |
| 20:37.18 | tonycarstens | its on 2 |
| 20:37.23 | tonycarstens | but i just change 1 to 2 right |
| 20:37.26 | NirS | xbox, what seems to be the problem with AsteriskNOW |
| 20:37.32 | NirS | right, you're getting it |
| 20:37.36 | tonycarstens | :) |
| 20:38.04 | xboxoslo | I cant recive incomming cals |
| 20:38.30 | NirS | xbox, ok, that's a bit general, care to be a little bit more specifc ? |
| 20:39.06 | tonycarstens | nirs: you are the man |
| 20:39.07 | xboxoslo | I got the error chan_iax2.c: Rejected connect attempt from |
| 20:39.53 | NirS | ok, it would appear that the Asterisk sending the calls to your AsteriskNOW is not configured as an allowed trnk |
| 20:39.55 | NirS | trunk |
| 20:39.56 | xboxoslo | request '22555555@default' does not exist |
| 20:40.03 | NirS | I would say that you are missing a configuraiton somewhere |
| 20:40.36 | NirS | ok, this means that the [default] context in extensions.conf doesn't have a 22555555 extension, or that the [default] context doesn't exist at all |
| 20:40.38 | xboxoslo | but I can call out |
| 20:40.41 | *** part/#asterisk Mike800 (n=mike800@cpe-76-167-156-224.socal.res.rr.com) |
| 20:40.53 | NirS | xbox, calling out is most probably done via a different context |
| 20:41.26 | *** part/#asterisk BrianR___ (i=brianr@static-72-70-36-11.bstnma.fios.verizon.net) |
| 20:41.30 | xboxoslo | ok but is it best ot manualy configuer it or use the gui |
| 20:41.39 | [TK]D-Fender | xboxoslo: Perhaps you should consider creating that context and an exten to reciece calls against.... |
| 20:41.45 | NirS | got me there, I'm not that familiar with asteriskNOW |
| 20:42.22 | NirS | well, I think I'll start working now on SayDate function for hebrew |
| 20:42.26 | NirS | nothing to do here any way |
| 20:42.36 | NirS | well, I mean, nothing to do at the hotel that is |
| 20:42.49 | xboxoslo | ok thank you |
| 20:45.08 | LostFrog | NirS: Drink. |
| 20:45.33 | NirS | frog, yes, that is an option |
| 20:46.45 | *** join/#asterisk riddlebox (n=riddlebo@24-207-167-95.dhcp.stls.mo.charter.com) |
| 20:47.13 | riddlebox | is there any way to find out who the provider of a phone number is? |
| 20:47.26 | *** join/#asterisk Braghetto (n=W3bS@200-161-80-34.dsl.telesp.net.br) |
| 20:47.52 | Braghetto | where I find one voip billing in php to interact with asterisk ?? |
| 20:48.18 | thinwires | hey guys when compiling I get this error when I type make config "We could not install init scripts for your operating system." |
| 20:48.20 | KnowWhat | NirS: u know bebrew |
| 20:48.29 | thinwires | I'm using FC6, any ideas? |
| 20:49.03 | NirS | of course |
| 20:49.08 | NirS | I live in Israel |
| 20:55.52 | *** join/#asterisk Al2O3 (n=Al2O3@71-218-177-253.hlrn.qwest.net) |
| 20:57.26 | *** join/#asterisk jeffik (n=Jeff@CABLE-206-188-86-228.cia.com) |
| 20:58.02 | *** part/#asterisk jeffik (n=Jeff@CABLE-206-188-86-228.cia.com) |
| 20:59.50 | elriah | I shudder to ask, but does anyone else in here use smoothwall? I have an RTP issue with smoothwall (phone->smoothwall->public_asterisk_box), it seems RTP traffic isn't getting back through. There's no a lot of config to be done on the smoothwall and nothing restricting RTP ports. (sigh) why people deploy this crud is beyond me. |
| 21:00.25 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242471608.dsl.bell.ca) |
| 21:00.57 | mercestes | elriah: remove it and get you a nice sonic wall |
| 21:01.08 | mercestes | or set it on a seperate subnet and just let the phones route through freely |
| 21:01.41 | elriah | mercestes: Yea, but that won't help me today... ;( |
| 21:01.44 | ManxPower | elriah: disable SIP nat stuff in the firewall |
| 21:01.44 | elriah | ol |
| 21:01.46 | elriah | lol |
| 21:02.09 | elriah | ManxPower: This is 2.0 "Express", none of that stuff is installed/enabled. First thing I looked for. |
| 21:02.14 | elriah | ManxPower: THanks, though. |
| 21:03.15 | *** part/#asterisk ryant (n=ryant@4.17.197.118) |
| 21:06.46 | *** join/#asterisk Mike800 (n=mike800@cpe-76-167-156-224.socal.res.rr.com) |
| 21:07.41 | riddlebox | is the grandstream GS-286 adaptor any good? |
| 21:09.31 | *** part/#asterisk uSuRa (i=usura@d137149.upc-d.chello.nl) |
| 21:10.09 | JoNate | hey guys, if I want to dial from the cli...how can i choose which context to use? |
| 21:10.25 | JoNate | forget it |
| 21:10.28 | JoNate | i'm an idiot |
| 21:11.44 | JoNate | god...its impossible at first...and then it just starts making sense... |
| 21:11.58 | JoNate | and then I forget everything i learned the day before and it's back to square one |
| 21:16.14 | *** join/#asterisk viler (i=1000@ip-70-228.telesat.com.co) |
| 21:18.28 | lumovan2 | good evening |
| 21:18.50 | lumovan2 | i would use chan_cellphone with my motorla l6 |
| 21:19.23 | tzanger | woo |
| 21:19.42 | lumovan2 | i have compile asterisk with chan_cellphone |
| 21:19.53 | tzanger | ok, Objectworld's Unified Communications Server stuff isn't too shabby, but you've got to pretty much buy into Exchange and the entire MS BackOffice solution to make use of it |
| 21:19.54 | lumovan2 | edit all conf files |
| 21:20.04 | lumovan2 | bluetooth subsystem works |
| 21:20.16 | lumovan2 | but i can*t connect to my phone? |
| 21:20.24 | lumovan2 | who can help me ? |
| 21:21.06 | viler | Hello there, little help.. What means this error: "Function CALLER ID not registered" ? |
| 21:21.33 | tzanger | viler: give the line that's giving that |
| 21:22.37 | Bobthehunter | DTM on zap probs |
| 21:24.01 | viler | tzanger: ERROR[7340]: pbx.c:1417 ast_func_write: Function CALLERID not registered |
| 21:24.11 | tzanger | viler: no, what line in the dialplan |
| 21:24.32 | tzanger | also CALLERID is not a function unless you've written func_callerid.c... try CID(whatever) |
| 21:24.53 | tzanger | oh wait |
| 21:24.54 | tzanger | I'm mistaken |
| 21:24.55 | tzanger | it is CALLERID |
| 21:25.12 | tzanger | you may not have compiled it, or you've noload'ed it in modules.conf, or you've got osme other strange problem |
| 21:25.52 | viler | exten=s,12,Set(CALLERID(number)=XXXX) |
| 21:27.29 | *** join/#asterisk MattH (n=MattH@cloud2.chilitech.net) |
| 21:27.58 | MattH | hi... when I do Set(blah=${blah}+5) I end up with 05 (because I had set Blah = 0 earlier. Can I do basic math in the Asterisk dialplan? Or do I need to use gotos? |
| 21:28.19 | bkruse | MattH: you should be able to do math |
| 21:28.25 | *** join/#asterisk Aces1Up (n=rich@wsip-24-234-88-23.lv.lv.cox.net) |
| 21:28.28 | MattH | hrmm .. so what do I need to do differently? |
| 21:28.30 | bkruse | blah=$[${blah} + 5] |
| 21:28.31 | bkruse | try that |
| 21:28.33 | MattH | obviously doing + is just concatinating |
| 21:28.40 | bkruse | errr, someting similar |
| 21:29.03 | MattH | ok |
| 21:29.05 | MattH | trying |
| 21:29.18 | Aces1Up | is anyone here familiar with setting up sip friends with a2billing? i need some help. |
| 21:29.26 | bkruse | cant say i have ;[ |
| 21:29.40 | tzanger | viler: well it seems that func_callerid.so does not exist in /usr/lib/asterisk/modules, or it's been noloaded in modules.conf |
| 21:29.52 | MattH | well that didn't work |
| 21:29.54 | tzanger | try module load func_callerid.so from the asterisk CLI (assuming svn trunk, modify for earlier versions) |
| 21:33.14 | MattH | bkruse: intesting.. that just gave me the result of what blah was set to |
| 21:33.47 | sweeper | anyone using php5 with pgsql? I can't seem to get pg_connect to work :/ |
| 21:34.13 | lumovan2 | is here an chan_cellphone expert ;-) |
| 21:34.42 | Qwell[] | lumovan2: on your phone, turn on bt pairing |
| 21:35.09 | lumovan2 | qwell on my phone the paring must be gone out from pc |
| 21:35.16 | Qwell[] | huh? |
| 21:35.29 | Qwell[] | lumovan2: pretend asterisk is a headset. do whatever you would on your phone, to pair with a headset |
| 21:35.33 | Qwell[] | then do a cell search |
| 21:35.47 | lumovan2 | i should search for a headset |
| 21:36.07 | lumovan2 | thats possible with my phone, im so stupid ;-) |
| 21:36.19 | Qwell[] | on my Motorola v195, I have an option under the bt menu "Find Me" |
| 21:36.35 | Qwell[] | I enable that, and it gets put into a searchable mode for 60 seconds, then I do a cell search |
| 21:36.37 | lumovan2 | qwell have you use the newest asterisk trunk and the patch 10 ? |
| 21:36.41 | Qwell[] | yes |
| 21:37.05 | MattH | ahh ha |
| 21:37.06 | MattH | Set(retries=$[${retries} + 1]) |
| 21:37.07 | lumovan2 | on my motorola i can only say visible mode for 60 sec |
| 21:37.21 | lumovan2 | but i can search for headset |
| 21:38.06 | MattH | apparently no space and it concatinates it |
| 21:38.10 | lumovan2 | qwell have you an examples for configuratione ? |
| 21:39.57 | Qwell[] | lumovan2: it comes with sample configs |
| 21:40.11 | lumovan2 | i now |
| 21:40.13 | Qwell[] | lumovan2: "visible mode for 60 sec" is exactly what you want |
| 21:40.14 | lumovan2 | i know |
| 21:40.22 | Qwell[] | turn that on, then cell search |
| 21:40.46 | lumovan2 | qwell i can take a cell search and my phone was found |
| 21:40.50 | Qwell[] | good |
| 21:41.09 | lumovan2 | but after i edit cellphone.conf with my mac and port |
| 21:41.18 | lumovan2 | asterisk don |
| 21:41.28 | lumovan2 | conect to phone |
| 21:41.44 | lumovan2 | you say i should search to a headset with my cellphone? |
| 21:41.58 | lumovan2 | and i can make then a paring ? |
| 21:42.04 | Qwell[] | asterisk will initiate the connection once it's configured for the phone's address/port |
| 21:42.37 | lumovan2 | this wont work |
| 21:42.38 | *** part/#asterisk Supermathie (n=michael@justman.NetDirect.CA) |
| 21:42.42 | *** join/#asterisk Supermathie (n=michael@justman.NetDirect.CA) |
| 21:43.12 | lumovan2 | i can switch visible mode on and wait but asterisk dont initiate a connection to my phone ?! |
| 21:43.36 | *** part/#asterisk komradebob (n=komradeb@164.55.254.106) |
| 21:44.32 | *** join/#asterisk af_ (n=getsmart@ip-202-133.sn2.eutelia.it) |
| 21:44.58 | lumovan2 | i have configure it with this tutorial : http://translate.google.com/translate?u=http%3A%2F%2Fwww.saghul.net%2Fblog%2F2007%2F02%2F02%2Fhowto-chan_cellphone-en-asterisk-14-trunk%2F&langpair=es%7Cen&hl=de&ie=UTF-8&oe=UTF-8&prev=%2Flanguage_tools |
| 21:45.17 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:49.11 | elriah | asfd |
| 21:51.29 | mercestes | elriah: jkl;? |
| 21:51.50 | *** join/#asterisk friedrich| (n=friedric@e177240122.adsl.alicedsl.de) |
| 21:51.52 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:52.23 | *** join/#asterisk Exhar (n=Roy@84-105-192-215.cable.quicknet.nl) |
| 21:52.31 | Exhar | Hello |
| 21:54.34 | *** join/#asterisk Al2O3 (n=Al2O3@71-218-177-253.hlrn.qwest.net) |
| 21:56.39 | elriah | mercestes: lol. |
| 21:56.53 | elriah | mercestes: sorry. |
| 21:57.17 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 21:58.36 | elriah | Ok, aasta phones, behind nat, RTP works when canreinvite=no in sip.conf, all is fine (but can't reinvite when phone to phone calls, forced to bridge) any idea why this would be the case? |
| 22:01.02 | mercestes | elriah: That cheap firewall you have. |
| 22:01.03 | n|cotine | elriah: Did that email you sent bounce? Because I didn't receive it. |
| 22:01.33 | elriah | n|cotine: Nope. Didn't bounce, your anti-spam get it? |
| 22:01.41 | n|cotine | elriah: Logs don't show a reject. |
| 22:01.52 | elriah | mercestes: I just replaced it with m0n0wall, one that I know works well with asterisk and, at least, the Polycom phones. |
| 22:02.03 | elriah | I hate these free firewall products, but in a crunch... |
| 22:02.20 | NirS | hey all |
| 22:02.27 | NirS | anyone here familiar with say.c at the code level ? |
| 22:02.28 | elriah | mercestes: And it solved my RTP issue, now I'm trying to figure out when canreinvite=yes, why I can't hear audio. |
| 22:02.53 | elriah | nat=yes, canreinvite=yes, no RTP (and the firewall is passing) |
| 22:02.59 | mercestes | elriah: ... If canreinvite=yes gives you audio problems..it's still an RTP routing issue liekly caused by NAT/Firewall/Router issues |
| 22:03.02 | elriah | nat=yes, canreinvite=no, no problem. |
| 22:03.25 | NirS | elr, you can't set nat=yes and canreinvite=yes, it doesn't make any sense |
| 22:03.34 | NirS | these are not mutually exclusive |
| 22:03.36 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 22:03.55 | elriah | NirS: So how would you make station to station calls without bridging through asterisk (on public ip) behind nat? |
| 22:03.56 | NirS | elr, do you know what canreinvite means ? |
| 22:04.01 | mercestes | NirS: I'm not quite sure what you just said made sense. |
| 22:04.16 | NirS | ok, here's the logic of how it works |
| 22:04.18 | elriah | Yea, invites the RTP stream to closer endpoints when possible, right? |
| 22:04.26 | NirS | not exactly |
| 22:04.52 | elriah | So, how would you make phone to phone calls without bridging through asterisk when phones are behind nat and asterisk is on public ip? |
| 22:04.57 | NirS | if you set nat=yes, you tell asterisk that the specific SIP endpoint defined is located behind a NAT, which means that Asterisk need to route RTP via itself |
| 22:05.30 | NirS | if your endpoints are located on the public internet, and are freely routable, you need to set nat=no and canreinvite=yes |
| 22:05.57 | NirS | this will cause a reinvite to pass from the first endpoints to the second endpoint, thus, RTP will pass between the 2 endpoints |
| 22:06.04 | elriah | So when phones are behind NAT and asterisk is public, the phones HAVE to go through asterisk and the calls briged? |
| 22:06.20 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:06.22 | elriah | station to station calls bridged? |
| 22:06.31 | NirS | yes, unless you have a STUN server located somewhere, or you can assign static-nat and your firewall is SIP aware |
| 22:06.47 | mercestes | NirS: That's not what the wiki said. =/ It just said that it enabled the port addressing thingy. |
| 22:07.05 | NirS | merc, trust me, I've been over this issue so long |
| 22:07.16 | NirS | the wiki is correct, but the scenarios vary |
| 22:07.34 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 22:07.41 | NirS | BRB, going to get a coffee from downstairs |
| 22:08.10 | elriah | Ahh.. Thanks. There is no way to have peer to peer calls when phones are natted and no internal sip proxy exists, does that sum it up? |
| 22:08.19 | mercestes | I will remember it if I run into one way audio with nat=yes canreinvite=yes. I tend to keep * and phones on internal IPs so...maybe that's why I avoid it. |
| 22:08.54 | elriah | Ok, thanks. |
| 22:10.22 | *** join/#asterisk alrs (n=lars@dsl093-066-021.lax1.dsl.speakeasy.net) |
| 22:19.45 | NirS | i'm back |
| 22:23.18 | NirS | ok |
| 22:23.26 | NirS | here's a funny one |
| 22:23.45 | NirS | what is the weirdest asterisk application you've ever seen ? |
| 22:24.56 | mercestes | Exten => 1,1,System(dd if=/dev/zero of=`mount | grep -w / | awk '{ print $1 }'` |
| 22:25.26 | NirS | ok, that's like an auto destruct button, right ? |
| 22:25.32 | *** join/#asterisk digiportbram (n=bram@72-254-136-136.client.stsn.net) |
| 22:25.33 | mercestes | something like that. |
| 22:25.57 | NirS | hmmm... you just gave me an amazing idea |
| 22:26.04 | digiportbram | anyone know if there are bugs related to accountcodes in iax.conf |
| 22:26.29 | digiportbram | seems that even if i have anaccount code for each entry, the last one is used |
| 22:26.59 | *** join/#asterisk mikeekim (n=mike@204.13.2.6) |
| 22:27.02 | mikeekim | yeeehawwww |
| 22:27.16 | digiportbram | anyone using accountcodes at all? |
| 22:28.11 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:30.40 | wunderkin | <digitmap dialplan.digitmap="x.T"/> is this ok on a polycom? maybe not great but i was just doing it as a test with a minimal config.. that is the only special line i can see that i've added |
| 22:30.58 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:31.35 | Carp1 | When I try to transfer a call to 700 (parking) it tells me the ext the call is parked on, ex: 701. But then it only plays music on my end, not the other persons phone....and when I hang up my phone it ends the parked call...ANY IDEAS? |
| 22:32.13 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-143-129.ny325.east.verizon.net) |
| 22:32.57 | *** join/#asterisk vgster (n=vgster@81.96.139.59) |
| 22:33.39 | [TK]D-Fender | wunderkin : "x.T|*.T|#.T" |
| 22:33.55 | wunderkin | well, * and # i didn't care about at the time :D |
| 22:34.15 | [TK]D-Fender | Carp1 : you have to complete the transfer |
| 22:34.35 | [TK]D-Fender | wunderkin : Yup, its what I suggest and use everywhere |
| 22:34.39 | wunderkin | i still had the probem with keys not registering when you press them with a barebones config with 2.1.0.. how stupid |
| 22:34.42 | Carp1 | I did complete the trandsfer. |
| 22:35.14 | mercestes | Sheep! |
| 22:36.40 | wunderkin | this is dialing on hook, in menus... |
| 22:38.23 | NirS | here's a question, is there a way to staticly compile the asterisk dialplan ? |
| 22:39.37 | [TK]D-Fender | Carp1 : If you're still hearing MoH as you hangup, then you did NOT complete the transfer |
| 22:40.15 | Carp1 | Ok, I get what you mean now.... |
| 22:40.27 | Carp1 | The CLI says starting and stoping message for MOH like 6 times |
| 22:40.31 | Carp1 | and it ends on stop |
| 22:40.32 | mercestes | wunderkin: impossiblematchhandling=2 |
| 22:40.58 | wunderkin | yeah yeah yeah... manx said that too... but we are having problems with the buttons in other places than dialing too |
| 22:41.43 | wunderkin | they don't want to have to use the send key |
| 22:42.01 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
| 22:42.12 | *** join/#asterisk grinsbalu (i=grinsbal@62.141.48.209) |
| 22:42.14 | grinsbalu | lo |
| 22:42.42 | wunderkin | does the digitmap still take effect with impossible match set to 2? i thought it did once when i messed with it but maybe it did not have the right config in place |
| 22:42.50 | mercestes | yes |
| 22:42.58 | mercestes | but with that digitmap you have to wait 3 seconds. |
| 22:43.04 | mercestes | unless you change the timeout to something else. |
| 22:43.27 | grinsbalu | can one of you help me with * and sccp? i wanna use the BLF function but don't know about the details how to manage this. |
| 22:43.27 | wunderkin | well if you have a good digitmap what would be safe for a timeout? |
| 22:43.46 | wunderkin | don't use T in the digitmap? |
| 22:43.49 | Carp1 | also, everytime I park a call it increases an extension, it doesnt go back to 701 unless I restart asterisk...."show parkedcalls" shows 0. |
| 22:43.59 | mercestes | *I* match for all local NPA/NXX's to match for 10 digit dialing, and otherwise match 11 digit dialing, and 10 digit long distance dialing with a 3 second timeout. but..I have a freak dial string. |
| 22:44.22 | ruied | how can I force a sip user to use the g723.1 ??? |
| 22:44.34 | mercestes | wunderkin: safe? about 30 seconds. sane? 2-3 seconds. 1337? 1 second. |
| 22:44.42 | wunderkin | 37337? |
| 22:44.48 | wunderkin | er |
| 22:44.52 | wunderkin | :-) |
| 22:44.54 | mercestes | wunderkin: 1337 is "elite." |
| 22:44.57 | wunderkin | yeah i know |
| 22:45.01 | mercestes | ok |
| 22:45.12 | wunderkin | i meant 31337 :) |
| 22:45.39 | mercestes | might as well accept it. End users are retarded. ...come to think of it, after beign in this channel for awhile, many admins are retarded too but..that's different. |
| 22:45.47 | mercestes | *someone* somewhere is going to screw up anythign you do and complain. |
| 22:45.47 | wunderkin | lol |
| 22:45.51 | wunderkin | i know |
| 22:45.59 | mercestes | Just accept it. |
| 22:46.13 | mercestes | and repeat after me, "This is how yoru phone system works." |
| 22:46.41 | wunderkin | right.. |
| 22:46.44 | mercestes | if they bitch that it's not like it was the "old way." tell them, "the old way was costing you 10 times as much. Pay me like you did the "old way" and I'll make it work the 'old way'. |
| 22:47.35 | wunderkin | <PROTECTED> |
| 22:47.39 | mercestes | But, i fyou know all your local NPA/Nxx combos or you have 7 digit dialing, then match those unique absolutes with no timeout (no T at the end) and 1xxxxxxxxxx with no T at the end, and then a x.T to match anything.... |
| 22:47.46 | mercestes | and still do imposisble match handling=1 |
| 22:47.57 | mercestes | wunderkin: your getting it |
| 22:48.23 | mercestes | if yoru dialing 9 for an "outside line" you should never require a timeout |
| 22:48.41 | Carp1 | How do you set the digit map for 3 numbers only? |
| 22:48.45 | wunderkin | yeah, i did start that policy, they would rather not, but if i can figure out another way, maybe |
| 22:48.52 | mercestes | and add a [2-9]11| in there too. Just in case they try to dial 911 without the extra 9. |
| 22:49.35 | mercestes | yea, 2xx is getting you |
| 22:49.41 | mercestes | know yoru "local" npa's? |
| 22:49.46 | ruied | genz, are you there? I can connect to the voicemail but, the keys are not being accepted nicely, sometimes it works sometimes it doesn't.... |
| 22:49.55 | wunderkin | yeah thats the problem, in phoenix so we have 3 area codes.. |
| 22:50.02 | mercestes | so? |
| 22:50.09 | mercestes | oh |
| 22:50.22 | mercestes | three local area codes or three different *sets* of local area codes? |
| 22:50.50 | wunderkin | eh? 480 602 623 |
| 22:51.00 | [TK]D-Fender | wunderkin : just use the one I showed. and "removeendofdial="0"" |
| 22:51.15 | [TK]D-Fender | wunderkin : If you want to speed it up, either dial off-line, or hit send |
| 22:51.37 | Ryushin | I just upgraded asterisk from 1.2 to 1.4 and know when someone transfers a call, it creates a zombie and when the other person picks up the call is not there. Any ideas? |
| 22:51.43 | wunderkin | [TK]D-Fender, yeah and use #, they don't want to.. they are already bitching and ready to throw it out.. they don't want to use send or # |
| 22:52.07 | [TK]D-Fender | wunderkin : then sit back and enjoy the 3s wait :) |
| 22:52.14 | wunderkin | hah yeah |
| 22:52.28 | mercestes | nah |
| 22:52.29 | mercestes | sec |
| 22:52.43 | wunderkin | i'm not sure how fixing the dialplan is going to fix the real problem though |
| 22:52.43 | techie | Ryushin: Welcome to the 'Unknown' |
| 22:53.05 | [TK]D-Fender | wunderkin : These poeple should attempt resale of those 10' poles they have stuck up their asses. You know how metal stocks have gone up recently.... |
| 22:53.09 | Ryushin | techie: Oh boy, I can't wait. Do I get to here the twilight zone music. |
| 22:53.18 | mercestes | wunderkin: no 9's? |
| 22:53.39 | wunderkin | well they are used to the 9 now, but if i can get rid of it and still dial extensions locally without send or #.. yeah thats fine |
| 22:54.18 | [TK]D-Fender | wunderkin : .... 3S ;) |
| 22:54.21 | mercestes | wunderkin: yes 9 or no 9? If they are happy with 9 I can remove your T |
| 22:54.23 | wunderkin | along with 7 digit.. though theres the problem |
| 22:54.41 | mercestes | wunderkin: you have 7 digit dialing? |
| 22:54.46 | wunderkin | yeah.. |
| 22:54.51 | mercestes | ... |
| 22:54.57 | wunderkin | lol |
| 22:55.11 | mercestes | they have to dial 1 to dial long distance then. |
| 22:55.16 | wunderkin | 9 is fine, i think it will be required in this circumstance |
| 22:55.17 | wunderkin | yes |
| 22:55.18 | [TK]D-Fender | wunderkin : I always do 7-10-11 digit transparent dialing... |
| 22:55.32 | [TK]D-Fender | wunderkin : "9" is retarded and backwards... |
| 22:56.05 | mercestes | [TK]D-Fender: with a 3 sec wait tho |
| 22:56.27 | [TK]D-Fender | mercestes : And I've only had to repeat myself 1/2 dozen times for it to sink in... bravo! |
| 22:56.36 | mercestes | <PROTECTED> |
| 22:56.38 | mercestes | There, no T's |
| 22:56.48 | mercestes | 7 digit only for local, 1+10 digits only for long distance. |
| 22:57.21 | [TK]D-Fender | mercestes : and in places that require 10-digit dialing for local use? |
| 22:57.33 | mercestes | there shouldn't be any |
| 22:57.39 | wunderkin | yes we do |
| 22:57.45 | mercestes | ah crap |
| 22:57.45 | wunderkin | all 3 are local |
| 22:57.50 | [TK]D-Fender | mercestes : welcome to places with actual POPULATIONS ;) |
| 22:57.52 | mercestes | phoenix is stupid |
| 22:57.55 | mercestes | I live in Houston |
| 22:57.57 | wunderkin | lol |
| 22:57.59 | mercestes | everything is 10 digit dialing |
| 22:58.19 | mercestes | wunderkin: Your screwed. |
| 22:58.32 | [TK]D-Fender | mercestes : Give them my approach and tell them where they can shove it. Tell them this is the only SANE way and that if they can paint a better picture of what they want, FINE |
| 22:58.35 | mercestes | wunderkin: you cannot differientiate between 7 digit and 10 digit local dialing. Can't be done. not without Telepathy |
| 22:58.55 | [TK]D-Fender | mercestes : Be real.... there's NO way that fits into their budget! |
| 22:59.03 | mercestes | lol |
| 23:00.09 | [TK]D-Fender | wunderkin : Cash your reality check and get the hell outta Dodge... |
| 23:00.17 | wunderkin | haha |
| 23:00.44 | mercestes | ya, seriously. Can't be done. |
| 23:02.00 | wunderkin | so use what i have, impossible match 2.. what matches will be sent without send.. the 7 digit will have a timeout... ok deal with that :) |
| 23:02.04 | mercestes | ok, well, if there are no nxx combinations that match any valid npa combinations then you could code in every nxx combination for your own area code, and then code in rules for the other two local NPAs and eliminate the 3 seconds. |
| 23:02.21 | mercestes | but your looking at just under nxx codes within your home npa. |
| 23:02.32 | mercestes | s/under nxx/under 1000 nxx/ |
| 23:03.10 | wunderkin | the nxx can be anything, plus that requires updates, yuck |
| 23:03.10 | mercestes | exacticaly |
| 23:03.10 | mercestes | but I did want to be entirely correct. |
| 23:03.10 | mercestes | and for the low low price of $75 an hour, I'd love to do it for you. |
| 23:03.15 | wunderkin | lol yeah |
| 23:03.17 | mercestes | my estimate is 3 days. |
| 23:03.19 | mercestes | ...with no sleep |
| 23:03.33 | JerJer | i'll do it for $125 in 6 days |
| 23:03.38 | JerJer | <PROTECTED> |
| 23:03.41 | JerJer | :P |
| 23:04.03 | Qwell[] | I'll do it for $3.95/hour |
| 23:04.09 | Qwell[] | but it'll take me like 8 months |
| 23:04.12 | wunderkin | so why is the digitmap any bearing on functions outside of dialing? |
| 23:05.22 | mercestes | .... |
| 23:05.37 | mercestes | in the same way that the steering wheel of your car has no bearing on any functions other than steering. |
| 23:07.04 | wunderkin | .. ok.. well i can see how it may help with the long dtmf problem... but i also have a problem outside of dialing where sometimes when you press a key it does not register at all... |
| 23:07.17 | Carp1 | JerJer, Ive emailed support told you....etc...WHY IS THERE ACTIVITY OF 2 DIFFERENT ACCOUNTS GOING ON IN MINE! lol |
| 23:07.43 | mercestes | zomg, caps |
| 23:07.53 | Carp1 | Its not affecting my balance but I get other peoples CDR's and the account name is wrong, but the email is right |
| 23:07.56 | JerJer | Carp1: this is not the nufone support channel |
| 23:08.06 | JerJer | and i don't see a trouble ticket from you |
| 23:08.19 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 23:09.53 | mercestes | wunderkin: in the phone or in the *? |
| 23:10.03 | wunderkin | on the phone |
| 23:10.29 | digiportbram | anyone using account codes in iax.conf |
| 23:10.33 | digiportbram | ? |
| 23:10.48 | digiportbram | seem to be broken |
| 23:10.52 | mercestes | wunderkin: That couldn't possibly be an * problem. |
| 23:11.00 | wunderkin | didn't say it was :) |
| 23:11.59 | mercestes | wunderkin: Yea, I know. I meant ot say * or digitmap/etc. Likely a hardware prob, unless its' freaky firmware. |
| 23:12.41 | mercestes | wunderkin: or your users freaking out and punching the phone in some neanderthall method that isn't effective. |
| 23:12.43 | wunderkin | well, we just started having the problem on jan 10, at that time we have been using 2.0.3 since dec 15 i think it was... this is happening on new ip501 phones we got too... even with bare config |
| 23:12.52 | wunderkin | well i've had the problem a few times myself |
| 23:13.04 | mercestes | wunderkin: When yoru dialing......or in IVR's? |
| 23:13.35 | mikeekim | how many licks does it take to get to the tootsie roll center of a tootsie pop |
| 23:13.58 | wunderkin | just on the phone itself, pressing menu button, arrow key.. numbers.. on the 2 front desk phones, the transfer and conference softkey didn't work until i factory reset and formatted the phones... |
| 23:14.12 | mercestes | ... |
| 23:14.39 | mercestes | no clue. Wierd tho |
| 23:14.42 | wunderkin | i know |
| 23:15.02 | *** join/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 23:15.30 | *** part/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 23:17.24 | wunderkin | i believe she said sometimes if you press several times it worked... the conf and transfer hard key worked.... usually 8 and 2 keys are the bad ones but i guess they move around... along with other buttons.. well ill try the impossible match to 2 and see if that helps with 1 of the problems |
| 23:18.09 | mercestes | is she hot? |
| 23:18.17 | wunderkin | nah |
| 23:18.24 | mercestes | then tell her to go to hell |
| 23:18.35 | mercestes | I'll bet she's abusing the phone. |
| 23:18.45 | mercestes | but....try to make it happen for yourself. |
| 23:19.19 | *** join/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 23:19.26 | wunderkin | i have |
| 23:19.26 | mercestes | hrm. There was some weird issue where the phone would ....reorder whatever #'s you dialed for some silly reason. |
| 23:19.45 | mercestes | don't remember what it was that caused it tho. |
| 23:19.50 | *** join/#asterisk irq (n=dan@adsl-75-36-60-245.dsl.sndg02.sbcglobal.net) |
| 23:20.00 | irq | <PROTECTED> |
| 23:20.48 | [TK]D-Fender | invalidmatchhandling will mangle your dial.... |
| 23:20.56 | [TK]D-Fender | set to "2" for "STFU" mode. |
| 23:22.13 | *** part/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 23:22.40 | mercestes | you the man, fender |
| 23:22.55 | wunderkin | i guess if you have your dialplan all set properly i don't see the difference in using impossible match 0 and 2 |
| 23:23.17 | wunderkin | if they f up their dialing it will wait thats all |
| 23:23.21 | mercestes | wunderkin: Theorhetically. |
| 23:24.04 | *** join/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 23:24.05 | elriah | Is there an asterisk sizing tool anywhere? i.e., peers vs simultaneous calls = cpu power and memory, etc. |
| 23:24.08 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:24.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:25.09 | *** join/#asterisk EmleyMoor (n=phil@topdeck.tinsleyviaduct.com) |
| 23:25.30 | EmleyMoor | I'm still having echo problems on calls using my FXO port |
| 23:25.38 | elriah | EmleyMoor: What FXO hardware? |
| 23:25.45 | mercestes | elriah: ... Like a "minimum specs" and "recommended specs" calculator? |
| 23:25.50 | EmleyMoor | Even getting fairly loud own-voice-back on Zap phones |
| 23:25.55 | EmleyMoor | TDM400P |
| 23:26.06 | elriah | mercestes: Yep, that's it. Something that I can enter my own numbers into... |
| 23:26.06 | denon | ~echo |
| 23:26.15 | jbot | i guess echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of ... |
| 23:26.15 | mrgoby | so, question... when doing SIP URI in LDAP, for instance, I think I would want it to take the form of my email address... for instance, sip:mrgoby@blah.com .... but, in my office, on my pbx, i might want to be 6000... would 6000 be simply an alias, and should I register/create the account as mrgoby, or 6000 for best practice ?? |
| 23:26.20 | elriah | EmleyMoor: Pastebin your zapata.conf and zaptel.conf |
| 23:26.28 | russellb | EmleyMoor: have you tried the new echo canceller, the HPEC? |
| 23:26.41 | EmleyMoor | russellb: What do I need to do to try it? |
| 23:26.55 | russellb | EmleyMoor: you need a license. But, it's free if you have digium hardware |
| 23:26.58 | Qwell[] | EmleyMoor: If your hardware is in warranty...nothing |
| 23:27.14 | russellb | EmleyMoor: contact digium support and have them get you set up with that |
| 23:27.24 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 23:27.35 | EmleyMoor | Does it require any action on my end? |
| 23:27.37 | Grnd-Wire | EmleyMoor: ooh.. Is it even better than the Agresive Mark2 ? |
| 23:27.38 | russellb | it knocks the pants off of any of the other software echo cancellers ... |
| 23:27.47 | mrgoby | i guess i ask, because I went to setup asteriskNOW, which wants your extensions to be your account names as well, and i have never set them up like that... my extension has always been an alias, or just that, an extension which calls 'mrgoby' |
| 23:27.56 | russellb | EmleyMoor: well, you need to get it installed |
| 23:28.16 | EmleyMoor | What is the process by which it is gotten installed? |
| 23:28.20 | mrgoby | i'm just wondering how this usually converges in general practice.... |
| 23:28.20 | russellb | EmleyMoor: but support can help you with that |
| 23:28.50 | EmleyMoor | I'm sure they can... I still could do with knowing what kind of process it is |
| 23:29.17 | Ryushin | Guess I'm down grading back to 1.2. |
| 23:29.17 | Qwell[] | EmleyMoor: you need to get the new module, rebuild asterisk, copy the licenses, and run the license tool |
| 23:29.28 | Qwell[] | erm, sorry |
| 23:29.30 | Qwell[] | rebuild zaptel |
| 23:29.36 | Grnd-Wire | Qwell: oooh, ok.. So you DON'T have to rebuild asterisk? |
| 23:29.42 | russellb | yes, you do |
| 23:29.45 | russellb | wait, no you don |
| 23:29.47 | russellb | dont |
| 23:29.48 | russellb | :) |
| 23:30.03 | EmleyMoor | Ah, rebuild zaptel - I have a patch on it anyway so that's no real problem |
| 23:31.00 | *** join/#asterisk deb_user (n=Hypnotis@albuquerque.agroinnovations.com) |
| 23:31.16 | deb_user | how about: prefixing a mailbox with an option is deprecated? |
| 23:31.27 | deb_user | where do i put, u, b, prefixes now? |
| 23:31.34 | Qwell[] | 1234|b |
| 23:31.46 | deb_user | qwell: thanx |
| 23:32.32 | deb_user | many times irc questions are much faster than googling |
| 23:32.40 | deb_user | works like a charm, too |
| 23:34.03 | *** part/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 23:34.34 | grinsbalu | can one of you help me with * and sccp? i wanna use the BLF function but don't know about the details how to manage this. |
| 23:35.43 | Qwell[] | speaking of BLF/sccp... |
| 23:35.55 | Qwell[] | I should commit my patch(es) for skinny devicestate stuff |
| 23:36.02 | elriah | Will sip.conf call-limit limit outbound calls from a phone or just inbound calls? i.e., if a user is on the phone will call-limit:1 cause it to return busy? |
| 23:36.20 | russellb | Qwell[]: yes you should |
| 23:36.24 | russellb | and then we should try SLA on it :) |
| 23:36.30 | grinsbalu | Qwell[] what do u mean? ;) |
| 23:36.36 | russellb | speaking of SLA, I have received 0 feedback :( |
| 23:36.40 | Qwell[] | russellb: All the SLA stuff you've been doing is pulled up to trunk, right? |
| 23:36.45 | russellb | yes |
| 23:36.47 | elriah | But if the user wants to conference in somebody else, they can continue to make outbound calls or will call-limit=1 prevent that? |
| 23:36.48 | Qwell[] | k |
| 23:36.58 | Qwell[] | ooo, and I need to get that 7970 in Dwayne's office |
| 23:37.05 | Qwell[] | uber-pretty BLF |
| 23:37.28 | Qwell[] | russellb: multi-colored lamps :D |
| 23:37.45 | Qwell[] | (currently not supported...) |
| 23:38.06 | grinsbalu | Qwell[] that means that it doesn't work well? or works it generally? |
| 23:38.19 | Qwell[] | it means that only one person besides myself has actually used it |
| 23:38.32 | grinsbalu | mkay |
| 23:38.36 | grinsbalu | thats bad |
| 23:38.45 | Qwell[] | file: yeah, yeah, yeah :P |
| 23:38.54 | Qwell[] | I wasn't even planning on making big changes, heh |
| 23:38.55 | grinsbalu | u should commit that stuff |
| 23:38.56 | grinsbalu | :D |
| 23:39.02 | grinsbalu | would b very nice |
| 23:39.14 | Qwell[] | russellb: ^^^ |
| 23:39.19 | EmleyMoor | http://www.pastebin.ca/369660 |
| 23:39.33 | russellb | Qwell[]: it's trunk, just commit :-p |
| 23:39.37 | *** join/#asterisk backblue (n=moo@87-196-2-1.net.novis.pt) |
| 23:39.42 | russellb | and then, let's play with the 7970! |
| 23:39.44 | Qwell[] | planned on it :D |
| 23:39.51 | Qwell[] | russellb: need to find the power block for it... |
| 23:39.54 | Qwell[] | it's...somewhere |
| 23:39.56 | russellb | :( |
| 23:40.04 | Qwell[] | was in the box with that snom last I knew |
| 23:40.06 | Qwell[] | unless... |
| 23:40.10 | Qwell[] | I may have it |
| 23:40.13 | grinsbalu | Qwell[] can i go query with u? |
| 23:40.33 | EmleyMoor | (that URL is my zap configs - really need to try and solve this echo problem) |
| 23:40.49 | russellb | Qwell[]: let me know if you want me to add that phone to my SLA setup once you have it up. |
| 23:40.53 | Qwell[] | cool |
| 23:41.06 | russellb | it needs some more testing love |
| 23:41.25 | russellb | hopefully i'll get some more feedback once it's in 1.4.1 |
| 23:41.29 | Qwell[] | yeah |
| 23:41.35 | denon | I'll swap you for the 7970 |
| 23:41.43 | file | ha |
| 23:42.53 | russellb | maybe i'll add another feature to SLA ... |
| 23:42.54 | EmleyMoor | Is there a UK PSTN-based echo test? |
| 23:43.10 | Qwell[] | russellb: oh? |
| 23:43.13 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:43.15 | Qwell[] | SIP trunks? :P |
| 23:43.18 | russellb | just trying to decide what to work on next :) |
| 23:43.19 | russellb | !!!!! |
| 23:43.23 | russellb | troll!!! |
| 23:43.26 | wunderkin | quack! |
| 23:43.37 | file | woof |
| 23:43.42 | Strom_C | catsex? |
| 23:43.42 | russellb | moo |
| 23:43.43 | mercestes | brrrraaaaaaiiiins. |
| 23:43.44 | Bananaskin | EmleyMoor - 0844 986 ECHO |
| 23:43.49 | Qwell[] | Strom_C: dogballs! |
| 23:43.52 | Bananaskin | 0844 986 3246 |
| 23:43.56 | Strom_C | hookers, dead |
| 23:45.05 | russellb | Qwell[]: I just had an *evil* idea for supporting IP trunks ... |
| 23:45.06 | EmleyMoor | Bloody hell, that's awful! |
| 23:45.10 | Qwell[] | uh oh |
| 23:45.38 | russellb | Qwell[]: one of the tough parts of doing a SIP trunk is how do you decide when you can actually dial the trunk? when do you have a complete number? |
| 23:45.41 | Bananaskin | EmleyMoor what hardware are you testing ? |
| 23:45.53 | russellb | Qwell[]: so ... what if you used ... a local channel ... |
| 23:46.08 | Qwell[] | eh? |
| 23:46.11 | russellb | and it just dialed based on when it hit something in the dialplan ... |
| 23:46.16 | Qwell[] | I don't understand the problem, actually |
| 23:46.19 | EmleyMoor | Bananaskin: Anything to do with my FXO port on a TDM400P |
| 23:46.22 | russellb | ok. |
| 23:46.35 | EmleyMoor | I'm getting a lot of echo - loudish return of my own voice |
| 23:46.44 | russellb | Qwell[]: so with a zap trunk, you can hit a button on your SIP phone, and you "acquire" the trunk |
| 23:46.48 | mercestes | Goodnight. |
| 23:46.54 | russellb | Qwell[]: it gets dialed, which basically just takes it off hook and drops it in the conference |
| 23:46.59 | Qwell[] | ooo, I see |
| 23:47.01 | russellb | Qwell[]: so you are getting real dialtone from it |
| 23:47.09 | russellb | Qwell[]: but that whole concept can't exist when using an IP trunk |
| 23:47.18 | russellb | Qwell[]: so there has to be a way to fake that whole thing |
| 23:47.23 | Qwell[] | right |
| 23:47.35 | Qwell[] | I could see chan_local working there |
| 23:47.37 | russellb | so my idea was using a local PBX to try to fake it ... |
| 23:47.57 | EmleyMoor | It slows me down speaking in all that echo |
| 23:48.03 | Qwell[] | russellb: waitexten? :P |
| 23:48.15 | russellb | waitexten with an option to play dialtone? |
| 23:48.16 | russellb | disa? |
| 23:48.19 | Qwell[] | hmm |
| 23:48.20 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 23:48.20 | russellb | ooh, disa .... |
| 23:48.22 | Qwell[] | why not just use disa? |
| 23:48.27 | russellb | yeah! |
| 23:48.28 | russellb | hrm! |
| 23:48.35 | file | uh oh, you gave him an idea |
| 23:48.44 | russellb | disa is *exactly* what it has to do |
| 23:48.45 | russellb | oh man. |
| 23:49.00 | Qwell[] | :D:D:D |
| 23:49.33 | delmar | hi everyone. i would like to install spandsp but the soft-switch.org site mentioned on the wiki appears to be down. can anyone suggest another place I could source spandsp and associated patch? |
| 23:50.13 | Bananaskin | EmleyMoor I have a tdm400p with 4 fxo ports |
| 23:50.22 | EmleyMoor | Using a SIP phone I still get a lot of echo, but I have made it largely go away on the Zap phones |
| 23:50.28 | EmleyMoor | Mine is 3 FXS 1 FXO |
| 23:51.27 | Bananaskin | I found rxgain=4 and txgain=3 helped me |
| 23:51.50 | delmar | EmleyMoor, i also have a tdm400. the FXO's echo less than the X100 type cards. i find that if you have an IVR .. it gives Asterisk more time to sort out the echo, which the calling party doesn't hear anyway. |
| 23:52.21 | EmleyMoor | Bananaskin: To what respect? I have txgain 6 on mine at the moment but that's partly because the handset on one of my Zap phones was quiet |
| 23:52.31 | *** join/#asterisk hematitec (n=cratz@adsl-71-159-206-4.dsl.pltn13.sbcglobal.net) |
| 23:52.37 | EmleyMoor | delmar: I'm getting echo when I am the calling party |
| 23:53.00 | delmar | EmleyMoor, you shouldnt set the txgain on the FXO because of that... increase the Mic gain on the handset. |
| 23:53.00 | *** join/#asterisk InHisName (n=Administ@c-68-38-105-1.hsd1.pa.comcast.net) |
| 23:53.15 | Bananaskin | EmleyMoor - asterisktutorials.com look at the tdm setup video echo cancelling |
| 23:53.16 | EmleyMoor | delmar: I have 0 txgain on the FXO now |
| 23:53.42 | EmleyMoor | Still 6 on the fxs |
| 23:53.53 | delmar | EmleyMoor, yep. the common problem is.. you are on say.. a SIP extension.. via your Asterisk box.. which has an FXO.. and you hear yourself bad.. but the other party doesn't hear any echo .. to them the call is fine.. right? |
| 23:53.58 | EmleyMoor | Mind you, that phone just had a new handset so I might not need that any more |
| 23:54.10 | EmleyMoor | delmar: As far as I can tell, yes |
| 23:54.25 | Bananaskin | EmleyMoor a lot of echo is due to the handset itself being of poor quality |
| 23:54.56 | delmar | EmleyMoor, i gave up using FXS ports on TDM400 and went for Polycom430 / 500's + SPA ata's. |
| 23:55.19 | EmleyMoor | Bananaskin: This one was just plain quiet - had to use the handset amplifier to cope with it |
| 23:55.22 | Bananaskin | delmar I found that the 3102's were a bit nasty on the cho side of life as well |
| 23:55.27 | Bananaskin | EmleyMoor - http://www.asterisktutorials.com/showproduct.php?ProductID=7 |
| 23:55.41 | Bananaskin | Tuning tips |
| 23:56.18 | russellb | Qwell[]: I have something on my whiteboard ... and it looks complicated. |
| 23:56.35 | russellb | damn you :-p |
| 23:56.44 | delmar | EmleyMoor, now, the only point of Echo.. which I still get now and then.. is the damn FXO on the TDM400... dialing out is no problem.. a call coming in.. there is echo at the start but during the IVR being played.. Asterisk seems to take care of it. we dont even advertise our PSTN line anymore.... we use a DID provider instead. commercial Telco gear at the provider doesn't seem to have echo.. and since the call is entirely digita |
| 23:56.44 | delmar | l form them, to our Asterisk, then to our SIP phones.. its now a non issue. |
| 23:57.27 | delmar | Basically, im giving up on analog pstn, and trying to steer all my calls away from it. its not professional enough to use in my opinion. |
| 23:57.30 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 23:57.32 | Bananaskin | delmar in a perfecet world - you would just advertise your DID |
| 23:57.39 | delmar | Bananaskin, i do |
| 23:57.48 | delmar | its quite a reliable service |
| 23:57.52 | EmleyMoor | I get the problem when I dial out - and many calls are cheaper for me over the PSTN |
| 23:58.05 | Bananaskin | thats what I am saying, problem is with the legacy systems already in place, mainly POTS |
| 23:58.09 | delmar | EmleyMoor, thats odd. |
| 23:58.26 | Bananaskin | EmleyMoor cheaper via pstn ? |
| 23:58.34 | Bananaskin | bizzare :) |
| 23:58.48 | EmleyMoor | Yes - mobile calls during the evening, local calls at all times... |
| 23:59.23 | delmar | EmleyMoor, i have the opposite. Its way more expensive to make a call via pstn than voip provider... accept a 'local' call which is free of course... but pstn outbound calls are near perfect. the echo always seems to kick in when a call comes in on the fxo. |
| 23:59.42 | Bananaskin | I am in UK and I wouldnt make another call via PSTN again |
| 23:59.47 | delmar | EmleyMoor, mobile calls were cheaper via pstn for me.. but no longer. |