| 00:00.04 | enkido1970 | that is a different point alltogether Maxn. Just because it is complicated, it does not make SIP a "better" platform |
| 00:00.07 | ManxPower | enkido1970: Honestly, that sounds a lot like H323, except it's not plain text and so is a bitch to debug without special tools |
| 00:00.11 | enkido1970 | see what 4G and the others are doing |
| 00:00.39 | ManxPower | But regardless H323 with Asterisk is not easy to do. |
| 00:00.40 | enkido1970 | not really Manx .. you can debug H323 in a multitude of ways, and IT IS TEXT .. just ASN-1 text |
| 00:01.10 | enkido1970 | look, I run 1million minutes each weekend, with asterisk and cisco. beautiful |
| 00:01.12 | enkido1970 | no probs |
| 00:01.25 | enkido1970 | the problem I'm having now is with * 1.4 and friends |
| 00:01.43 | ManxPower | Ah, with 1.4 Nevermind. I have no suggestions then |
| 00:01.45 | Juggie | ok, well, moving away from this argument. |
| 00:01.48 | Juggie | whats your problem |
| 00:01.49 | enkido1970 | the 323 channel is compiling, just some freak within the makefiles not putting together the .so files |
| 00:02.10 | enkido1970 | Juggie ^^^ |
| 00:02.23 | Juggie | are you getting a chan_h323.so or whatever its called? |
| 00:02.44 | enkido1970 | no .. that's the bitch. the make under channels/h323 compiles fine |
| 00:02.44 | Juggie | or just the .o |
| 00:02.50 | enkido1970 | just the .o |
| 00:02.54 | enkido1970 | no .so |
| 00:02.58 | Juggie | hm. |
| 00:03.03 | Juggie | ok, must be a makefile problem. |
| 00:03.03 | ManxPower | I didn't know that chan_h323 was updated for use with 1.4 |
| 00:03.24 | *** join/#asterisk betatester (n=tester@pool-71-251-229-244.rcmdva.fios.verizon.net) |
| 00:03.31 | enkido1970 | I am trying to build the addons, and still .. h323 is failing to compile alltogether |
| 00:03.32 | ManxPower | enkido1970: BTW, why are you trying to use a non-released version of Asterisk? |
| 00:03.36 | enkido1970 | here is a little dump |
| 00:03.46 | ManxPower | ~pastebin |
| 00:03.50 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 00:03.56 | enkido1970 | I don't think so .. this is siply the 1.4beta3 from the SVN |
| 00:04.03 | Juggie | ManxPower, looks like h323 is still in trunk |
| 00:04.10 | Juggie | so i guess it was ported. |
| 00:04.14 | enkido1970 | aha |
| 00:04.18 | file | it was updated... immensely |
| 00:04.19 | Juggie | Strom_C, still not compiling is a viable issue. |
| 00:04.27 | Juggie | file, ahh just the guy we need. |
| 00:04.29 | ManxPower | Juggie: or nobody got around to updating it. |
| 00:04.34 | enkido1970 | boys, where is the branch for the 1.4 addons ? |
| 00:04.40 | file | no, PCadach put A LOT of work into it |
| 00:04.41 | Juggie | ManxPower, it was last updated 2 weeks ago |
| 00:04.47 | ManxPower | file: any reason to use chan_h323 rather than the H323 channel driver from asterisk-addons? |
| 00:05.12 | Juggie | enkido1970, please provide a pastebin of your compile or whatever |
| 00:05.15 | file | ManxPower: if one works better then the other for the specific use and load ... then yes |
| 00:05.15 | Juggie | so we can see. |
| 00:05.15 | ManxPower | enkido1970: I don't think there is asterisk-addons for 1.4 |
| 00:05.25 | enkido1970 | two ticks |
| 00:05.29 | Juggie | ManxPower, there most certainly is. |
| 00:05.56 | JT | enkido1970: you are going to run one million minutes a weekend on beta software?? |
| 00:05.56 | enkido1970 | Manx, the reason I went that path is because of issues with the addons h323 |
| 00:05.58 | *** join/#asterisk Soul (n=Soul@87-196-35-89.net.novis.pt) |
| 00:06.05 | ManxPower | Juggie: Nifty. |
| 00:06.14 | enkido1970 | I had no probs with the 1.2 branch |
| 00:06.28 | enkido1970 | infact, I am just going to ditch the 1.4 babe and revert back |
| 00:06.36 | enkido1970 | have a deadline to meet in 8 hours |
| 00:06.38 | Juggie | ManxPower, http://svn.digium.com/view/asterisk-addons/branches/1.4/ |
| 00:06.41 | ManxPower | enkido1970: I guess it's time to file an official bug report. |
| 00:06.49 | ManxPower | enkido1970: It sucks to be you. |
| 00:06.49 | Juggie | enkido1970, a log of your compile would be nice |
| 00:06.55 | Juggie | so we can make sure its taken care of |
| 00:07.02 | Juggie | it would appear to be a linking problem. |
| 00:07.09 | Juggie | considering you get your .o but not your .so |
| 00:07.10 | enkido1970 | Juggie, will get right on it after I finish the 1.2 build |
| 00:07.20 | file | with NOISY_BUILD turned on |
| 00:07.25 | enkido1970 | yup .. |
| 00:07.27 | enkido1970 | will do |
| 00:07.37 | Juggie | file will probally spot the problem in like 30 seconds |
| 00:07.42 | enkido1970 | :D |
| 00:07.46 | enkido1970 | I'm sure he will |
| 00:07.47 | ManxPower | I never can understand why people want to run beta in production, expecially with Asterisk's history of horrid bugs. |
| 00:07.49 | Juggie | hes good like that. |
| 00:08.10 | rob0 | file: don't let them down! |
| 00:08.14 | enkido1970 | well .. good poing Manx .. |
| 00:08.18 | Strom_C | ManxPower: it's the "hemorhhaging edge technology" factor |
| 00:08.19 | file | lawl |
| 00:08.27 | enkido1970 | my motivation was memory leaks |
| 00:08.42 | Juggie | file, i think i've cracked my Swiss Challet addiction, i havnt had it in 2 weeks :P |
| 00:08.43 | enkido1970 | lawl |
| 00:08.45 | ManxPower | Strom_C: *nod* many more people are into massive pain that I ever suspected. |
| 00:08.55 | Qwell | Juggie: funny, they just opened one here, and I'm going right now |
| 00:08.55 | enkido1970 | loooooool |
| 00:08.59 | file | Juggie: ungood |
| 00:09.07 | file | Qwell: Swiss Chalet?!? |
| 00:09.09 | Juggie | hah really?! |
| 00:09.11 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 00:09.14 | Qwell | no |
| 00:09.14 | Juggie | dont tease file like that :) |
| 00:09.21 | Strom_C | ManxPower: this explains why there's a market for S&M gear |
| 00:09.34 | enkido1970 | ok boys, I'm out of here for tonight. I'll do the posting on the bugtracker, and post a dump |
| 00:09.39 | Juggie | Qwell, what did they open. |
| 00:09.43 | Qwell | nothing |
| 00:09.48 | Juggie | you jack ass :) |
| 00:09.50 | enkido1970 | ciao |
| 00:09.53 | ManxPower | Strom_C: and explains why it is so damn expensive. $60 for a pair of leather restraints! |
| 00:09.59 | Juggie | thats why file wont move :P |
| 00:10.02 | Strom_C | haha |
| 00:10.07 | *** part/#asterisk enkido1970 (n=1@host81-155-14-191.range81-155.btcentralplus.com) |
| 00:10.34 | Juggie | file: me and some friends drove all the way from ottawa to kingston (1.5hrs) yesterday for some mary browns :) |
| 00:10.51 | dlynes_laptop | ewwww |
| 00:10.54 | Juggie | and she did have the best legs in town |
| 00:11.02 | Omer | how do i allow multipuple ports in sip.conf |
| 00:11.02 | Omer | ? |
| 00:11.10 | dlynes_laptop | I'm surprised Mary Brown's hasn't closed down yet |
| 00:11.12 | ManxPower | Omer: you can't. |
| 00:11.18 | Omer | why ? |
| 00:11.19 | Juggie | 8ball says, you dont |
| 00:11.27 | ManxPower | the port= option in sip.conf specifies the REMOTE PORT |
| 00:11.33 | Omer | what if i have 2 differen voip careers with different ports |
| 00:11.33 | file | Omer: because chan_sip wasn't written to support that |
| 00:11.36 | ManxPower | Omer: because nobody has written support for it. |
| 00:11.41 | Omer | and few rempote users with diff ports |
| 00:11.57 | Juggie | Omer, why would people be contating you on a port besides the one you specify |
| 00:12.07 | Omer | hmmm |
| 00:12.22 | Omer | well some people needs conectivity on different ports |
| 00:12.41 | dlynes_laptop | Omer: on their end, right? i.e. to deal with firewall issues? |
| 00:12.45 | Omer | 5060 is standard |
| 00:12.50 | Omer | yes |
| 00:12.51 | Juggie | to deal w/ isp's blocking sip? |
| 00:12.54 | Omer | yes |
| 00:12.57 | ManxPower | Omer: Asterisk does not support multiple SIP ports on the Asterisk server. It supports specifying the port of the remote device, but you almost never need it. |
| 00:12.59 | Omer | exactly |
| 00:13.12 | Juggie | Omer, i have two suggestions. |
| 00:13.17 | Juggie | #1 move everyone to a non standard port. |
| 00:13.23 | Omer | hmm ok |
| 00:13.37 | file | I think either 1. I'm confused or 2. We're all confused |
| 00:13.42 | ManxPower | #3 Don't use Asterisk |
| 00:13.54 | Omer | i cant do 3 |
| 00:13.54 | Juggie | #2, build a second seperate copy of asterisk, with a different config path and binary path etc... and run a second copy to handle your 'special' users. |
| 00:14.00 | JT | umm |
| 00:14.06 | JT | what about the third option |
| 00:14.10 | Juggie | thats all i got. |
| 00:14.12 | Omer | but do consider this point |
| 00:14.13 | JT | use iptables to port forward |
| 00:14.22 | JT | (or similar) |
| 00:14.23 | Omer | hmm yeah |
| 00:14.31 | Juggie | yeah, that probally wont work though. |
| 00:14.31 | Omer | that can be done |
| 00:14.38 | Juggie | if * returns the packet w/ a source of 5060 |
| 00:14.39 | Omer | or ill use some softswitch |
| 00:14.47 | ManxPower | JT: the 3rd option is what I call the race car option. You must use the right tool for the job. You would not try driving a Ford Festiva in a NSACAR rase. |
| 00:14.50 | ManxPower | race |
| 00:15.04 | JT | ok, or run another instance of asterisk |
| 00:15.07 | JT | perhaps in xen |
| 00:15.15 | dlynes_laptop | Does OpenSER or SER allow you to listen on multiple ports? |
| 00:15.27 | file | sure. |
| 00:15.38 | dlynes_laptop | So why not use one of those as a front-end to asterisk in this case, then? |
| 00:15.45 | npc105 | Anyone ever had a problem with MONITOR_EXEC not being recognized or honored by Monitor()? |
| 00:15.49 | npc105 | I am trying to get sox to join the two legs and compress the single file into a ogg, however it never calls the MONITOR_EXEC commmand |
| 00:15.55 | JT | Juggie: you sure iptables can't reqrite it? |
| 00:16.10 | dlynes_laptop | JT: it would need to rewrite the SIP headers, too |
| 00:16.12 | Juggie | JT, when * sends its response, it will send it from 5060 |
| 00:16.39 | JT | hmm |
| 00:16.40 | Juggie | so it would depend the other end either 1) using the specified port or 2) trying to reply on the source port it received the sip packet from |
| 00:17.12 | dlynes_laptop | JT: which is basically what a SIP proxy does |
| 00:17.24 | bsdfreak | where's fred |
| 00:17.25 | JT | or another instance of asterisk |
| 00:17.41 | Juggie | yeah, openser or another copy of * would be the best solution. |
| 00:18.16 | JT | it would be a nice feature addition in * though |
| 00:18.21 | ManxPower | I think a different server would be easier than a 2nd copy of Asterisk on the same server |
| 00:18.49 | JT | ManxPower: it shouldn't be that hard in xen |
| 00:18.59 | JT | somtime an extra physical server isn't an option |
| 00:19.23 | ManxPower | an extra server is always an option. |
| 00:19.43 | JT | oh come on |
| 00:19.46 | JT | it costs money |
| 00:19.47 | *** join/#asterisk mceGEEK (n=mceGEEK@70-89-196-165-inkleaf-mn.hfc.comcastbusiness.net) |
| 00:19.51 | JT | take space |
| 00:19.56 | JT | makes heat |
| 00:20.00 | JT | uses electricity |
| 00:20.11 | Strom_C | contains molecules |
| 00:20.18 | Strom_C | makes the baby jesus cry |
| 00:20.22 | JT | yep |
| 00:20.23 | Strom_C | etc etc etc etc etc etc etc etc etc |
| 00:20.35 | Strom_C | oh, and don't forget "infuriates PeTA" |
| 00:20.45 | Strom_C | and something about mayonnaise |
| 00:21.08 | JT | you've got to admit that getting another server to get around this problem is more a hack then a solution |
| 00:21.25 | ManxPower | JT: So is every other option that was listed. |
| 00:21.33 | wwalker | if it infuriates PETA, then you should absolutely do it as often as inhumanely possible!! |
| 00:21.40 | Strom_C | hahahaha |
| 00:21.52 | ManxPower | PETA = People Eating Tasty Animals? |
| 00:22.39 | wwalker | The evil bastards took away the People for Eating Tasty Animals site. bastards... terrorists too. almost as bad as the IRS |
| 00:22.46 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@adsl-11-92-66.mia.bellsouth.net) |
| 00:23.34 | TheCops | Someone already had echo via SIP to SIP with the same codec ? |
| 00:23.44 | TheCops | I have problem with this scenario |
| 00:24.03 | JT | ManxPower: a hack that takes a whole extra computer while it sometimes may be easier, is not as nice as one done in software |
| 00:25.15 | hmmhesays | anyone speak german in here? |
| 00:25.29 | JT | nein |
| 00:26.02 | hmmhesays | 99 red balloons has a germain verse in the american punk version |
| 00:26.07 | ManxPower | TheCops: that problem can ONLY be cause by the phones |
| 00:26.13 | hmmhesays | german even |
| 00:26.30 | Qwell | hmmhesays: well, yeah, from the original :p |
| 00:26.57 | hmmhesays | i'm listening to the goldfinger remake, and I need to sing the one german verse, but I have no idea how to pronounce this |
| 00:27.07 | Strom_C | 99 luftballons |
| 00:27.08 | TheCops | ManxPower, hrmm..I mean, this is not 2 phone. I have a Tekelec 7000 PBX at one end, asterisk, and the SIP device. all linked in SIP |
| 00:27.08 | Qwell | position? |
| 00:27.59 | Qwell | hmmhesays: What position is it at? :P |
| 00:28.15 | Strom_C | hmmhesays: http://www.eightyeightynine.com/music/nena-99luftballoons.html |
| 00:28.23 | Strom_C | shazzam |
| 00:28.43 | file | I have that... |
| 00:28.46 | file | in german and english |
| 00:29.09 | hmmhesays | i suppose I could download the original nena version |
| 00:29.17 | hmmhesays | does she sing it pretty clearly? |
| 00:29.31 | Strom_C | as far as I remember, yes |
| 00:29.46 | hmmhesays | in the goldfinger version he's pretty much just yelling it |
| 00:30.35 | Qwell | which word? |
| 00:31.23 | hmmhesays | 4th verse is in german on the goldfinger version |
| 00:32.13 | Qwell | yeah, it's different from the orig lyrics :p |
| 00:32.35 | Qwell | maybe not |
| 00:32.51 | Qwell | brb |
| 00:34.04 | *** join/#asterisk _cleric_ (n=dacleric@p548207D1.dip0.t-ipconnect.de) |
| 00:36.33 | hmmhesays | goldfinger lyrics are the same i believe |
| 00:38.26 | *** join/#asterisk quadrata (n=jhuntwor@ool-44c61466.dyn.optonline.net) |
| 00:38.44 | quadrata | greetings |
| 00:39.05 | Qwell | hmmhesays: they are the same :p |
| 00:39.24 | quadrata | perhaps someone could throw me a cluebat - I was more familiar with asterisk-1.2.x, but now I'm trying to get up to speed with the 1.4.x-betas |
| 00:39.34 | quadrata | it seems the installation of asterisk-sounds has changed? |
| 00:39.46 | ManxPower | quadrata: did you read UPGRADE.txt? |
| 00:39.54 | quadrata | in the Asterisk source? |
| 00:39.57 | quadrata | no |
| 00:40.01 | Strom_C | of course not |
| 00:40.06 | ManxPower | quadrata: it is a good place to start |
| 00:40.17 | quadrata | thanks - didn't know it was there |
| 00:40.25 | quadrata | besides, technically I'm not upgrading |
| 00:41.02 | ManxPower | You are upgrading your info about Asterisk |
| 00:41.25 | quadrata | ManxPower: yes, I realize the logic there now ;) |
| 00:44.10 | quadrata | thanks ManxPower - that clears up some questions |
| 00:44.14 | quadrata | I had the asterisk build scripted and was trying to sort out the appropriate changes now |
| 00:49.04 | *** join/#asterisk Winkie (n=urmom@host86-130-187-253.range86-130.btcentralplus.com) |
| 00:49.13 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:50.21 | *** join/#asterisk RoyK (n=roy@217-175-235.100710.adsl.tele2.no) |
| 00:53.28 | mat2 | hi everyone |
| 00:54.04 | mat2 | i just upgraded to 1.4 beta, and having problems with my tdm400 card now.. |
| 00:54.37 | mat2 | when i run ztcfg I get: |
| 00:54.37 | mat2 | asterisk ~ # ztcfg -vvvvvvv |
| 00:54.37 | mat2 | Notice: Configuration file is /etc/zaptel.conf |
| 00:54.37 | mat2 | line 223: Unable to read Zaptel version information. |
| 00:54.37 | mat2 | Zaptel Version: |
| 00:54.38 | mat2 | Echo Canceller: ô¯û·0¬ÿ¿«ÿ¿Ãý·ÄÏÿ·ÈÛÿ·«ÿ¿¥2ü· |
| 00:54.54 | file | you upgraded zaptel as well? |
| 00:54.58 | mat2 | yes |
| 00:56.12 | mat2 | should i try installing again? |
| 00:57.37 | mat2 | am i supposed to do a modprobe zaptel after? |
| 01:02.10 | masked | rmmod zaptel |
| 01:02.13 | masked | make install |
| 01:02.16 | masked | modprobe zaptel |
| 01:02.25 | masked | (as root) |
| 01:03.00 | JT | why do you need a make install in the middle? |
| 01:05.16 | masked | JT: strictly you don't but it'd be good practice to remove the one currently running in memory before replacing it's disk version and loading it |
| 01:05.50 | JT | i really don't think it makes a difference |
| 01:05.55 | JT | he has already compiled |
| 01:06.03 | masked | JT: i know, point taken. |
| 01:06.05 | JT | so he'd just need to reload the modules and run ztcfg |
| 01:06.33 | mat2 | ok.. ill give that a shot.. |
| 01:10.02 | *** join/#asterisk Grnd-Wire (n=groundwi@71-217-117-57.tukw.qwest.net) |
| 01:10.41 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 01:10.43 | Grnd-Wire | Greetings guys! Anyone have any current documentation on getting zaptel to compile on FC6 ? The wiki at voip-info.org doesn't have anyone on Fedora Core 6 [yet].. |
| 01:10.55 | asdx | is there a voip telephone that runs linux? |
| 01:11.47 | *** join/#asterisk brian (i=brian@unaffiliated/brian) |
| 01:11.48 | brian | hi |
| 01:13.00 | JT | asdx: snom |
| 01:14.02 | Grnd-Wire | heya |
| 01:14.54 | Grnd-Wire | Has anyone successfully compiled zaptel on Fedora Core 6 yet? |
| 01:15.04 | asdx | JT: thanks |
| 01:15.36 | asdx | JT: how much does it cost? |
| 01:15.52 | JT | depends on the model |
| 01:15.56 | rob0 | Grnd-Wire: I don't know, but jbot has a ~fedorabug factoid, I think. |
| 01:16.12 | rob0 | ~fedorabug |
| 01:16.21 | quadrata | anyone run into trouble with $(sbindir) not being configured in the Makefile of 1.4.0-beta3? |
| 01:16.25 | rob0 | or not |
| 01:16.33 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 01:16.34 | rob0 | ~centosbug |
| 01:16.35 | jbot | extra, extra, read all about it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 01:16.46 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 01:17.07 | quadrata | ./configure --prefix=/usr --exec-prefix=/usr results in the asterisk utils being installed in / |
| 01:19.02 | quadrata | hrm, actually, it looks right from the included makeopts file |
| 01:19.14 | quadrata | but it's definitely not installing the binaries correctly |
| 01:21.00 | asdx | how much does a TDM400P cost? |
| 01:21.21 | Grnd-Wire | rob0: hm - It's not finding kernel sources that I know are there.. Does that sound familiar to you? |
| 01:21.25 | oink | Anyone knows the main differences between 7906G and 7911G phones ? |
| 01:21.36 | Grnd-Wire | rob0: heh - It sounds ffamiliar to me.. every freaking release it's different... :/ |
| 01:21.38 | Qwell | oink: about 5 |
| 01:21.46 | Grnd-Wire | haha |
| 01:21.47 | oink | Qwell: Thanks ;-) |
| 01:21.54 | Grnd-Wire | $5? 5 features? |
| 01:21.58 | Grnd-Wire | :D |
| 01:23.16 | oink | What about a feature / usability comparison between Cisco 7911G and Snom 300 ? |
| 01:23.33 | oink | If anyone here has tested both phones |
| 01:24.41 | Grnd-Wire | oink: Checkout www.voip-info.org Search around a little bit, and you'll find all sorts of info on the phones themselves, as well as what it takes to configure them with various versions of Asterisk |
| 01:25.08 | Grnd-Wire | oink: I feel it's a tad disorganized, but there's SOO much info there - I don't think I could exactly organize it better myself.. |
| 01:25.25 | masked | quadrata: i had a problem installing in a non-root environment, it would still try to put things in /var/lib |
| 01:26.07 | quadrata | yeah, but I mean this one is totally wrong - the utils don't go to a bin dir |
| 01:26.35 | quadrata | the problem is that $(ASTSBINDIR) doesn't seem to be getting set |
| 01:26.39 | quadrata | so the utils end up as /muted , /streamplayer , etc |
| 01:26.49 | masked | bugger. |
| 01:27.02 | quadrata | yeah, well I can manually move them for now :P |
| 01:27.08 | quadrata | but that's not going to be good |
| 01:27.09 | oink | Grnd-Wire: I went there already ;-) Thanks though, I guess I should test these phones and judge by myself |
| 01:28.37 | Grnd-Wire | oink: Unfortunately - I'm going to end up buying an Aastra, and a Snom phone as well.. I have a Grandstream GX-2000 here I bought off of ebay used (less than retail price).. |
| 01:29.24 | Grnd-Wire | oink: I don't even have my asterisk installed yet, but these phones are pretty damn cool considering how much they cost.. They don't FEEL cheap, which is a really important consideration when designing a sellable product. |
| 01:29.33 | rob0 | Qwell is wrong. 7911G minus 7906G is 5G !! |
| 01:30.51 | Qwell | If G were a unit, it'd be G7911 |
| 01:30.52 | masked | wow thats like the generation beyond the next generation from 3G thats called telstra's next-g |
| 01:30.52 | masked | stupid foxtel advertisement |
| 01:30.52 | rob0 | I think it's a suffix, like K or M, except it means kilodollars (Grand). |
| 01:31.57 | Grnd-Wire | hee hee |
| 01:34.13 | *** join/#asterisk FunnyManVA (n=coriley@c-75-75-99-240.hsd1.va.comcast.net) |
| 01:35.46 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:36.14 | shmaltz | anyone here have the release notes for the Polycom sip 2.0.1 or 2.0.2? |
| 01:36.27 | shmaltz | err 2.0.3 |
| 01:52.40 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 01:55.48 | *** join/#asterisk gnu_linux_geek (n=mark@cpe-71-74-141-159.neo.res.rr.com) |
| 01:56.11 | gnu_linux_geek | hello everyone |
| 02:01.04 | *** part/#asterisk mat2 (n=mat@c-24-5-141-132.hsd1.ca.comcast.net) |
| 02:01.26 | TheCops | Someone have the Polycom SIP Software 2.0.3? |
| 02:03.57 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 02:06.05 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 02:06.16 | iceyp | hey guys, can someone suggest a web based sip client please |
| 02:06.27 | *** join/#asterisk gnu_linux_geek (n=mark@cpe-71-74-141-159.neo.res.rr.com) |
| 02:07.29 | gnu_linux_geek | hello everyone, when is then release of asterisk due? any CentOS experts here btw? |
| 02:07.31 | *** join/#asterisk jerlique2 (n=jerlique@lnk2.adl.adsl.esc.net.au) |
| 02:09.43 | gnu_linux_geek | sorry s/then/the next / |
| 02:09.53 | gnu_linux_geek | s/then/the next/ |
| 02:10.22 | gnu_linux_geek | whoops I did that wrong |
| 02:12.33 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 02:13.03 | iceyp | anyone know of a sip web client? |
| 02:14.18 | iceyp | or iax |
| 02:15.39 | gnu_linux_geek | iceyp: have you checked voip-info.org? I think I saw some info there about web clients |
| 02:16.10 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 02:16.22 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:20.14 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 02:22.52 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 02:31.24 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 02:31.42 | gnu_linux_geek | cya all later |
| 02:31.46 | *** part/#asterisk gnu_linux_geek (n=mark@cpe-71-74-141-159.neo.res.rr.com) |
| 02:40.15 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 02:43.41 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 02:47.38 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 02:49.52 | *** join/#asterisk dsd_ (n=dsd@gentoo/developer/dsd) |
| 02:54.02 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 03:04.35 | *** join/#asterisk AJaymn (i=AJayMn@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 03:10.08 | dsd_ | i am running asterisk on my local box, just trying to get a hello-world over SIP thing working to start with. registration happens ok, and the asterisk console says it is playing hello-world, but all i get is silence. any suggestions for where to look next? |
| 03:10.28 | JT | nat |
| 03:10.35 | JT | an echotest would be more useful |
| 03:10.42 | JT | or 2 party test |
| 03:10.56 | dsd_ | cant be nat, i'm running this on my local box |
| 03:11.18 | JT | the sip client is on the local box too? |
| 03:11.24 | dsd_ | indeed |
| 03:11.42 | dlynes_laptop | dsd_: are you using a sangoma card? |
| 03:11.47 | JT | try an echo test? |
| 03:11.57 | *** join/#asterisk _Soul_ (n=Soul@87-196-70-84.net.novis.pt) |
| 03:12.15 | JT | dlynes_laptop: sip - asterisk, dunno where hardware would come in? |
| 03:12.21 | dsd_ | dlynes_laptop: this is a simple hello world playback thing.. i dont think my telephony hardware matters at this point |
| 03:12.41 | dsd_ | i'll try an echo test |
| 03:13.06 | JT | have you checked that the hello world soundfile even exists? |
| 03:13.38 | dsd_ | yes, it does |
| 03:13.42 | dlynes_laptop | JT: because if the hwec hung on initialization, and a sip call is made into the box, no audio will get played back because asterisk will try to use zaptel timing if it thinks a zaptel timing device exists |
| 03:14.01 | JT | ah ok |
| 03:14.05 | dsd_ | dlynes_laptop: ah good point |
| 03:14.10 | dlynes_laptop | I've only experienced that problem with sangoma |
| 03:14.25 | dlynes_laptop | It's usually because the pid lock file is there on bootup |
| 03:14.31 | dlynes_laptop | so sangoma thinks it's already running |
| 03:14.40 | dsd_ | i have a cheapy wildcard X100P |
| 03:14.45 | dlynes_laptop | so it loads the driver, but doesn't initialize using wanrouter |
| 03:14.51 | dsd_ | how can i check that its producing a timing signal? |
| 03:14.55 | dlynes_laptop | dsd_: make sure dmesg didn't have any issues, then |
| 03:15.17 | dsd_ | no problems there |
| 03:15.39 | JT | dsd_: zttest |
| 03:16.09 | dsd_ | yep close to 100% accuracy |
| 03:16.41 | Juggie | dsd_, do you have a t1 card installed without a t1 configured? |
| 03:16.43 | dlynes_laptop | dsd_: are you getting any errors in your full log? |
| 03:16.48 | JT | dlynes_laptop: be more specific |
| 03:16.50 | JT | err |
| 03:16.53 | JT | dsd_ |
| 03:17.08 | dsd_ | --- Results after 7 passes --- |
| 03:17.08 | dsd_ | Best: 99.987793 -- Worst: 99.987793 -- Average: 99.987793 |
| 03:17.09 | Juggie | er, nm, i just read above. |
| 03:17.21 | JT | that's a decent score |
| 03:17.27 | JT | below 99.97 is bad |
| 03:17.36 | dsd_ | actually the zaptel one is on a remote box.. but i was having this problem there, so i decided to move the server locally |
| 03:17.49 | Juggie | huh? |
| 03:17.49 | dsd_ | on this system i have no telephony hardware |
| 03:18.01 | Juggie | did you install zaptel? |
| 03:18.09 | dlynes_laptop | and the local machine is having issues as well? |
| 03:18.17 | dsd_ | on the one with the X100P - yes. on the local machine - no |
| 03:18.30 | Juggie | dsd_, the one w/ the x100p i suspect is a lack of timeing. |
| 03:18.37 | Juggie | try this |
| 03:18.49 | Juggie | assuming you have the scripts all installed |
| 03:18.57 | dsd_ | yes local machine has the same problem, hello-world is silent, BUT i didnt configure the timing source thing.. havent configured the ztdummy(?) source |
| 03:19.09 | Juggie | where did hello-world come from? |
| 03:19.21 | Juggie | is that part of *? |
| 03:19.22 | JT | so what are the echo test results? |
| 03:19.36 | dsd_ | asterisk includes the hello-world sample sound file |
| 03:19.44 | Juggie | make sure your sip audio is working |
| 03:19.46 | Juggie | first |
| 03:20.10 | dsd_ | it works with the ekiga echo server |
| 03:20.30 | JT | what are the echo test results on your asterisk machine? |
| 03:20.41 | dsd_ | one sec :) |
| 03:21.02 | *** join/#asterisk raetin (n=raetin@136.218.121.70.cfl.res.rr.com) |
| 03:21.02 | Grnd-Wire | hmm - How do you do echo tests? |
| 03:21.15 | JT | Echo() |
| 03:21.52 | dsd_ | yep echo test works (locally) |
| 03:21.54 | EyeCue | <3 echo() |
| 03:22.14 | JT | so you have an issue with playback perhaps |
| 03:22.27 | dsd_ | strange |
| 03:22.34 | JT | try record() then playback() (the same file) |
| 03:22.41 | raetin | Hey, I am running asterisk 1.4 over an internet connection to a VOIP provider -- am getting a lot of echo over land lines. Any suggestions? Most of what I find about fixing it is changing zaptel headers which I don't think would affect my setup? |
| 03:23.48 | *** join/#asterisk jerlique (i=jerlique@203.25.185.109) |
| 03:24.06 | *** join/#asterisk jerlique (i=jerlique@203.25.185.111) |
| 03:24.46 | dsd_ | JT: doesnt work |
| 03:25.35 | JT | is much showing on the console? |
| 03:25.40 | JT | make sure the verbosity is at least 5 |
| 03:26.00 | dsd_ | no errors |
| 03:26.37 | dsd_ | when it plays back hello-world it exits surprisingly quickly |
| 03:26.37 | dsd_ | i recorded as wav, and the playback of that seemed to last the same length as the recording |
| 03:28.18 | dsd_ | echo test works to the remote system as well |
| 03:31.14 | *** join/#asterisk ipso (n=ipso@S010600a0d1b92a08.ok.shawcable.net) |
| 03:33.31 | dsd_ | even though Playback() appears broken, i should be able to use my X100P regardless right? |
| 03:33.38 | ManxPower | raetin: Echo must be fixed where the VOIP and PSTN interface. i.e. your provider |
| 03:34.10 | ManxPower | If you get echo from a provider your best solution is to change providers. |
| 03:34.35 | raetin | ManxPower: thanks |
| 03:34.39 | JT | ManxPower: what about lines remote of the provider? they can't control those |
| 03:34.42 | ManxPower | dsd_: put "cat /proc/interrupts" on pastebin.ca |
| 03:34.52 | raetin | On that note, anyone have experience with broadvoice? |
| 03:35.09 | dsd_ | http://pastebin.ca/274986 |
| 03:35.19 | ManxPower | JT: No, he cannot control those, which is why the best solution is to use a provider with decent echo canceling |
| 03:35.22 | dsd_ | deffo getting interrupts on wcfxo |
| 03:35.30 | JT | heh |
| 03:35.38 | dsd_ | also dialling my number is making my phone ring |
| 03:35.40 | ManxPower | dsd_: that looks fine. |
| 03:35.48 | raetin | or have any suggestions on a good provider for asterisk (for 7-8 lines) |
| 03:35.52 | dsd_ | (cant answer as it will cost too much) |
| 03:36.09 | ManxPower | dsd_: The 2nd port on the X100P is hardwared to the first port. You cannot use it for Asterisk |
| 03:36.25 | ManxPower | raetin: All VOIP providers suck. Teliax seems to suck less. |
| 03:36.43 | raetin | ugh |
| 03:36.54 | raetin | am I going to regret doing this for a smb? |
| 03:36.59 | dsd_ | ManxPower: hmm.. let me tell you what i'm trying to do first |
| 03:37.30 | dsd_ | i want to connect to my overseas computer over SIP, which is connected to a phone line over there via a X100P |
| 03:37.36 | ManxPower | raetin: We use Asterisk in this setup: Telco -> PRI -> Asterisk -> SIP phones. As you can see we do not send calls over the internet |
| 03:37.50 | dsd_ | and i then want to basically use that phone line for outgoing calls, while being overseas |
| 03:38.06 | ManxPower | raetin: Asterisk works fine for an SMB, but don't expect to save much MONEY doing it. |
| 03:38.27 | dsd_ | i'm aware that my X100P setup is limited in that i couldnt, say, dial into it from one phone line and then make an outgoing call |
| 03:38.36 | raetin | ManxPower: Dunno, our long distance bill is pretty big (not a call center though) |
| 03:39.05 | ManxPower | raetin: then what you need to do is get better LD rates. You can get them for as low as 7cents/min or less without VoIP. |
| 03:39.15 | ManxPower | I think I saw a 3cents/min somwhere |
| 03:39.31 | ManxPower | raetin: Then send your toll calls over VoIP and all other calls over local phone lines. |
| 03:39.56 | raetin | hmm |
| 03:40.00 | ManxPower | Heck, some telcos even have "unlimited long distance" packages for SMBs |
| 03:40.11 | dlynes_laptop | dsd_: it'll do waht you need...the secondary port is just a passthrough port so that if you want to use it for faxing out when asterisk isn't using it you can |
| 03:40.38 | ManxPower | raetin: Asterisk is a GREAT system. I manage about 6 or 7 Asterisk systems |
| 03:40.49 | raetin | oh yeah, I'm liking asterisk |
| 03:41.04 | raetin | honestly I think it is a bit overcomplicated to setup, but otherwise it is good |
| 03:41.13 | ManxPower | Local phone lines will ALWAYS be more reliable than VoIP over the internet |
| 03:41.19 | *** join/#asterisk ipso (n=ipso@S010600a0d1b92a08.ok.shawcable.net) |
| 03:41.44 | raetin | was thinking about getting 4 lines with two providers to help out with reliability |
| 03:42.04 | ManxPower | raetin: two Internet providers too? |
| 03:42.25 | raetin | in the somewhat near future, yes |
| 03:42.27 | ManxPower | No matter how reliable a provider is, the internet is not reliable. |
| 03:42.32 | raetin | dsl & cable |
| 03:43.18 | raetin | used to have a T-1, but opted "down" for the cable after the T-1 in our area proved to be only semi-reliable |
| 03:43.22 | dlynes_laptop | The Internet is totally reliable...why just yesterday, Telus reliably didn't work for about 3 or 4 hours |
| 03:43.32 | raetin | hehe |
| 03:43.39 | raetin | yeah |
| 03:43.45 | ManxPower | I'm lucky in that %90 of my customer's calls are in-state and we have unlimited calling within Louisiana and Mississippi from our local telco |
| 03:44.01 | dlynes_laptop | I can always depend on Telus to totally foul up customer's expectations |
| 03:44.10 | raetin | yeah, our total phone bills top 800 bucks a month |
| 03:44.33 | ManxPower | raetin: Talk to a couple of CLECs, you can get a better deal. |
| 03:44.50 | dsd_ | can i make asterisk dial a number over zaptel, and record that call to a local sound file? either in addition to sending it over SIP, or instead |
| 03:45.05 | raetin | and I despise the telco we work with -- every time we need to get something done it gets screwed up royally |
| 03:45.08 | ManxPower | dsd_: "show appliation monitor" |
| 03:45.10 | raetin | usually by sprint really |
| 03:45.33 | raetin | who is being resold by our telco more or less |
| 03:45.48 | ManxPower | We had the dream relationship with our telco. For many years we the telco's 2nd largest customer. |
| 03:45.55 | dsd_ | ManxPower: thanks! reading |
| 03:45.55 | *** join/#asterisk ipso (n=ipso@S010600a0d1b92a08.ok.shawcable.net) |
| 03:45.55 | ManxPower | (the largest was the telco's ISP) |
| 03:46.17 | dlynes_laptop | raetin: for a second there, I thought you were going to say your telco was Telus :) |
| 03:46.33 | ManxPower | raetin: where are you located? |
| 03:46.37 | raetin | hehe, no, it is FDN (FL digital networks) |
| 03:46.41 | raetin | Orlando |
| 03:46.58 | ManxPower | raetin: you don't want your telco to be a reseller, you want them to be a facility's based CLEC |
| 03:47.26 | ManxPower | raetin: are most of your calls in-state, inter-state or internationallal? |
| 03:47.51 | raetin | Well, I'm not familiary with all the telco terms -- but FDN might be a CLEC, I know they have a lot of their own equipment than handles calls & so forth |
| 03:48.38 | raetin | i'd say about 30%'ish in state, the rest being long distance or incoming on an 800 number |
| 03:48.39 | ManxPower | raetin: IF they are reselling service then they are not facilities based. |
| 03:49.01 | raetin | sprint owns all the wire in the area |
| 03:49.06 | raetin | or embark rather |
| 03:49.14 | ManxPower | raetin: I'll bet you can get a T-1 to a LD provider for free, you will get VERY low rates. |
| 03:49.40 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 03:49.42 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 03:50.09 | raetin | any suggestions on looking for them? |
| 03:50.09 | ManxPower | raetin: yes, but a Fac CLEC will ONLY be using the telco's physical plant (wires). A reseller can't provide their own dialtone |
| 03:50.30 | raetin | ah yeah, I think FDN is a CLEC then |
| 03:50.32 | ManxPower | raetin: First question is how much equipment have you purchased for your Asterisk PBX? |
| 03:51.01 | raetin | one server, which is being dual purposed to a few other very light jobs as well |
| 03:51.35 | ManxPower | raetin: good, then you are not tied into any one specific card. |
| 03:51.42 | raetin | ah, nope |
| 03:51.55 | dsd_ | cool, i can make calls! |
| 03:52.05 | raetin | that server could only take one zaptel card I think though |
| 03:52.17 | raetin | being a 1U |
| 03:52.28 | *** join/#asterisk bmg505 (n=leon@c1-116-1.rndf.isadsl.co.za) |
| 03:52.29 | ManxPower | raetin: I would consider this two projects. The first is to find good pricing for LD and local lines, then use Asterisk to save more money by using VoIP for overflow when your local ines are busy, for example |
| 03:52.47 | *** join/#asterisk ipso (n=ipso@S010600a0d1b92a08.ok.shawcable.net) |
| 03:54.17 | ManxPower | raetin: I would talk to other business people and friends and find out who they use and if they are happy with them. Also look in google. |
| 03:54.32 | raetin | alrighty |
| 03:54.38 | raetin | thanks much :) |
| 03:55.07 | ManxPower | Once you make the contact, send them 3 months of your phone bills. Tell them they must save you X percent or not to bother calling you back,. |
| 03:55.15 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 03:55.22 | ManxPower | raetin: My plans for my clients are to use VoIP for overflow calls. |
| 03:55.34 | ManxPower | i.e. all local lines in use, sent it over VoIP |
| 03:57.06 | raetin | hmmm |
| 03:57.49 | ManxPower | have a local VOIP number, have the telco call-forward-on-busy to the VoIP number |
| 03:59.23 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 03:59.38 | Newbie___ | hi all, where can i get a CA area code to work with * ? |
| 04:00.26 | ManxPower | Newbie___: We have no idea what you just said. That usually means the answer is "Since you are using Trixbox or FreePBX you should go to #freepbx for help." |
| 04:01.02 | Newbie___ | ManxPower: ok |
| 04:01.27 | raetin | ugh, I tried setting this up with trixbox, it was nice until I finished and it just didn't want to work right :) lol |
| 04:01.29 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 04:02.06 | ManxPower | raetin: I feel that Frixbox/Freepbx are WORSE for users. With them you have to learn to troubleshoot Asterisk AND you have to learn their config file design |
| 04:03.19 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:03.49 | [TK]D-Fender | ManxPower : Not really, those who start with it and find frustration either dump it entirely and go plain-* or dump anything * based for a good long while. If they can't figure out how to use freePBX's dumbed down interface, what prayer do they have tofixing its flaws? |
| 04:04.34 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 04:04.55 | raetin | Actually, when I found out that Trixbox stores config in mysql & then destroys & recreates config files rather than actually parsing the config files, I pretty much tossed it |
| 04:05.00 | [TK]D-Fender | Trixbox is for lazy poeple who just to get the job done the easiest way possible and not have to learn anything. |
| 04:05.20 | CunningPike | ManxPower: Trixbox radar is working well tonight ;) |
| 04:05.46 | ManxPower | CunningPike: It's in the wording and terms. |
| 04:05.53 | [TK]D-Fender | CunningPike : Yeah, that was just EERIE. |
| 04:06.06 | ManxPower | The word "trunk" usually indicates TrixBox. Especially "sip trunk" |
| 04:06.48 | raetin | Mentioning it in the channel got a "I disagree! Relational vs. Flat file? Relational is so much more flexible!" Which is a bit interesting to say when you are converting relational to flat |
| 04:06.54 | [TK]D-Fender | ManxPower : I don't recall him using that word before you ID'd him. |
| 04:07.05 | ManxPower | [TK]D-Fender: not in this case. |
| 04:07.49 | ManxPower | In this case I figured anyone that used the phrase "CA area code" when he meant "Canadian phone number", he must be a Trixbox user |
| 04:08.26 | CunningPike | Could have been California... |
| 04:08.41 | [TK]D-Fender | ManxPower : CA is California, Canada has MANY area codes. Heck, Montreal has abou 4 now :) |
| 04:08.48 | ManxPower | When someone puts a sip.conf on pastebin.ca that only contains "#include sip_additional.conf" I just want to slap them into next week. |
| 04:09.03 | ManxPower | [TK]D-Fender: California has many area codes too. |
| 04:09.08 | [TK]D-Fender | ManxPower : Yeah. thats a dead giveaway.... we know the signs.... |
| 04:09.23 | [TK]D-Fender | ManxPower : That too :) Was just giving a single city-example :) |
| 04:09.29 | ManxPower | CunningPike: Odd that it never occured to me that he might have meant California |
| 04:09.36 | raetin | lol |
| 04:09.36 | dsd_ | thanks for the help everyone, much appreciated |
| 04:09.40 | *** part/#asterisk dsd_ (n=dsd@gentoo/developer/dsd) |
| 04:10.24 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:10.26 | raetin | hmm |
| 04:10.31 | Qwell | mog: !!! |
| 04:11.02 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 04:11.45 | ManxPower | How many Buddhist monks does it take to change a lightbulb? |
| 04:11.53 | raetin | ManxPower: just as an example, say I setup asterisk w/ 4 voip lines (primary for LD) and 4 lines over POTS. If the internet goes down, is asterisk going to automatically notice and fall back on the POTS line or are users going to get not avail, etc. types of messages? |
| 04:12.01 | ManxPower | None, the Lightbulb must change itself. |
| 04:12.06 | raetin | hehe |
| 04:12.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:12.24 | ManxPower | raetin: no, but you should be able to make it do so with a good dialplan |
| 04:12.26 | [TK]D-Fender | raetin : You need to do everything in your diaplan yourself |
| 04:12.43 | [TK]D-Fender | ManxPower : How many psychologists does ti take to change a lightbulb? |
| 04:12.58 | ManxPower | I basically wrote some very terrible macros to make managing the dialplan |
| 04:13.00 | raetin | ah, ok, I can look that up then |
| 04:13.10 | ManxPower | easier |
| 04:13.21 | CunningPike | [TK]D-Fender: Tell me about your mother |
| 04:13.53 | [TK]D-Fender | Answer : just 1, but the lightbulb has to really want to change. |
| 04:14.09 | ManxPower | Q: How many radical lesbian feminists does it take to change a light bulb? |
| 04:14.21 | ManxPower | A: THATS NOT FUNNY!!!!!! |
| 04:14.30 | [TK]D-Fender | ... |
| 04:14.34 | raetin | hehe |
| 04:14.52 | file | [TK]D-Fender: I don't want to know your name |
| 04:15.38 | JT | because you may be obliged to provide it under threat of court order? :) |
| 04:15.54 | dlynes_laptop | CunningPike: good evening |
| 04:16.05 | [TK]D-Fender | file : I just want |
| 04:16.06 | CunningPike | hey, dlynes_laptop |
| 04:16.13 | file | ! ! ! |
| 04:17.48 | dlynes_laptop | CunningPike: working on a dial plan app |
| 04:18.04 | CunningPike | dlynes_laptop: So I see in #asterisk-dev |
| 04:18.10 | CunningPike | dlynes_laptop: What's it for? |
| 04:18.27 | dlynes_laptop | CunningPike: You know how some offices want all incoming calls to ring on all phones? |
| 04:18.50 | dlynes_laptop | CunningPike: Well, when they pick up a line, they're usually calling out on the last available line appearance |
| 04:19.00 | dlynes_laptop | CunningPike: So, I never know what lines are free on each phone |
| 04:19.17 | dlynes_laptop | CunningPike: it's kind of like a crap shoot |
| 04:19.18 | CunningPike | dlynes_laptop: I see |
| 04:19.33 | dlynes_laptop | CunningPike: so I'm writing an application to allow me to define a regex for each phone |
| 04:19.53 | dlynes_laptop | CunningPike: and it'll grab the first available line appearance defined by that regex for the named peers |
| 04:20.08 | CunningPike | dlynes_laptop: Cool |
| 04:20.40 | dlynes_laptop | CunningPike: Chanisavail or ischanavail or whatever it is, is completely useless |
| 04:20.47 | dlynes_laptop | It's not even remotely reliable |
| 04:20.50 | CunningPike | dlynes_laptop: Yes, it is, for that purpose |
| 04:21.00 | [TK]D-Fender | dlynes_laptop : how so? |
| 04:21.13 | dlynes_laptop | [TK]D-Fender: it always tells me the line is free |
| 04:21.17 | dlynes_laptop | [TK]D-Fender: even when it isn't |
| 04:23.11 | [TK]D-Fender | dlynes_laptop : using the "s" option? Never seen that fail before.... |
| 04:24.03 | [TK]D-Fender | dlynes_laptop : If you call it plain and the phone can accept another call (CW) the it'll tell you its "available". As well it should, since the outcome would be that the callee is ignoring th call. |
| 04:24.55 | dlynes_laptop | [TK]D-Fender: ${AVAILSTATUS} always returns AST_DEVICE_NOT_INUSE |
| 04:25.14 | dlynes_laptop | [TK]D-Fender: all my line appearances are set to calllimit=1 |
| 04:25.46 | dlynes_laptop | [TK]D-Fender: that availstatus should get set regardless of whether I use the 's' option, should it not? |
| 04:26.20 | dlynes_laptop | [TK]D-Fender: besides that, using chanisavail will just give me a large, overly complicated, unmaintainable dial plan for what I need to do |
| 04:26.29 | [TK]D-Fender | dlynes_laptop : "s" returns "not available" if the device hasany channel in a state other than "idle" |
| 04:27.19 | [TK]D-Fender | dlynes_laptop : Now THAT may well be true.... depends how many poeple you need to do this for. Mind you you could parse sometthing out quick in AGI at worst without making it a compiled app |
| 04:27.33 | dlynes_laptop | [TK]D-Fender: yeah, but I don't know agi |
| 04:27.36 | dlynes_laptop | [TK]D-Fender: and I do know c |
| 04:28.01 | dlynes_laptop | [TK]D-Fender: and also from what I hear, agi is quite unstable |
| 04:29.10 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-227.mtco.com) |
| 04:29.44 | [TK]D-Fender | dlynes_laptop : For the size of application you need it really shouldn't be hard at all, and for frequency and concurrency of execution should pose no threat to stability. |
| 04:30.10 | [TK]D-Fender | dlynes_laptop : You just need to pass a list of phones to validate, and return a modded string of actual devices to ring. |
| 04:30.56 | dlynes_laptop | [TK]D-Fender: each phone will typically have up to five lines, and there will typically be up to eight phones |
| 04:30.57 | [TK]D-Fender | dlynes_laptop : Sounds like a remarkably easy parsing job. You could do it with CUT's entirely in dialplan if you wanted without TOO much pain, but it would look a little bulky. |
| 04:31.26 | [TK]D-Fender | dlynes_laptop : You use 1 reg per line on a multi-line phone? |
| 04:31.34 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 04:31.51 | ManxPower | dlynes_laptop: I believe calllimit messes up ChanIsAvail |
| 04:31.55 | dlynes_laptop | [TK]D-Fender: one sip account per line appearance, yes |
| 04:32.11 | ManxPower | dlynes_laptop: That is what I do too. Makes things SO much easier |
| 04:32.30 | dlynes_laptop | [TK]D-Fender: I was using registrations, but I'm getting rid of those now that I find sip registrations tend to make the phones less reliable |
| 04:32.58 | dlynes_laptop | if a registration is a little late coming, asterisk detects the phone as being out of service |
| 04:33.11 | ManxPower | dlynes_laptop: turn off qualify= That should fix that |
| 04:33.12 | [TK]D-Fender | dlynes_laptop : That is just....ULGY. I'd offer you some crack, but I think you've overdone it already :) |
| 04:33.13 | dlynes_laptop | if the phone boots up and its buggy firmware forgets to register, it's deemed out of service |
| 04:33.46 | [TK]D-Fender | dlynes_laptop : Sounds like you need to pick better phones... |
| 04:35.28 | ManxPower | ...all 100+ of them. We make one big happy VoIP family! |
| 04:36.40 | wunderkin | ill be happy with polycom once my problem gets fixed :/ |
| 04:38.07 | dlynes_laptop | qualify is turned off, for what it's worth |
| 04:38.08 | [TK]D-Fender | wunderkin : Oh yeah... I told yo, RMA the damned phone :) |
| 04:38.49 | ManxPower | As far as I can tell the answer to most polycom issues is "Use SIP firmware 1.6.7" |
| 04:39.11 | wunderkin | [TK]D-Fender, yeah... i submitted the logs... the reseller offered to exchange the phones... i have other ones i could swap with for the office... but who is to say that those don't have a problem either since we haven't been using the 'extras' :D |
| 04:39.31 | wunderkin | ManxPower, this problem seems to be hardware related |
| 04:40.18 | [TK]D-Fender | Well I run about 30 or so personally and have used a large number of firmwares and never had anything like you've mentioned. |
| 04:40.20 | ManxPower | The only time we have ever had Polycoms die is when some of them spent a couple of days wet from flooding. |
| 04:40.46 | TheCops | [TK]D-Fender hi! |
| 04:40.48 | wunderkin | i guess i should get off of my butt and swap them tomorrow blah but there is a problem that if they have an outgoing unanswered call and answer an incoming call it resets.. which has been hard to get them to STOP doing that so for every reboot, i have to check if that was the cause |
| 04:41.13 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 04:41.56 | ManxPower | wunderkin: what firmware rev? |
| 04:42.04 | wunderkin | 1.6.7, 2.0.1, 2.0.2 |
| 04:43.05 | wunderkin | i added that on my list of problems to polycom but hard telling if they can do anything about it |
| 04:44.04 | ManxPower | weird. |
| 04:44.21 | wunderkin | im not sure what people think should happen when they do that but.. meh :) |
| 04:44.36 | [TK]D-Fender | wunderkin : Its toast. Turn it in. |
| 04:44.51 | wunderkin | rofl.. yes dr fender... this other problem is global though |
| 04:44.54 | ManxPower | one would assume they would put the unanswered outgoing call on hold first |
| 04:45.02 | [TK]D-Fender | wunderkin : Or ask for some PB&J to go with it :) Everything in life is better witha little PB&J |
| 04:46.03 | ManxPower | I think it's time for bed. |
| 04:46.04 | wunderkin | i tought them that they should just press the line key or whatever and call and it would be put on hold.. but how can you put an unanswered call on hold? thinking about the internals, sip... |
| 04:46.23 | ManxPower | no idea |
| 04:46.26 | wunderkin | :D |
| 04:46.35 | ManxPower | press the hold button? |
| 04:46.39 | wunderkin | lol |
| 04:47.29 | wunderkin | i guess the phone could still process it, that would be weird though when the person picked up and hears hold music |
| 04:48.08 | *** part/#asterisk raetin (n=raetin@136.218.121.70.cfl.res.rr.com) |
| 04:48.09 | *** join/#asterisk Del_Mon (n=de_mon@fl-69-69-142-236.dyn.embarqhsd.net) |
| 04:48.10 | ManxPower | What would you rather have it do? |
| 04:48.23 | wunderkin | slap the user? |
| 04:50.15 | [TK]D-Fender | wunderkin : You can just pull another call and whatever one you're on does go on hold. |
| 04:50.43 | wunderkin | yes.. if it is answered |
| 04:52.16 | wunderkin | if the call is not answered, it resets.. does it not for you or have you not been in that situation? |
| 04:53.33 | [TK]D-Fender | wunderkin : You can't put an unanswered call on hold. Your phone has to answer it, and can then proceed to put it on hold. |
| 04:53.45 | wunderkin | exactly |
| 04:55.49 | wunderkin | the phone resets if the luser tries to answer another call and they already have an outgoing call that is not answered yet |
| 04:56.37 | wunderkin | solution: don't do that.. lol :P but they don't stop |
| 04:57.28 | dlynes_laptop | wunderkin: solution: when your phone makes a call, answer the call on asterisk, and then do a dial to make the outgoing call |
| 04:57.38 | dlynes_laptop | wunderkin: then you should be able to put it on hold while it's ringing |
| 04:57.50 | *** part/#asterisk quadrata (n=jhuntwor@ool-44c61466.dyn.optonline.net) |
| 04:58.15 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 04:58.24 | wunderkin | i was thinking about that |
| 04:59.28 | dlynes_laptop | yeah, but it's easier to complain about your customers |
| 04:59.28 | dlynes_laptop | I know :) |
| 04:59.50 | wunderkin | :-) |
| 05:04.49 | wunderkin | i should have done that earlier.. thanks |
| 05:06.22 | *** join/#asterisk g0tw00d (n=rchace@68-113-159-200.dhcp.nplt.ne.charter.com) |
| 05:15.41 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 05:23.37 | *** join/#asterisk parag_ast (n=Parag@dxb-b112143.alshamil.net.ae) |
| 05:24.37 | parag_ast | Good Morning All |
| 05:25.42 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 05:27.59 | Marshall16 | anyone got an iax2 account i can use? |
| 05:28.33 | rob0 | iaxtel.com :) |
| 05:28.49 | rob0 | I also use IAX with FWD. |
| 05:35.29 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 05:35.52 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 05:36.24 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 05:40.43 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:46.10 | *** join/#asterisk qwertz (n=qwertz@165.206.27.217.static.versanetonline.de) |
| 05:52.38 | *** join/#asterisk visba (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net) |
| 05:52.46 | *** part/#asterisk visba (n=dca[lapt@c-24-8-53-17.hsd1.co.comcast.net) |
| 05:55.03 | Grnd-Wire | hmm.. So has anyone made the BLF work before on Grandstream phones? |
| 05:58.18 | Grnd-Wire | hmm - Is it possible to turn up the speaker volume on the GXP-2000 ? It seems really quiet to me.. |
| 06:03.59 | *** join/#asterisk [B]abyCool16 (n=java@server.ivinskis.kursenai.lm.lt) |
| 06:03.59 | *** join/#asterisk MuRat (n=java@server.ivinskis.kursenai.lm.lt) |
| 06:05.00 | dlynes_laptop | Grnd-Wire: gxp-2000 has that problem too? |
| 06:05.07 | dlynes_laptop | Grnd-Wire: thought it was only the budgetones |
| 06:07.54 | *** join/#asterisk Del_Mon (n=de_mon@fl-69-69-138-2.dyn.embarqhsd.net) |
| 06:10.18 | *** join/#asterisk LoneShadow (n=duh@c-67-188-116-104.hsd1.ca.comcast.net) |
| 06:10.43 | Grnd-Wire | dlynes_laptop: oh, so you know about this? umm - I dunno - This is a trial for me, so I'm trying to learn about all the annoying problems it has.. :D |
| 06:11.48 | dlynes_laptop | I know nothign about the gxp2000 |
| 06:11.55 | dlynes_laptop | just the grandstream budgetone 102's |
| 06:12.37 | Grnd-Wire | So they have really low speakre volume in the handset eh? |
| 06:15.19 | Grnd-Wire | Tell me about that, cause I was hoping to try one of those as well - for the really cheap customers. :P |
| 06:15.23 | *** join/#asterisk asberger (n=nb@cpe-76-170-93-79.socal.res.rr.com) |
| 06:18.51 | *** join/#asterisk arley (n=arley@66-214-136-53.dhcp.gldl.ca.charter.com) |
| 06:31.49 | *** join/#asterisk gnu_linux_geek (n=mark@cpe-71-74-141-159.neo.res.rr.com) |
| 06:45.38 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:50.04 | *** join/#asterisk bulatitoy (n=rmn@adsl-70-231-146-111.dsl.snfc21.sbcglobal.net) |
| 06:51.06 | bulatitoy | anybody tried linksys wip300? |
| 07:04.38 | *** join/#asterisk AndyCap_ (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 07:06.15 | *** join/#asterisk Brumle (n=brumle@brumle.com) |
| 07:09.13 | *** join/#asterisk xnon (n=xnon@200.8.5.123) |
| 07:20.03 | *** join/#asterisk shinux__ (n=shinux@196.220.28.250) |
| 07:20.49 | *** join/#asterisk NeScHe`mUziK`diN (n=AceofBas@88.224.163.67) |
| 07:26.25 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:27.12 | *** join/#asterisk apardo (n=apardo@87.217.147.146) |
| 07:29.32 | *** join/#asterisk ang (n=ang@caracas-1765.adsl.interware.hu) |
| 07:37.37 | *** join/#asterisk apardo_ (n=apardo@87.217.145.95) |
| 07:41.54 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 07:42.09 | *** join/#asterisk olinuXP (n=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 07:42.40 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 07:43.50 | *** join/#asterisk apardo (n=apardo@87.217.145.95) |
| 07:45.45 | *** join/#asterisk apardo (n=apardo@87.217.145.95) |
| 07:50.21 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:50.35 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 07:58.06 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 08:01.13 | *** join/#asterisk antonrd (i=antonrd@sleepless.wan-dki.net) |
| 08:09.09 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 08:11.21 | *** join/#asterisk LoneShadow (n=duh@c-67-188-116-104.hsd1.ca.comcast.net) |
| 08:16.06 | *** join/#asterisk [hC] (n=hardcore@70.68.154.154) |
| 08:19.41 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 08:23.29 | *** join/#asterisk danp (i=danp@elmer.glueless.net) |
| 08:26.36 | *** join/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it) |
| 08:27.40 | *** join/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it) |
| 08:28.42 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 08:30.49 | *** part/#asterisk gnu_linux_geek (n=mark@cpe-71-74-141-159.neo.res.rr.com) |
| 08:38.46 | DrCron | was 1.4 a total recode of asterisk? |
| 08:41.07 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 08:41.54 | mosty | i don't know, but no |
| 08:43.25 | *** join/#asterisk KermitTheFragger (n=ktf@118-197.bbned.dsl.internl.net) |
| 08:49.03 | hads | DrCron: No |
| 08:49.33 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 08:53.31 | DrCron | is there a way to pass an sip uri over iax? |
| 08:53.56 | mosty | drcron: what do you mean exactly? |
| 08:54.59 | DrCron | I'm trying to set up my asterisk config so that if, for example i entered SIP:music@vostrom.com in an iax client and dialed, i would connect to that sip address |
| 08:55.20 | DrCron | right now, asterisk interprets the @ to mean context |
| 08:55.41 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 08:55.53 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:55.54 | DrCron | so it tries to find extension music in context vostrom.com, which doesnt exist |
| 08:56.48 | mosty | drcron: why don't you use a phone that supports sip then? |
| 08:57.48 | monsted | mosty: because he wants IAX phones, maybe? |
| 08:58.03 | DrCron | or i want all my calls to go through my asterisk box |
| 08:58.15 | DrCron | i could use sip, but would rather not |
| 08:58.28 | mosty | use a phone that supports both, i mean |
| 08:59.13 | DrCron | thats an option, and i can do it that way if nothing else works, but that seems, well, a bit stupid |
| 09:00.00 | mosty | it seems logical to me, use iax for iax and use sip for sip |
| 09:00.08 | DrCron | i thought '?' was suposed to indicate context, and beyond that, that clients shouldnt have the ability to request context that overides the one set in iax.conf |
| 09:08.40 | *** join/#asterisk olinux (i=olinux@ip68-107-4-202.sd.sd.cox.net) |
| 09:09.14 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.89) |
| 09:09.20 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:11.31 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:11.40 | *** join/#asterisk banan_ (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 09:15.44 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:15.52 | FuriousGeorge | anyone know where to get DID from Brazil? |
| 09:16.11 | FuriousGeorge | local telcos cant issue international DID right? |
| 09:16.22 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:17.05 | mosty | FuriousGeorge: they can |
| 09:20.33 | FuriousGeorge | mosty: is this something you need to setup wholesale? I'm looking for S. American DID |
| 09:20.44 | FuriousGeorge | maybe some non-US N American |
| 09:22.49 | mosty | no, you can buy single DID's |
| 09:23.09 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 09:23.31 | FuriousGeorge | Ive found a few consumer level services. Im looking for something I can interface with * obviously :) |
| 09:24.59 | FuriousGeorge | https://secure.vozbrasil.com/vozbrasil/ |
| 09:27.36 | *** join/#asterisk Hello2007 (n=wardeh20@mail.splendor.net) |
| 09:28.12 | Hello2007 | how can i restart asterisk every midnight? |
| 09:28.58 | mosty | use a cron job |
| 09:29.10 | Hello2007 | what do i have to write in it? |
| 09:29.21 | mosty | man 5 crontab |
| 09:29.38 | Hello2007 | hehehe |
| 09:29.50 | Hello2007 | well i m trying but its not working |
| 09:29.51 | FuriousGeorge | Hello2007: there is a good example on the WIKI |
| 09:30.15 | Hello2007 | i try it |
| 09:30.22 | Hello2007 | does it work? |
| 09:31.43 | pif | you guys use mozphone ? |
| 09:32.36 | FuriousGeorge | Hello2007: im gonna go out on a limb and say it probably does what the article says it will |
| 09:32.56 | FuriousGeorge | its just a matter of asterisk -rX "restart now" |
| 09:33.27 | *** join/#asterisk emanus (n=emanuele@adsl-241-27.38-151.net24.it) |
| 09:34.38 | FuriousGeorge | i actually restart my snom sip phones right after *, too |
| 09:36.05 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-44-36.w90-5.abo.wanadoo.fr) |
| 09:36.37 | Hello2007 | */3 * * * * /usr/sbin/asterisk -rx "stop now" |
| 09:36.55 | Hello2007 | is this correct? |
| 09:37.04 | Hello2007 | for stopping it |
| 09:37.43 | *** join/#asterisk apardo (n=apardo@87.217.145.95) |
| 09:42.50 | FuriousGeorge | Hello2007: just stick an sh script in cron.daily |
| 09:43.10 | FuriousGeorge | my default cron settings do everything in daily at 3am |
| 09:43.12 | DrCron | yhea, but stop gracefully might be a better idea |
| 09:44.08 | Hello2007 | no problem ,it worked |
| 09:44.25 | Hello2007 | i m just testing right now |
| 09:44.42 | DrCron | gracefully waits for chans to close iirc, and now, well, its more of a sledgehammer |
| 09:46.56 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 09:47.24 | *** join/#asterisk _toggy (n=scound@host-81-191-169-198.bluecom.no) |
| 09:47.42 | *** join/#asterisk Mannetjie (n=Mannetji@vc-196-207-32-235.3g.vodacom.co.za) |
| 09:47.47 | _toggy | Anyone got the 1.2.13-devel pack (sourcecodes)? |
| 09:47.54 | FuriousGeorge | DrCron: true, but Ive notived sometimes when * misbehaves a channel will remain open permanently |
| 09:48.23 | FuriousGeorge | so it wouldnt reboot at all, in my script i do gracefully first then now |
| 09:48.27 | *** join/#asterisk henrique (n=henrique@200-161-80-249.dsl.telesp.net.br) |
| 09:49.20 | _toggy | anyone got em? |
| 09:49.36 | Mannetjie | hi all |
| 09:50.12 | Mannetjie | I have a beginners question |
| 09:50.18 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 09:50.54 | FuriousGeorge | tell the beginner to ask for himself |
| 09:50.57 | FuriousGeorge | we wont bite |
| 09:51.13 | Mannetjie | heheh, thanks FuriousGeorge |
| 09:51.26 | pif | Mannetjie : RTFM! |
| 09:51.42 | pif | just kidding |
| 09:52.02 | Mannetjie | if I have a 4 port analog card and 1 line from my telecom provider (analog) |
| 09:52.26 | Mannetjie | can I plug the line into 1 of the ports and use the 3 remaining ones for analog extentions? |
| 09:52.58 | Mannetjie | heheheh pif, I am so busy reading at this stage it's just not true, hehehehe |
| 09:53.20 | FuriousGeorge | Mannetjie: sure you can, thats what those cards are for |
| 09:53.35 | mosty | mannetjie: depends what modules you have in that analogue card |
| 09:53.41 | pif | furious, depends if the port is fxo/fxs |
| 09:54.02 | pif | fxo == telco , fxs == handset |
| 09:54.03 | mosty | mannetjie: a phone port is different to a line port |
| 09:54.08 | FuriousGeorge | i guess i shouldnt assume he knows that :) |
| 09:54.10 | FuriousGeorge | ~fxofxs |
| 09:54.14 | jbot | well, 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. |
| 09:54.29 | Mannetjie | will it be possible with the Digium TDM10B card? |
| 09:54.53 | _toggy | anyone got a link for the 1.2.13-devel pack? |
| 09:54.55 | mosty | mannetjie: you just have to get the right modules for it |
| 09:54.57 | Mannetjie | aaahhh I was wondering what the fxo, fxs stand for |
| 09:55.12 | FuriousGeorge | Mannetjie: here is what you need to understand |
| 09:55.35 | FuriousGeorge | TDM10B means a TDM400P card with 1 FXS and 0 FXO, iirc |
| 09:55.43 | FuriousGeorge | (revision b) |
| 09:55.56 | FuriousGeorge | so you want a TDM31B |
| 09:55.56 | Mannetjie | aaahhhh I see |
| 09:55.57 | DrCron | so you can add fxs modules |
| 09:55.59 | FuriousGeorge | which is more |
| 09:56.11 | hads | FuriousGeorge: Corrext, except B os for bundle |
| 09:56.19 | FuriousGeorge | ah |
| 09:56.21 | parag_ast | Anybody is having idea for providing QoS for IAX2 protocol |
| 09:56.23 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 09:56.23 | DrCron | do you already own a TDM10B? |
| 09:56.23 | hads | Hmm. I can't type tonite. |
| 09:56.47 | FuriousGeorge | hads: i believe you anyway, though I think there are two versions of the TDM400P card |
| 09:56.53 | Mannetjie | DrCron, no not yet, I just want to find out what would be the best to do |
| 09:57.07 | hads | FuriousGeorge: They are up to about Rev J or something. |
| 09:57.14 | Mannetjie | This is what I want to do : |
| 09:57.26 | FuriousGeorge | Mannetjie: but yes, you can purchace the TDM400P and FX/O or /S modules separately |
| 09:57.42 | FuriousGeorge | hads: i revise: there were two the last time I bought one :) |
| 09:57.46 | *** join/#asterisk tmyneii (n=localhos@161.53.107.100) |
| 09:57.58 | hads | :) |
| 09:58.13 | Mannetjie | I want to input a specific analog extention into asterisk |
| 09:58.49 | Mannetjie | then when a call is put through to the extention asterisk must forward this to a VOIP extention |
| 09:58.54 | Mannetjie | is that possible? |
| 09:59.05 | mosty | yes |
| 09:59.09 | FuriousGeorge | Mannetjie: yeah, thats what a PBX does |
| 09:59.26 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:59.27 | Mannetjie | I figured that, |
| 09:59.28 | FuriousGeorge | Mannetjie: read up on the DIALPLAN, its like a simple scripting language to route calls |
| 09:59.29 | Mannetjie | :) |
| 09:59.31 | DrCron | thats more or less what asterisk is designed to do |
| 09:59.34 | FuriousGeorge | ~dialplan |
| 09:59.37 | jbot | well, dialplan is the thing configured in extensions.conf |
| 09:59.57 | Mannetjie | FuriousGeorge, so I can buy the card and then just add fxs modules to it, great :) |
| 10:00.25 | DrCron | if you want each phone to have its own extension, ip phones are much cheaper then adding fxs modules |
| 10:00.48 | FuriousGeorge | Mannetjie: and while its not recommended, if you are just fooling around you can even run a few different phones (on the same "extension") off of 1 PBX |
| 10:00.51 | FuriousGeorge | er |
| 10:00.52 | FuriousGeorge | FXS |
| 10:01.00 | Mannetjie | I am looking into ip phones at this stage DrCron |
| 10:01.05 | shellshark | http://pastebin.ca/275213 |
| 10:01.06 | FuriousGeorge | ~s/PBX/FXS |
| 10:01.24 | FuriousGeorge | shellshark: hi |
| 10:01.39 | shellshark | does anyone have any ideas why calls never go out prov3? |
| 10:03.37 | shellshark | FuriousGeorge: heya |
| 10:06.11 | DrCron | so, if i understand, iax phones cant place calls into asterisk if they use the uname@server addressing scheme, because asterisk thinks the @is a request for a diffrent context, and that cant be changed |
| 10:09.38 | FuriousGeorge | shellshark: because 1 and 2 always work? |
| 10:10.14 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 10:10.22 | FuriousGeorge | DrCron: what IAX phone doesnt work with asterisk? |
| 10:10.30 | FuriousGeorge | dont you think thats kinda silly? |
| 10:11.18 | DrCron | it works fine for numeric dialing, but if you want to dial an address with an @ asterisk gets a bit grumpy |
| 10:11.37 | *** join/#asterisk CleanerX (n=nix@p54A39C4C.dip0.t-ipconnect.de) |
| 10:12.49 | DrCron | ie: it cant handle extensions with an @ sign in them |
| 10:13.43 | shellshark | FuriousGeorge: i figured it out... prov3 is SIP, not IAX2... would explain that ;) |
| 10:14.14 | shellshark | an extension should not have an @ in it |
| 10:14.24 | shellshark | a URI should though |
| 10:14.25 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:15.07 | DrCron | yhea, but asterisk sees everything i put into this phone as a extension, not as a uri |
| 10:15.25 | DrCron | well, actually this softphone, and the 2 others i tried |
| 10:15.29 | FuriousGeorge | DrCron: asterisk is supposed to be between your client and whatever you are dialing |
| 10:15.50 | zoa | what softphone did you try ? |
| 10:15.58 | zoa | idefisk should be able to handle it i think |
| 10:16.06 | shellshark | so it's trying to dial as "IAX2/iax2://guest@server/extension" ? |
| 10:16.39 | *** join/#asterisk syn (i=syn@kenobi.sceen.net) |
| 10:16.41 | syn | hello |
| 10:16.51 | shellshark | heya |
| 10:17.27 | syn | is it possible to limit the number of simultaneous calls to a queue agent to 1 ? |
| 10:17.45 | syn | without using something like call-limit in sip.conf which makes transfers impossible since a second call must be done |
| 10:19.44 | syn | Lady <= ad bot |
| 10:19.55 | DrCron | i want to be able to dial either SIP:user@server << ie sip uri, or IAX2:user@server <<iax2 uri |
| 10:19.59 | syn | LARAx17[f] <= same |
| 10:22.34 | qwertz | Hi, is it possible to dial more than one sip phone in a single dial command (like Dial(SIP/100&SIP/101))? |
| 10:23.17 | syn | qwertz: yes |
| 10:23.33 | syn | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 10:25.49 | mosty | syn: to do that, i think you need to use call-limit on the account, but use more than one account on the phone |
| 10:27.09 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 10:28.08 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 10:28.17 | syn | mosty: eww |
| 10:28.40 | mosty | syn: i agree :( |
| 10:28.50 | syn | our other option is to hack queue.c directly ... |
| 10:29.04 | syn | i don't know which would be fastest |
| 10:29.13 | syn | or faster, at least |
| 10:29.22 | mosty | it would certainly be quicker to get multiple accounts working |
| 10:29.36 | syn | i don't know |
| 10:29.37 | mosty | but be my guest if you want to hack queue.c - i may even help you |
| 10:29.40 | syn | this relies on phone configuration |
| 10:29.52 | syn | we have several hunderds of phones :/ |
| 10:30.17 | qwertz | syn, thanks had a && instead of a & |
| 10:30.25 | mosty | print out instructions, and get the operators to do the config |
| 10:30.35 | syn | mosty: eh :) |
| 10:30.51 | DrCron | so, um, no help on getting uri dialing working? |
| 10:31.08 | syn | mosty: i'm not sure :/ |
| 10:31.15 | syn | i don't trust they do things correctly |
| 10:31.18 | syn | a dirty hack will do |
| 10:31.20 | mosty | app_queue does suck, i just don't have the time to figure out how it works and how to fix it all by myself |
| 10:31.49 | syn | is it improved in 1.4 ? |
| 10:31.55 | mosty | i haven't looked |
| 10:31.58 | syn | ok |
| 10:32.02 | syn | thanks for your answers |
| 10:32.49 | mosty | i just thought of another possibility- you could write an agi script to do it. it wouldn't be as efficient, but you could get it up and running soon enough |
| 10:33.12 | syn | i fear it'd be too slow |
| 10:33.54 | syn | anyway, it won't be the first dirty hack i'm doing, i'm getting used to it :) |
| 10:34.17 | mosty | how many incoming calls per minute do you have? |
| 10:34.36 | syn | depends :)) |
| 10:34.55 | mosty | how many are you planning on as a maximum? |
| 10:35.12 | syn | per minute hmm |
| 10:35.18 | syn | i'd say 50 |
| 10:35.28 | mosty | that is quite a lot |
| 10:35.31 | syn | yes |
| 10:35.45 | syn | almost one per second |
| 10:35.58 | syn | yes, that's the maximum |
| 10:36.05 | syn | of course, usually it's much lower |
| 10:36.17 | mosty | if you could find a nice language which doesn't have much startup time it could work though |
| 10:36.32 | syn | that's the hard part |
| 10:36.40 | mosty | though you would need some sort of efficient persistent storage |
| 10:36.43 | syn | the platform is embeded |
| 10:37.11 | syn | no python or perl |
| 10:37.30 | syn | anyway, i just wanted to know if there was already something existing |
| 10:37.31 | mosty | well i'd say you're back to C |
| 10:37.35 | syn | yes |
| 10:38.14 | syn | let's code then |
| 10:38.17 | syn | have a nice day :) |
| 10:38.24 | *** part/#asterisk syn (i=syn@kenobi.sceen.net) |
| 10:38.36 | oink | let's write portable code |
| 10:48.28 | _toggy | anyone have a link for the asterisk-devel 1.2.13 pack? |
| 10:49.20 | *** join/#asterisk Osochebol (n=Osochebo@210.245.33.1) |
| 10:50.58 | Mannetjie | ok peeps I have to run, thanks for the help!! |
| 10:51.10 | *** join/#asterisk imyousuf (n=IceChat7@203.208.196.140) |
| 10:53.31 | *** part/#asterisk imyousuf (n=IceChat7@203.208.196.140) |
| 10:57.17 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 10:58.55 | mendol | when i got "Dial failed due to trunk reporting BUSY - giving up" it means i did smth wrong with my configuration, or it may be my voip provider problems? |
| 11:00.33 | mosty | how many simultaneous calls does your provider allow? and how many are you using? |
| 11:01.03 | mendol | it allowes unlimited atm, and im testing 1 sip trunk |
| 11:01.56 | mosty | so the first one fails? |
| 11:02.10 | mendol | yes |
| 11:03.33 | mosty | can you test by registering a sip phone to the service without asterisk? |
| 11:03.54 | mendol | yep |
| 11:04.06 | mendol | got linksys pap2t |
| 11:04.18 | in-pt | Hi..can anyone tell me how to solve the "too many open files" error |
| 11:04.31 | DrCron | what os? |
| 11:04.45 | in-pt | fedora core 3 |
| 11:06.21 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 11:06.24 | KermitTheFragger | in-pt: what does /etc/security/limits.conf say ? |
| 11:06.59 | in-pt | * - nofile 2048 |
| 11:07.19 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 11:07.20 | in-pt | which i had changed from 1024 to 2048..but still have problem |
| 11:07.37 | KermitTheFragger | what does running ulimit on the command line say ? |
| 11:07.56 | KermitTheFragger | as the user which gets the error ofcourse |
| 11:07.56 | in-pt | ulimit -n says 2048 |
| 11:08.07 | in-pt | and i have 20 sip phones only |
| 11:09.34 | zoa | change it to 65534 |
| 11:09.47 | zoa | but make sure its changed for the same threa too |
| 11:09.58 | zoa | so check it from the CLI |
| 11:10.46 | mendol | mosty: it works with pap2t :-/ but with my asterisk i got busy signal with outgoing calls |
| 11:11.04 | in-pt | zoa: i didnt understands "threa too" |
| 11:11.20 | KermitTheFragger | im reading somewhere "For example, if the limit is set at 1024 (a common default value) Asterisk can handle approxiately 150 SIP calls simultaneously." |
| 11:12.17 | *** join/#asterisk MatsK (n=22cd8bdb@gw-sthlm01.rebtel.com) |
| 11:12.55 | in-pt | KermitTheFragger: i also headr that therefore after inceasing upto 2048 i dont think that it would solve this..but needs to be done something else |
| 11:14.00 | mosty | mendol: what does the asterisk console say? have you registered asterisk succesfully? |
| 11:14.07 | mendol | yes |
| 11:14.24 | mendol | tho using default configuration it doesnt work :-/ |
| 11:14.39 | mosty | what does the console say? |
| 11:14.47 | KermitTheFragger | in-pt: which version are you using, btw ? |
| 11:15.07 | in-pt | asterisk-1.2.4 |
| 11:15.46 | KermitTheFragger | and ulimit -a says nothing strange or low ? |
| 11:16.51 | KermitTheFragger | you are running the ulimit -a as the user that runs asterisk, right ? |
| 11:16.59 | in-pt | core file size is 0 and rest are unlimited values and some are 2048 |
| 11:17.09 | in-pt | what is core file size ? |
| 11:17.42 | KermitTheFragger | for core dumps |
| 11:17.46 | KermitTheFragger | iirc |
| 11:18.05 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 11:18.06 | KermitTheFragger | so its the max size of a core dump iirc |
| 11:18.08 | *** join/#asterisk emiquelito (n=emiqueli@200-155-185-1.static.spo.ifx.net.br) |
| 11:18.22 | KermitTheFragger | shouldnt be a problem |
| 11:18.41 | in-pt | ok |
| 11:18.44 | KermitTheFragger | are you running SE-Linux ? |
| 11:18.50 | in-pt | no |
| 11:18.57 | KermitTheFragger | hmm |
| 11:19.13 | in-pt | but running iptables |
| 11:19.34 | KermitTheFragger | iptables does not enforce limits on open files |
| 11:19.46 | mendol | too many things i see in this debug, it register but i got errors in both inc/out connections. |
| 11:20.05 | in-pt | ok |
| 11:20.53 | *** join/#asterisk MatsK (n=22cd8bdb@gw-sthlm01.rebtel.com) |
| 11:20.54 | mendol | i got "call established" then "hangup" when i try to make outgoin call |
| 11:22.01 | mendol | <PROTECTED> |
| 11:22.02 | mendol | <PROTECTED> |
| 11:22.05 | KermitTheFragger | in-pt: i need to run to a meeting, be back in about 2 hours |
| 11:22.35 | in-pt | so what you can suggest me about it |
| 11:22.57 | in-pt | ok its better we will talk later |
| 11:23.08 | *** join/#asterisk apardo_ (n=apardo@87.217.145.95) |
| 11:23.48 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 11:24.44 | mendol | <PROTECTED> |
| 11:28.42 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-44-36.w90-5.abo.wanadoo.fr) |
| 11:29.32 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 11:29.35 | Chris-NB | hi |
| 11:29.54 | Chris-NB | anyone using asterisk for sending faxes? |
| 11:30.50 | *** join/#asterisk fourcheeze (n=rich@office.callmaster.co.uk) |
| 11:32.10 | dwmw2 | Chris-NB: not recently. I've used OpenPBX more recently for it. |
| 11:33.40 | Chris-NB | dwmw2, and what have u used for sending faxes? |
| 11:33.46 | Chris-NB | dwmw2, which app |
| 11:34.11 | dwmw2 | app_txfax |
| 11:34.24 | dwmw2 | which is part of the standard build from SVN, or the Fedora packages |
| 11:35.00 | dlynes_laptop | Chris-NB: I think it's borked in anything newer than asterisk 1.2.9.1 |
| 11:35.08 | *** join/#asterisk imyousuf (n=IceChat7@203.208.196.140) |
| 11:36.32 | dlynes_laptop | Chris-NB: basically you want the latest spandsp-0.0.2prexx.tar.gz file |
| 11:36.46 | dlynes_laptop | Chris-NB: You can download it from www.soft-switch.org |
| 11:37.03 | dlynes_laptop | Chris-NB: you'll also need to grab the stuff from the asterisk subdirectory |
| 11:37.10 | dlynes_laptop | Chris-NB: donm't bother with the testing directory though |
| 11:38.34 | *** join/#asterisk xain (n=aadilism@2-237-154-202.wol.net.pk) |
| 11:38.51 | xain | hi |
| 11:40.26 | Chris-NB | dlynes_laptop, so you recomend the use of spandsp? |
| 11:40.47 | dlynes_laptop | Well, like i said |
| 11:40.58 | dlynes_laptop | you can't use anything newer than asterisk 1.2.9.1 from what I understand |
| 11:41.02 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 11:41.11 | dlynes_laptop | but I'm successfully using it for receiving faxes at the moment |
| 11:41.18 | dlynes_laptop | I'm having issues trying to send faxes |
| 11:41.21 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:41.28 | dlynes_laptop | Haven't figured out what my problem is yet, though |
| 11:42.07 | dlynes_laptop | but the receiving is working flawlessly |
| 11:43.29 | dlynes_laptop | but, otoh |
| 11:43.48 | dlynes_laptop | I don't see the need for being able to send faxes over voip for most people |
| 11:46.14 | DrCron | well, that or another way to send faxes without a fax machine |
| 11:46.56 | Nivex | remote printers over a VPN. It amazed me how many faxes Red Hat sent to itself despite having this configuration. |
| 11:47.04 | DrCron | meh |
| 11:47.17 | dlynes_laptop | we have a need for it, but I would imagine most offices don't |
| 11:47.18 | DrCron | send it to a 3rd party |
| 11:47.27 | DrCron | fax is the main reason for that |
| 11:47.35 | Nivex | PDF + email? |
| 11:47.46 | monsted | fax needs to die :) |
| 11:47.48 | dlynes_laptop | Isn't that for receiving faxes? |
| 11:47.54 | Nivex | monsted: amen brother! |
| 11:48.09 | dlynes_laptop | 99% of the people I send pdf attachments to have adobe acrobat reader |
| 11:48.11 | monsted | dlynes_laptop: instead of printing something and faxing it, send it directly to the recipient by email :) |
| 11:48.16 | dlynes_laptop | if they don't, I tell them to install it |
| 11:48.37 | dlynes_laptop | monsted: Yeah...I'm more looking at faxing for fax blasting |
| 11:48.44 | DrCron | iirc, fax is closer to a legal doc iirc |
| 11:48.59 | dlynes_laptop | monsted: much easier to find local fax numbers than to find local email addresses |
| 11:49.23 | Chris-NB | dlynes_laptop, I don't want to send faxes over IP. I'll send them via ISDN, but without a Fax machine |
| 11:49.56 | dlynes_laptop | Chris-NB: you're going to be sending a lot? |
| 11:50.09 | dlynes_laptop | Chris-NB: and you want the 'fax machine' available as a windows 'printer'? |
| 11:50.10 | DrCron | for pdf's to be legaly binding you need certificates, and signing, and auth... ugly |
| 11:50.33 | Chris-NB | dlynes_laptop, I don't know. I've only check and test it. |
| 11:51.05 | DrCron | printer -> fax directly is dificult |
| 11:51.22 | dlynes_laptop | Chris-NB: if you wnat to send a lot of faxes and you want to use it as a windows printer (a la winfax), take a look at hylafax (www.hylafax.org) and the iaxmodem application that comes with spandsp |
| 11:51.27 | DrCron | print to queue, then assign fax number is easier iirc |
| 11:51.54 | *** part/#asterisk imyousuf (n=IceChat7@203.208.196.140) |
| 11:51.56 | Chris-NB | dlynes_laptop, k, I'll look at that |
| 11:53.49 | dlynes_laptop | Chris-NB: I forgot iaxmodem is a separate project: sf.net/projects/iaxmodme |
| 11:53.51 | dlynes_laptop | Chris-NB: I forgot iaxmodem is a separate project: sf.net/projects/iaxmodem |
| 11:54.01 | Chris-NB | dlynes_laptop, ok, thanks |
| 11:54.12 | Chris-NB | dlynes_laptop, I'll check it |
| 11:55.02 | DrCron | the hard bit is setting the number to send to |
| 11:56.02 | DrCron | at least, if you want to do it in the print window |
| 11:56.27 | dlynes_laptop | DrCron: have you used hylafax? |
| 11:57.42 | monsted | DrCron: hyla takes care of all of that |
| 11:57.59 | DrCron | ah, there is a windows client |
| 11:58.01 | dlynes_laptop | monsted: that's why i suspected he's never actually used hylaxfax |
| 11:58.02 | DrCron | nm then |
| 11:58.05 | dlynes_laptop | erm hylafax |
| 11:58.12 | DrCron | never had a need |
| 11:58.15 | *** join/#asterisk Skarmeth (n=Skarmeth@201009104156.user.veloxzone.com.br) |
| 11:58.24 | mosty | i used hylafax years ago, it was great |
| 11:59.08 | dlynes_laptop | I used it many many many years ago |
| 11:59.15 | dlynes_laptop | adn even then it had an option for a phone number |
| 11:59.24 | dlynes_laptop | but it was a bit of a pain to set up then |
| 11:59.30 | dlynes_laptop | I would imagine now it's quite easy to set up |
| 12:00.24 | mosty | it's easy in debian, was back then too |
| 12:00.27 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 12:00.34 | DrCron | hmm, looking like i'm gonna have to actually get a scaner now |
| 12:00.38 | dlynes_laptop | mosty: i'm talking about 8 or 9 years ago |
| 12:00.56 | mosty | i'm talking about 7 or 8 years ago |
| 12:01.01 | dlynes_laptop | mosty: back before 57.6K modems |
| 12:01.02 | DrCron | wanted to replace the crappy fax machine we have |
| 12:01.29 | mosty | 56k modems didn't really help with faxing |
| 12:01.41 | monsted | ISDN did |
| 12:02.03 | dlynes_laptop | We were just using a 14.4k sportster |
| 12:02.04 | mosty | isdn didn't help in this country, the only telco providing it priced themselves out of the market |
| 12:02.11 | monsted | you could even convince some ISDN faxes to do 64kbps faxing |
| 12:02.24 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-44-36.w90-5.abo.wanadoo.fr) |
| 12:02.29 | dlynes_laptop | mosty: that's what happened here, too |
| 12:02.44 | DrCron | and in most of the usa |
| 12:02.51 | dlynes_laptop | mosty: the telcos wanted to promote adsl instead |
| 12:03.11 | DrCron | yup |
| 12:03.45 | mosty | adsl wasn't implemented here for quite a while, our only affordable choices were 28/33/56k dialup or cable if you lived in a major city |
| 12:03.58 | monsted | well, ISDN and ADSL aren't really competitors :) |
| 12:04.06 | mosty | and cable gave you 100M a month, you could use that all in 5 minutes |
| 12:04.59 | DrCron | well, adsl was a fraction of the price |
| 12:05.00 | dlynes_laptop | Well, i think the only reason telcos decided to bring adsl out of the closet (adsl's been around since the late 70's/early 80's), was because they were losing market share to the cable providers selling broadband internet |
| 12:05.11 | DrCron | still is a fraction of the price |
| 12:05.41 | dlynes_laptop | before dsl and cable broadband came out, we were using a dual-gang isdn |
| 12:05.51 | dlynes_laptop | and that was expensive as hell |
| 12:06.25 | DrCron | yup |
| 12:06.42 | DrCron | you would think that isdn prices would come down |
| 12:07.13 | monsted | ISDN is quite a lot more expensive than POTS, mostly due to scale |
| 12:07.41 | DrCron | more then adsl as well? |
| 12:07.45 | monsted | no |
| 12:07.54 | dlynes_laptop | monsted: well, that's kinda obvious |
| 12:08.04 | dlynes_laptop | monsted: isdn is affordable in europe, but not in north america |
| 12:08.29 | qwertz | Hi, does anybody know if the current stable bristuff * is already patched with the pickup function in app_devstate.c - at least I can't apply the patch. |
| 12:09.17 | monsted | we pay DKK 159 for ISDN2 and DKK 129 for POTS |
| 12:09.50 | monsted | not too bad |
| 12:11.05 | dlynes_laptop | How much is DKK 129 in euros? |
| 12:11.52 | DrCron | hmm, got to find a faxmodem for my server |
| 12:12.08 | *** join/#asterisk Osochebol (n=Osochebo@58.186.23.89) |
| 12:12.14 | dlynes_laptop | DrCron: you don't have a zaptel-enabled asterisk server? |
| 12:13.13 | DrCron | nope |
| 12:13.21 | DrCron | running ip only |
| 12:15.25 | DrCron | so fax over ip would be nice |
| 12:16.16 | DrCron | unless someone out there has a spare zaptel card |
| 12:17.03 | DrCron | of course, that would mean i have to get another server up, running linux... ick |
| 12:19.53 | monsted | dlynes_laptop: 16 euros |
| 12:20.05 | dlynes_laptop | 16 euros? |
| 12:20.09 | dlynes_laptop | are you kidding? |
| 12:20.27 | monsted | wait, make that 21 :) |
| 12:20.29 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 12:20.31 | dlynes_laptop | even 21 |
| 12:20.37 | dlynes_laptop | that's still dirt cheap |
| 12:20.56 | dlynes_laptop | We pay minimum $35 Cdn |
| 12:20.59 | monsted | note that it *doesn't* include free local calls |
| 12:21.05 | dlynes_laptop | oh yeah |
| 12:21.20 | dlynes_laptop | forgot about that difference between europe and north america |
| 12:21.50 | monsted | (but you can buy that for 10 euro extra) |
| 12:22.04 | dlynes_laptop | ah...so iow |
| 12:22.24 | dlynes_laptop | there's no point in even getting pots where you are, if you need two lines |
| 12:27.31 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 12:27.32 | key2 | hi |
| 12:27.44 | key2 | anyone aware of a opensource TURN server ? |
| 12:30.34 | DrCron | does asterisk 1.4.x handle the username@server any diffrently then 1.2.9? |
| 12:30.59 | DrCron | ie: would upgrading help with URI handling? |
| 12:31.35 | _toggy | anyone been abel to get an Eicon 4bri isdn card working with trixbox? |
| 12:32.03 | _toggy | <PROTECTED> |
| 12:32.44 | *** join/#asterisk henrique (n=henrique@200-161-80-249.dsl.telesp.net.br) |
| 12:36.53 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:37.08 | *** join/#asterisk sergee (i=opera@195.94.224.197) |
| 12:39.03 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:40.53 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 12:46.04 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@8.10.2.50) |
| 12:47.10 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 12:47.52 | *** part/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 12:51.00 | *** join/#asterisk lorinc (n=ang@caracas-4570.adsl.interware.hu) |
| 12:51.54 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 12:58.01 | *** join/#asterisk carlus (n=carlus@201-1-72-212.dsl.telesp.net.br) |
| 13:00.11 | qwertz | Is it only possible to use the cutting of variables with EXTEN (EXTEN:5)? |
| 13:00.38 | cy3o3 | sup ya'llz |
| 13:00.48 | nibbler_de | qwertz: yup - there was some sort of application for that but it's deprecated |
| 13:00.55 | cy3o3 | anyone got an iaxtel number to do some quick testing? |
| 13:03.34 | qwertz | nibbler_de, thanks for the info, I'm trying to get the SIP account id of a hidden caller id to log into a queue - do you know another way to get the sip id? |
| 13:04.14 | queuetwo | Will installing digium's g.729 codec improve IAX performance? |
| 13:05.10 | *** join/#asterisk AuPix (n=root@mail.aupix.com) |
| 13:05.17 | Nivex | eek! it's root! |
| 13:06.05 | zoa | no it wont |
| 13:06.36 | DrCron | i know some people have gotten uri dialing working, but is that from sip phones only? |
| 13:07.03 | *** join/#asterisk Osochebol (n=Osochebo@58.186.23.89) |
| 13:08.43 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:12.02 | MrChimpy | grrr! dial limits going funny for me on 1.2.13! |
| 13:12.43 | *** join/#asterisk Shini- (n=e-MaSTeR@uni31-2-82-228-91-114.fbx.proxad.net) |
| 13:12.47 | MrChimpy | -- Executing Dial("Zap/1-1", "Zap/g1/phonenumber||L(60000:30000:5000)") in new stack |
| 13:13.07 | *** join/#asterisk Slabber (n=simon@194.7.163.103) |
| 13:13.11 | MrChimpy | it picks up 60000 as timelimit, 30000 as play_warning |
| 13:13.18 | MrChimpy | but hangs up after 30s |
| 13:13.21 | MrChimpy | no warnings. |
| 13:14.10 | Slabber | Hello, I am having troubles with native call-transfer and call-forwarding from Cisco phones after upgrading from 1.2.x to 1.4beta3 - is this a feature change?? |
| 13:14.37 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 13:17.31 | *** join/#asterisk bitwise_ (n=bitwise@ipa51.14.tellas.gr) |
| 13:20.19 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 13:21.21 | *** part/#asterisk Shini- (n=e-MaSTeR@uni31-2-82-228-91-114.fbx.proxad.net) |
| 13:22.11 | *** join/#asterisk Yoshinori (i=Yoshinor@201-43-11-81.dsl.telesp.net.br) |
| 13:23.53 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 13:26.27 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-44-36.w90-5.abo.wanadoo.fr) |
| 13:34.26 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 13:34.44 | *** join/#asterisk SoftIce_ (n=softice@c1-103-5.ndn.isadsl.co.za) |
| 13:35.02 | SoftIce_ | hi, device wctdm is in standard zaptel install |
| 13:35.08 | SoftIce_ | or do I need to install bristuff? |
| 13:35.31 | zoa | standard zaptel |
| 13:35.58 | SoftIce_ | thanks and can you tell me what module hdlc is used for? |
| 13:36.38 | zoa | http://en.wikipedia.org/wiki/HDLC |
| 13:37.36 | SoftIce_ | so its not needed for the wctdm card to work? |
| 13:38.07 | zoa | i dont think so |
| 13:39.14 | DrCron | is there a way to do internal scheduling in asterisk? for wakeup calls and the like? |
| 13:40.00 | DrCron | or would that have to be handled via an external scheduler ie: cron |
| 13:41.13 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) |
| 13:41.34 | zoa | yes |
| 13:41.37 | zoa | look for call files |
| 13:44.26 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-44-36.w90-5.abo.wanadoo.fr) |
| 13:48.33 | DrCron | can you make a call file without a channel, if you dont have any hardware ptsn connections, and want it to use the dialplan |
| 13:51.49 | Hello2007 | when i specify a codec in the extension does it overwrite the codec specified in the sip trunk or it is the oppositr? |
| 13:59.14 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:02.04 | Hello2007 | can i specify that fo internal calls use g711 and for outbound sip calls use g729??? |
| 14:02.10 | zoa | DrCron: yes |
| 14:02.16 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-44-36.w90-5.abo.wanadoo.fr) |
| 14:02.19 | *** join/#asterisk shark (n=shark@213.16.111.131) |
| 14:02.45 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-215-87.cablep.bezeqint.net) |
| 14:04.35 | DrCron | and can you have a call file call more then one extension? |
| 14:05.33 | *** join/#asterisk santibiotico (n=santi@37.Red-83-36-42.dynamicIP.rima-tde.net) |
| 14:06.32 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 14:07.48 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-44-36.w90-5.abo.wanadoo.fr) |
| 14:13.14 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 14:13.49 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-41-226.socal.res.rr.com) |
| 14:14.12 | *** join/#asterisk Menace- (i=Menace-@66.181.104.31) |
| 14:16.23 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 14:18.39 | *** join/#asterisk lorinc (n=ang@caracas-4570.adsl.interware.hu) |
| 14:22.18 | *** join/#asterisk Skarmeth (n=Skarmeth@201009104156.user.veloxzone.com.br) |
| 14:34.30 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 14:37.33 | *** join/#asterisk |dennis| (n=Dennis@200.32.233.82) |
| 14:38.40 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:39.11 | *** join/#asterisk |dennis| (n=Dennis@200.32.233.82) |
| 14:39.29 | *** join/#asterisk dasenjo (n=dasenjo@190.24.176.4) |
| 14:40.15 | *** part/#asterisk sergee (i=opera@195.94.224.197) |
| 14:41.07 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-12-222.w86-221.abo.wanadoo.fr) |
| 14:44.33 | *** join/#asterisk dummie (n=dummie@200.32.233.82) |
| 14:44.42 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-12-222.w86-221.abo.wanadoo.fr) |
| 14:46.19 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 14:46.20 | *** join/#asterisk dummie (n=dummie@200.32.233.82) |
| 14:46.47 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 14:48.08 | *** join/#asterisk dummie (n=dummie@200.32.233.82) |
| 14:48.56 | *** join/#asterisk |dennis| (n=dummie@200.32.233.82) |
| 14:49.23 | *** join/#asterisk |dennis| (n=dummie@200.32.233.82) |
| 14:50.13 | *** join/#asterisk |dennis| (n=dummie@200.32.233.82) |
| 14:55.58 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:56.22 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 15:00.26 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:02.47 | *** part/#asterisk rtcg (n=chatzill@mail.richardthecomputerguy.com) |
| 15:05.38 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 15:06.51 | *** join/#asterisk nvicf (n=vincent@201.250.167.253) |
| 15:07.05 | nvicf | hello, anyone knows what model is the pri card that has 2 ports? |
| 15:07.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:08.51 | *** join/#asterisk g0tw00d (n=rchace@68-113-159-200.dhcp.nplt.ne.charter.com) |
| 15:09.02 | *** join/#asterisk dasenjo_ (n=dasenjo@190.24.178.135) |
| 15:10.34 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 15:10.51 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-41-226.socal.res.rr.com) |
| 15:10.59 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 15:12.56 | *** join/#asterisk danbrwn (n=chatzill@216.77.58.40) |
| 15:15.10 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-12-222.w86-221.abo.wanadoo.fr) |
| 15:15.31 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 15:16.16 | *** join/#asterisk NoPlug (n=hughes@46.142-65-87.adsl-dyn.isp.belgacom.be) |
| 15:16.29 | *** join/#asterisk nays85 (i=nays85@66.252.12.130) |
| 15:20.01 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 15:20.14 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 15:20.54 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:21.56 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 15:22.11 | key2 | <PROTECTED> |
| 15:25.27 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:28.36 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 15:29.06 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 15:29.33 | *** part/#asterisk naitram (n=chatzill@216.77.58.40) |
| 15:29.48 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 15:30.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:31.02 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:31.51 | *** join/#asterisk Slabber (n=simon@194.7.163.103) |
| 15:32.10 | Slabber | Can someone please help? |
| 15:32.25 | mosty | ask a specific question |
| 15:32.52 | Slabber | Has anybody got attended transfers working with 1.4? |
| 15:33.15 | *** join/#asterisk stuq (n=stuq@74-32-63-190.dsl1.mdl.ny.frontiernet.net) |
| 15:33.36 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 15:33.46 | Slabber | there seems to have been a change in chan_sip.c revision 31274 and it broke my call transfers using Cisco phones |
| 15:34.27 | Slabber | 'SIP attended transfer: Error: No target channel' |
| 15:34.30 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-12-222.w86-221.abo.wanadoo.fr) |
| 15:36.26 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@8.10.2.50) |
| 15:36.31 | *** join/#asterisk stuq_ (n=stuq@user-12lcqia.cable.mindspring.com) |
| 15:36.57 | Slabber | maybe something to do with 'REFER' messages but I'm not sure how to handle these properly |
| 15:37.26 | Slabber | the changes also broke call-forwarding but I fixed it using a 'catch' statement in the Macro where the initial 'Dial' command was |
| 15:38.41 | file | 1. Try a checkout from the 1.4 branch 2. File a bug report with all the information you can (console output, debug information, sip debug, yada) |
| 15:39.39 | Slabber | will do |
| 15:39.47 | *** join/#asterisk foxxtrot (n=craig@c-67-185-148-19.hsd1.or.comcast.net) |
| 15:40.54 | *** join/#asterisk dusty_ (n=dusty@dns.suspicious.org) |
| 15:41.31 | *** part/#asterisk dusty_ (n=dusty@dns.suspicious.org) |
| 15:43.15 | *** join/#asterisk ka05 (n=graeme@host81-149-13-214.in-addr.btopenworld.com) |
| 15:43.55 | bsdfreak | ye |
| 15:44.22 | ka05 | Can anyone point me in the right direction to setup dtmf tones? |
| 15:44.23 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 15:44.42 | blitzrage | ka05: huh? |
| 15:44.49 | blitzrage | dtmfmode= ? |
| 15:44.59 | ka05 | for automated phone systems |
| 15:45.06 | ka05 | so that key presses are registered |
| 15:45.38 | blitzrage | I'm confused... |
| 15:45.52 | blitzrage | you mean like an auto attendent? |
| 15:47.11 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:47.54 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.104.Dial1.SanJose1.Level3.net) |
| 15:53.42 | santibiotico | when i dial an external number, the phone starts the dial tone immediately, but the other peer rings a few seconds (10 more or less) later |
| 15:54.01 | *** part/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 15:54.04 | Strom_C | what do you mean "starts the dial tone"? |
| 15:54.32 | file | Strom_C: a more beautiful vision I have never seen, if you don't mind me saying... a lifelong ambition to fulfill my dream - what have you done to me? |
| 15:54.43 | *** join/#asterisk klasstek_ (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:54.52 | santibiotico | is there any way to get the tone in the dialing peer and the ring tone synchronized? |
| 15:55.14 | Strom_C | file: you're not making sense |
| 15:55.20 | santibiotico | Strom_C: i mean the beeeep beeeep that means the other peer is ringing |
| 15:55.26 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-158.lsanca.dsl-w.verizon.net) |
| 15:55.33 | santibiotico | i don't know how to call it in english, sorry :P |
| 15:55.33 | Strom_C | santibiotico: take the "r" flag off of your Dial() command |
| 15:55.41 | santibiotico | ok |
| 15:55.46 | santibiotico | thanks Strom_C |
| 15:55.47 | Strom_C | santibiotico: audible ringing, usually |
| 15:55.59 | Strom_C | santibiotico: you should never use that flag unless you absolutely need to |
| 15:56.36 | Strom_C | clobbering the signaling that the remote party is sending you is almost always a bad idea |
| 15:57.41 | Strom_C | MrChimpy: 1.2 svn is usually a Good Idea(tm) |
| 15:58.12 | *** join/#asterisk dasenjo (n=dasenjo@190.24.177.171) |
| 15:58.24 | MrChimpy | i'm about to put a callback type app into production which reallly needs the Dial limits stuff which 1.2.13 breaks |
| 16:00.21 | *** join/#asterisk santiago (n=santiago@190.24.177.171) |
| 16:00.46 | *** join/#asterisk X-Rob_ (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 16:01.25 | G4335 | hm.. if i want to connect a single analog fax device to an asterisk-box (customer requirement..) - what kind of hardware would be recommended to do this? |
| 16:01.28 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 16:03.12 | Strom_C | G4335: best thing to do is to connect the fax to a channel bank, bring in a CAS or ISDN T1/E1, and bridge all calls across the same multi-span T1/E1 card. |
| 16:03.47 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 16:04.27 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 16:04.28 | G4335 | isn't that a bit much effort for connecting just one single analog device...? |
| 16:05.07 | Strom_C | G4335: that's the best way to ensure reliability with faxing |
| 16:05.24 | Strom_C | otherwise, you're better off running the fax completely independently of the PBX |
| 16:06.17 | *** join/#asterisk darkskiez_ (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 16:06.21 | *** join/#asterisk doolph (n=doo@200.46.148.58) |
| 16:06.33 | G4335 | hrm.. |
| 16:06.39 | doolph | sup |
| 16:09.22 | RoyK | Strom_C: or get something that can do t.38 gatewaying |
| 16:09.38 | *** join/#asterisk NeScHe`mUziK`diN (n=RIZA@85.108.150.19) |
| 16:09.53 | RoyK | asterisk can probably do t.38 gateway one day, but it'll be a day some time from now |
| 16:09.59 | RoyK | couple of years or something :P |
| 16:10.50 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:11.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:12.02 | Strom_C | hey mog |
| 16:12.49 | zoa | royk |
| 16:12.50 | zoa | dont think so |
| 16:13.24 | mog | hey Strom_C |
| 16:13.29 | *** join/#asterisk andresmujica (n=andresmu@201.245.228.228) |
| 16:13.31 | zoa | i have t.38 gateway on asterisk |
| 16:13.41 | RoyK | zoa: you do? |
| 16:13.49 | Juggie | zoa has everything. |
| 16:13.50 | zoa | yes |
| 16:13.53 | *** join/#asterisk asternic (n=asternic@ip-177.houseware.com.ar) |
| 16:13.58 | RoyK | zoa: closed source, then? |
| 16:14.06 | file | what DOESN'T zoa have? |
| 16:14.26 | zoa | closed source yes |
| 16:14.41 | RoyK | openpbx t.38 gateway is almost there.... |
| 16:14.54 | *** join/#asterisk _Vile (n=mattk@bc182112.bendcable.com) |
| 16:15.01 | HarryR | RoyK, YATE's t.38 gateway is complete and aparently working very well |
| 16:15.34 | in-pt | Hi..what i needs to do to read and write voicemail conf in mysql db |
| 16:15.35 | mog | zoa you know you want to free the source |
| 16:15.37 | RoyK | HarryR: on sip? |
| 16:15.45 | Juggie | mog, jack in the box! |
| 16:15.57 | RoyK | HarryR: or h.323? |
| 16:15.58 | mog | i actually ate at jacks on saturday |
| 16:16.26 | HarryR | RoyK, likely to be on SIP, go join #yate and ask some questions as I'm not using T38 with YATE atm |
| 16:16.37 | zoa | on h323 its a bitch i think |
| 16:17.05 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 16:17.48 | rbd | hey guys, any way to totally disable SIP challenge-response authenication for each phone (or all phones)? |
| 16:17.51 | zoa | need to go now |
| 16:18.11 | Strom_C | rbd: why? |
| 16:18.50 | rbd | Strom_C: we are testing an app we're developing (that issues SIP commands) against asterisk and no authenication is a good first phase |
| 16:19.22 | Strom_C | I /believe/ insecure=very will turn that off, although I may be wrong |
| 16:20.23 | *** join/#asterisk _Vile (n=mattk@bc182112.bendcable.com) |
| 16:20.47 | mog | they didnt have to unlock the doors for me though Juggie |
| 16:21.21 | Qwell[] | mog: Where's the fun in that? |
| 16:22.10 | Juggie | hah thats no fun |
| 16:22.14 | Juggie | its only good @ 2am |
| 16:22.23 | Qwell[] | no, no, it's good all day |
| 16:22.27 | Juggie | between 10am and 12pm its pretty horrible food :) |
| 16:22.27 | Qwell[] | it's just better at 2am |
| 16:22.28 | asternic | Hello |
| 16:22.35 | Juggie | er, 12am |
| 16:22.51 | *** join/#asterisk _cleric_ (n=dacleric@p54820E96.dip0.t-ipconnect.de) |
| 16:23.03 | Qwell[] | Juggie: They make bad food for 2 hours a day? |
| 16:23.58 | rbd | Strom_C: okay, It may only apply to asterisk 1.0.9 and earlier but I will try it. thanks |
| 16:24.45 | Strom_C | 1.0.9? |
| 16:24.56 | asternic | Hello all.. Is anyone using trunk? |
| 16:26.15 | in-pt | Hi..what i needs to do to read and write voicemail conf in mysql db |
| 16:26.32 | in-pt | can anyone give me a clue where to look for that?? |
| 16:27.09 | asternic | in-pt: extconfig.conf |
| 16:27.30 | asternic | voicemail => odbc,asterisk,users |
| 16:28.07 | *** join/#asterisk vexorg (n=vexorg@h209-17-153-98.gtcust.grouptelecom.net) |
| 16:28.12 | asternic | look into res_odbc.conf |
| 16:28.24 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-231.range81-152.btcentralplus.com) |
| 16:28.33 | *** part/#asterisk asternic (n=asternic@ip-177.houseware.com.ar) |
| 16:28.41 | *** join/#asterisk lemos (n=esgrovas@62.48.215.118) |
| 16:28.45 | lemos | sup guys |
| 16:28.48 | lemos | one quick question |
| 16:29.00 | lemos | what's the port parameter on the sip.conf file for? |
| 16:29.08 | lemos | i don't quite get it its usefulness |
| 16:29.22 | lemos | I've gone through the wiki, but I still can't make it out |
| 16:29.25 | doolph | its the port where your sip is listening |
| 16:29.44 | lemos | doolph: but that's the bindto or whatever parameters on the global section, isn't? |
| 16:29.50 | lemos | doolph: I'm speaking peer wise |
| 16:31.47 | in-pt | asternic: where i can see sample confs for that..and where is the db schema..these files dont have that ? |
| 16:32.15 | in-pt | i just want to save username, pass, email in mysql to send voicemail |
| 16:32.39 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 16:37.55 | *** join/#asterisk shinux__ (n=shinux@196.220.30.248) |
| 16:38.41 | file | lemos: what if you need to contact the remote peer on a different port? |
| 16:39.43 | *** join/#asterisk Stalker_ (n=Miranda@83.234.35.238) |
| 16:40.42 | Stalker_ | hi people! |
| 16:40.49 | *** join/#asterisk apardo (n=apardo@87.217.145.95) |
| 16:42.48 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 16:42.52 | Katty | morning! |
| 16:43.15 | *** join/#asterisk spyda (n=scott@hera.copi-rite.com) |
| 16:43.49 | Katty | get out. |
| 16:43.54 | in-pt | what is the channel of asterisk-addons ?? |
| 16:44.01 | Qwell[] | in-pt: there isn't one |
| 16:44.04 | Qwell[] | here is fine |
| 16:44.08 | MrChimpy | joy. |
| 16:44.25 | Qwell[] | MrChimpy: I believe there is a patch on the bug tracker |
| 16:45.13 | in-pt | Qwell[] i want to know what changes in needs to do in apps/Makefile to enable vm support with mysql? |
| 16:45.45 | MrChimpy | qw: patch was apparently applied to svn, but it doesn't work |
| 16:45.49 | file | MrChimpy: leave a note on the existing bug, 8541 |
| 16:45.55 | MrChimpy | unless there's another |
| 16:46.00 | file | k |
| 16:46.19 | *** join/#asterisk stuq (n=stuq@74-32-63-190.dsl1.mdl.ny.frontiernet.net) |
| 16:47.29 | *** part/#asterisk spyda (n=scott@hera.copi-rite.com) |
| 16:48.00 | MrChimpy | 8386 looks like the same thing, and murf applied his patches apparently |
| 16:48.14 | Stalker_ | how to buy the asterisk business edition ? |
| 16:48.50 | file | MrChimpy: he did, and he tested it - I was on the phone when he did |
| 16:49.24 | MrChimpy | this is nuts then :) |
| 16:49.35 | file | more like complicated |
| 16:49.56 | file | one thing goes to another that you didn't expect... which causes an issue to occur... but if you change that, then it breaks something else... |
| 16:50.07 | *** join/#asterisk henrique (n=henrique@200-161-80-249.dsl.telesp.net.br) |
| 16:50.23 | MrChimpy | yep. i'm not touching anything beyond trying to trace where a problem is |
| 16:50.45 | *** join/#asterisk stuq_ (n=stuq@user-12lcqia.cable.mindspring.com) |
| 16:50.53 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:51.02 | *** join/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it) |
| 16:51.03 | MrChimpy | did this pop up in 1.2.13 specifically? should I drop back more than .12? |
| 16:51.25 | *** part/#asterisk UlbabraB (n=UlbabraB@88-149-155-155.f5.ngi.it) |
| 16:51.27 | file | I don't remember off the top of my head |
| 16:51.44 | MrChimpy | ok, i'll do some diff browsing. thanks for the help. |
| 16:51.47 | *** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.51) |
| 16:51.57 | tdonahue-laptop | hi all |
| 16:52.09 | MrChimpy | it was ok in .7, so only 6 revs to try :) |
| 16:54.00 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:0:240:f4ff:fe6b:7650) |
| 16:55.28 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 16:55.34 | tdonahue-laptop | is there any way to semi-permanently busy out a zap channel from the console, we are having line issues and want to busy out that line so calls don't come in on it |
| 16:55.36 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 16:56.17 | Strom_C | set up a call between it and console/dsp? |
| 16:57.06 | *** join/#asterisk mat2 (n=mat@69.111.138.74) |
| 16:57.24 | mat2 | good morning everyone |
| 17:00.07 | Strom_C | everybody's hugging |
| 17:00.21 | file | isn't it crazy? |
| 17:00.31 | Strom_C | bonkers, even |
| 17:00.50 | file | yay |
| 17:00.53 | Katty | i feel so loved. sniffle. |
| 17:01.12 | Katty | my polycoms do not love me, however )= |
| 17:01.16 | Katty | i'm having a flashy light issue. |
| 17:01.20 | Katty | it makes me all sad inside. |
| 17:02.19 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 17:02.39 | Corydon-w | Flashing when it ought not? |
| 17:03.12 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 17:03.27 | nays85 | can anyone recommend a billing system for ~5000 residential customers? |
| 17:03.35 | *** join/#asterisk shinux__ (n=shinux@196.220.27.43) |
| 17:03.37 | file | it's otay |
| 17:03.42 | Katty | Corydon-w: it doesn't flash when there's voicemail )= |
| 17:03.54 | Katty | Corydon-w: but i can take my working phone, and give it to another person...and register it someone else... |
| 17:04.05 | Corydon-w | Katty: are you using a voicemail context other than "default"? |
| 17:04.07 | mat2 | I just upgraded to 1.4beta and my Zap interface is no longer working |
| 17:04.07 | Katty | Corydon-w: and it won't work. but their nonworking phone, registered as me, works. |
| 17:04.13 | Katty | Corydon-w: yesh. |
| 17:04.30 | Katty | Corydon-w: do i need to go snooping about voicemail.conf? |
| 17:04.34 | Corydon-w | Katty: check your voicemail= line in sip.conf and make sure it specifies @othercontext |
| 17:04.37 | *** join/#asterisk PolinA (n=Miranda@83.234.35.238) |
| 17:04.42 | Katty | Oooo. |
| 17:05.22 | mat2 | has anyone else had problems after upgrading? |
| 17:05.22 | Corydon-w | Katty: if you don't specify, it assumes the default context |
| 17:05.46 | Katty | Corydon-w: i love you. |
| 17:05.58 | Katty | Corydon-w: you found my error!! |
| 17:06.04 | Corydon-w | woohoo! |
| 17:06.12 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 17:07.03 | mat2 | ztcfg shows the channels are configured: |
| 17:07.03 | mat2 | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 17:07.04 | mat2 | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
| 17:07.04 | mat2 | 2 channels configured. |
| 17:07.11 | *** join/#asterisk RoyK (n=roy@ti211310a080-15179.bb.online.no) |
| 17:07.27 | mat2 | but i get the following error: |
| 17:07.27 | mat2 | [Dec 11 05:13:02] WARNING[12546]: channel.c:2870 ast_request: No channel type registered for 'Zap' |
| 17:07.27 | mat2 | [Dec 11 05:13:02] WARNING[12546]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
| 17:07.37 | *** join/#asterisk _X-Rob_ (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 17:07.46 | in-pt | can anyone please tell me where to look for saving voicemails in mysql ? |
| 17:07.59 | *** join/#asterisk jarrod (i=nobody@dont.juniperyour.net) |
| 17:08.15 | jarrod | how do i change the caller-id on anonymous calls to not read 'asterisk' ? |
| 17:08.31 | in-pt | mat2: have you configured the channel in zapata.conf ? |
| 17:10.01 | Katty | Corydon-w: i've been poking about the phones and the ftp server for days over this issue >.< |
| 17:10.30 | *** join/#asterisk alamantia (i=anthonyl@nat/digium/x-d501695f265fa048) |
| 17:10.43 | mat2 | ; define channels |
| 17:10.44 | mat2 | context=incoming ; Incoming calls go to [incoming] in extensions.conf |
| 17:10.44 | mat2 | signalling=fxs_ks ; Use FXS signalling for an FXO channel |
| 17:10.44 | mat2 | channel => 1 ; PSTN attached to port 2 |
| 17:10.51 | mat2 | in-pt: yes |
| 17:11.05 | mat2 | ignore the comment on that channel line :) |
| 17:11.12 | mat2 | it is connected to channel 1 |
| 17:12.01 | Corydon-w | mat2: please use pastebin.ca instead of pasting multiple lines to the channel |
| 17:12.10 | in-pt | can you pastebin your zaptel.conf, zapata.conf and extensions.conf related to zap |
| 17:12.29 | mat2 | ok.. sorry corydon :) |
| 17:13.39 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 17:14.25 | lemos | file: why would I need to do that? Isn't that port random?. The port contacts asterisk with a source random port, right? So what's the usefulness for this parameter? |
| 17:15.32 | Corydon-w | lemos: the port parameter is for specifying the remote port for control packets |
| 17:15.58 | Corydon-w | lemos: note that RTP uses its own set of ports, but the RTP ports are set up using the control |
| 17:16.57 | *** join/#asterisk apardo (n=apardo@84.76.197.85) |
| 17:17.22 | file | and the standard SIP port is 5060, it's not random |
| 17:18.30 | mat2 | in-pt: http://www.paste.me.uk/326.html |
| 17:20.00 | ka05 | |
| 17:20.03 | ka05 | quit |
| 17:21.22 | *** join/#asterisk jarrod (i=nobody@dont.juniperyour.net) |
| 17:21.59 | *** join/#asterisk apardo (n=apardo@84.76.197.85) |
| 17:22.17 | in-pt | mat2: fxsks=1-2 is not good change to fxoks=1 |
| 17:22.50 | jarrod | where does asterisk set the callerid name to 'asterisk' on incoming private calls? i'd like to modify this setting |
| 17:23.39 | *** join/#asterisk gr1ncheux_ (n=devine@AToulouse-257-1-47-83.w90-5.abo.wanadoo.fr) |
| 17:24.19 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:24.35 | mat2 | in-pt: are you supposed to use fxsks for an fxo channel? |
| 17:24.36 | doolph | I think its the callerid "asterisk" |
| 17:25.09 | jarrod | when calls come in that are private, 'asterisk' is in the calleridname field |
| 17:25.18 | mat2 | in-pt: o get an error message when modprobe wctdm |
| 17:25.22 | in-pt | mat2: change in zapata.conf also |
| 17:25.29 | mat2 | ok |
| 17:25.47 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 17:27.58 | mat2 | im getting errors just loading the modules. |
| 17:28.18 | mat2 | my card is fxo, and it says in the error messages should be fxs signalling |
| 17:28.28 | doolph | what version of zaptel are you using? |
| 17:28.34 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 17:28.37 | mat2 | 1.4 beta |
| 17:28.43 | doolph | ok |
| 17:28.53 | mat2 | 1.4.0-beta2 |
| 17:28.54 | mat2 | :) |
| 17:28.54 | doolph | what kind of card is it |
| 17:29.08 | mat2 | digium tdm 2 fxo |
| 17:29.21 | doolph | are you loading wcfxo ? |
| 17:29.21 | in-pt | do you have fxs card or fxo card connected ? |
| 17:29.25 | mat2 | in slot 1-2 |
| 17:29.42 | mat2 | wctdm |
| 17:29.47 | doolph | did you try modprobe zaptel & modprobe wctdm ? |
| 17:29.52 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 17:30.51 | doolph | you can alse see if it is attached well cat /proc/interruptors |
| 17:30.51 | doolph | i mean /proc/interrupts |
| 17:30.51 | mat2 | yes.. modules load fine, and ztcfg shows they are configured |
| 17:30.51 | doolph | ok use scanzap tool |
| 17:30.52 | doolph | to configure it |
| 17:30.52 | doolph | i mean zapscan |
| 17:31.00 | doolph | zapscan.bin something like that |
| 17:31.06 | mat2 | card has been working fine for 1 year until the upgrade this weekend |
| 17:31.18 | doolph | then edit /etc/zapte.conf and remove buggy lines |
| 17:31.24 | doolph | also /etc/asterisk/zapata.conf |
| 17:31.37 | doolph | zapscan will configure it for you |
| 17:33.24 | mat2 | when i run zapscan it say "Skipping Zap Scanning" |
| 17:33.53 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:35.10 | *** join/#asterisk Cableguy (n=cabl3guy@S01060004e28f1b52.cg.shawcable.net) |
| 17:37.14 | doolph | umm |
| 17:37.25 | Cableguy | Hello all, anybody available to answer a few questions about Asterisk? Specifically concerning hardware requirements? |
| 17:37.29 | doolph | i dont know then |
| 17:37.37 | doolph | Cableguy just ask |
| 17:37.56 | codefreeze | MrChimpy: re: 8386 and 8541--- still a problem? |
| 17:40.09 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 17:41.01 | *** join/#asterisk ^rommy^ (n=mynnoryk@222.124.24.107) |
| 17:41.14 | ^rommy^ | helo everybody.. |
| 17:42.21 | ^rommy^ | helo |
| 17:42.41 | doolph | hi |
| 17:42.50 | *** join/#asterisk avalone (n=avalone_@dial-075.vl-cen-as3.avtlg.ru) |
| 17:43.22 | *** join/#asterisk bitwise (n=bitwise@ipa214.10.tellas.gr) |
| 17:43.27 | ^rommy^ | halow |
| 17:43.30 | ^rommy^ | hi.. |
| 17:43.39 | wunderkin | .... |
| 17:43.52 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:45.27 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:45.39 | *** join/#asterisk shinux__ (n=shinux@196.220.29.101) |
| 17:50.50 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 17:52.17 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 17:53.51 | *** join/#asterisk henrique (n=henrique@200-161-80-249.dsl.telesp.net.br) |
| 17:58.29 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:59.50 | *** part/#asterisk dasenjo (n=dasenjo@190.24.177.171) |
| 18:00.05 | *** join/#asterisk jarrod (i=nobody@dont.juniperyour.net) |
| 18:00.33 | jarrod | what is the best solution for running two asterisk softswitches, in an active/passive failover state? |
| 18:02.26 | *** join/#asterisk asternic (n=asternic@ip-177.houseware.com.ar) |
| 18:10.23 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 18:10.32 | *** join/#asterisk scotsmac (n=kmk@82-40-19-233.cable.ubr03.uddi.blueyonder.co.uk) |
| 18:11.40 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 18:12.43 | *** part/#asterisk scotsmac (n=kmk@82-40-19-233.cable.ubr03.uddi.blueyonder.co.uk) |
| 18:12.46 | *** join/#asterisk tparcina (n=tparcina@195-29-117-97.adsl.net.t-com.hr) |
| 18:15.08 | *** join/#asterisk roguebug (n=roguebug@dip-109-134.bras.dsl.breisnet.com) |
| 18:15.11 | roguebug | hi |
| 18:16.20 | mat2 | is chan_zap.so supposed to be installed during asterisk installation, or zaptel? |
| 18:16.30 | roguebug | i configured asterisk 1.4.0.3beta wit a --prefix=/my_prexix_directory. still, make install gives me an error: |
| 18:16.31 | roguebug | mkdir: cannot create directory `/var/lib/asterisk/static-http': Permission denied |
| 18:16.45 | roguebug | where can i set a prefix for that? |
| 18:17.06 | roguebug | s/1.4.0.3beta/1.4.0beta3/g |
| 18:17.30 | Qwell[] | roguebug: edit makeopts.in, add ${prefix} before those two lines, near the top |
| 18:17.34 | Qwell[] | then re-run configure |
| 18:17.48 | Qwell[] | (those two, meaning the /var/ and /etc/ paths) |
| 18:20.09 | roguebug | ok thx Qwell[] |
| 18:20.24 | Qwell[] | roguebug: there is already a bug on the tracker - just haven't had time to make a patch for it |
| 18:21.16 | Qwell[] | roguebug: If you want to upload a patch, please feel free... bug number is 8555 |
| 18:21.42 | Qwell[] | (assuming you have a disclaimer on file already, or can send one in...) |
| 18:23.47 | in-pt | MySQL RealTime: Failed to connect database server vm on 127.0.0.1 (err 1045). Check debug for more info |
| 18:23.48 | roguebug | i don't see any mention of /etc or /var in makeopts.in but in makeopts. can i assume that this is the file to change? |
| 18:23.57 | in-pt | whats the reason for that error in asterisk cli |
| 18:23.59 | Qwell[] | no, it's makeopts.in, i'm fairly sure |
| 18:24.18 | in-pt | i have setup correct username and pass in res_mysql.conf file |
| 18:24.58 | roguebug | my makeopts.in doesn't have any explicitly named directories, just @SOME_KEYWORD@ type entries |
| 18:25.13 | Qwell[] | one sec, I'll upload a patch |
| 18:25.22 | Qwell[] | ahh, okay, it isn't makeopts.in |
| 18:27.00 | roguebug | but makeopts? |
| 18:28.09 | Qwell[] | you CAN edit makeopts as a temp fix, but it'll hose itself next time you run configure |
| 18:28.35 | *** join/#asterisk Modcuts (n=Moducts@88-109-229-212.dynamic.dsl.as9105.com) |
| 18:28.53 | asternic | hello! is someone using svn-trunk and zap channels? |
| 18:33.28 | roguebug | Qwell: so what's the clean fix? |
| 18:33.54 | roguebug | Qwell[] even (my tab completion is a bit overzealous) |
| 18:34.32 | *** join/#asterisk ltd (n=z@202-161-1-26.dyn.iinet.net.au) |
| 18:35.14 | Qwell[] | roguebug: I just talked it over with kpfleming, and it isn't really a bug. |
| 18:36.02 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 18:36.46 | Qwell[] | roguebug: as a hack, you can edit makeopts like I said - but, it is a hack |
| 18:38.09 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 18:39.05 | asternic | A quick question... just checking to see if I'm the only one with the problem. I upgraded to SVN-trunk to try the new asterisk-gui. After making some updates to the dialplan everything seems to be working, with one strange issue: when I dial zap/g1/XXX the channel reported by asterisk is always Zap/0-0. Am I the only one? |
| 18:39.21 | roguebug | ok thx Qwell [] |
| 18:39.59 | roguebug | i'm gonna go mad if i don't find that xchat setting that gives me back my bash-like autocomplete |
| 18:40.19 | wasim | for anybody in Pakistan, PTA and FIA arrent the CEO of Cogilient for using Asterisk ... alledgedly doing termination ... another case of the Big BAD PTCL punishing small companies |
| 18:40.37 | wasim | most everybody who knows them vouches they weren't ... |
| 18:41.49 | Qwell[] | arrent? arrest? |
| 18:42.40 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 18:45.19 | *** join/#asterisk websae_ (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 18:46.29 | *** join/#asterisk naitram_ (n=chatzill@216.77.58.40) |
| 18:47.46 | *** join/#asterisk groogs_ (n=greg@d38-54-164.commercial1.cgocable.net) |
| 18:47.59 | PupenoR | In the dial plan, how do I echo a variable I set ? |
| 18:48.16 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 18:50.53 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) [NETSPLIT VICTIM] |
| 18:50.53 | *** join/#asterisk normie (i=david@fatboy.mcnally.dk) [NETSPLIT VICTIM] |
| 18:50.53 | *** join/#asterisk JonR800 (i=jon@p1mp.org) [NETSPLIT VICTIM] |
| 18:50.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) [NETSPLIT VICTIM] |
| 18:50.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:50.54 | *** join/#asterisk StyleWarz (i=stylewar@gayanalfisting.biz) [NETSPLIT VICTIM] |
| 18:50.54 | *** join/#asterisk oink (i=ziga@al.co.ve) [NETSPLIT VICTIM] |
| 18:50.54 | roguebug | whew ultrasplit |
| 18:50.54 | *** join/#asterisk findlay (n=justin@67.137.24.115) [NETSPLIT VICTIM] |
| 18:50.54 | *** join/#asterisk JT_ (n=jon@unicomsystems.com) [NETSPLIT VICTIM] |
| 18:50.54 | *** join/#asterisk psion (n=123@22.80-202-239.nextgentel.com) |
| 18:50.55 | *** join/#asterisk SheriF_SpacE (n=sherif@chatty.shellshark.net) [NETSPLIT VICTIM] |
| 18:50.55 | *** join/#asterisk bradoaks (n=bradoaks@dog.lost-habit.com) [NETSPLIT VICTIM] |
| 18:50.55 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM] |
| 18:50.55 | PupenoR | I mean, echo to the console, print it. |
| 18:50.55 | CunningPike | Ugh |
| 18:50.55 | *** join/#asterisk SwK[Work] (n=SwK@br0.asteriasgi.com) [NETSPLIT VICTIM] |
| 18:50.55 | *** join/#asterisk Mw3 (i=mw3@ip59934bd1.bp-1031.rubicom.hu) [NETSPLIT VICTIM] |
| 18:50.56 | *** join/#asterisk klasstek_ (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) [NETSPLIT VICTIM] |
| 18:50.56 | *** join/#asterisk NeScHe`mUziK`diN (n=Limon@85.102.155.142) |
| 18:51.37 | CunningPike | Hmm - just had a user send me a voicemail recording which had a small portion of it sounding really speeded up - just a few seconds in the middle of it. Anyone know what could cause that? |
| 18:52.02 | *** join/#asterisk Corydon76-home (i=purple@c-68-53-162-99.hsd1.tn.comcast.net) [NETSPLIT VICTIM] |
| 18:52.03 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 18:52.04 | *** join/#asterisk technoid_ (n=technoid@adsl-68-254-38-80.dsl.bltnin.sbcglobal.net) [NETSPLIT VICTIM] |
| 18:52.07 | *** join/#asterisk intralanman (n=lanman@pool-71-253-247-137.nrflva.east.verizon.net) [NETSPLIT VICTIM] |
| 18:52.16 | *** join/#asterisk queuetue (n=scott@MTRLPQ02-1177745854.sdsl.bell.ca) [NETSPLIT VICTIM] |
| 18:52.19 | *** join/#asterisk leprechau (i=leprecha@c-68-53-17-135.hsd1.tn.comcast.net) [NETSPLIT VICTIM] |
| 18:52.29 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) [NETSPLIT VICTIM] |
| 18:52.37 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) [NETSPLIT VICTIM] |
| 18:52.44 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) [NETSPLIT VICTIM] |
| 18:52.46 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM] |
| 18:52.47 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) [NETSPLIT VICTIM] |
| 18:52.54 | *** join/#asterisk Dont_Hate_bia (n=te_lo_me@8.10.2.50) |
| 18:52.56 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) [NETSPLIT VICTIM] |
| 18:53.14 | *** join/#asterisk Dibbler (n=Dibbler@dsl-217-155-254-174.zen.co.uk) [NETSPLIT VICTIM] |
| 18:53.21 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM] |
| 18:53.28 | *** join/#asterisk puddingpimp_work (n=puddingp@gateway.quickcircuit.co.nz) [NETSPLIT VICTIM] |
| 18:53.30 | *** join/#asterisk Skarmeth (n=Skarmeth@201009104156.user.veloxzone.com.br) [NETSPLIT VICTIM] |
| 18:53.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) [NETSPLIT VICTIM] |
| 18:53.42 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) [NETSPLIT VICTIM] |
| 18:53.43 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) [NETSPLIT VICTIM] |
| 18:53.47 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) [NETSPLIT VICTIM] |
| 18:53.47 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 18:54.01 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) [NETSPLIT VICTIM] |
| 18:54.04 | *** join/#asterisk ccesario (n=ccesario@matriz.isic.com.br) [NETSPLIT VICTIM] |
| 18:54.12 | SplasPood | [boot] is doing OnJoin spam |
| 18:54.23 | *** part/#asterisk Qwell[] (i=qwell@unaffiliated/qwell) |
| 18:54.23 | *** join/#asterisk Qwell[] (i=qwell@unaffiliated/qwell) |
| 18:54.23 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 18:54.44 | Qwell[] | /msg me the text? |
| 18:54.55 | *** join/#asterisk Dont_Hate_bia (n=te_lo_me@8.10.2.50) |
| 18:55.05 | doolph | uh |
| 18:55.37 | roguebug | Qwell[]: ok after that little "hack" in makeopts, the /var/lib/asterisk/static-http is prefixed nicely and the files installed there fine. however, farther down in the make install, i get other errors. now it even tries to write files to the root dir : |
| 18:55.55 | roguebug | /usr/bin/install: cannot create regular file `/stereorize': Permission denied |
| 18:56.22 | *** join/#asterisk wasim (n=wasim@203.81.230.108) |
| 18:56.22 | Qwell[] | dunno what to tell you.. it was a hack |
| 18:56.29 | roguebug | same for /streamplayer , /aelparse and /muted |
| 18:56.54 | Qwell[] | the "right" thing to do, would be to not use --prefix, and use `DESTDIR=/path/to/install/dir make install`, then chroot to that dir |
| 18:57.31 | CunningPike | Just had a user send me a voicemail recording which had a small portion of it sounding really speeded up - just a few seconds in the middle of it. Anyone know what could cause that? We have no other audio problems..... |
| 18:58.02 | mat2 | i deleted my chan_zap.so and then reinstalled asterisk(1.4beta3), but it doesnt seem to have recreated the chan_zap module. is it supposed to be in the asterisk package? |
| 18:58.16 | Qwell[] | mat2: install zaptel, then rerun configure, and make install |
| 18:58.40 | doolph | mat2 your problem aint asterisk its zaptel |
| 18:58.48 | mat2 | rerun configure on asterisk? |
| 18:58.54 | mat2 | after installing zaptel |
| 18:58.54 | Qwell[] | after installing zaptel, yes |
| 18:58.56 | mat2 | ok |
| 18:59.54 | *** mode/#asterisk [+b *!*n=Limon@85.102.155.*] by Qwell[] |
| 18:59.54 | *** kick/#asterisk [[boot]!i=qwell@unaffiliated/qwell] by Qwell[] (pr0n spam) |
| 19:00.29 | Qwell[] | google groups, sort by date |
| 19:01.25 | Corydon-w | SplasPood: dates back to the era of BBS's and word filters |
| 19:02.25 | SplasPood | Corydon-w: Hrm, seems plausible.. |
| 19:02.50 | Corydon-w | Not just plausible; I know for a fact that's where it originated |
| 19:03.47 | Corydon-w | It was a major reason for connecting to a BBS in the first place |
| 19:03.52 | mat2 | Qwell: should I do a make clean before reinstalling asterisk too? |
| 19:03.54 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:03.57 | Qwell[] | mat2: nah |
| 19:04.23 | SplasPood | Corydon-w: heh porn is the major reason for ANYTHING |
| 19:04.30 | Qwell[] | If you were uClibc, where would you hide dl? :P |
| 19:04.39 | SplasPood | Porn and videos of people getting hurt |
| 19:04.41 | Qwell[] | nm |
| 19:04.56 | Corydon-w | No, porn and pirates |
| 19:06.37 | Corydon-w | BBS's were prior to the era of digital video |
| 19:06.38 | roguebug | Corydon: what about pirates? |
| 19:06.38 | Corydon-w | How much video can you really view over a 14.4k modem? |
| 19:06.38 | roguebug | Corydon: if we had more pirates, there would be no global warming! |
| 19:10.32 | asternic | I ran a BBS back then... good times |
| 19:11.21 | wasim | yeah, PCBoard! |
| 19:11.28 | wasim | and galactica |
| 19:11.29 | SplasPood | Corydon-w: yes I know they were... so replace videos with 'photos' for back then. |
| 19:11.50 | wasim | MBBS rocked too, nice ansi color screens! |
| 19:12.07 | SplasPood | I remember hacking all the text strings in MBBS to make it look as much like some unix system as possible |
| 19:12.21 | SplasPood | I have this giant archive of BBS software floating around somewhere |
| 19:12.36 | wasim | shit, 2 lines in the whole country, me and SQ used to logon and talk to each other ... |
| 19:12.43 | wasim | then someone else got a modem a few years later |
| 19:12.48 | Qwell[] | SplasPood: in your closet with your bell bottoms? |
| 19:13.04 | SplasPood | Qwell[]: heh /export/software/BBS, actually :P |
| 19:14.05 | justinu | wasim: pcboard was great |
| 19:14.12 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 19:14.14 | Un1x | anyone know of how i can edit pdfs? |
| 19:14.22 | wasim | justinu: yeah, and you got cheap USR too |
| 19:14.24 | Qwell[] | Un1x: khexedit |
| 19:14.56 | SplasPood | wasim: I think they did that for all BBS operators |
| 19:15.04 | *** join/#asterisk shinux__ (n=shinux@196.220.29.101) |
| 19:15.28 | justinu | wasim: ahh, the infamous sysop discount... good times |
| 19:15.40 | Qwell[] | "discount" |
| 19:15.51 | Qwell[] | weren't they still > $50? :P |
| 19:15.52 | asternic | I still have my sysop email account active.. |
| 19:16.11 | justinu | yeah... they were like half price... so it was $500 instead of $1000 |
| 19:16.27 | wasim | yah, back them moving bits down copper was bloody expensive per bit |
| 19:16.44 | PupenoR | Dialplan applications have no return value, right ? |
| 19:17.47 | russellb | well, they do ... |
| 19:18.03 | russellb | but, it's kind of a silly concept to present to users |
| 19:18.06 | SplasPood | I'd like to get my hands on a courier or two... I gave all my old ones way |
| 19:18.10 | SplasPood | s/way/away |
| 19:18.17 | PupenoR | russellb: can I save it to a value ? how ? |
| 19:18.18 | russellb | you just need to know whether they continue, hang up, or jump to a different externsion or priority |
| 19:18.21 | PupenoR | to a variable. |
| 19:18.21 | russellb | you can't |
| 19:18.34 | SplasPood | PupenoR: What's your application? |
| 19:18.46 | SplasPood | wasim: heh |
| 19:18.54 | russellb | a bunch of applications set a STATUS variable on exit, i.e. DIALSTATUS |
| 19:19.04 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 19:19.07 | PupenoR | russellb: oh, the return of an application defines what to do next, doesn't it ? |
| 19:19.09 | Katty | mew. |
| 19:19.13 | russellb | PupenoR: yes |
| 19:19.29 | PupenoR | thanks, that's clarifing. |
| 19:19.48 | russellb | you're welcome |
| 19:19.51 | PupenoR | SplasPood: various... the names wouldn't help, I am developing them. |
| 19:19.59 | SplasPood | not the names |
| 19:20.01 | SplasPood | I mean |
| 19:20.05 | SplasPood | wtf are you trying to do? :) |
| 19:20.59 | *** join/#asterisk klasstek_ (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 19:21.32 | *** join/#asterisk naftali5 (n=naftali5@ool-44c05121.dyn.optonline.net) |
| 19:22.45 | PupenoR | SplasPood: I prefer more polite questions. Thank you. |
| 19:24.20 | wunderkin | wtf are you trying to do, please? :D |
| 19:24.36 | Qwell[] | I demand that you please tell me wtf you're doing, immediately |
| 19:24.54 | Qwell[] | :) |
| 19:24.58 | rob0 | Or else. :) |
| 19:25.02 | Strom_C | I insist on knowing exactly wtf you're doing! Pretty please, with sugar on top :) |
| 19:25.13 | rob0 | ~abuse PupenoR |
| 19:25.21 | jbot | ACTION smacks PupenoR across the face. "Take that, sucker!" |
| 19:25.25 | mercestes | Katty! *hugs* |
| 19:25.31 | russellb | ~lart rob0 |
| 19:25.50 | russellb | be nice to people writing code. |
| 19:25.51 | russellb | :-p |
| 19:26.44 | *** join/#asterisk knathraak (n=zach@151.196.142.242) |
| 19:27.13 | mat2 | Qwell: i did the reinstall, but the chan_zap module is still not there |
| 19:27.52 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 19:28.05 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 19:28.09 | doolph | mat2 did your zaptel got installed succesfully? |
| 19:29.28 | mat2 | no errors with the zaptel installation |
| 19:29.28 | *** join/#asterisk shinux__ (n=shinux@196.220.29.101) |
| 19:31.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:31.37 | knathraak | i am wondering if there is a way to configure a voice mail box so that some administrative options are not available to the user. |
| 19:31.43 | *** join/#asterisk henrique (n=henrique@200-161-80-249.dsl.telesp.net.br) |
| 19:31.47 | Qwell[] | knathraak: You'll have to edit the code |
| 19:32.34 | knathraak | Qwell[]: edit the source and recompile? |
| 19:33.03 | Qwell[] | pretty much |
| 19:34.49 | knathraak | Qwell[]:hmm... not optimal.. okay here's another (related) question...the user has the option of moving messages to other folders besides new and old (including "friends", I believe, and another). are these folders configurable? ones other than new and old are really nonsensical for our situation |
| 19:38.04 | Katty | mercestes: who are you? >.< |
| 19:38.50 | file | Katty: I don't have a speed dial entry for you on my PBX!!! |
| 19:39.18 | Katty | rut roh. |
| 19:39.28 | *** join/#asterisk santiago (n=santiago@190.24.177.171) |
| 19:39.31 | Katty | do you want iax? |
| 19:39.32 | file | that makes me sad |
| 19:39.33 | file | sure |
| 19:39.39 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 19:45.07 | *** join/#asterisk Scrye (i=1127@god.damnit.us) |
| 19:45.21 | *** join/#asterisk _cleric_ (n=dacleric@p54822D1B.dip0.t-ipconnect.de) |
| 19:45.33 | Scrye | is there a channel that could help me setup a sangoma T3 card? |
| 19:45.51 | Scrye | more specifically, how wanpipe?.conf is parsed |
| 19:46.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:47.23 | shido6 | screw it..... what are you having trouble with? |
| 19:47.52 | Scrye | im trying to figure out how to make multiple interfaces that use only partial channels |
| 19:48.14 | Scrye | i have a T3 card and the telco is going to bridge all my T1s into channels on the T3 |
| 19:48.33 | Scrye | so my first t1 will be on channels 1-24, second on 25-48 and so on |
| 19:48.42 | shido6 | um... |
| 19:48.43 | Scrye | i just cant find an example to start from |
| 19:48.48 | shido6 | which t3 card do you have? |
| 19:48.50 | Scrye | sangoma |
| 19:48.53 | Scrye | a301 |
| 19:48.53 | shido6 | url? |
| 19:48.57 | shido6 | ok |
| 19:49.06 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 19:49.16 | Scrye | http://www.sangoma.com/datasheets/p_aft-et3-specs |
| 19:49.28 | shido6 | that supports CLEAR CHANNEL |
| 19:50.06 | brad_mssw | ... why do I think I remember hearing that particular card was not supported by asterisk a while ago (6+mos) ... things could have changed by now though |
| 19:50.31 | mat2 | Qwell: is it possible there is something missing in asterisk1.4beta that is required to build the chan_zap module? |
| 19:50.43 | Scrye | shido6: and that means? |
| 19:51.48 | SplasPood | PupenoR: Thats what the :) implied, but I'm sorry I even bothered to care about you problem too, THanks! |
| 19:51.49 | brad_mssw | Scrye: http://lists.digium.com/pipermail/asterisk-users/2006-December/174175.html |
| 19:52.00 | shido6 | 24 ds0's... 1.34 Mbps |
| 19:53.04 | *** join/#asterisk henrique_ (n=henrique@201-27-71-35.dsl.telesp.net.br) |
| 19:53.06 | Scrye | ok now i know the ISP is on glue |
| 19:53.13 | Katty | file: i love you! |
| 19:53.15 | file | zomg, I Just talked to Katty |
| 19:53.16 | *** join/#asterisk Lannister (i=Lanniste@adsl-63-200-88-82.dsl.scrm01.pacbell.net) |
| 19:53.26 | file | Katty: <3 |
| 19:53.27 | Scrye | they specifically mentioned clearchannel |
| 19:53.29 | Katty | file: was it everything you thought it would be? |
| 19:53.34 | file | Katty: yes!!! |
| 19:53.37 | Lannister | hey hey |
| 19:53.57 | Lannister | my work has me researching voip vs analog, but i cant find much comparative info on google, just sales pitches |
| 19:54.01 | Katty | file: yay. |
| 19:54.15 | Lannister | we have a system of 20 analog phones ...i dont know how many lines exactly and we are looking to replace the system |
| 19:55.19 | Lannister | we had a homebrew pbx at my last work and there were a fuckton of echo and clarity issues at first |
| 19:55.47 | Lannister | is that common with voip? |
| 19:56.08 | Qwell[] | echo is only possible on analog. packets don't change |
| 19:56.19 | Katty | echo is horror )= |
| 19:56.29 | Katty | it makes me all sad inside. |
| 19:56.31 | Qwell[] | You *can* however create echo in some circumstances, with a bad handset or whatever, where it is analog |
| 19:56.31 | Lannister | oh it was voip, and there was an echo |
| 19:57.14 | Lannister | hmm...if we were going from analog to voip, what is involved? |
| 19:57.23 | rob0 | /exec -out echo "Cheer up Katty. :)" |
| 19:57.35 | Katty | rob0: that echo is okay. |
| 19:58.46 | *** join/#asterisk DA_ (n=da@72.20.33.113) |
| 19:58.49 | brad_mssw | Lannister: depends on how much $$ you want to spend, and if you plan on giving everyone voip phones, or if you plan on keeping your existing analog phones and bringing them back into a TDM2400P or similar |
| 19:59.11 | Qwell[] | rob0: ut: command not found |
| 19:59.23 | *** join/#asterisk grabeez (n=gaving@neptune.enter.net) |
| 19:59.39 | *** join/#asterisk richmanmM (n=richmanm@70.89.184.1) |
| 19:59.39 | rob0 | Sorry, I guess it's an irssi(1) thing. |
| 20:00.24 | Lannister | can we keep our existing phones and somehow replace the whole controlling box thingy thats in the phone closet...i have no idea what im talking about obviously |
| 20:00.46 | Qwell[] | Lannister: what type of phones are they? |
| 20:00.52 | Lannister | at&T MLS-12 |
| 20:00.56 | Qwell[] | not analog? |
| 20:00.58 | Lannister | analog |
| 20:01.07 | Qwell[] | then you should be able to, sure |
| 20:01.15 | Qwell[] | just need some way to interface with them |
| 20:01.27 | Qwell[] | how exactly are they connected to the old pbx? |
| 20:01.30 | Lannister | supposedly the phone setup is old and almost nobody will work on it |
| 20:01.33 | Lannister | so they want a "new" one |
| 20:01.34 | Qwell[] | straight into it, or to a channelbank? |
| 20:01.35 | *** part/#asterisk Scrye (i=1127@god.damnit.us) |
| 20:01.46 | Lannister | dunno, how can i check |
| 20:01.47 | Qwell[] | if channelbank, you can reuse it |
| 20:01.49 | Qwell[] | go look, heh |
| 20:01.52 | Lannister | hehe |
| 20:01.53 | Lannister | ok sec |
| 20:02.04 | Lannister | btw, looking for bids |
| 20:02.05 | JT | well it sounds like they want to change all the handsets anyway |
| 20:02.11 | Qwell[] | is there an external box that they connect to, which connects over T1 to the old pbx? |
| 20:02.28 | Lannister | sec, i will go describe the phone box and/or take pics |
| 20:03.12 | brad_mssw | Lannister: you're probably wanting a TDM2400P with a bunch of FXS modules to plug your analog phones into, so you don't have to replace those |
| 20:03.24 | Qwell[] | brad_mssw: not if it's connected to a channel bank |
| 20:03.24 | brad_mssw | Lannister: http://www.digium.com/en/products/hardware/tdm2400p.php |
| 20:03.32 | Qwell[] | then he can just get a t1 card |
| 20:03.58 | JT | eh |
| 20:04.07 | JT | i think they want a "new" phone system |
| 20:04.14 | JT | not one that appears the same to users |
| 20:04.33 | brad_mssw | Qwell[]: hmm, could be connected to a channel bank I guess, but his old system was probably pretty expensive if that's the case |
| 20:05.28 | *** join/#asterisk mcrichmanM (n=richmanm@70.89.184.1) |
| 20:06.34 | *** join/#asterisk soylentgreen (n=fgast@bb1-fe0.only640k.org) |
| 20:06.35 | *** part/#asterisk psion (n=123@22.80-202-239.nextgentel.com) |
| 20:06.36 | Lannister | k, got pics sec |
| 20:07.47 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 20:08.32 | naftali5 | hang on that phone is an avaya/att digital phone propietary |
| 20:08.37 | naftali5 | he'll need new phones |
| 20:08.45 | Lannister | might |
| 20:08.49 | Lannister | some old wierdness |
| 20:08.56 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 20:09.32 | Qwell[] | then yeah, you're screwed |
| 20:09.53 | *** join/#asterisk te_lo_meto_mami (n=te_lo_me@8.10.2.50) |
| 20:10.55 | Lannister | sorry, they are cell phone pix: http://users.cwnet.com/vvortex3/phone/ |
| 20:11.09 | naftali5 | on a happier note, they are reselling on ebay for about $10-$40 each |
| 20:11.23 | Lannister | cool |
| 20:11.23 | Qwell[] | naftali5: $20 says they were about $400 each new :) |
| 20:11.47 | Qwell[] | yikes, those are some mighty ugly phones |
| 20:11.50 | naftali5 | they will not work with asterisk/ so $10 is $10 :( |
| 20:11.53 | Lannister | so i dont know what all that stuff is going on in the closet...im the network admin here too lol, i just know zero about phone systems |
| 20:11.56 | Qwell[] | Lannister: yeah, you need new phones it looks like |
| 20:12.07 | Lannister | in order to get any modern phone system? |
| 20:12.26 | Lannister | bastards |
| 20:12.27 | Qwell[] | in order to get anything besides the AT&T pbx you have. Those are apparently proprietary phones |
| 20:12.40 | Lannister | sounds like some microsofty nonsense |
| 20:12.52 | Qwell[] | let me just say though... nice touch with the uber-cool powerstrip in lower.jpg |
| 20:13.05 | Lannister | hehe |
| 20:13.14 | Strom_C | Don't worry; I'm sure Avaya will be more than happy to sell you a new PBX for some ungodly amount of money |
| 20:13.17 | Qwell[] | gotta <3 the lack of surge protector there |
| 20:13.28 | JT | most office phones are proprietary, traditionally |
| 20:13.34 | Qwell[] | file: I win |
| 20:13.38 | Lannister | so ...voip or analog? |
| 20:13.48 | Strom_C | Lannister: the best way to go these days is voip |
| 20:13.49 | Qwell[] | Lannister: if you're gonna buy new phones, I say join the 21st century |
| 20:13.50 | JT | good luck even buying analogue |
| 20:13.52 | Lannister | we are looking for best overall value of cost vs quality, but really dont need any odd voip features |
| 20:14.02 | JT | your other option would be propretary digital, Lannister |
| 20:14.03 | file | eep |
| 20:14.07 | Qwell[] | Lannister: new phones can be had for...$200ish |
| 20:14.09 | file | sure you don't want the PRI card instead? |
| 20:14.11 | Qwell[] | for very decent phones |
| 20:14.12 | JT | almost no-one will sell you an analogue pabx that size |
| 20:14.29 | Qwell[] | Lannister: There is one caveat though |
| 20:14.38 | Strom_C | Lannister: have a look at the cisco 7940/7941 or the polycom ip430 |
| 20:14.43 | Qwell[] | Lannister: If you aren't wired for ethernet properly...there is an additional cost there |
| 20:14.44 | brad_mssw | Lannister: wouldn't go analog these days |
| 20:14.46 | hmmhesays | hmm isn't there registry settings for wine? |
| 20:14.51 | Qwell[] | hmmhesays: winecfg :p |
| 20:14.52 | Lannister | we are wired for ethernet |
| 20:15.00 | Qwell[] | hmmhesays: or drive_c/../*.reg |
| 20:15.10 | Lannister | so the opposite of analog being voip? |
| 20:15.14 | JT | no |
| 20:15.16 | JT | damnit |
| 20:15.19 | Qwell[] | not "opposite" |
| 20:15.24 | JT | the apposite of analogue is digital |
| 20:15.26 | Lannister | sorry, wrong word choice |
| 20:15.31 | JT | voip is packetised digital |
| 20:15.37 | Lannister | oh ok |
| 20:15.38 | hmmhesays | Thanks |
| 20:15.44 | brad_mssw | Lannister: well, i'd go for standardized voip, like SIP |
| 20:15.50 | file | Qwell[]: you are SO linear |
| 20:15.52 | Lannister | see im not understanding how this will actually work because like ...if its voip where does the bandwidth come from |
| 20:16.02 | Lannister | and how is it routed out to non internet lines |
| 20:16.06 | Lannister | err analog lines |
| 20:16.11 | JT | voip runs over ethernet |
| 20:16.21 | JT | how you connect to the phone network is up to you |
| 20:16.25 | JT | it can be over digital phone lines |
| 20:16.31 | JT | ananlogue phone lines |
| 20:16.33 | Lannister | so its only voip inside of the building |
| 20:16.37 | JT | voip over Internet |
| 20:16.40 | Strom_C | Lannister: you could always hire a consultant |
| 20:16.41 | JT | it's how you set it up |
| 20:16.45 | Lannister | well we will |
| 20:16.51 | Lannister | we just want to get raped as little as possible |
| 20:16.57 | hads | heh |
| 20:17.01 | Lannister | maybe even have some lube on hand |
| 20:17.29 | Lannister | we're in sacramento, ca |
| 20:17.36 | Lannister | we have a 20k quote |
| 20:17.39 | Strom_C | article V subsection 3C of my contract is labeled "MINIMIAL RAPE" |
| 20:17.45 | Lannister | that seems rediculous |
| 20:17.45 | Strom_C | Lannister: I'm in los angeles |
| 20:17.49 | Lannister | i'll learn how to do it for 20k lol |
| 20:17.57 | hads | That's not too bad. |
| 20:17.58 | Qwell[] | 20k? how many users? |
| 20:18.02 | Lannister | 20ish |
| 20:18.16 | *** join/#asterisk dasenjo (n=dasenjo@190.24.177.171) |
| 20:18.19 | Qwell[] | (20 * 200) + 3000 + consultant |
| 20:18.43 | Lannister | we actually have 17 employees but want a little bit of room |
| 20:19.17 | hads | That's the thing about ethernet, it's just ethernet so more room is easy. |
| 20:19.22 | Qwell[] | Strom_C: how does it usually work? % of total price of hardware, or flat fee, or? |
| 20:19.37 | Strom_C | Qwell[]: i'm not sure what you mean |
| 20:19.40 | Strom_C | my consulting fees? |
| 20:19.42 | Qwell[] | consultant fees |
| 20:19.45 | Strom_C | hourly |
| 20:19.50 | Qwell[] | gotcha |
| 20:20.14 | Qwell[] | so, yeah, can't be much more than $10k I'd imagine... unless you had to fly somebody out |
| 20:20.24 | Lannister | so like...some sort of voip controlling box would go in that closet ...and we'd set up all of our 20 odd voip phones to talk to it...then from that point i dont really understand how calls get out, or get routed when coming in and such |
| 20:20.26 | Qwell[] | assuming the phones are $200, and the server is $3000 :P |
| 20:20.42 | Strom_C | the last proposal I did for a 17-station client came out at about $10k including consulting fees |
| 20:20.48 | Qwell[] | though, you probably are connecting via T1's... |
| 20:21.00 | Qwell[] | for 17 users...I'd hope you have a single T |
| 20:21.07 | Lannister | do we? |
| 20:21.12 | JT | Lannister: using hardware it can connect via PRIs (T1s) or analogue lines |
| 20:21.16 | Lannister | i dunno what we have in regards to the phone system |
| 20:21.18 | Qwell[] | if not, your consultant would probably tell you to, and you'd save money in the long run :D |
| 20:21.29 | Qwell[] | Strom_C: pretty accurate so far? heh |
| 20:21.30 | JT | or voip too |
| 20:21.34 | *** join/#asterisk Dr-Linux|home (n=Dreamer@202.59.73.131) |
| 20:21.43 | Lannister | can you tell if we are using T1s or analog lines from teh pics at http://users.cwnet.com/vvortex3/phone/ ? |
| 20:21.46 | Strom_C | Qwell[]: pretty much |
| 20:21.58 | Qwell[] | Lannister: nah, didn't see anything obvious. it's just a cable |
| 20:22.03 | Dr-Linux|home | anybody ever use OpenSpeech voice recognition system with asterisk? |
| 20:22.11 | Lannister | im not sure how to tell |
| 20:22.23 | Qwell[] | unless the ortronics is a csu/dsu or something... who knows |
| 20:22.27 | Lannister | so asterisk is basically a way to avoid buying the prebuilt $3000 controlling box? |
| 20:22.31 | JT | Lannister: how many phones do you have? |
| 20:22.36 | Lannister | 17 offices |
| 20:22.41 | Lannister | but we probably want 20 phones |
| 20:22.43 | Qwell[] | 17 OFFICES? |
| 20:22.48 | Qwell[] | That's much different than 17 users |
| 20:22.50 | Lannister | no 17 individual fofices |
| 20:22.55 | Qwell[] | in the same location? |
| 20:22.56 | Lannister | yes |
| 20:22.58 | Katty | Qwell[]: :< |
| 20:22.59 | Qwell[] | okay, heh |
| 20:23.04 | Qwell[] | because that would've seriously changed everything |
| 20:23.05 | *** join/#asterisk dacleric (n=dacleric@p548219A5.dip0.t-ipconnect.de) |
| 20:23.08 | *** join/#asterisk backblue (n=moo@87-196-103-134.net.novis.pt) |
| 20:23.11 | JT | how many phones do you have NOW? |
| 20:23.17 | Lannister | probably like 18 |
| 20:23.29 | JT | yeah you have maybe a dozen analogue lines to the PSTN |
| 20:23.35 | JT | looking at the picture of the PBX |
| 20:23.38 | Lannister | ok |
| 20:23.45 | Lannister | so we can support like 12 concurrent calls? |
| 20:23.51 | Lannister | is that how it works? |
| 20:23.56 | JT | very rough estimate |
| 20:24.01 | Lannister | k |
| 20:24.02 | JT | going on cables into linecards |
| 20:24.03 | Qwell[] | depends on how you're connected to the PSTN... |
| 20:24.04 | *** join/#asterisk converx (n=locid@206-248-176-51.dsl.teksavvy.com) |
| 20:24.04 | JT | i could be wrong |
| 20:24.07 | file | "it all depends" |
| 20:24.25 | Lannister | k |
| 20:24.26 | JT | but there's a lot more than 18 connections there |
| 20:24.26 | Lannister | this stuff is neat, i wish i could set it up but the company cant afford to pay for my fuckups |
| 20:24.26 | Qwell[] | IF you have a dozen analog lines from the telco, you'll save a bit of money by switching to a partial T1... |
| 20:24.27 | converx | how to enable odbcstorage in asterisk 1.4 ? |
| 20:24.30 | hads | Find your phone bill |
| 20:24.36 | Qwell[] | converx: in menuselect, look under voicemail options |
| 20:24.41 | Katty | file: my laptop keeps turning itself off :< |
| 20:24.42 | Qwell[] | enable ODBC_STORAGE in there |
| 20:24.50 | converx | thx. |
| 20:24.50 | JT | was thinking of phone bill |
| 20:25.15 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:25.24 | Lannister | i'll go get the bill |
| 20:25.26 | *** join/#asterisk graabein (n=gunnar@nat.sigmasoft.com) |
| 20:26.37 | file | Katty: :( |
| 20:27.09 | Katty | file: i think it's hottttness is a smidgen much. |
| 20:27.16 | Katty | file: or it's goin on a power trip |
| 20:28.19 | *** join/#asterisk xnon (n=xnon@200.8.5.123) |
| 20:28.34 | hmmhesays | heh |
| 20:29.05 | roguebug | what's the difference in 1.2.13 between the normal tarball and the one with "netsec" in the name? |
| 20:30.27 | Qwell[] | the netsec branch is for use with hardware that supports midcom - ie Ranch Networks firewalls |
| 20:31.07 | Qwell[] | firewalls? routers? whatever they are |
| 20:31.08 | Lannister | i got the bill, trying to figure out at&t's statement |
| 20:31.48 | roguebug | ok thx Qwell[] |
| 20:33.23 | Lannister | wtf, doesnt say anything about the number of lines or whether we have a T! |
| 20:33.30 | Qwell[] | Lannister: PRI? |
| 20:33.37 | Lannister | PRI? |
| 20:33.42 | wunderkin | it just says please pay $4242.42? |
| 20:33.43 | Qwell[] | does it say "PRI" anywhere? |
| 20:33.47 | JT | Lannister: does it charge line rental per phone number? |
| 20:34.08 | Lannister | it only has 2 phone numbers |
| 20:34.14 | naftali5 | Lannister: look at the tag on the fourth box from the left on the phone system, and post |
| 20:34.20 | Qwell[] | Lannister: can more than 2 people make a call at once? |
| 20:34.26 | Lannister | yes |
| 20:34.26 | JT | maybe you only have 2 phone lines, lol |
| 20:34.31 | Qwell[] | then it's at least a partial PRI |
| 20:34.32 | Lannister | many people can call at once |
| 20:34.36 | Qwell[] | undoubtedly |
| 20:34.47 | Lannister | what is a pri? |
| 20:34.47 | Qwell[] | and by "call", I do mean outside of your office |
| 20:34.53 | Lannister | yes |
| 20:35.03 | JT | primary rate interface |
| 20:35.05 | JT | t1/e1 |
| 20:35.06 | Lannister | oh ok |
| 20:35.06 | converx | which file is menuselect defined? |
| 20:35.08 | JT | digital circuit |
| 20:35.14 | Lannister | this is neat shit |
| 20:35.47 | JT | Lannister: does the bill have the name of any of the line services being paid for? |
| 20:35.52 | Lannister | so i suppose if it were individual lines it would bill per number |
| 20:35.58 | Lannister | so we have a partial T1 then? |
| 20:36.00 | JT | i would've thought it'd say something |
| 20:36.13 | mercestes | Hey, if I wanted to edit the "voicemail notification" email, where would I do that? |
| 20:36.17 | Lannister | not really, just talks about taxes and shit |
| 20:36.31 | JT | you sure it's your main bill? |
| 20:36.53 | Lannister | apparently we pay like $200/mo to at&t if that helps |
| 20:37.02 | JT | umm seems low |
| 20:37.17 | JT | maybe it's just your fax or modem line bill :P |
| 20:37.30 | naftali5 | file /etc/asterisk/voicemail.conf |
| 20:37.32 | backblue | $200 mo? bah |
| 20:37.32 | Lannister | no, it lists the main line |
| 20:37.38 | JT | hmm |
| 20:37.55 | backblue | i tought you would say something like $15000/mo |
| 20:37.56 | Lannister | fourth box from the left, sec |
| 20:38.25 | *** join/#asterisk ang (n=ang@caracas-1031.adsl.interware.hu) |
| 20:39.40 | file | eep |
| 20:40.26 | Lannister | lucent 206e module |
| 20:40.34 | *** join/#asterisk robin__sz (n=robin@rapid2.gotadsl.co.uk) |
| 20:42.06 | *** join/#asterisk Winkie (n=urmom@host86-130-187-123.range86-130.btcentralplus.com) |
| 20:42.44 | PupenoR | Does the console statment in logger.conf refeer to the console when you run asterisk -c or to asterisk -r ? |
| 20:42.46 | Lannister | so ok...having voip or analog phones doesnt really make much difference internally? |
| 20:42.50 | *** join/#asterisk penghb (n=nnnpengh@202.108.130.138) |
| 20:43.13 | naftali5 | seems you only have analog lines around 6-8 |
| 20:43.14 | JT | weird setup with that pabx |
| 20:43.19 | monsted | Lannister: only for the features you can usually tack onto the ip pbx |
| 20:43.23 | JT | but that module does 6 phone extensions, and 2 lines |
| 20:43.48 | JT | so if all modules are like that, then you have 5 lines |
| 20:44.02 | naftali5 | look at his pic seems to be only 2 and 2 in that one |
| 20:44.08 | naftali5 | not all his modules are the same |
| 20:44.39 | JT | count the ones up the top row |
| 20:44.41 | *** join/#asterisk khris (n=nnnnnnkr@mrtg.sisgroup.com.au) |
| 20:45.16 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 20:45.26 | Lannister | ok... see i dunno i just had a bad experience with a custom asterisk box at my last job, it had echo and clarity problems that took awhile to sort out |
| 20:45.33 | Lannister | maybe it was bad hardware and cheap phones ...dunno |
| 20:48.08 | Lannister | ok so im starting to visualize this |
| 20:48.35 | Lannister | 18 voip phones... some voip controlling box ...from there im not sure what controls the interface to the external analog lines |
| 20:49.16 | monsted | just get a cisco router and use call manager express ;) |
| 20:49.54 | Lannister | really? |
| 20:51.28 | brad_mssw | Lannister: get quality voip phones which can be provisioned easily, like polycom ip301 and ip501's for higher-end personnel ... then all you need is to figure out how to bring the external lines into asterisk ... if they're truly analog/pots/pstn, you'd need something like a digium TDM400P or TDM2400P depending on number of lines, otherwise, if it's a partial T1, use a Sangoma T1/E1 card |
| 20:51.35 | JT | ok, my line estimate is now 8 analogue lines |
| 20:52.02 | JT | brad_mssw: nothing stopping them from changing the form of the inbound lines if economical |
| 20:52.02 | backblue | Lannister: dont loose your money, pay someone to do it, or buy a solution to someone. |
| 20:52.13 | brad_mssw | JT: very true |
| 20:52.16 | JT | digital lines should always be preferred over analogue |
| 20:52.49 | backblue | JT: analogue lines are very good, for adsl. |
| 20:53.00 | backblue | much better than rdis |
| 20:53.14 | JT | hrm, the pics aren;t very clear, but maybe there's only 4 lines?! |
| 20:53.26 | Lannister | brad: k ...but i've seen asterisk config before |
| 20:53.27 | JT | you can always have an analogue line or two for fax and modem + adsl |
| 20:53.45 | Lannister | im more interested in an easily configurable in an afternoon solution |
| 20:54.01 | JT | isdn is far superior for phone, but his numbers may not justify it |
| 20:54.02 | JT | Lannister: |
| 20:54.07 | JT | Lannister: "you're dreaming" |
| 20:54.08 | Lannister | with like nice fuzzy web interfaces |
| 20:54.11 | Lannister | hehehe |
| 20:54.14 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 20:54.23 | JT | setting up a phone system for an entire office does not take one afternoon |
| 20:54.28 | brad_mssw | Lannister: sure, depends on what features you want though, and what experience level you have ... if you want a full automated attendant with a deep menuing system, that could take a while to create |
| 20:54.38 | Lannister | dont need all that |
| 20:54.45 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 20:54.49 | Lannister | ability to take calls and make outgoing calls, voicemail and ability to transfer |
| 20:54.53 | Lannister | thats it |
| 20:55.53 | Lannister | if there were some way i could do this myself, i'd gladly do it just for the learning experience if there is some prebuilt solutions that I could just plug in with reasonably simple config |
| 20:56.02 | brad_mssw | Lannister: i'd say your biggest config issue is your incoming phone lines ... no idea how big a company you are or how much traffic you do, but you could always look into using a voip provider (junction networks or vonage business plus), so you don't have to worry about the hardware end |
| 20:56.13 | shido6 | you can do it yourself |
| 20:56.45 | Lannister | well i'd prefer to interface with the existing analog lines |
| 20:57.02 | Lannister | well ok, we dont know if they are analog or digital but yeah |
| 20:57.06 | *** join/#asterisk RoyK (n=roy@ti211310a080-15179.bb.online.no) |
| 20:57.30 | FuriousGeorge | i hate when this happens... zt hook failed: |
| 20:57.30 | brad_mssw | Lannister: if you're looking to do a proof-of-concept before jumping head-first, use Asterisk@Home/trixbox/whatever-its-named-now ... |
| 20:57.36 | Lannister | im gonna go stare at it for awhile, then go home adn get some left over curry chicken |
| 20:57.43 | FuriousGeorge | i havent seen that since i started restarting * daily |
| 20:57.53 | FuriousGeorge | i wonder if the cable is bad |
| 20:58.27 | nays85 | does asterisk still cache dns lookups from *.conf indefinitely? |
| 20:59.45 | converx | can someone help with enabling odbcstorage in asterisk 1.4? |
| 21:00.51 | *** join/#asterisk CleanerX (n=nix@p54A39C4C.dip0.t-ipconnect.de) |
| 21:02.23 | RoyK | converx: realtime? |
| 21:02.45 | FuriousGeorge | any one know if this error "zt hook failed:" when a zap channel attempts to dial out, could be caused by a bad cable? there is a dialtone the whole time, but DTMF does not appear to be recognized |
| 21:02.47 | converx | with voicemssages table |
| 21:03.05 | *** part/#asterisk tparcina (n=tparcina@195-29-117-97.adsl.net.t-com.hr) |
| 21:03.08 | FuriousGeorge | and beeps come in over the line, this is a second hand account unfortunately |
| 21:07.20 | JT | brad_mssw: i think his original idea to get a onsultant was better than the using trixbox idea |
| 21:07.46 | brad_mssw | JT: yeah, depends on how much $$ he wants to spend ... |
| 21:08.03 | JT | he will spend less with a consultant |
| 21:08.09 | JT | a good one |
| 21:09.33 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 21:11.42 | converx | the file example_menuselect-tree -- what does it do? |
| 21:12.20 | *** join/#asterisk alamantia (i=anthonyl@nat/digium/x-2b1b40cb05596a5f) |
| 21:18.32 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 21:18.35 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 21:19.33 | variable_office | people are having problems hearing me on my asterisk system. I can hear them fine. by logic, that should be a problem with upload, but download is the one that is ever full. any suggestions on the problem? |
| 21:19.43 | variable_office | this is g711, and asterisk -r reports no errors |
| 21:20.29 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:20.36 | Cableguy | Hi all, just had a question about hardware requirements. I have 1 Shaw Digital Phone (VOIP) line feeding my house and 10 ethernet lines running throughout my house. What hardware would I need to install in my asterisk box to allow for proper functionality? I currently only have regular phones in the house, not VOIP phones. |
| 21:21.56 | Cableguy | Currently the lines are punched down on a distribution block. |
| 21:23.43 | *** join/#asterisk chkdsk (n=chkdsk@62.159.49.140) |
| 21:23.57 | Cableguy | I'm a little confused on what I use to input my Shaw line into and what is used to leave the asterisk box to go to the distribution block. It's not just a dialup modem is it? |
| 21:24.36 | Cableguy | <PROTECTED> |
| 21:24.36 | Cableguy | <Cableguy> Currently the lines are punched down on a distribution block. |
| 21:24.53 | Cableguy | opps sorry.. I'm new to irc. :) |
| 21:27.06 | monsted | don't repeat the question - we saw it the first time and someone will probably answer at some point |
| 21:27.09 | Cableguy | Yeah, I'm appologize about that.. I'm still learning IRC too. Thanks. :) |
| 21:27.41 | *** join/#asterisk Aximas (n=Aximas@gw-100.selfnet.de) |
| 21:27.45 | monsted | if i read your question right, you need FXS ports |
| 21:28.35 | Cableguy | Is that a PCI card that I would install into my asterisk server? |
| 21:28.47 | monsted | lots of options |
| 21:29.28 | monsted | plain pci cards, network-attached gateways, E1/T1 pci cards and channel banks |
| 21:30.04 | mrichmanM | If i have added extensions in the config files and reloaded asterisk why would i still need to use database put in the asterisk cli for the same extensions? |
| 21:32.42 | Cableguy | Well, the scope of this project is to create a office presence for my home business. I would like to take the one VOIP line feeding my house and have asterisk be able to split the house in two. Office lines and home lines. If I have 10 lines in the house, would I be looking for a card that has one input and 2 outputs? |
| 21:32.50 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:34.21 | *** join/#asterisk blitzrage (n=blitzrag@dsl017-122-217.mci1.dsl.speakeasy.net) |
| 21:35.25 | Cableguy | I'm learning the types of terminology when it comes to phone technology. Am I explaining what I'm wanting to do clearly enough? |
| 21:35.25 | monsted | since you're getting IP from the telco, you don't need an "input" |
| 21:36.16 | monsted | in analogue terms, an FXO port is something that connect to a phone line and an FXS ports connects to phones |
| 21:36.27 | monsted | you'd want either two or more FXS ports |
| 21:38.18 | mrichmanM | Is there a way to have asterisk automatically load its database from the config files? |
| 21:39.26 | blitzrage | mrichmanM: you mean loads its configs from the database? |
| 21:39.43 | Cableguy | So in this situation, it does not matter whether my service is a traditional phone line or a VOIP line? |
| 21:39.54 | monsted | Cableguy: it does |
| 21:40.08 | monsted | Cableguy: if your service is a POTS line, you need an FXO port |
| 21:40.26 | mrichmanM | I mean that I can't use an extension until i have done database put multiple times |
| 21:40.29 | *** join/#asterisk ztel (n=scott@70.103.238.2) |
| 21:40.51 | mrichmanM | and if i have already added it to extensions.conf why do i need to do a database put |
| 21:41.11 | blitzrage | mrichmanM: because you need to tell the DB about it |
| 21:41.34 | mrichmanM | is there a way to have it do it automatically |
| 21:41.36 | ztel | Hello everyone. Has anyone had any luck configuring BLF on polycom 601's? |
| 21:41.45 | Cableguy | got it. No I do not have a POTS line. I have VOIP supplying service to the house now. So I would require a FXS ports, right? |
| 21:41.45 | blitzrage | mrichmanM: nope |
| 21:41.54 | blitzrage | mrichmanM: at least not from what I can tell you are trying to do |
| 21:42.11 | monsted | Cableguy: FXS ports for the phones, yes |
| 21:42.12 | blitzrage | makes no sense to me really |
| 21:42.27 | monsted | Cableguy: one per extension, not necessarily one per phone |
| 21:42.57 | *** part/#asterisk blitzrage (n=blitzrag@dsl017-122-217.mci1.dsl.speakeasy.net) |
| 21:42.59 | mrichmanM | the scenario is that i add the extension in extensions.conf but before i can use that extension i also must add the info to the asterisk db using database put |
| 21:43.07 | mrichmanM | it seems redundant |
| 21:43.54 | Cableguy | Monsted, how do I put your name in front of my comment like you do with mine? |
| 21:44.03 | mrichmanM | is there a better way for me to add extensions |
| 21:44.07 | monsted | err, type it in? :) |
| 21:44.23 | Cableguy | lol |
| 21:44.32 | monsted | Cableguy: my client uses nick completion by hitting "cab<tab>" |
| 21:44.48 | Cableguy | ahh got it. thanks :) |
| 21:45.22 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 21:47.19 | mrichmanM | am I doing something wrong or am i just whining? |
| 21:49.52 | *** join/#asterisk Aximas (n=Aximas@gw-100.selfnet.de) |
| 21:50.33 | Cableguy | monsted: ok back to the question - so if I wanted 2 extensions in the house ( and I punched down 5 lines per extension onto 2 seperate phone blocks ), then the card would need 3 ports on it ( card would probably have 4 but I would only need 3), one for the incoming VOIP service and 2 for the outgoing extensions. Am I understanding this right? |
| 21:52.51 | monsted | no |
| 21:52.54 | mrichmanM | blitzrage: does that make sense? |
| 21:53.02 | monsted | the incoming voip "line" doesn't use a port |
| 21:58.13 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 21:58.43 | CunningPike | Cableguy: The 'incoming' port would actually be an ethernet port to connect the VOIP side of your gateway to the analog side |
| 21:59.19 | CunningPike | Cableguy: Or, if you use a card, you only need 2 FXS ports |
| 21:59.48 | Cableguy | ok.. now this is where I was getting confused. How does the asterisk server interact with your outside service? For example, if somebody calls your place, how does the asterisk server answer the line? I'm sorry for such a simple question, I must not be understanding the topology. I install VOIP phones in residential settings for a living but do not understand some of the more advanced stuff. I hope you can be patient with me. |
| 22:01.53 | Cableguy | CunningPike: AHH ok there is a difference with Shaw VOIP and every other VOIP service out there. Let me explain: |
| 22:02.33 | Cableguy | Coaxial cable connects to the back of the VOIP module |
| 22:02.57 | ztel | Any of you fine folks have any experience with polycom? More specifically has anyone here configured polycom to use blf's? I know it can be done but the docs are sukcing and polycom is sucking harder. |
| 22:03.04 | ztel | support wise |
| 22:03.17 | mrichmanM | blitzrage: Sorry my wireless disconnected so i missed any response you might have sent is there a better way for me to add extensions? |
| 22:03.19 | CunningPike | ztel: All you need are hints, and then set up watched buddies on your phones |
| 22:04.22 | CunningPike | ztel: It's quite simple - exten => 1234,hint,SIP/1234. Then, set up a directory entry in the watching phone with a contact of 1234 and set Watched Buddy to yes |
| 22:05.15 | Cableguy | A small RJ11 jumper leaves the VOIP module into a transition block. A CAT3 line leaves the transisiton block and terminates to the house phone block. This VOIP service doesn't use an residential internet connection to operate. |
| 22:06.05 | CunningPike | Cableguy: Ah. I don't know how you'd do that - I guess you'd have to treat your Shaw service like POTS (from a pure cabling perspective) and connect an FXO port to it, but there' |
| 22:06.21 | CunningPike | Cableguy: There's no guarantee that it would behave like POTS |
| 22:06.36 | converx | how do i enable odbcstorage in asterisk 1.4? |
| 22:06.44 | ztel | Okay, so I have hints working, BLF is setup and working on snom's and xten, so hints are good. But.. so you are refereing to the phonedirectory config then right? Not the reg under phone1.cfg? |
| 22:07.09 | CunningPike | Cableguy: Why do you need asterisk in the middle? Why not just cable it like you say? |
| 22:07.41 | CunningPike | ztel: Yes - or do it right on the phone |
| 22:07.49 | Cableguy | CunningPike: May we go private? |
| 22:07.56 | CunningPike | Cableguy: Sure |
| 22:07.59 | ztel | okay, I was going about it all wrong then :)... thanks! |
| 22:08.51 | CunningPike | ztel: No prob - hope you get it working |
| 22:09.10 | CunningPike | Cableguy: Did you /identify ? |
| 22:11.18 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 22:13.18 | Cableguy | um no, what is that? |
| 22:13.32 | joe | puzzled: ping |
| 22:14.07 | puzzled | joe: pong |
| 22:14.32 | Cableguy | CunningPike: What does the /identify command do? |
| 22:14.56 | mrichmanM | is there a way to export the database and reimport it? |
| 22:15.04 | mrichmanM | into asterisk |
| 22:15.18 | CunningPike | Cableguy: It identifies you as the legitimate holder of your nick on freenode - you can't /msg without it |
| 22:15.50 | *** join/#asterisk groogs____ (n=chatzill@cbl-66-102-80-249.wtccommunications.ca) |
| 22:16.07 | Cableguy | CunningPike: No I have not doen that.. how do I complete that task? |
| 22:16.20 | CunningPike | Did you register your nick? |
| 22:17.10 | zapp-branigan | hi when i make a internal call in iax i hear a echo who can remove the echo ? i don't use zaps |
| 22:17.12 | CunningPike | Cableguy: Type '/msg nickserv register <password>' where <password> is your chosen password |
| 22:17.21 | CunningPike | Cableguy: Leave out the quotes, natch |
| 22:18.15 | Cableguy | CunningPike: ok done |
| 22:18.35 | Crescendo_ | How can I configure a Cisco IP Phone on the LAN so when I take it to the WAN it'll work fine? |
| 22:18.49 | CunningPike | Cableguy: /msg away, my friend |
| 22:19.03 | CunningPike | Cableguy: I should see your private messages now |
| 22:19.29 | Cableguy | CunningPike: Opps it said it was already registered, let me try another nick. I change my nick with the /nick command right? |
| 22:19.47 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 22:20.03 | Lannister | so my boss is still afraid of voip phones for our internal office network and interfacing with our "6" phone lines. What are issues that may come up? Are clarity or echo issues common? How can I avoid these issues? |
| 22:20.12 | *** join/#asterisk Crabman (n=Ant1@88.164.61.94) |
| 22:20.14 | Cableguy | CunningPike: It said that message in the freenode window. |
| 22:20.20 | CunningPike | Cableguy: Correct - and, by a funny coincidence, I think I know the owner of Cableguy |
| 22:20.57 | Cableguy | CunningPike: lol that's funny. |
| 22:21.17 | Cableguy | CunningPike: Ok let me try another nick |
| 22:21.19 | CunningPike | Lannister: Plan, test, the usual |
| 22:21.32 | *** part/#asterisk Crabman (n=Ant1@88.164.61.94) |
| 22:22.45 | *** join/#asterisk ^eugenics (i=chinamil@ip-174.net-81-220-169.nice.rev.numericable.fr) |
| 22:23.28 | *** join/#asterisk stephane (n=stephane@gw.sortilege.net) |
| 22:24.13 | Cabl3guy | CunningPike: ok looks like I'm registered. I double clicked your name and started typing. Did you get the message? |
| 22:27.01 | *** join/#asterisk Eliran_Itzhak_ (n=eliran@bzq-82-81-15-57.red.bezeqint.net) |
| 22:28.02 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 22:28.25 | *** part/#asterisk ztel (n=scott@70.103.238.2) |
| 22:28.29 | *** join/#asterisk KuJaX (n=one@c-67-164-204-41.hsd1.ut.comcast.net) |
| 22:28.58 | zapp-branigan | hi when i make a internal call in iax i hear a echo who can remove the echo ? i don't use zaps |
| 22:29.15 | Strom_C | zapp-branigan: what kind of station equipment is on either end of the call? |
| 22:29.35 | zapp-branigan | i use asterisk 1 line |
| 22:29.38 | *** join/#asterisk [hC] (n=hardcore@70.68.154.154) |
| 22:29.43 | zapp-branigan | i speak wih a friend |
| 22:29.48 | *** join/#asterisk wunderkin- (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 22:29.56 | Strom_C | zapp-branigan: but what kind of telephone set are you using |
| 22:30.07 | zapp-branigan | i use the iaxlite |
| 22:30.12 | zapp-branigan | software |
| 22:30.12 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 22:30.13 | Strom_C | softphone? |
| 22:30.15 | Strom_C | ah ok |
| 22:30.23 | Strom_C | are you using a headset, or just a speaker and a microphone |
| 22:30.43 | zapp-branigan | a headset |
| 22:30.47 | zapp-branigan | the two |
| 22:30.53 | Strom_C | i'd blame your headset |
| 22:31.02 | Strom_C | your friend is also using a headset? |
| 22:31.07 | zapp-branigan | yes |
| 22:31.20 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 22:31.37 | zapp-branigan | can be change some from iax? |
| 22:31.47 | Strom_C | that sentence makes no sense |
| 22:32.06 | zapp-branigan | :( |
| 22:32.36 | zapp-branigan | in zap i read we can recompile one file |
| 22:32.42 | Strom_C | that's zaptel |
| 22:32.46 | zapp-branigan | no |
| 22:33.00 | zapp-branigan | i ask that |
| 22:33.10 | zapp-branigan | if is something like zaptelç |
| 22:33.21 | Strom_C | here, why don't you call me and i'll see if i hear an echo |
| 22:33.33 | zapp-branigan | ok |
| 22:34.54 | zapp-branigan | how can i write you ? |
| 22:35.01 | JT | Cabl3guy: looking at your siutation, i think the only reason you would need asterisk would be if you require differen extensions within your house |
| 22:35.18 | Strom_C | zapp-branigan: what do you mean? I sent you an IAX2 address |
| 22:35.19 | JT | Cabl3guy: and yes, using an FXO port would be the easiest way to connect to the voip service |
| 22:35.49 | Lannister | its odd that this old phone system seems to have 2 external lines hooked up to every 6 internal lines |
| 22:35.59 | zapp-branigan | i must register to write in the prevailed ? |
| 22:36.01 | Lannister | does that mean that only those 6 internal lines can use those 2 external lines |
| 22:36.14 | Lannister | or can it balance between the several odd pbx boxes |
| 22:36.33 | zapp-branigan | Strom_c who i can write in the prevailed |
| 22:36.34 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 22:36.39 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-15-57.red.bezeqint.net) |
| 22:36.43 | rob0 | I tried a telephone headset (with adapter of course) for a softphone, and my results were very bad. Can someone recommend a good headset, pref. not too expensive? |
| 22:36.47 | hmmhesays | anyone ever do any user auth with ser? |
| 22:36.53 | zapp-branigan | Private messages from unregistered users are currently blocked due to spam problems, but you can always message a staffer. Please register! ( http://freenode.net/faq.shtml#privmsg ) |
| 22:37.03 | *** join/#asterisk WAudette (n=WAudette@c-71-237-146-239.hsd1.or.comcast.net) |
| 22:37.42 | [hC] | any ideas on why a bunch of polycom phones, mainly 501's running sip 1.6.7 (no nat) would lose registration often? |
| 22:37.42 | Strom_C | zapp-branigan: you need to register your nick |
| 22:38.24 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 22:40.40 | *** join/#asterisk knobenheimer (n=knoben@c-71-205-184-144.hsd1.mi.comcast.net) |
| 22:40.56 | JT | Lannister: i don't think so, they are probably shared amongst all extensions |
| 22:41.03 | JT | subject to the internal programming |
| 22:41.08 | *** join/#asterisk Scrye (i=1127@god.damnit.us) |
| 22:41.31 | Scrye | anyone know of a cheap DS3/DS1 multiplexor? |
| 22:41.37 | *** part/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:41.40 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:41.41 | JT | rob0: i got a cheap headset (about USD$8) from the computer markets |
| 22:41.43 | JT | worked fine |
| 22:42.38 | JT | Scrye: cheap i dunno, try ebay |
| 22:44.03 | rob0 | JT: So maybe just a lousy sound card? |
| 22:44.12 | JT | also a possibility |
| 22:44.16 | JT | it must be full deplex |
| 22:44.25 | rob0 | 00:08.0 Multimedia audio controller: ESS Technology ES1983S Maestro-3i PCI Audio Accelerator (rev 10) |
| 22:44.28 | JT | otherwise you'kll have issues |
| 22:44.32 | JT | ess, poo poo |
| 22:44.40 | JT | i dunno if that's FDX or not |
| 22:44.49 | rob0 | ah, okay. It's an old Dell laptop BTW. |
| 22:44.55 | JT | hrm |
| 22:45.07 | JT | yeah most newer stuff is full duplex |
| 22:45.18 | JT | i guess you could try a usb soundcard or usb phone handset |
| 22:45.33 | JT | although softphone handsets are annoying compared to a headset |
| 22:45.38 | JT | headsets rule |
| 22:45.38 | rob0 | or, an ATA and analog phone :) |
| 22:45.56 | JT | still ties you to a handset :P |
| 22:46.12 | rob0 | nah, there are analog phones with headsets. |
| 22:46.35 | JT | there are, but they usually cost |
| 22:46.47 | rob0 | But ideally I want a small number of gadgets, and yes, I want to keep the cost down. |
| 22:47.13 | Scrye | any brands in mind for that multiplexer |
| 22:47.47 | JT | i dunno, a decent one like adtran |
| 22:47.55 | JT | not sure who the bignames are in ds3 muxes |
| 22:49.01 | Strom_C | adtran? :) |
| 22:49.06 | rob0 | I guess eventually I'll have to invest in a real laptop, instead of getting by on old junk. |
| 22:49.23 | JT | Strom_C: heh, i was guessing |
| 22:49.37 | JT | as we don't use ds3s here afaik |
| 22:49.40 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 22:50.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 22:52.38 | Crescendo_ | How can I configure a Cisco IP Phone on the LAN so when I take it to the WAN it'll work fine? |
| 22:55.59 | mrichmanM | can i use add extension instead of database put to make the database aware of the new extensions in extensions.conf? |
| 22:56.31 | Strom_C | mrichmanM: usually you just reload extensions.conf and the new extensions just show up |
| 22:56.39 | Strom_C | are you using some weird GUI or something? |
| 22:57.24 | *** join/#asterisk aao_pwner (i=asd@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:57.31 | mrichmanM | they were originally created with amp but we are no longer using that due to problems |
| 22:57.47 | Strom_C | ugh |
| 22:58.04 | Strom_C | best thing to do, honestly, is rebuild everything from scratch at this point |
| 23:03.18 | *** join/#asterisk holmier (i=holmier@bsd.org.pl) |
| 23:07.57 | Lannister | FUCK of all people if you call aT&T and dont choose a menu option, it tells you to hang up and call again |
| 23:08.09 | JT | haha |
| 23:08.11 | JT | arseholes |
| 23:09.09 | Cabl3guy | JT: Thanks for the input, I'm just chatting with CunningPike right now. I'f I have further questions I'll be sure to look for you. Thanks. |
| 23:09.46 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:10.24 | EmleyMoor | Is there a way to have three lists of incoming CLI to be treated differently? ... |
| 23:10.55 | EmleyMoor | e.g. whitelist always get through, greylist only get through at certain times, blacklist never get through |
| 23:11.10 | *** join/#asterisk dj-fu (n=deejay@203.173.191.8) |
| 23:11.27 | EmleyMoor | (all others get through at a wider range of times but not at all times) |
| 23:12.59 | Strom_C | sure |
| 23:13.03 | Strom_C | you can use gotoif statements |
| 23:13.07 | Strom_C | and gotoiftime |
| 23:13.15 | dj-fu | hi there, i'm having some issues with my asterisk setup here at work and need a hand. We have two servers, one runs the asterisk stuff, another which is a firewall/dhcp box. I've just had to rebuild the dhcp box and it's giving a different ip range to the voip phones, subsequently they do not 'register' to the voip box |
| 23:13.23 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 23:13.26 | EmleyMoor | Yes - I'm just wondering about the implementation of the lists themselves |
| 23:13.41 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 23:13.44 | Lannister | roflmao at the auto attendant at 18002882020 |
| 23:14.18 | Lannister | its making me talk to it |
| 23:14.20 | dj-fu | i've never configured/looked at asterisk at all so just wondered if anyone could point me to the docs to reconfigure these new ip addresses for the voip phones |
| 23:14.23 | Lannister | and it doesnt recognize what im saying |
| 23:14.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 23:14.53 | Strom_C | dj-fu: usually it's the phones you have to reconfigure |
| 23:14.55 | Strom_C | not asterisk |
| 23:15.39 | EmleyMoor | Lannister: You what? |
| 23:15.40 | dj-fu | hrm |
| 23:16.37 | dj-fu | Strom_C: they are getting a dhcp ip address and aren't registering to the asterisk so I figured that asterisk isn't configured to allow those IP's? |
| 23:16.45 | dj-fu | wish I knew something about this shit |
| 23:17.13 | EmleyMoor | dj-fu: SIP phones? |
| 23:17.17 | Strom_C | is the asterisk box getting a dhcp address as well? |
| 23:17.25 | dj-fu | no, asterisk box is static on the same range |
| 23:17.27 | dj-fu | (10.1.1.2) |
| 23:17.41 | dj-fu | dhcp is 10.1.1.10-30 |
| 23:18.02 | dj-fu | EmleyMoor: yeah, Linksys |
| 23:18.13 | dj-fu | spa941 |
| 23:18.14 | Lannister | ok, got tons of info on my phone lines, i have 5 separate analog lines in hunt mode. So what hardware do i need to interface these lines with an asterisk box? Does anyone sell a simplified solution? |
| 23:18.26 | EmleyMoor | Was about to ask if NAT was involved |
| 23:18.55 | dj-fu | hm |
| 23:19.11 | Strom_C | Lannister: you can interface to the pots lines with a digium tdm2400p |
| 23:19.14 | EmleyMoor | Lannister: A TDM24xx card with at least 2 FXO midules?# |
| 23:19.19 | EmleyMoor | modules |
| 23:19.21 | Lannister | they're called pots lines? |
| 23:19.24 | Strom_C | POTS |
| 23:19.28 | Strom_C | plain old telephone service |
| 23:19.36 | dj-fu | under sip registry in asterisk info through the web interface it's showing this - 713/713 192.168.1.128 D 5061 Unmonitored, shouldn't it be the 10.* range? |
| 23:19.39 | Strom_C | this is a technical term :) |
| 23:19.48 | JT | bingo |
| 23:19.53 | Strom_C | dj-fu: WEB INTERFACE? |
| 23:19.54 | Lannister | hahaha |
| 23:19.57 | JT | i was right about 5 lines |
| 23:20.06 | dj-fu | Strom_C: i'm stabbing in the dark here man. |
| 23:20.16 | dj-fu | the asterisk was setup when I got here, things have been broken for a long time |
| 23:20.34 | Strom_C | dj-fu: what is the name of the web interface? |
| 23:20.34 | Lannister | ok cool does anyone sell a box thats all ready to go that can interface with my 5 pots lines and my 20 voip phones? so that I can just configure it and the hardware/OS is already set up |
| 23:20.44 | dj-fu | asterisk@ome |
| 23:20.45 | dj-fu | home* |
| 23:20.56 | Strom_C | dj-fu: oh christ, and you're running a BUSINESS on that? |
| 23:21.08 | dj-fu | unfortunately - it was like that when I got here |
| 23:21.13 | dj-fu | I'm just getting told to fix it |
| 23:21.21 | Strom_C | here's what you do |
| 23:21.24 | Strom_C | (1) scrap it |
| 23:21.28 | Strom_C | (2) rebuild it correctly |
| 23:21.39 | JT | well in the meantime i think he needs it fixed |
| 23:21.39 | dj-fu | is there a wiki or a tut somewhere that I can follow? |
| 23:21.39 | Strom_C | (3) strangle and stab whoever set it up initially |
| 23:21.51 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 23:22.22 | rob0 | EmleyMoor: I think a brownlist would look good over there next to the coffee table. |
| 23:22.32 | *** join/#asterisk DavoFrom818 (n=Vito310@cpe-76-173-56-41.socal.res.rr.com) |
| 23:22.55 | DavoFrom818 | hey guys |
| 23:23.09 | CoffeeIV_ | If I have a PRI, not connected to a cellular service, is it possible to use the SMS() dialplan application to send text messgaes ? Do I have to sign up for some service with the cellular providers to do it ? |
| 23:23.25 | dj-fu | Strom_C: can you point me in the right direction? what should I be running my business on? heh |
| 23:23.36 | EmleyMoor | I don't want certain numbers being given the option to actually try the phones, others only at very limited times, a handful always and all the rest at limited times |
| 23:23.53 | Strom_C | dj-fu: a real linux distribution with real asterisk |
| 23:23.55 | DavoFrom818 | how do these guys do free 411? can i add 411 feature to my box? and have it look up the yellowpages? |
| 23:23.58 | Strom_C | not this asterisk@home crap |
| 23:24.09 | Strom_C | dj-fu: read the book |
| 23:24.12 | Strom_C | ~thebook |
| 23:24.14 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:24.14 | Strom_C | good primer |
| 23:24.22 | dj-fu | ta |
| 23:24.31 | DavoFrom818 | Strom_C asterisk@home is oldddd! why dont u get Trixbox |
| 23:24.42 | dj-fu | I just rebuilt this gentoo box, the firewall/dhcp etc - would that be suitable? |
| 23:25.39 | *** join/#asterisk [hC] (n=hardcore@70.68.154.154) |
| 23:25.42 | Strom_C | oh, trixbox blows just as badly :) |
| 23:25.55 | Strom_C | dj-fu: sure |
| 23:26.17 | dj-fu | bugger. I'll have to swap that 4 port super thing |
| 23:26.27 | *** join/#asterisk apardo (n=apardo@87.217.144.168) |
| 23:26.27 | Strom_C | ? |
| 23:26.47 | dj-fu | the pci card, I dunno what it is |
| 23:26.50 | dj-fu | looks like a supermodem |
| 23:27.11 | DavoFrom818 | can asterisk do speech recognition so the user can speak the phone number? |
| 23:28.57 | holmier | anyone has a good source for isdn hfc pci cards for asterisk? need a quite good number of them |
| 23:29.18 | holmier | (one port) |
| 23:29.19 | naftali5 | Davo: http://www.lumenvox.com/partners/integrator/digium/asterisk.aspx |
| 23:29.26 | Strom_C | DavoFrom818: asterisk business edition + lumenvox can |
| 23:30.05 | naftali5 | Strom_c: doesn't need buiseness edition, even trixbox integrated it, still lumenvox costs |
| 23:30.28 | JT | dj-fu: umm it's probably an FXO card |
| 23:30.31 | JT | like a TDM400P |
| 23:30.43 | JT | worth a little more than a modem |
| 23:30.56 | JT | if you already have a server for it, why make another one |
| 23:31.25 | JT | anyway, it's not a 1day task for the new user to rebuild it, i think you should try and make your existing system work in the meantime |
| 23:31.49 | Strom_C | if you need it yesterday: |
| 23:31.51 | Strom_C | ~hafc |
| 23:31.55 | jbot | hafc is, like, hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 23:32.01 | dj-fu | l0l |
| 23:32.20 | dj-fu | I'd rather pretend to know what I'm doing and get paid for that |
| 23:32.21 | JT | Strom_C: well the only thing that has changed is his dhcp server |
| 23:33.35 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:33.53 | dj-fu | there's nothing in asterisk that would lock my phones to only come from certain addresses is there? |
| 23:34.16 | JT | yes, in sip.conf they might be set to static IPs |
| 23:34.28 | JT | did you change the ip range in DHCP? |
| 23:35.06 | dj-fu | yeah. DHCP was previously 192.168.1.*, now it's 30 address 10.1.1.10-30 |
| 23:35.11 | dj-fu | err, 20 address |
| 23:35.18 | JT | why was it changed? |
| 23:35.35 | JT | you'll need to reconfigure a bunch of things, i think it's safe to say |
| 23:36.25 | dj-fu | the dhcp box died |
| 23:36.29 | dj-fu | and I rebuilt it from scratch |
| 23:36.48 | *** join/#asterisk richmanmM (n=richmanm@70.89.184.1) |
| 23:37.51 | dj-fu | I can't find any static IP's |
| 23:37.57 | dj-fu | blargh |
| 23:38.37 | JT | why did you change the ip range then? it was likely to break stuff :P |
| 23:38.52 | JT | can't you change it back |
| 23:39.07 | holmier | dj-fu: maybe you just need enable forwarding bu echo-ing some value to the /proc? |
| 23:39.09 | Lannister | so my friends are telling me "fuck voip over analog lines" |
| 23:39.16 | Lannister | and to use pri |
| 23:39.21 | Lannister | lies? truths? |
| 23:39.28 | JT | pri is the best |
| 23:39.45 | Lannister | how do i convert from my 6 analog lines to digital pri |
| 23:39.50 | holmier | it depends on your needs :) |
| 23:39.55 | holmier | which is best |
| 23:41.03 | Lannister | i need the ability to have up to 6 concurrent outgoing conversations ...and X concurrent sessions within my own phone network of 20 voip phones |
| 23:42.17 | EmleyMoor | Is there an example of a "three lists" CLI handler anywhere? |
| 23:43.02 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 23:43.21 | EmleyMoor | (I justg h |
| 23:43.33 | EmleyMoor | (I just had a call from someone I want to greylist!) |
| 23:44.23 | dj-fu | thanks for your help guys |
| 23:44.49 | Un1x | Hey, gus quick question how do i record a call |
| 23:44.51 | Un1x | what is the command |
| 23:44.59 | naftali5 | moniotor() |
| 23:45.03 | naftali5 | monitor() |
| 23:45.18 | Un1x | and it will record the calls? |
| 23:46.13 | EmleyMoor | I've read a bit about blacklisting but I want multiple shades |
| 23:46.35 | Un1x | http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor |
| 23:46.40 | Un1x | is the conference important |
| 23:46.41 | Un1x | with meetme? |
| 23:48.40 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 23:51.47 | DavoFrom818 | naftali5 any free routes to it? |
| 23:53.38 | *** join/#asterisk realman (n=alraiky@dsl62-149-92-22.saudi.net.sa) |
| 23:53.53 | realman | hi |
| 23:54.22 | JT | holmier: technically PRIs are the best, just may or may not be economical |
| 23:54.44 | realman | any one can help for CID setting |
| 23:55.21 | JT | http://www.voip-info.org/wiki/view/Setting+Callerid |
| 23:56.31 | realman | I am missing the zap setting for saudi |
| 23:56.47 | realman | I am trying for more than a week |
| 23:57.30 | JT | what sort of lines? |
| 23:58.03 | realman | PSTN with DTMF singnalling |
| 23:58.28 | JT | are you trying to set outbound CLI? |
| 23:58.55 | JT | like what it will display on outside phones when you clal them |
| 23:58.55 | realman | the inbound CID |
| 23:58.57 | JT | call |
| 23:59.02 | JT | hrm |
| 23:59.06 | JT | what hardware |
| 23:59.11 | EmleyMoor | realman: You mean to receive it? |
| 23:59.34 | realman | TDM400P |
| 23:59.48 | realman | nop I can't recivr CID |