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| 02:02.53 | igcewieling | ha! I put some things where my cat likes to sleep at the end of my desk, so she stuck her paw in my glass of water. |
| 02:08.34 | drmessano | I can see that |
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| 02:24.11 | DannyA | hey all. does anyone know of a linux program/daemon that will do constant pcap capture and stream the data somewhere? |
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| 08:08.51 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 12:29.21 | DarkPonay | Hi everyone, is someone using the patch usecallmanager.nz for supporting Cisco phones? |
| 12:34.27 | Samot | Many are from what I can tell. |
| 12:40.39 | DarkPonay | Oh ok, i'm facing up some difficulties with it, and as i am a bit new to asterisk, probably misconfigured it since i'm trying everything i find to face the problems i have... |
| 12:43.05 | Samot | Well what's the problem? |
| 12:46.28 | DarkPonay | I often have "Retransmission timeout reached on transmission", so i think i didn't configure Asterisk properly, and/or added usecallmanager entries in a good way. It is a new asterisk 16.10 installation, so i was trying to make a sort of lab, using exclusively the documentation |
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| 12:47.19 | Samot | Well that means Asterisk is trying to send the transmission again because it didn't get a reply. |
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| 12:49.38 | DarkPonay | And since i plan to use some video phones (i have for testing two 8845 and two 7975) i have strange behaviours when using video (like a very very low resolution) and it is a mess when i try to conference between video and non video devices (all cisco with SIP firmware) |
| 12:50.22 | DarkPonay | Samot, yeah that's what i've found but everything is on the same lan, so i suppose my config is wrong somewhere |
| 12:50.59 | Samot | Possible. |
| 12:51.14 | Samot | Honestly, I don't use Cisco phones and never used that patch so can't really say if that is the issue |
| 12:52.18 | DarkPonay | Even without the patch, i had some "bad file descriptor" or "database I/O error" |
| 12:53.03 | Samot | That sounds like a bad install/setup |
| 12:53.37 | DarkPonay | i reinstalled the system and asterisk multiple times... As i am new i am using some tutorials to perform the install. |
| 12:54.21 | Samot | How are you installing Asterisk? |
| 12:54.24 | DarkPonay | i have another VM with asterisk 16 and chan-sccp which is installed in the same way and wich is working fine... |
| 12:54.25 | Samot | From source or packages? |
| 12:54.40 | DarkPonay | from source, because i neek to apply the patch |
| 12:54.45 | DarkPonay | need* |
| 12:55.02 | Samot | Well let me ask this, how many of these phones do you have? |
| 12:56.48 | DarkPonay | on the test asterisk SIP machine, only 4. But on my working SCCP system i have 25 of them that i would like to migrate to SIP, since we will replace majority of them by new Cisco only SIP models |
| 13:00.11 | DarkPonay | i like doing tests with 4 phones, so i can test different scenario like conferencing, transfers, queues, etc |
| 13:01.19 | Samot | Well I was asking for other reasons. |
| 13:01.39 | Samot | Cisco phones on non-Cisco systems is not the best choice. |
| 13:02.06 | DarkPonay | I perfectly know, and i agree with you so much ! :D |
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| 13:03.27 | Samot | So if you don't patch Asterisk first, do the basics work? |
| 13:03.39 | Samot | Without the patch, simple register and calls should still work. |
| 13:03.45 | Samot | Just can't use most of the features of the phone. |
| 13:03.49 | DarkPonay | The very basic works, yes. |
| 13:04.02 | Samot | So this only screws up once the patch is installed? |
| 13:04.47 | DarkPonay | No no it is almost the same, it works with and without, but i have the sip retransmit errors, sometimes the bad file descriptor when a phone registers... |
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| 13:06.52 | DarkPonay | I'm following this install procedure http://usecallmanager.nz/patching-asterisk.html which is very basic as you can see. I tested to do it exactly like this. |
| 13:08.31 | DarkPonay | here i modified it a little, like "make menuselect" to enable some different things and selecting messages in my languages |
| 13:09.33 | DarkPonay | and i am not performing "make samples" because there are too many files, and i prefer using "make basic-pbx" and start my modifications from there |
| 13:13.10 | DarkPonay | i'm doing all this on a basic Debian 10 installation |
| 13:13.38 | Samot | What are the actual errors you are getting? |
| 13:16.34 | DarkPonay | Sometimes the "SIP retransmit", "bad file descriptor", problems with video (wrong resolution, activated when i call a non-video phone), and some bugs related to the dialplan added with the patch... |
| 13:16.56 | DarkPonay | Then it is special because these phones works in TCP mode |
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| 14:42.52 | avb | gentlemen, is it normal that CDR(xxxx) gets/sets are not working in the hangup handlers? |
| 14:43.10 | avb | or Im missing something |
| 14:54.22 | sibiria | did you enable CDRs at all? |
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| 14:54.56 | sibiria | the cdr backend is needed for these functions to return data, even if you are not using the actual cdr LOGGING modules |
| 14:55.31 | avb | sibiria: yes, its enabled |
| 14:55.51 | avb | everything is working outside of the hangup handler |
| 14:56.26 | avb | but if you do hangup_handler_push, and inside the hangup handler you use Set(CDR(xxx)=yyy) -- it has no effect |
| 14:57.10 | avb | <PROTECTED> |
| 14:57.10 | avb | Call Detail Record (CDR) settings |
| 14:57.10 | avb | ---------------------------------- |
| 14:57.10 | avb | <PROTECTED> |
| 14:57.10 | avb | <PROTECTED> |
| 14:57.13 | avb | <PROTECTED> |
| 14:57.15 | avb | <PROTECTED> |
| 14:57.18 | avb | * Registered Backends |
| 14:57.20 | avb | <PROTECTED> |
| 14:57.23 | avb | <PROTECTED> |
| 14:57.25 | avb | <PROTECTED> |
| 14:57.28 | avb | <PROTECTED> |
| 14:59.52 | igcewieling | avb: don't flood. did you read this section of the sample config? https://pastebin.com/hfLNieD4 |
| 15:02.15 | avb | i did |
| 15:02.18 | avb | igcewieling: > By enabling this option, the CDR will be ended before executing |
| 15:02.21 | avb | ; the "h" extension and hangup handlers so that CDR values such as "end" and |
| 15:02.24 | avb | ; "billsec" may be retrieved inside of of this extension. |
| 15:02.24 | avb | oh, sorry |
| 15:02.36 | avb | endbeforehexten is set to default |
| 15:02.51 | avb | so its set to 'no' |
| 15:03.06 | avb | i have tried to force set it to no as well |
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| 15:48.06 | avb | looks like related to the Local/ channels im using all over |
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| 16:50.40 | igcewieling | Local/ channels are a hassle. I avoid them everywhere except for outgoing efaxes which are processed using spool files. |
| 16:55.11 | file | they are both useful, and painful |
| 17:00.00 | igcewieling | The one annoying thing about Local channels is they get a new log-id |
| 17:00.17 | igcewieling | s/The// |
| 17:02.47 | file | I added multistream support to them last week |
| 17:02.49 | file | it hurt my head |
| 17:03.28 | igcewieling | multistream support? |
| 17:04.23 | file | Asterisk 16 and above has support for media streams on channels, but stuff has to be updated to take advantage of it/be aware of it |
| 17:04.42 | file | so I added it so that if there's a Local channel in between a PJSIP channel and ConfBridge then video conferencing still works |
| 17:05.02 | igcewieling | Ah, that is helpful. |
| 17:05.12 | file | it hurt my head particularly because streams have a state - send/receive, send only, receive only, inactive, removed |
| 17:05.36 | file | and that state is from the perspective of Asterisk to the other side... so with a Local channel in the mix... they are reversed on each side... |
| 17:05.47 | file | it was slightly confusing until I remembered that. |
| 17:51.40 | avb | igcewieling: i got two hassles in one. originate ami + Local channel :) |
| 17:52.23 | avb | then it goes to the Queue with a Local/ channel again... |
| 17:52.49 | avb | dumped CDRs and now writting everything to mysql directly via ODBC |
| 17:53.04 | avb | live became way easier right away |
| 17:54.52 | avb | too bad that Originate can accept only one context and a channel |
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| 19:55.27 | forgotmynick | Hi. I have this "problem" since the beginning which I've never got around to looking into. Say if an agent receives a call and has to "transfer it", the agent needs to speak to the team before transferring the call (a process that will change end of this year). So the agent will directly dial the teams queue number, wait for an answer, speak to them, transfer the call and disconnect. If the team take a while to |
| 19:55.27 | forgotmynick | answer, the call gets dropped - what can I do? chan_sip.c: Disconnecting call 'SIP/1002-000000eb' for lack of RTP activity in 301 seconds. |
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