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| 01:00.21 | sweettea | for some reason the polycom phone's message waiting light wont go away, what should i check? |
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| 08:37.08 | davlefou | hi, this don't work : frame.c:821 ast_parse_allow_disallow: Cannot allow unknown format 'alaw&ulaw&ilbc' |
| 08:37.21 | davlefou | is it ; i need to use? |
| 08:42.07 | wdoekes | ; starts a comment |
| 08:42.10 | wdoekes | try , instead |
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| 08:51.46 | the-duke | hi, what is a reason that asterisk is not trying to register with a peer again? I have the case that provider responded with handle_response_register: Forbidden - wrong password on authentication for REGISTER and it seems no additional attempts have been made |
| 09:01.47 | davlefou | wdoekes, thanks. |
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| 09:30.43 | catphish_ | is there any way to see what codec(s) each active channel us using? |
| 09:32.07 | catphish_ | i need a way to monitor how many concurrent channels are using a licenced codec to make sure i maintain enough licences to cover the peak usage |
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| 10:21.31 | the-duke | hi, what is a reason that asterisk is not trying to register with a peer again? I have the case that provider responded with handle_response_register: Forbidden - wrong password on authentication for REGISTER and it seems no additional attempts have been made |
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| 10:28.32 | catphish_ | the-duke: this channel tends to be dead until about 4 hours from now |
| 10:28.53 | catphish_ | quite america-timezone |
| 10:32.19 | dan_j | Hi. In a queue, calling a macro on answer and setting a variable, sets the variable on the agent channel. Is there any way to set a variable on the caller's channel from within the dialplan? |
| 10:38.27 | the-duke | Thank you catphish_ |
| 10:50.46 | dan_j | What's the purpose of the 'channel' argument in the SHARED() function ? |
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| 11:04.19 | dan_j | Never mind. Got that answer. |
| 11:04.56 | dan_j | Next question, when using queues, how can I pass a variable to the macro being executed on answer? |
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| 12:30.52 | marceloamorim | guys, using a goip, it's necessary enable 100rel to transfer a call properly, without losing one side audio? |
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| 14:38.50 | marceloamorim | using a goip ( as gateway gsm)** |
| 14:39.06 | marceloamorim | ast13.cert4 cannot compile without pjsip? |
| 14:40.07 | [TK]D-Fender | You shouldn't be touching cert unless you have a Digium support contract |
| 14:41.10 | marceloamorim | so should I install 13.8.0 as my main ast server? |
| 14:42.19 | [TK]D-Fender | Generally yes |
| 14:42.30 | marceloamorim | ok, ty sir |
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| 14:43.47 | marceloamorim | I'm having a problem using vpn over tcp when I transfer a call using gsm gateway from dinstar. and I don't have enough knowledge to sip trace and find the problem |
| 14:44.49 | marceloamorim | so I will try to update even I didn't find related problem on changelog |
| 14:46.25 | [TK]D-Fender | How much is "not enough"? |
| 14:47.19 | [TK]D-Fender | And "having a problem" doesn't say a single thing about what the sympttoms are, what messages you get from the system, what you hear or anything at all. |
| 14:48.35 | marceloamorim | one way audio after I transfer |
| 14:49.05 | [TK]D-Fender | <[TK]D-Fender> How much is "not enough"? <- |
| 14:50.19 | marceloamorim | I should know all steps from sip, so I could know the head of packets and see to where the audio was sent. |
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| 14:53.00 | marceloamorim | and the other problem is, this is intermittent problem, so is even harder to find (for me of course) |
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| 14:58.59 | catphish_ | i'm seeing what appears to be a limit of about 500 characters on my dialplan app parameters? is there a specific limitation there? |
| 14:59.29 | catphish_ | for example, a long string passed to Dial() seems to be truncated, initially i thought this applied only to some apps, but it appears to be more global |
| 15:18.29 | [TK]D-Fender | it is |
| 15:18.40 | [TK]D-Fender | and it's just a hard-coding issue |
| 15:19.41 | catphish_ | [TK]D-Fender: can i change it easily with a constant? |
| 15:19.57 | [TK]D-Fender | dunno... |
| 15:20.03 | catphish_ | looks |
| 15:22.05 | catphish_ | although i have absolutely no idea where to look :( |
| 15:39.04 | catphish_ | i suspect this is the blameful data structure: https://github.com/asterisk/asterisk/blob/1.8/pbx/pbx_realtime.c#L344 |
| 15:46.04 | dan_j | catphish_: There doesn't seem to be a limit on static files. Only with realtime. |
| 15:46.16 | dan_j | I had that issue with realtime before I switched back to static files. |
| 15:46.34 | catphish_ | dan_j: i think static files have a limit on the parser of about 4k |
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| 15:56.56 | dan_j | catphish_: per extension or total? |
| 15:57.02 | catphish_ | per line |
| 15:57.06 | catphish_ | i believe |
| 15:57.30 | dan_j | catphish_: 4k is quite large amount |
| 15:57.51 | dan_j | for one line |
| 15:58.05 | dan_j | catphish_: what are you trying to achieve? |
| 15:58.37 | catphish_ | passing large numbers of destinations to dial() |
| 16:00.14 | dan_j | What are the destinations? Do they change regularly or are they fairly constant? |
| 16:00.23 | dan_j | You have a couple of options. |
| 16:00.51 | dan_j | 1. Make small groups of destinations and then call them all using Dial(Local/ channels |
| 16:01.21 | dan_j | 2. Create a queue. Put all the destination as queue members. Then you've just got one queue to call. |
| 16:02.55 | dan_j | If you are a programmer, you could look at resolving the issue with realtime if there is one and submitting it to be included in future releases. |
| 16:07.09 | catphish_ | i already posted a link to where i believe the limitation lies |
| 16:07.38 | catphish_ | i will test changing it |
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| 17:16.18 | syadnom | anyone here running elastixMT? |
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| 17:22.04 | malcolmd | boo |
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| 18:28.07 | ellenor | sweettea, voicemail |
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| 19:37.45 | hexanol | someone has already seen this kind of segfault on Asterisk 13.6.0: https://www.zerobin.net/?5b2be5cc7abc71e6#WYrvQvxoRg6DbIocbThSL748iTsTePhUwuONBS0bxns= ? |
| 19:38.22 | hexanol | the line offset are wrong btw, i.e. this is a patched app_queue unfortunately |
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| 19:38.45 | hexanol | I've taken a look on the tracker, but did not found anything that helped me |
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| 19:52.36 | newtonr | I don't recall one like that |
| 19:53.03 | newtonr | If you can reproduce it in a default app_queue with the latest 13 then I'd file an issue |
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| 19:57.38 | hexanol | newtonr: I agree but I don't know how to reproduce it yet |
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