I have a working ooh323 channel through an * system between Avaya and Siemens except for one tiny detail. The Siemens system doesn't answer the call before a timeout on the * side if the call is directed to voice mail. I have tried changing the rtptimeout= value in the ooh323.conf file and I have tried exten=>_1XX,1,Dial(ooh323/${EXTEN}@farend|120) both to no avail. Anyone think of somewhere else a connection timer might be messing me up in this? Thanks.
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RTP timeout ooh323
- Thread starter vaoldschl
- Start date
So, what you are saying is that if Asterisk gets a call from outside or from Avaya, and tries to send the call to an extension on the Siemens, the call times out before it goes to the Siemens voicemail. Where does the call end up? AsteriskMail?
- Thread starter
- #3
I've stripped the dialplan in * to nothing except my extension assignments and it appears to be waiting about 30 seconds and then clearing the channel. The call just disconnects. If I turn on autofallthrough and watch a debug it runs across "timeout reached but no value 't'" or somesuch verbage just before destroying the channel. Also, there are no outside calls; the only place * can get a call from right now is Avaya to Siemens or vice versa. Thanks.
Autofallthrough in extensions.conf is set to yes by default. Asterisk 1.0 (and earlier) behavior was to wait for an extension to be dialed after there were no more extensions to execute. "autofallthrough" changes this behavior so that the call will immediately be terminated with BUSY, CONGESTION, or HANGUP based on Asterisk's best guess. If you are writing an extension for IVR, you must use the WaitExten application if "autofallthrough" is set to yes. Not sure how it effects on regular extension, but would set to no. The "t" asterisk is looking for is basically how to terminate the call.
- Thread starter
- #6
I have autofallthrough=no and I've tried =yes as well. Both ways disconnect call after 30 secs. WaitExten is used to wait for user input on an IVR so that wouldn't work to extend the ringing time would it? If it would then would it look like this?
exten => s,1,Answer
exten => s,n,WaitExten(secs)
Also, if I do a show dialplan I get my entries and then a bunch of stuff coming from pbx_ael that I don't need. Where is that coming from?
exten => s,1,Answer
exten => s,n,WaitExten(secs)
Also, if I do a show dialplan I get my entries and then a bunch of stuff coming from pbx_ael that I don't need. Where is that coming from?
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