vulcan wrote:
I don't know if dial options are causing it but you can try with just the basic:
|60|L(%timeout%:60000:30000)
Secondly, there is no "I" option and "g" is meaningless because when the AGI ends, A2B hangs up the channel. Why do you need the "o" option?
I tested the progressinband=yes and I can see asterisk sending and receiving the 183 message. SIP 183 results in more load on asterisk, that is why it is not enabled by default.
Dear Vulcan,
Sorry for delay answer, as apparently I have recorded my answer here, but I forgot to send it, so, just now I sow that...
About the Dialcommand:
Quote:
i: Asterisk will ignore any forwarding requests it may receive on this dial attempt. (new in 1.4) Useful if you are ringing a group of people and one person has set their phone to forwarded direct to voicemail on their cell or something which normally prevents any of the other phones from ringing.
So, here, i need to avoid the call forwarding, even, I have set callforward directive in sip, to no, but, I had some security inconvenient years ago, where user place a call, then forward it to other destination, and place another at the same time, and that let the account in serous negative balance. So, to solve that, I have enabled in file and in realtime the callforward=no, and the i directive in the dialcommand to avoid that.
Then:
Quote:
g: When the called party hangs up, continue to execute commands in the current context at the next priority.
Here, i'm obligating asterisk to go to the next priority in the dial context, for the wholesale service, as the a2b agi, didn't handle me the congestion messages, and ny others... so, wholesale are requiring this to understand what's gone, so my dialplan for wholesale is:
Quote:
[wholesale]
exten => _X.,1,Progress()
exten => _X.,n,Wait(1)
exten => _X.,n,AGI(a2billing.php,7)
exten => _X.,n,Congestion()
exten => _X.,n,Hangup()
Because if not, asterisk will not go to the next priority, and congestion message will never be delivered... I'm not aware if new versions of the agi are having this, even, now I'm running v. 2.0.7, but I have set this since v. 1.8, so, in that version, it wasn't included... and I haven't reviewed this...
Then, about the issue itself...
Quote:
I tested the progressinband=yes and I can see asterisk sending and receiving the 183 message. SIP 183 results in more load on asterisk, that is why it is not enabled by default.
It's true, now, I have asked my customer to re-test it, and they are getting properly the 183, but when I test it myself, I'm not getting this, or can't reproduce it. I guess it's due to my client, I was testing with csipsimple for android, and media5fone, both are not having this, so, I guess it's a client side missing, but, I can see that the issue it's solved, as my end client are getting that now... waiting for there final approval.
The issue, now, is the progressinband message are heavy in asterisk production environment, so, the question is how to set this only for concrete client who are requiring this, and let it off for the rest to avoid system overload. For now, I have set it as no by default, and just enable it for that concrete peer... will see if that will work...
Regards,