|Summary:||ASTERISK-02561: No Audio / Fast busy on incoming call on T100P|
|Reporter:||Donny Kavanagh (donnyk)||Labels:|
|Date Opened:||2004-10-08 01:43:05||Date Closed:||2011-06-07 14:05:17|
|Description:||When asterisk handles the call (aka not bridged with sip or another channel on the pri) no audio is heard and the line drops with a fast busy.|
****** ADDITIONAL INFORMATION ******
I have had asterisk setup for a few weeks now, i am able to call between registered sip clients, call outbound through the pri no problem.
I am also able to call inbound to our did's and have asterisk handle the call and bridge it to a sip client or a sip server or whatever.
If i call from my sip phone to asterisk, i am able to access voice mail, play audio files etc.
here is the config for the line in extensions.conf
exten => 6261,1,Answer()
exten => 6261,2,Wait(30)
exten => 6261,3,Background(ivrmenu)
exten => 6261,4,Wait(30)
i put in the waits to make sure the it wasnt the play causing it to flip out.
If i call that extension from sip, it works fine, plays the file.
If i call a did which goes through the pri into asterisk i see this:
-- Executing Answer("Zap/5-1", "") in new stack
-- Accepting call from '613<num removed>' to '6261' on channel 0/5, span 1
-- Executing Wait("Zap/5-1", "30") in new stack
-- Channel 0/5, span 1 got hangup
== Spawn extension (sip, 6261, 2) exited non-zero on 'Zap/5-1'
-- Hungup 'Zap/5-1'
(i didnt hang up i just waited on the line, and i heard nothing)
if i call another extension however which is mapped to a sip server (phone) i get:
-- Executing Dial("Zap/6-1", "SIP/siptest|20") in new stack
-- Called siptest
-- Accepting call from '613<num removed>' to '6260' on channel 0/6, span 1
-- SIP/siptest-f4ca is ringing
-- SIP/siptest-f4ca answered Zap/6-1
-- Channel 0/6, span 1 got hangup
== Spawn extension (sip, 6260, 1) exited non-zero on 'Zap/6-1'
-- Hungup 'Zap/6-1'
In this case i hear the audio from the sip on the other end, and all works as expected (i hung up here)
I've expirenced the same problem with 1.0.0 and 1.0.1
I've tried irc and searched online, no one had any ideas (except to do an answer as i already am), and what i did find online was just one guy who also had found no solution, so i'm pretty sure its a bug. Someone throw me a bone here.
I did make a post on the mailing list about it but never got any feed back as of yet.
|Comments:||By: Mark Spencer (markster) 2004-10-08 01:46:15|
This is a technical support issue and should be handled through Digium technical support. Please read the bug guidelines before placing further bug reports.
By: Donny Kavanagh (donnyk) 2004-10-08 02:43:30
Reminder sent to markster
Sorry to be a bother, just a quick question about bug 2604.
Why would this be a digium issue, if the card is clearly able to put audio onto the line as long as the call is a bridged call? Its only if i attempt to make asterisk handle the call, rather then forward it to another number through the pri, or a sip client that it does not work.
Intreastingly enough, if i answer a call and call MusicOnHold, it does not work from the pri. However if i call out to an external number (through the pri) or call in to a sip phone (using a did), and put the call on hold, then music on hold works and asterisk is able to put audio on the pri line.
If i'm totally missing something here, please excuse my ignorance, i just checked out digium's website and its not apparent if my issue would fall under the free assistance or not. Thanks again for your time.