| 00:00.08 | brdude | Oh god! |
| 00:00.22 | brdude | Please tell me there was at least some documentation. |
| 00:03.28 | dym | well |
| 00:03.29 | dym | :D |
| 00:03.30 | dym | i would |
| 00:03.32 | dym | if there was |
| 00:03.33 | dym | :D |
| 00:03.42 | dym | i was on my own |
| 00:03.46 | dym | so why document things? |
| 00:03.51 | dym | i ofc had comments in the dialplans |
| 00:03.57 | dym | but my style comments |
| 00:04.02 | dym | doubt people could make much sense of it |
| 00:04.48 | nightrid3r | hehehe :D |
| 00:04.54 | brdude | lol. |
| 00:05.25 | nightrid3r | if there's 1 thing programers hat its documentation :) |
| 00:05.53 | brdude | Nice, I'll make sure that in future jobs anything that requires any asterisk knowled to look at the documentation before hand or to ask for a realy high salary. |
| 00:05.59 | *** join/#asterisk tzafrir_laptop (~tzafrir@2001:41b8:203:dc11:21c:23ff:fe20:991f) |
| 00:06.10 | nightrid3r | or both |
| 00:06.41 | brdude | I know I hate documentation too, but eversince I've started doing it for my self I can't even remember how many times it's saved my ass. |
| 00:07.28 | dym | well |
| 00:07.29 | dym | brdude: |
| 00:07.33 | dym | no matter what you end up with |
| 00:07.40 | dym | i strongly advise you to "learn" asterisk. |
| 00:07.51 | dym | even if you do end up with a gui |
| 00:07.55 | brdude | dym, I definetely will. |
| 00:07.57 | dym | you will know how the gui works |
| 00:08.06 | dym | * is a wonderful piece of software |
| 00:08.10 | brdude | As soon as that book comes in I'll be eating it up. |
| 00:08.15 | dym | even though it killed plenty of my nights |
| 00:08.24 | dym | i have a joyful time with it |
| 00:08.28 | dym | still had* |
| 00:08.34 | dym | like woman really. |
| 00:08.35 | dym | :D |
| 00:08.43 | brdude | lol |
| 00:08.47 | brdude | :D |
| 00:09.02 | dym | learn how they work - program them to your benefit |
| 00:09.14 | dym | you could install a gui on them, but then you dont know what you end up with |
| 00:09.46 | brdude | I definetely will be doing it from the ground up. |
| 00:09.54 | dym | excellent! |
| 00:09.58 | dym | report back here with results! |
| 00:10.14 | brdude | I still have to use the GUI but I'll just have to figure out a way to make them play nice. |
| 00:10.22 | brdude | Sir yes sir! |
| 00:10.24 | dym | if i was you |
| 00:10.32 | dym | i'd start without gui |
| 00:10.37 | dym | do some stuff |
| 00:10.40 | dym | register some phones |
| 00:10.43 | dym | write some dialplans |
| 00:10.45 | dym | get into it |
| 00:10.49 | dym | then -> erase everything |
| 00:10.50 | dym | and install gui |
| 00:11.01 | dym | because then you already know the basics |
| 00:11.52 | brdude | Nice will do. |
| 00:11.57 | dym | great |
| 00:12.08 | brdude | Although I might have to do things a little diferently. |
| 00:12.17 | dym | damn.. when i started i bought HFC hardware worth 2k Eur |
| 00:13.25 | brdude | I might have to do a GUI install to show management it works and it's what we should use. Errase everything install withough GUI play around, errase everything again and then do a final install with the GUI. |
| 00:13.32 | brdude | dym, why? |
| 00:13.57 | dym | brdude: cause we used to have an internal ISDN system |
| 00:14.05 | dym | before we switched to SIP |
| 00:14.19 | dym | ISDN phones, which needed to have their own S0 connection to the asterisk |
| 00:14.30 | dym | and powered by it, too |
| 00:14.49 | dym | i actually fried a HFC card :D |
| 00:14.54 | dym | worth 400 Eur :D |
| 00:14.55 | brdude | that sucks. |
| 00:15.09 | dym | funtimes |
| 00:15.37 | WIMPy | How did you fry that? |
| 00:15.51 | dym | think it was a wrong jumpersetting |
| 00:16.04 | dym | and a cable inserted from the NTBA |
| 00:16.08 | dym | :D |
| 00:16.21 | WIMPy | I wouldn't expect that to fry it. |
| 00:16.29 | dym | for some strange reason it did |
| 00:16.33 | dym | BN4S0 |
| 00:16.37 | dym | remember like it was yesterday |
| 00:17.03 | brdude | We have a PRI in place, I'm thinking in using the same company as our SIP Trunk provider. They seem to be able to help with the whole migration. |
| 00:17.10 | dym | WIMPy: stop eavesdropping and fix my CALLERID=num problem :D |
| 00:17.29 | dym | oh |
| 00:17.31 | WIMPy | I've alredy put power in to the cards in both polarities at the same time. That didn't cause harm. |
| 00:17.49 | dym | PRI and SIP Trunk |
| 00:17.54 | WIMPy | What callerid problem? |
| 00:17.58 | dym | WIMPy: Also BeroNet? |
| 00:18.06 | dym | or random HFC |
| 00:18.36 | dym | Well, i cant seem to catch the incoming callerID from my sipgate trunk |
| 00:18.52 | dym | $CALLERID(num) is beeing a dick |
| 00:19.17 | dym | that should cut the number from the SIP-TO Header |
| 00:19.27 | dym | < >@blabla part |
| 00:19.29 | WIMPy | NFI. Both cards I have were sold to me as Junghanns, but I'm pretty sure at least one of them isn't. |
| 00:19.38 | dym | right |
| 00:19.45 | WIMPy | TO? |
| 00:19.52 | dym | err |
| 00:19.53 | dym | sorry |
| 00:19.54 | dym | FROM |
| 00:20.11 | WIMPy | Probably. |
| 00:20.35 | dym | probably what? :D |
| 00:20.41 | WIMPy | However Sipgate like to use the customer number as callerid if nothing else is configured. |
| 00:20.50 | dym | nah |
| 00:20.52 | dym | incoming calls |
| 00:20.54 | dym | throught the trunk |
| 00:21.00 | WIMPy | Yes |
| 00:21.16 | dym | how are you supposed to configure an incoming callerid before it passes the trunk? |
| 00:21.34 | WIMPy | When I get calls from other sipgate customers I usually see theyr customer ID, icl. e0. |
| 00:22.08 | dym | Well |
| 00:22.15 | dym | according to this: http://www.sipgate.de/beta/public/static/sipgate.de/media/Das_Produkt_sipgate_Trunking.pdf |
| 00:22.21 | WIMPy | Incomming, before it passes the trunk? That's obviouselt somethin the other end does. |
| 00:22.32 | dym | its supposed to be in ${EXTEN} |
| 00:22.35 | dym | which is bull |
| 00:22.43 | dym | WIMPy: exactly. |
| 00:22.55 | dym | The trunk is supposed to pass the number to the next "hop" |
| 00:22.58 | WIMPy | So you want the called number, not the callerid? |
| 00:23.15 | dym | well, yes CALLERID(num) as mentioned before. |
| 00:23.30 | WIMPy | Something doesn't fit here. |
| 00:24.02 | WIMPy | Which number are you looking for? |
| 00:24.43 | dym | The number of the incoming caller |
| 00:24.54 | *** join/#asterisk radic (~radic@tmo-096-110.customers.d1-online.com) |
| 00:26.08 | WIMPy | Well, as I said, I often get customer IDs there. Let me try to call from a phone. |
| 00:26.21 | dym | thanks cutiepie |
| 00:26.50 | WIMPy | Works as expected. |
| 00:27.23 | dym | well |
| 00:27.27 | dym | i just checked my logs |
| 00:27.43 | dym | i also get From: "NUMBER"<sip:NUMBER@213.239.205.120>;tag=as750271a9 |
| 00:27.58 | dym | where NUMBER is the complete Number of the imcoming caller |
| 00:28.05 | WIMPy | Maybe you need trustrpid for that? |
| 00:28.13 | dym | mhh |
| 00:28.22 | dym | should be enabled |
| 00:28.23 | dym | sec |
| 00:28.37 | dym | 'tis |
| 00:29.20 | dym | am i making a mistake? |
| 00:29.28 | dym | should it not be in $CALLERID(num) ? |
| 00:29.38 | WIMPy | That's the one. |
| 00:29.54 | dym | odd |
| 00:30.02 | dym | maybe i need to escape the variable |
| 00:30.13 | WIMPy | Where? |
| 00:30.17 | dym | im using it within a jabber message |
| 00:30.27 | WIMPy | nfi |
| 00:30.39 | WIMPy | Put a verbose in to your dialplan. |
| 00:30.54 | dym | Jabbersend(asterisk,patrick@jabber.openroot.de,Incoming Call from $CALLERID(num)) |
| 00:31.08 | dym | maybe the entire message needs quotes |
| 00:31.11 | dym | *tries* |
| 00:31.18 | WIMPy | The {} are missing! |
| 00:31.47 | dym | as in ${CALLERID(num)} ? |
| 00:31.56 | WIMPy | yes |
| 00:32.20 | WIMPy | They aren't optional as in the system shell. |
| 00:32.25 | WIMPy | (sometimes) |
| 00:32.33 | dym | lol |
| 00:32.36 | dym | works |
| 00:32.37 | dym | :D |
| 00:32.42 | dym | DOH |
| 00:32.43 | dym | thanks |
| 00:35.11 | brdude | Nice, glad you got that resolved. |
| 00:36.42 | brdude | Ok another stupid question. Has anyone used "visual dialplan"? Mayble I'd be able to used that for management insead of having a GUI installed on the server. |
| 00:37.22 | *** join/#asterisk sp0rkie (~sp0rk@adsl-68-17-173-100.bhm.bellsouth.net) |
| 00:38.10 | dym | not really |
| 00:38.11 | dym | sorry |
| 00:38.40 | brdude | It's all good. You guys have helped me plenty already. |
| 00:38.55 | brdude | But I though it was worth a shot. |
| 00:40.39 | dym | :) |
| 00:41.02 | dym | god - i went to an audition today and couldnt get my microphone to synchronize... |
| 00:41.05 | dym | so embarrasing |
| 00:51.16 | *** join/#asterisk wonderworld (~ww@port-92-201-113-243.dynamic.qsc.de) |
| 00:52.06 | *** join/#asterisk radic (~radic@tmo-097-125.customers.d1-online.com) |
| 01:01.53 | linuxgecko | can i create channels /jabber connections on the fly? (ie, not in hardcoded into the dialplan) ?? |
| 01:04.12 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 01:09.09 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 01:15.22 | BenC[UK] | Is there a variable for which number fails in a multiple number Dial() ? |
| 01:15.44 | BenC[UK] | ie.. if I calla number and get busy... I get to my busy extension, which calls an AGI script, but i need to know which number was busy |
| 01:16.16 | WIMPy | All of them. |
| 01:16.54 | WIMPy | It will continue as long as at least one destination is reacable. |
| 01:17.32 | WIMPy | But o BOB feature would be quite nice, i think. |
| 01:22.00 | *** part/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 01:30.07 | kaushal | Hi |
| 01:30.35 | kaushal | I have edited chan_dahdi.conf |
| 01:30.45 | kaushal | and set group accordingly |
| 01:30.56 | kaushal | when i try to initiate the call |
| 01:31.26 | kaushal | channel originate DAHDI/g0/XXXXXXXXXX Application MP3Player /home/kaushal/obd-demo.mp3 |
| 01:31.42 | kaushal | [Jul 30 06:58:53] NOTICE[14450]: channel.c:5256 __ast_request_and_dial: Unable to request channel DAHDI/g0/XXXXXXXXXX |
| 01:31.51 | kaushal | Am i missing anything |
| 01:32.05 | kaushal | Shall i pastebin chan_dahdi.conf |
| 01:32.24 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:34.42 | kaushal | WIMPy: hi |
| 01:35.20 | *** join/#asterisk [ctrl][alt][del] (~Jabber@pdpc/supporter/active/ctrl-alt-del) |
| 01:41.12 | kaushal | checking in again for the query |
| 01:42.17 | WIMPy | Did you restart? |
| 01:44.08 | *** join/#asterisk cookieliu (~cookie@27.115.97.2) |
| 01:44.25 | kaushal | WIMPy: yeah i did it |
| 01:45.01 | WIMPy | And the channels in that group are usable? |
| 01:45.11 | kaushal | yes |
| 01:45.33 | WIMPy | And no more information with verbose/debug? |
| 01:45.43 | kaushal | WIMPy: shall i pastebin the configs and CLI output ? |
| 01:45.49 | *** join/#asterisk tzafrir_laptop (~tzafrir@static-8-143-241-92-p2mp.broadband.blic.net) |
| 01:46.04 | WIMPy | Let's have a look. |
| 01:46.17 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 01:46.29 | kaushal | WIMPy: please let me know what details you need ? |
| 01:46.47 | WIMPy | Start with the cli output. |
| 01:47.13 | kaushal | ok |
| 01:47.20 | kaushal | WIMPy: i have pasted it already |
| 01:47.49 | brdude | ok where can I get a cheap sip trunk to test my system? I tried o go to les.net and have signed up fine and verified that my sistem is working but I can't find where to sign up for a plan online. |
| 01:47.54 | WIMPy | That's the only line? |
| 01:49.25 | kaushal | WIMPy: http://pastebin.ubuntu.com/654846/ |
| 01:50.41 | WIMPy | Ok, chan_dahdi.conf, please. And you should fix that permission thing. |
| 01:50.46 | kaushal | ok |
| 01:53.01 | kaushal | WIMPy: http://pastebin.ubuntu.com/654847/ |
| 01:54.26 | WIMPy | Looks ok to me. |
| 01:54.46 | kaushal | WIMPy: I have edited it by hand |
| 01:54.51 | WIMPy | Anything in 'dahdi show status' or 'dahdi show channels'? |
| 01:54.53 | kaushal | does it hurt ? |
| 01:55.37 | WIMPy | Or doesn't it like 0 as group? |
| 01:55.56 | sp0rkie | does anyone have a few minutes to help me out with a realtime issue that i've been trying to work out for the greater part of this week? :\ |
| 01:56.11 | WIMPy | ~ask |
| 01:56.11 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 01:56.29 | *** join/#asterisk nightrid3r (~kvirc@91.176.150.94) |
| 01:56.59 | WIMPy | kaushal: Do you really want immediate=no? |
| 01:57.47 | kaushal | WIMPy: ok |
| 01:58.24 | kaushal | WIMPy: http://pastebin.ubuntu.com/654849/ |
| 01:58.30 | sp0rkie | sorry. i'm attempting to get SIP realtime working. when to do 'realtime mysql status', it shows as connected and i see the connection on the query log. there are no warnings or notifications regarding realtime (besides the normal 'found config file' and such). when i attempt to connect a softphone, it gives a warning saying 'cannot find peer.' and asterisk didn't even attempt to query the |
| 01:58.30 | sp0rkie | database. |
| 01:59.12 | WIMPy | Looking good. Have you tried group 1? |
| 01:59.17 | kaushal | yes |
| 01:59.23 | kaushal | still the same issue |
| 01:59.32 | WIMPy | So it's not the 0. |
| 01:59.54 | WIMPy | Would be great if it told us more. |
| 02:00.07 | kaushal | yeah |
| 02:00.32 | WIMPy | Have you tried to place a call from the outside to that line? Or to dial out in a specific channel? |
| 02:00.40 | WIMPy | on |
| 02:01.06 | kaushal | http://pastebin.ubuntu.com/654851/ |
| 02:02.38 | kaushal | WIMPy: also is there a way to set group using wancfg_dahdi ? |
| 02:02.58 | WIMPy | No idea. |
| 02:03.03 | kaushal | ok |
| 02:03.20 | WIMPy | Didn't have a Sangoma card so far. |
| 02:03.47 | *** join/#asterisk cookieliu (~cookie@27.115.97.2) |
| 02:05.03 | WIMPy | So does it work apart from the group setting? |
| 02:05.06 | kaushal | when i edit it again using wancfg_dahdi i have http://pastebin.ubuntu.com/654853/ |
| 02:05.17 | kaushal | and this works perfectly fine |
| 02:05.33 | kaushal | but when i edit it by hand to set group it doesnot work |
| 02:06.25 | WIMPy | Interesting. |
| 02:07.24 | WIMPy | Maybe 0 shouldn;t be mixed with others. It's just a wild guess, but you could try to use 1-4 instead of 0-3. |
| 02:07.39 | kaushal | ok |
| 02:09.36 | kaushal | No luck |
| 02:10.04 | kaushal | WIMPy: the permission issue has been fixed |
| 02:10.04 | kaushal | WIMPy: Please suggest further |
| 02:10.13 | WIMPy | Did you build a special chan_dahdi without group support? |
| 02:10.33 | WIMPy | I'm running out of ideas :-( |
| 02:10.44 | kaushal | build chan?_dahdi ? |
| 02:14.30 | *** join/#asterisk picard276 (~chatzilla@rrcs-24-199-53-187.west.biz.rr.com) |
| 02:14.37 | picard276 | hey guys having a 911 emergency |
| 02:14.46 | picard276 | went to restart asterisk and when i did an amportal restart got this |
| 02:14.59 | picard276 | usr/sbin/safe_asterisk line 145: 4718 segmentation fault? |
| 02:15.12 | WIMPy | It's pretty strange that just changing the group numbers breaks the thing. |
| 02:16.20 | WIMPy | picard276: Start it with asterisc -cvvvddd and see how far it gets or if there's any clue, where it barfs. |
| 02:16.37 | picard276 | hold one sec Wimpy |
| 02:16.45 | WIMPy | Otherwise inspect the core with gdb. |
| 02:16.59 | picard276 | everything looks fine then after |
| 02:17.05 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 02:17.05 | *** join/#asterisk b0rf (~b0rf@ip98-164-100-237.no.no.cox.net) |
| 02:17.05 | *** join/#asterisk disgrntld81 (~asdf@CPE-70-92-225-30.wi.res.rr.com) |
| 02:17.05 | *** join/#asterisk WHYS (~drumm@137.28.94.209) |
| 02:17.06 | picard276 | etc/asterisk/chan_skype.conf |
| 02:17.14 | picard276 | it says segmentation fault |
| 02:18.40 | WIMPy | It's either the last module you see or the next one, wich might be hard to determine. But at least you know the good ones. |
| 02:19.02 | *** join/#asterisk fobus912 (~fobus912@41.249.41.218) |
| 02:19.12 | fobus912 | Hi All |
| 02:19.36 | fobus912 | have any test ipv6 implementation |
| 02:19.39 | fobus912 | with Tahi |
| 02:19.44 | fobus912 | ? |
| 02:19.47 | WIMPy | I have seen crashes because of bad config files, so you can try to move them away and then back and see when the problem goes or comes back. |
| 02:20.37 | picard276 | ok now i deleted that skype config (did not need it for any reason.. |
| 02:20.43 | picard276 | and now it goes to asterisk ready |
| 02:20.53 | picard276 | then under it *CLI>segmentation fault? |
| 02:21.06 | WIMPy | If you don't need it you shouldn;t load it. |
| 02:21.18 | WIMPy | What? |
| 02:21.23 | ChannelZ | yeah it's probably a mismatched module |
| 02:21.39 | picard276 | hmm |
| 02:21.48 | picard276 | ok i did an apply config in my freepbx to see if it would givem e any more inof |
| 02:21.53 | picard276 | info |
| 02:22.35 | picard276 | says unable to connect to asterisk manager from var/lib/retrieve_conf |
| 02:24.35 | picard276 | it also says unable to connect to manager: connection refused... (meaning asterisk is prob.. not running right? |
| 02:24.57 | picard276 | did an ampotal restart |
| 02:25.04 | picard276 | this error |
| 02:25.24 | WIMPy | I'm not sure where you are now. Can you start Asterisk now or does it still segfault? |
| 02:27.23 | picard276 | <PROTECTED> |
| 02:27.28 | kaushal | WIMPy: it worked :) |
| 02:27.40 | picard276 | WIMPy that was on an amportal restart |
| 02:27.49 | kaushal | WIMPy: I get this issue [Jul 30 07:57:08] NOTICE[20652]: app_mp3.c:127 timed_read: Poll timed out/errored out with 0 |
| 02:28.02 | WIMPy | kaushal: How? |
| 02:28.38 | kaushal | i removed chan_dahdi.conf* |
| 02:28.49 | kaushal | and used wancfg_dahdi |
| 02:28.53 | kaushal | it worked |
| 02:29.05 | WIMPy | o.O |
| 02:29.28 | WIMPy | Doesn't it use the normal chan_dahdi? |
| 02:29.30 | picard276 | WIMPy any suggestions? |
| 02:29.52 | kaushal | WIMPy: Also is there a way to map PRI Calling Pilot Number ? |
| 02:30.16 | WIMPy | picard276: See above. Or disable autoload in modules.conf and manually load them until it crashes. |
| 02:30.30 | WIMPy | Pilot? |
| 02:32.30 | picard276 | see above? |
| 02:33.18 | WIMPy | Try the config files one by one or use gdb on the core file. See |
| 02:33.22 | WIMPy | ~collectdebug |
| 02:33.22 | infobot | hmm... collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 02:35.45 | WIMPy | picard276: However I'm not really sure it will be of any helf to you if you find something, as FreePBX will probably reset your configs on the next click you make, anyway. |
| 02:37.45 | ChannelZ | Hurray for automation! |
| 02:40.45 | *** part/#asterisk sp0rkie (~sp0rk@adsl-68-17-173-100.bhm.bellsouth.net) |
| 02:51.01 | brdude | Anyone have any sugestions for a cheap SIP trunk I can use to test my asterisk box? |
| 02:51.15 | *** join/#asterisk sp0rkie (~sp0rk@adsl-68-17-173-100.bhm.bellsouth.net) |
| 02:56.47 | *** part/#asterisk fobus912 (~fobus912@41.249.41.218) |
| 02:57.22 | p3nguin | qwell: That's certainly an unusual one. I assume it means "I would not." |
| 03:03.00 | *** join/#asterisk ketema (~ketema@ketema.net) |
| 03:03.19 | ChannelZ | brdude: google for "sip test numbers" |
| 03:03.37 | brdude | ChannelZ thanks. |
| 03:03.39 | *** join/#asterisk cookieliu (~cookie@27.115.97.2) |
| 03:07.21 | ChannelZ | SIP/ideasip.com/301 is good for an echo test |
| 03:28.18 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 03:49.28 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 03:58.20 | *** join/#asterisk gogasca (~gogasca@166.205.138.239) |
| 04:18.01 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 04:20.45 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 04:23.24 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 04:32.51 | *** join/#asterisk linuxgeek (~uppal@61.5.195.13) |
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| 04:33.59 | NephFL | Hello |
| 04:34.17 | linuxgeek | hey |
| 04:35.01 | dym | brdude: /away detached |
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| 04:37.10 | NephFL | I'm trying to figure out an issue with freepbx endpoint manager, it isn't adding the extension and other values to the provisioning file, any idea on a reference for troubleshooting it? |
| 04:40.50 | sunfone | I would ask on #freepbx |
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| 05:18.20 | brdude | dym, later. |
| 05:24.15 | ChannelZ | yawns |
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| 09:55.17 | dym | hah brdude |
| 09:55.25 | dym | Dont know why i highlighted you :D |
| 09:55.27 | dym | accidently |
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| 12:46.26 | tzafrir_laptop | any idea if anybody reverse-engeneered the "TDM over IP" protocol of IP mux4 and the likes? |
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| 12:48.07 | coppice | tzafrir_laptop: I doubt it would take much to reverse engineer, but I think Pulsewan and RAD have patents on how they do their respective TDMoIPs |
| 12:48.23 | coppice | nothing like interoperability, eh? |
| 12:49.08 | tzafrir_laptop | right |
| 12:52.37 | coppice | This bit is odd in their description: |
| 12:52.39 | coppice | "Compensates for packet delay variation of up to 32 msec for E1 or 24 msec for T1 circuits" |
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| 15:06.07 | moex | in a context.. lets say "context1", I want to set some channel vars before going through my dialplan. What is the best way to do that? I tried "exten => s,1,Set(var=value) as the first statment .. Doesnt seem to be picked up by the dialplan though |
| 15:06.55 | moex | I tried a wildmatch _X. which seems to picks it up, but then, it wont go through the rest of the dialplan (as it has found a match) |
| 15:06.57 | moex | any ideas? |
| 15:09.04 | jwiggins | moex, pastebin it |
| 15:10.51 | moex | do you want me to show the actual extensions.conf or the output of dialplan show? |
| 15:11.07 | moex | I only have a few lines anyways.. |
| 15:11.09 | jwiggins | extensions.conf is easier for me to read |
| 15:11.15 | moex | ok, 1 minute |
| 15:12.37 | moex | http://pastebin.com/3uzUkdQK |
| 15:14.13 | jwiggins | do you have an output from the CLI? |
| 15:14.43 | moex | if I dial lets say "11", it doesnt match |
| 15:15.00 | jwiggins | really? |
| 15:15.19 | moex | grrr.. now I'm doubting myself.. let me try again |
| 15:15.31 | jwiggins | no context found? |
| 15:15.49 | moex | ok, so it calls "11" but it doesnt match the "exten =>s....." |
| 15:16.20 | moex | what is supposed to happen is, it sets context_prefix=1, then when it cincludes the "stdext" it will dial 111 |
| 15:16.20 | Gugge | and it shouldnt |
| 15:16.24 | Gugge | because you didnt dial s |
| 15:16.25 | jwiggins | everything looks good with your statement, of course if you dial 11 it should just do a Dial(SIP/11) |
| 15:16.43 | jwiggins | no it won't hit the s |
| 15:16.47 | jwiggins | like Gugge just said |
| 15:17.00 | moex | I think I'm a bit confused.. I thought the s should be hit first.. hmm |
| 15:17.09 | moex | ok.. so what is the best way to set those vars first thing |
| 15:17.16 | jwiggins | and I don't believe you need the "SIP/" encased in quotes |
| 15:17.19 | Gugge | put it in every extension |
| 15:17.29 | moex | I want to set them per context not per extension |
| 15:17.32 | Gugge | or call a macro from every extension |
| 15:17.33 | moex | is that possible? |
| 15:17.35 | Gugge | no |
| 15:17.47 | jwiggins | yup what Gugge said, there is no real way to say do all of this first then dial the extension |
| 15:18.00 | jwiggins | you can make a macro to do all of your s, dial |
| 15:18.05 | moex | I should have come here first.. been pulling my hair for 2 hours :D |
| 15:18.53 | jwiggins | like Macro(setupcall) |
| 15:19.01 | moex | thanks jwiggins and Gugge.. I'll go read the macro's part |
| 15:21.42 | jwiggins | moex, are you trying to build a multi office dial plan? |
| 15:22.05 | moex | sort of |
| 15:22.22 | moex | Like, I'm tryin gto set the prefixes, etc. before hand so that I can create a generic dialplan |
| 15:22.28 | jwiggins | you will deff. need a lot of macros that deal with passing the office_id argument |
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| 15:22.50 | jwiggins | yup |
| 15:23.13 | jwiggins | make a macro file you can call from to perform all your major jobs, such as dialing an outside line |
| 15:23.46 | moex | I think the concept is not really sinking in yet, so I cant really ask for help.. lol.. I think I have to read first before I make a fool out of myself :$ |
| 15:23.50 | jwiggins | then setup each office in its own extensions file and define them in your primary extensions.conf so you don't make a unmanageable extensions.conf file |
| 15:24.10 | jwiggins | you wont find a whole lot of info on mutli-tenancy |
| 15:24.31 | Gugge | moex: how about you make _XX in the A1 dialplan, and then end that with a goto to the stdext, insteed of including it in A1 ? |
| 15:24.38 | BugKhaM | is there a way to check which codec module is used as there are multiple codec_g729-xxx.so in my /usr/lib/asterisk/modules |
| 15:24.48 | jwiggins | just think of it like an object oriented project where each office is another object |
| 15:25.21 | Gugge | or you could use setvar on the sip peers |
| 15:25.26 | jwiggins | BugKhaM, in use on a given channel, or available? |
| 15:25.59 | moex | hmm.. I dont think I really understand that.. hehe no worries, I'll go do my reading first.. I came here thinking it was just a simple Set issue.. apparently not |
| 15:30.19 | BugKhaM | jwiggins: on a given channel i suppose but "module show like 729" also multiple 729 codecs available |
| 15:33.20 | jwiggins | On a given channel you can use "core show channel X" |
| 15:33.29 | jwiggins | and it should show what codec the call is in |
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| 15:38.30 | BugKhaM | jwiggins: yeah, it will just tell what codec is used i.e. g729 but it doesn't tell "g729 from codec_g729-xxx.so or codec_g729-yyy.so" |
| 15:39.16 | WIMPy | Look ate the modules usage count. |
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| 15:43.58 | BugKhaM | WIMPy: you mean from "module show"? they all have zero "Use Count" |
| 15:44.49 | WIMPy | While on a call that is |
| 15:45.06 | BugKhaM | WIMPy: i see |
| 15:47.42 | jwiggins | oh you are trying a open source and a non open source g729 codec I take it? |
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| 15:52.14 | BugKhaM | jwiggins: just wanna compare which codec module is working better |
| 15:53.25 | BugKhaM | WIMPy: I saw that now but the "Use Count" just stayed for a few seconds at the beginning of the call, is that normal? |
| 15:54.37 | WIMPy | No. |
| 15:54.49 | WIMPy | If it's in use you should see it there. |
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| 16:34.30 | Kobaz | anyone know how to get polycom phones to dial *6exten for directed pickup instead of the default of *6 |
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| 16:34.52 | Kobaz | i dont want to pick up any ringing phone, i want to pick up the specific ringing phone that the user selected on the button |
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| 16:46.37 | dym | Kobaz: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup |
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| 17:11.58 | Kobaz | dym: that doesn't have any information about the polycom side |
| 17:13.37 | dym | http://lists.digium.com/pipermail/asterisk-users/2011-February/258733.html |
| 17:13.49 | dym | first hit on google for "asterisk polycom pickup" |
| 17:15.25 | Kobaz | and that's my thread |
| 17:15.36 | Kobaz | legacy pickup works fine |
| 17:15.47 | Kobaz | no one has ever answered how to get directed pickup working |
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| 17:16.39 | Kobaz | i can't get the polycom to send *6<exten> |
| 17:16.43 | Kobaz | it just sends *6 |
| 17:16.51 | high-rez | I have a weird problem with a T1 card. The driver doesn't seem to want to load: wcopenpci: [00] busy wait FAILED at 671... wcopenpci: [00] Firmware version 0 not supported by this driver -- anyone seen this before ? |
| 17:21.54 | WIMPy | Missing firmware? |
| 17:22.22 | Kobaz | WIMPy: newest |
| 17:22.29 | Kobaz | oh, that was for him |
| 17:22.35 | WIMPy | yes :-) |
| 17:23.06 | WIMPy | Polycoms didn't cross my way so far. They seem pretty rare over here. |
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| 17:52.42 | high-rez | WIMPy: Got all of the firmware. She just doesn't wanna work :/ |
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| 17:55.03 | p3nguin | moex: You can't pick a random extension in a context to set some variables. You have to use whatever extension you're wanting to set the variables within. s is not a catch-all; s is literally extension s. |
| 17:56.07 | sundancer | hi all.. i have asterisk up and running for several years now, everything ok... now i would like that anyone can send INVITE to somenumber@my.asterisk.ipv4.address and that INVITE should end up with ringing some extension i have defined |
| 17:56.15 | sundancer | but i have no idea how to implement that |
| 17:56.25 | p3nguin | moex: And if that extension starts spawning off new channels, the variable will not get inherited by the new channels unless you precede the variable name with an underscore or two, depending on how much inheritance is required. |
| 17:56.30 | sundancer | which context and do i need allowguests=yes in sip.conf ? |
| 17:57.49 | p3nguin | sundancer: To accept anonymous calls, you have to have allowguests=yes in the general section of sip.conf, and you have to have a matching extension in the context that you have defined in the general section of sip.conf. |
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| 17:58.33 | p3nguin | So if you put context=misc_calls in general section, anonymous calls will go into misc_calls looking for the dialed extension. |
| 17:59.16 | p3nguin | And, by the way, extensions don't produce sounds, so extensions can't ring. |
| 17:59.45 | sundancer | thanx, yes i expressed that oddly... :) |
| 18:01.00 | moex | p3nguin, thanks for that info (that's where I was confused).. I actually started thinking differently.. As I name my contexts lets say "A1-A4", so I basically use ${CONTEXT:1} .. so for A1 it will produce "1" and so on.. that is sort of what I needed with the variables |
| 18:01.34 | moex | I'm still trying to wrap my head around some of the logic though :D |
| 18:01.51 | p3nguin | Which part are you still having trouble with? |
| 18:02.24 | moex | nothing specific.. I'm just going about trying stuff, and finding out how it works untill I feel comfy enough to actually write a real dialplan |
| 18:03.28 | p3nguin | I would recommend not trying anything until you understand samples and examples. Then when you're ready to try it and it doesn't work like you hoped, you'll know what to do to fix it. |
| 18:04.15 | moex | well.. I do understand them (sorta).. it only gets complicated when I try to use the noodle!! :$ |
| 18:13.14 | ChannelZ | Does anyone have a Comcast email address? |
| 18:15.10 | ChannelZ | Actually nevermind. I think this guy is just dumb and doesn't even know his own address.. he gave @comcast.com but I'm pretty sure all customers are @comcast.net |
| 18:15.26 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 18:24.18 | moex | quick question.. if I have this "call setup" sip_phone -> asterisk -> dahdi -> normal_phone.. when I hang up the normal phone, it just gives the "busy beep" on the sip_phone.. is there a way around this? |
| 18:24.45 | p3nguin | What do you want to happen? |
| 18:24.50 | moex | normal_phone = a dialed number |
| 18:24.58 | moex | well.. dahdi should hangup.. shouldnt it? |
| 18:25.09 | moex | seems like the line stays open |
| 18:25.18 | WIMPy | What kind of dahdi interface? |
| 18:25.35 | moex | FXO TDM800P or something like that |
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| 18:26.29 | WIMPy | Call states usually don;t work with analog stuff. |
| 18:26.54 | moex | is there a way I can make it check for that "busy tone", and hang up when it hears it? |
| 18:27.25 | p3nguin | The person using the SIP phone could just hang up his handset. |
| 18:27.41 | WIMPy | 'callprigress' or something. But it might hang up calls if someone talks like a busy signal. |
| 18:28.12 | moex | p3nguin, true. |
| 18:28.19 | moex | WIMPy, thanks.. I'll check that out |
| 18:29.47 | p3nguin | Sometimes it's easier to train users to do normal things, as opposed to rewriting the existing technology. |
| 18:30.43 | moex | p3nguin, its actually for my home.. but I'm actually trying to understand the whole concept first |
| 18:32.28 | p3nguin | You spend THAT MUCH money on an analog card for Asterisk in a home phone system?! |
| 18:33.29 | ChannelZ | or cheap chineese clone |
| 18:33.49 | moex | sorry? lol |
| 18:33.54 | p3nguin | Even the clones are expensive, aren't they? |
| 18:34.07 | WIMPy | That's what you do if you live in an area where you can't get the much cheaper BRI stuff, or so I heard. |
| 18:34.25 | p3nguin | That's crazy talk. |
| 18:35.16 | ChannelZ | I don't think ISDN is very cheap in the US |
| 18:35.22 | ChannelZ | at least last time I looked |
| 18:36.13 | p3nguin | If he's using a regular phone line into a card, I think the cheaper route would have been to get one of those $70 ATAs with an FXO port. |
| 18:36.13 | WIMPy | No, that's obviousely the reason Analog stuff still exists. |
| 18:38.48 | *** join/#asterisk Karen_m (~karen@d50-99-60-236.abhsia.telus.net) |
| 18:39.12 | Kobaz | hmm |
| 18:39.24 | Karen_m | I got a new polycom phone, it says to download the activation file into the phone's configuration directory and reboot the phone. How do you do this? |
| 18:39.31 | Kobaz | is there anything different in 1.8 versus 1.6 to get voicemail notifications on a sip phone? |
| 18:39.49 | moex | ok, so busytone did the trick |
| 18:40.14 | moex | this is actually pretty neat.. |
| 18:40.24 | DrDigital | if dialing an external phone number in like an extension don't you have to put a # in the begging or at the end? |
| 18:40.59 | DrDigital | oops thought i was in #freepbx |
| 18:41.18 | ChannelZ | Kobaz: not that I'm aware of.. all still works for me |
| 18:43.10 | Kobaz | hmm |
| 18:43.17 | Kobaz | i don't get voicemail notifications with 1.8.5 |
| 18:43.35 | Kobaz | <PROTECTED> |
| 18:43.42 | Kobaz | it's set up |
| 18:43.43 | ChannelZ | With SIP debug on it sends no notify? |
| 18:44.42 | ChannelZ | I dunno what's going on, someone else was here yesterday having problems too. But it's working for me.. the only difference is all my stuff is in @default so I just specify the mailbox name, but I even tried it with the @default and it still worked. |
| 18:47.38 | Kobaz | hmm |
| 18:47.49 | Kobaz | core restarted, now it's sending notifies |
| 18:48.04 | Kobaz | sounds like a bug |
| 18:48.07 | ChannelZ | guess it was just confused |
| 18:59.37 | Karen_m | can someone PLEASE tell me how you download a license file onto a soundpoint phone? |
| 19:00.56 | WIMPy | The manual that came with it? |
| 19:01.16 | ChannelZ | TFTP I imagine |
| 19:03.51 | Karen_m | there is no manual that came with it.. it says, download license key. 7. Load the activation file into the phone's configuration directory and reboot the phone (please ensure that the appropriate feature flags are set to "1" In the SIP.CFG file). |
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| 19:06.54 | ChannelZ | How did you upload configs to it in the first place? |
| 19:07.19 | WIMPy | places a bet on "not". |
| 19:08.44 | ChannelZ | config-whaa? |
| 19:10.32 | Karen_m | i have not uploaded configs at all |
| 19:10.34 | Karen_m | it's a brand new phone |
| 19:10.40 | Karen_m | i'm trying to configure it as well |
| 19:11.03 | ChannelZ | well google "polycom soundpoint configuration" - surely they have a million PDFs on their own site |
| 19:11.07 | WIMPy | Brand new, but without any manual? |
| 19:11.45 | ChannelZ | http://support.polycom.com/PolycomService/support/us/support/voice/index.html |
| 19:22.06 | Karen_m | sigh, all those docs don't tell you how to configure it. I go to hit setup, it asks me for a password. NO documentation came with this thing |
| 19:22.10 | Karen_m | great reseller :) |
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| 19:23.36 | Karen_m | the problem is, it says it has an ip, but doesn't respond :) |
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| 19:28.33 | Karen_m | 456 |
| 19:42.58 | sundancer | today i found bruteforce attack to my asterisk where they were trying to find valid extension and then try different passwords.. is there a way not to tell attacker about valid extension ? |
| 19:43.27 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:43.32 | WIMPy | 'alwaysauthreject' |
| 19:43.47 | sundancer | tnx |
| 19:43.56 | WIMPy | And don't use sample or defalut configurations. |
| 19:47.55 | ChannelZ | or extension numbers for SIP device names |
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| 20:00.20 | sundancer | any other steps i can take in order to harden bruteforcer's success? |
| 20:02.03 | ChannelZ | do you really need SIP access from *anywhere* that you have to leave it totally open? |
| 20:02.40 | WIMPy | Did you look at README-SERIOUSLY...? |
| 20:04.09 | WIMPy | I guess the general rule is: Don;t make anything available to the internet before you fully understand the whole configuration. |
| 20:04.31 | WIMPy | And set strictrtp. |
| 20:04.37 | sundancer | ChannelZ: not really, i can VPN to asterisk box and then REGISTER over VPN |
| 20:04.52 | ChannelZ | so firewall off port 5060. Problem solved |
| 20:04.56 | sundancer | WIMPy: will reread :) |
| 20:06.09 | dym | lol |
| 20:06.14 | dym | ideal suggestion, WIMPy |
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| 20:22.58 | BenC[UK] | Evening guys, I have a bit in my dialplan to send a number to an AGI script if its busy... |
| 20:23.08 | BenC[UK] | how can I get the number that was busy in the agi script? |
| 20:23.28 | BenC[UK] | I send in two numbers to the dial plan together to be called |
| 20:26.31 | WIMPy | It Dial returns BUSY, all destinations were busy or unreachable. |
| 20:26.42 | WIMPy | I think I said so yesterday. |
| 20:33.18 | BenC[UK] | ah yeah, you did, let me check again , I must be doing something wrong |
| 20:33.28 | BenC[UK] | as one of the numbers should be valid |
| 20:34.56 | WIMPy | If at least one device has been ringing when the timeout is reached you should see NOANSWER. |
| 20:35.06 | BenC[UK] | you're right, I see a no answer |
| 20:35.20 | BenC[UK] | I am doing some tests.. |
| 20:35.27 | BenC[UK] | I am sending a valid number, and an invalid one |
| 20:35.33 | BenC[UK] | I need to get teh status on the invalid one.. |
| 20:35.35 | BenC[UK] | is that possible? |
| 20:35.41 | BenC[UK] | I need to know if a number fails |
| 20:36.20 | ChannelZ | In a multi-dial it doesn't really keep track of that for each dial target, at least as far as the dial status is concerned |
| 20:37.21 | BenC[UK] | is there a way around it? can I use two dials in the extension or something? |
| 20:37.39 | ChannelZ | not if you want them to go simultaneously I dont think |
| 20:38.04 | moex | has anyone seen a chan_sip.c __sip_xmit blah hostname:port returned -1: Host is down warning on the cli? |
| 20:38.10 | ChannelZ | I don't know if AMI might spit out something useful |
| 20:41.20 | moex | hmm.. sip unregister <extension> seems to have fixed it |
| 20:41.31 | ChannelZ | moex: not unless the host is down |
| 20:41.40 | moex | yeah.. the host was infact down |
| 20:41.48 | moex | but shouldnt it unregister that extension automatically |
| 20:41.59 | ChannelZ | depends |
| 20:42.11 | ChannelZ | Is it dynamic or is it configured with a static host= |
| 20:42.18 | moex | dynamic |
| 20:43.30 | WIMPy | AMI should give information per channel. |
| 20:43.53 | WIMPy | Or you could dial all destinations via local channels and save the dialstatus there. |
| 20:44.11 | WIMPy | No. It does. :-) |
| 20:51.19 | BenC[UK] | OK, I will check that then, sounds good |
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| 21:49.09 | NephFL | hello |
| 21:53.22 | ChannelZ | Hi. |
| 21:53.56 | NephFL | My digium card decided to stop working today...and I have no idea why... |
| 21:54.08 | ChannelZ | Depression |
| 21:54.22 | NephFL | I believe it... |
| 21:54.49 | NephFL | the dahdi tool and cfg seem to say it is working fine |
| 21:55.00 | NephFL | but calls are failing |
| 21:55.57 | ChannelZ | What kind of card is it first of all |
| 21:56.15 | NephFL | aex800 |
| 21:56.29 | NephFL | 8 fxo ports and echo canellation board |
| 21:57.02 | ChannelZ | Does the console show anything interesting? |
| 21:57.29 | NephFL | Just text... |
| 21:57.42 | ChannelZ | Does the text say anything interesting? |
| 21:57.42 | NephFL | yesterday the system was rebooting randomly |
| 21:57.53 | ChannelZ | You say "calls are failing" but that's not much to go on |
| 21:57.54 | NephFL | today, it had a pcie error, so I pulled a secondary nic |
| 21:58.31 | NephFL | have also swapped motherboard and cpu |
| 21:59.34 | NephFL | I'll grab that error |
| 22:00.38 | NephFL | CHANUNAVAIL HANGUPCAUSE: 0 |
| 22:01.50 | ChannelZ | do your phone lines actually work? |
| 22:01.52 | NephFL | the analog card was working after changing the hardware, but not after pulling NIC, but not sure what caused the actual problem... |
| 22:01.54 | NephFL | yes |
| 22:02.11 | NephFL | tested each, one has a stutter tone so i disconnected it... |
| 22:02.17 | NephFL | but it was working yesterday.... |
| 22:02.38 | NephFL | but yes, good dialtone, and shows connected in dadhi_tool |
| 22:02.43 | ChannelZ | stutter tone.. voicemail from your telco? |
| 22:02.46 | NephFL | yes |
| 22:03.36 | ChannelZ | incoming calls get ignored? |
| 22:04.21 | NephFL | incoming call to main number rings and eventually goes to voicemail...other labelled numbers do various things... |
| 22:05.40 | ChannelZ | and you don't see "starting simple switch on ...." when you call |
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| 22:26.16 | NephFL | I just rechecked the lines and called the main number and am getting ivr |
| 22:29.13 | ChannelZ | which means what |
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| 23:24.53 | kaushal | what does group=0 mean in chan_dahdi.conf ? |
| 23:25.40 | WIMPy | It sets the group dor Dial(dahdi/g0/... or G or r or R. |
| 23:25.54 | kaushal | WIMPy: ok |
| 23:26.03 | kaushal | so does it mean callgroup ? |
| 23:26.34 | WIMPy | What is your definition of a callgroup? |
| 23:26.38 | WIMPy | It's for line hunting. |
| 23:26.58 | kaushal | ok |
| 23:27.15 | kaushal | WIMPy: Any docs which mentions about it |
| 23:28.07 | WIMPy | Only tthe sample config, as far as I know. |
| 23:28.11 | kaushal | ok |
| 23:28.58 | kaushal | so whats the difference between group=0 and callgroup=0 |
| 23:29.04 | kaushal | not sure i still understand |
| 23:29.53 | WIMPy | callgroup? Hmm. Is that for group pickup? |
| 23:30.10 | kaushal | WIMPy: let me pastebin the config |
| 23:32.42 | WIMPy | yes, it's the opposite to pickupgroup if I got that right. |
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| 23:50.56 | kaushal | WIMPy: http://sprunge.us/SZRC |
| 23:54.03 | WIMPy | Looks familiar |
| 23:54.16 | kaushal | WIMPy: yeah |
| 23:54.32 | kaushal | WIMPy: so what does g0 mean ? |
| 23:54.40 | kaushal | by definition |
| 23:55.31 | WIMPy | Use the first available channel of group 0. |
| 23:55.59 | kaushal | ok |
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