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| 01:01.52 | Apocryphal | Okay, so audio works if the call recipient holds and unholds... |
| 01:02.00 | Apocryphal | What is this sorcery... |
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| 01:45.43 | phix | so, where does one purchase codecs from? Besides Digium (pricy) |
| 01:50.24 | [TK]D-Fender | You don't |
| 01:50.37 | [TK]D-Fender | because the owner isn't going to sell a handful to you any cheaper |
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| 02:01.40 | WIMPy | If you need G.729, you can watch out for transcoder cards. They can sometime be cheaper than software licenses. |
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| 02:09.18 | [TK]D-Fender | Depending if you can support it and reach that profitability point |
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| 02:30.13 | ruben23 | hi guys i have my asterisk server installed but i was task to find a specific call logs for a particular number like 143567834, how do i search for the logs of this particular call any idea guys and help Please |
| 02:32.42 | WIMPy | grep |
| 02:34.50 | [TK]D-Fender | Open in some GUI file viewer/editor. Ctrl-F. |
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| 04:11.27 | tompaw | Did anyone ever succeeded in playing a sound via ari? |
| 04:11.45 | ruben23 | hi does Master.csv log files, have a realtime log update somehow written..? |
| 04:11.52 | tompaw | No matter what I do, which format I use, I always get the error: "res_stasis_playback.c: 1439871073.155347: Playback failed for sound:fpl_p2" |
| 04:11.59 | tompaw | Debug shows nothing at all. |
| 04:13.19 | ruben23 | any idea guys..? if mastr.csv have written realtiem logs..? |
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| 04:18.53 | tompaw | Actually, console log shows: -- <PJSIP/flowroute-00006237> Playing 'fpl_p2.g729' (language 'en') |
| 04:19.04 | tompaw | ...and nothing else with debug & verbose = 4 |
| 04:22.17 | tompaw | The only format that gives me ANY feedback is mp3. |
| 04:36.27 | tompaw | Same file played directly from dialplan works perfectly fine... |
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| 04:42.05 | tompaw | Well, looks like an ARI issue. |
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| 06:02.31 | kolko | Hello. I think about add radius support for auth sip peers. Write own realtime module (like res_config_*.c) - it is right way? Or need patch chan_sip? |
| 06:03.31 | kolko | And why asterisk now not support radius? Maybe, this is bad way? But, for integration with billing systems - it porably good choice |
| 06:13.15 | kleszcz | kolko: http://www.voip-info.org/wiki/view/CW+Radius+for+Asterisk |
| 06:15.20 | kolko | My google was broken soo hard |
| 06:15.39 | kolko | Big thanks! (no fun with coding :( ) |
| 06:15.57 | kleszcz | no problem ;] |
| 06:17.29 | kolko | But i cant find aaa_ibsng_r.so, or any sourse code |
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| 06:23.58 | kolko | This module cant authorize sip-peers (login/password are set in sip.conf in example), auth can be with "debit cards" and all made in extension - this i can make with AGI+perl/python/php (many examples). CDR COLLECTOR - not need, asterisk can do it himself |
| 06:25.39 | kolko | Only way (how I can see in sourses) - write realtime module with radius requests inside |
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| 08:42.14 | Zogot | hey, im looking at CEL events from Asterisk 11, did something change in Asterisk 13 that a normal conversation between Alice and Bob isn't a bridge in 11? |
| 08:42.58 | Zogot | is there bridge events in CEL in asterisk 11? |
| 08:44.45 | Zogot | my mistake, different event names in 11 from 13 |
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| 09:19.05 | Zogot | asterisk 13 can't remove from the menuselect pjsip. At the moment we would only be looking to use chansip, is it capable and stable to not have pjsip with asterisk 13? |
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| 10:19.30 | yun1989 | hello all |
| 10:19.30 | *** join/#asterisk doome_ (~doome@82.150.48.146) |
| 10:19.52 | yun1989 | i have one problem i have one trunk between asterisk and Intelbras CIP 850 |
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| 10:21.02 | yun1989 | and it's works ok but i have one delay of 5 seconds to have audio |
| 10:21.08 | yun1989 | anyone can help me ? |
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| 11:34.29 | wasanzy | hello |
| 11:35.30 | wasanzy | I am not able to connect to SIP server, but I can ping, and when I use nmap I can see that the port is open. but I get this anytime I do sip show peers: |
| 11:36.37 | wasanzy | outgoing IP Yes Yes 5060 UNREACHABLE |
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| 12:45.55 | phix | nasty |
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| 13:01.05 | bananapie | Hi. When I call '101,1,Dial(SIP/user1&SIP/user2,90 ) Is there a way to set a variable on the channel 'SIP/user1' without using a local channel or using 'setvar' in sip.conf, all this without setting the variable to channel 'SIP/user2'? |
| 13:01.39 | [TK]D-Fender | nope |
| 13:01.48 | bananapie | Thanks |
| 13:01.53 | [TK]D-Fender | All or none... |
| 13:02.15 | [TK]D-Fender | Actually... maybe |
| 13:02.44 | [TK]D-Fender | I think if you run a macro/gosb on the answering channel before it answers you could check the channel name and do it based on that... |
| 13:02.48 | [TK]D-Fender | kludgy but might work |
| 13:03.13 | bananapie | Macros is the part of asterisk I know the least. I guess it's time for me to learn them. |
| 13:03.14 | bananapie | thanks |
| 13:03.53 | bananapie | macros are deprecated :( |
| 13:04.17 | [TK]D-Fender | not going away.... |
| 13:04.19 | bananapie | ok |
| 13:04.35 | [TK]D-Fender | use a gosub. practically the same thing minus the bitching |
| 13:04.43 | *** join/#asterisk Draecos (~Draecos@58-7-58-83.dyn.iinet.net.au) |
| 13:05.44 | phix | [TK]D-Fender: any good guis for asterisk? |
| 13:06.13 | bananapie | phix like trixbox? |
| 13:06.25 | [TK]D-Fender | FreePBX is the only "free" one worth looking at", everything else is commercial |
| 13:06.41 | [TK]D-Fender | trixbox is a whole distro, and one dead for nearly half a decode |
| 13:06.42 | *** join/#asterisk bulkorok (~Adium@89.245.151.228) |
| 13:06.43 | [TK]D-Fender | decade* |
| 13:06.52 | phix | bananapie: no, to manage calls, transfer, see who is callingm etc.. |
| 13:07.00 | bananapie | really? I didn't know trixbox was dead :| |
| 13:07.10 | bananapie | phix, sorry, I don't know of any gui to recommend to you |
| 13:07.24 | [TK]D-Fender | phix: Your definition of GUI should have gotten some serious clarification |
| 13:07.38 | phix | I dont want to use trixbox or a linux distribution with asterisk bundled, I would like to keep using ubuntu / debian |
| 13:07.41 | [TK]D-Fender | phix: What you want is a "receptionist console" type thing |
| 13:07.59 | [TK]D-Fender | phix: And there are bolt-on things for that. |
| 13:08.07 | [TK]D-Fender | FOP2, iSymphony, etc |
| 13:08.08 | phix | [TK]D-Fender: Yeah but a "console" being something a receptionist with low computer experience can use |
| 13:08.18 | phix | [TK]D-Fender: $$$$?? |
| 13:08.29 | [TK]D-Fender | pretty cheap |
| 13:08.34 | bananapie | The macro is called when the user answers ( Dial M ), so I think I'll keep using local channels. I should learn C at some point and write a patch to asterisk and submit it. |
| 13:09.31 | phix | [TK]D-Fender: link me |
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| 13:09.46 | [TK]D-Fender | http://www.google.ca |
| 13:10.14 | phix | Oh, you're a Canadian from Canadia? |
| 13:10.23 | pjensen00 | www.google.fr |
| 13:10.39 | phix | www.google.com.au imho |
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| 13:12.53 | bananapie | Canada! |
| 13:13.11 | phix | Oh Canada!! |
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| 13:22.43 | bananapie | hey, if you haven't decided who you are voting for. This might help |
| 13:22.43 | bananapie | http://imgur.com/MSWs2Lz |
| 13:34.13 | phix | I would like to vote for Al Gore |
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