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| 05:42.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 07:08.35 | bananatreedad | is somebody familiar with the term actionurl? |
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| 08:03.41 | markusl | in the context of snom/unify phones, yes. |
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| 08:21.10 | LnxBil | Hi everyone |
| 08:22.35 | LnxBil | I'd like to know how you guys integrate home-office/on-the-go users which have also stationary phones but need to be available under one extension. I figured that this is not possible in asterisk. The phone with the latest 'register' rings |
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| 08:42.35 | markusl | LnxBil: you need a seperate sip peer for each phone and ring them in parallel |
| 08:43.06 | markusl | it requires a bit of work in the dialplan but is possible. |
| 08:43.54 | markusl | or you could put something in front that can do call forking like a sip proxy, sbc or freeswitch. |
| 08:44.09 | markusl | but i guess it's way easier to just add another sip peer and ring them both |
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| 09:01.01 | LnxBil | markusl: Okay, that's what we thought, too. |
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| 09:55.52 | stefan27 | is there any static code analyzers on dialplan that detects possible syntax errors? |
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| 10:07.49 | Zogot | hey all. have the unusual issue where a second dial right after another dial, doesn't fire a second app_start in CEL |
| 10:10.31 | Zogot | gist is taking ages, its a large dump of the output from the manager lol |
| 10:10.46 | Zogot | 1 sec, il have the examples |
| 10:14.00 | Zogot | ok, this is one that isn't working: http://codebeautify.org/jsonviewer/212df9 |
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| 10:14.12 | Zogot | if you open up the debugs section, you'll see every event from CEL |
| 10:14.28 | Zogot | its sorted by LINKEDID -> UNIQUEID |
| 10:15.28 | Zogot | http://codebeautify.org/jsonviewer/26b92a |
| 10:15.30 | Zogot | is a working one |
| 10:15.40 | Zogot | you can see in that one, there is 2 app_starts for the Dial application |
| 10:15.46 | Zogot | by adding a NoOp between the 2 dials |
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| 10:19.38 | imihaylov | Hello guys! I have problem with Asterisk ( Asterisk 11.7.0~dfsg-1ubuntu1) call files. When moved to /var/spool/asterisk/outgoing_done/ Asterisk doesn't write any status. How to make Asterisk write status of the test call from file? |
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| 10:22.53 | Zogot | imihaylov: theres no Status line when Archive: yes? |
| 10:23.46 | imihaylov | #Zogot Yes. The file is exactly the same as in outgoing directory |
| 10:25.58 | imihaylov | and no mater if I pick up or leave it ringing till the wait time expires |
| 10:27.34 | Zogot | imihaylov: im gonna eat lunch, il see if i can reproduce |
| 10:27.40 | Zogot | once i get back |
| 10:27.51 | imihaylov | thanks Zogot! |
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| 10:53.46 | Zogot | imihaylov: mine is adding Status: to the call file in outgoing_done |
| 10:53.52 | Zogot | imihaylov: what is your call file? |
| 10:56.10 | imihaylov | Channel: SIP/out_trunk/number-to-call \ Application: Playback \ WaitTime: 10 \ MaxRetries: 0 \ CallerID: 'Asterisk test' <99999> \ Archive: Yes |
| 10:56.15 | imihaylov | \ is new line |
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| 11:15.13 | Ice_Strike | timeout is specified on the queue, so if timeout is expired - what event it will return on AMI? |
| 11:18.41 | imihaylov | <Ice_Strike> is that a question to me ? |
| 11:20.17 | Ice_Strike | No |
| 11:20.23 | Ice_Strike | Anyone that know the answer. |
| 11:22.56 | Ice_Strike | I think it might be QueueCallerAbandon |
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| 12:04.07 | WIMPy | Isn't "Abandon" when the caller hangs up? |
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| 12:12.36 | Ice_Strike | Not sure |
| 12:13.38 | Ice_Strike | "Raised when an caller abandons the queue" you might be right. |
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| 12:36.17 | imihaylov | Ok I managed to get the status but why it says Expired even though I hang up and in asterisk cli I get Call failed to go through, reason (5) Remote end is Busy |
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| 12:36.52 | imihaylov | Queued call to SIP/trunk/number expired without completion after 0 attempts |
| 12:39.04 | WIMPy | Number of retries expired? |
| 12:39.19 | WIMPy | "expired" |
| 12:45.13 | imihaylov | MaxRetries = 0 |
| 12:45.19 | imihaylov | WaitTime = 15 |
| 12:45.43 | imihaylov | it calls me , i hang up and in the archive file the status is expired |
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| 15:29.08 | pjensen00 | Question about call recordings in Stasis. If I mute a channel (not the recording) does that pause the recording? It seems like that's what's happening. |
| 15:34.01 | mjordan | It won't pause the recording. |
| 15:34.07 | mjordan | Mute only affects the audio stream on the channel. |
| 15:34.58 | mjordan | If you want to pause the recording, you can use the Recordings resource to pause the recording: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Recordings+REST+API#Asterisk13RecordingsRESTAPI-pause |
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| 15:36.53 | cervajs2 | hi. any ideas why i have this problem? ERROR[9551]: pjsip:0 <?>: sip_endpoint.c Error processing packet from x.x.x.x:22866: Rx buffer overflow (PJSIP_ERXOVERFLOW) |
| 15:37.14 | cervajs2 | asterisk13+chan_pjsip+sipml5+chrome |
| 15:37.26 | cervajs2 | pjproject 2.3 |
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| 15:40.53 | mjordan | cervajs2: pjsip has a max packet size that can be exceeded by some webrtc communication, depending on what all is offered |
| 15:41.01 | mjordan | you can bump the max packet size in your site_config file in pjproject |
| 15:43.01 | mjordan | I've used #define PJSIP_MAX_PKT_LEN 12288 before and had it work well with WebRTC endpoints. |
| 15:43.07 | cervajs2 | ok.compiling 2.4. will try |
| 15:43.09 | mjordan | (I suspect this is your problem, at any rate) |
| 15:43.55 | igcewieling | can't wait for Asterisk 15 |
| 15:44.20 | file | yeah, the current max size of a packet is... 4000 |
| 15:44.25 | file | or rather, the default |
| 15:44.34 | file | and you DO need to rebuild Asterisk |
| 15:44.53 | cervajs2 | ok.tnx |
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| 20:04.50 | mub | I'm having an issue with ratio/adaptive/predictive dialing. This is on an OSDial box which I'm not accustomed to. |
| 20:05.11 | mub | I've created a test campaign and a test list with only my cell number in it. |
| 20:05.22 | mub | http://pastebin.com/tQsWmR71 |
| 20:05.33 | mub | sorry for agi debugging being on |
| 20:06.02 | mub | Manual dialing works fine, any other mode fails however |
| 20:06.20 | mub | I get the call on my cell, answer it, hear nothing, then it hangs up on me |
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| 20:09.05 | mub | It seems that a whole lot of people have this problem, but I'm getting a bunch of "Conference number '8601000' not found!" in my /var/log/asterisk/sip-debug |
| 20:09.28 | mub | and /etc/asterisk/meetme.conf and osdial_meetme.conf are both empty |
| 20:10.56 | mub | No one seems to mention that |
| 20:17.36 | reveal | mub: try looking at /etc/asterisk/osdial_extensions_conferences.conf |
| 20:20.47 | mub | reveal: This is all that's in it: |
| 20:20.51 | mub | exten => _487488.,1,Set(CDR(accountcode)=${EXTEN:6}) |
| 20:20.52 | mub | exten => _487488.,2,Goto(osdial,487488,1) |
| 20:20.54 | mub | exten => _487489.,1,ChanSpy(,g(${EXTEN:6})qES) |
| 20:20.56 | mub | exten => _487489.,2,Hangup() |
| 20:21.12 | reveal | <PROTECTED> |
| 20:21.12 | reveal | ; quiet entry and leaving conferences for OSDIAL 8601000 - 8601049 |
| 20:21.13 | reveal | ; quiet monitor extensions for meetme rooms (for room managers) 8601000 - 8601049 |
| 20:21.17 | reveal | right should be in there |
| 20:21.33 | mub | Nope, no comments |
| 20:21.55 | reveal | do you hangup when it connects as an agent |
| 20:22.53 | mub | I have a few times, but usually I don't. The result appears to be the same |
| 20:23.55 | reveal | are you using softphones like xlite or zoiper |
| 20:24.35 | reveal | im assuming you also read http://forum.osdial.org/viewtopic.php?f=5&t=643 |
| 20:26.28 | Vince-0 | mub, |
| 20:26.40 | Vince-0 | conference not found is a common error of Vicidial, from which OSdial is forked |
| 20:26.53 | reveal | im assuming they are the same |
| 20:27.13 | Vince-0 | yes, its not a problem |
| 20:28.14 | reveal | i ave osdial as well and it doesnt work at all |
| 20:28.35 | Vince-0 | stick to vici |
| 20:29.41 | reveal | couldnt get osdial to integrate with our pri wont dial our analog phones |
| 20:30.33 | reveal | mub: what about making sure your campaign allows inbound groups |
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