system
September 14, 2012, 2:07pm
1
I’m experiencing a rather wired problem.
I’ve got phone A and B.
When either initialize a call to the other, all things goes as planed, and voice is clear, no problem there.
But when either call a queue, they get the MOH, but when the other “pick up” the call, no sound is hearable, and csip on andoid indicates that there are no packagdes sent.
I’m running asteriskNow 2,0,0
The log for a queue call:
[Sep 14 15:37:30] – Executing [7001@outgoing:1] Verbose(“SIP/0004133079F3-00000088”, “2,“haandholdt” <0004133079F3> entering the support queue”) in new stack
[Sep 14 15:37:30] == “haandholdt” <0004133079F3> entering the support queue
[Sep 14 15:37:30] – Executing [7001@outgoing:2] Answer(“SIP/0004133079F3-00000088”, “”) in new stack
[Sep 14 15:37:30] – Executing [7001@outgoing:3] Playback(“SIP/0004133079F3-00000088”, “hello-world”) in new stack
[Sep 14 15:37:30] – <SIP/0004133079F3-00000088> Playing ‘hello-world.ulaw’ (language ‘da’)
[Sep 14 15:37:32] – Executing [7001@outgoing:4] Queue(“SIP/0004133079F3-00000088”, “support,600”) in new stack
[Sep 14 15:37:32] – Started music on hold, class ‘default’, on SIP/0004133079F3-00000088
[Sep 14 15:37:32] == Using SIP RTP CoS mark 5
[Sep 14 15:37:32] – SIP/40878701mobil-00000089 is ringing
[Sep 14 15:37:36] – SIP/40878701mobil-00000089 answered SIP/0004133079F3-00000088
[Sep 14 15:37:36] – Stopped music on hold on SIP/0004133079F3-00000088
[Sep 14 15:37:36] – Remotely bridging SIP/0004133079F3-00000088 and SIP/40878701mobil-00000089
[Sep 14 15:38:29] == Spawn extension (outgoing, 7001, 4) exited non-zero on ‘SIP/0004133079F3-00000088’
log for a “normal” direct call
== Using SIP RTP CoS mark 5
[Sep 14 17:03:36] – Executing [708@outgoing:1] Gosub(“SIP/0004133079F3-00000099”, “subDialAndVoicemail,start,1(SIP/4437e64cad4e&SIP/sofusIpad&SIP/40878701mobil,708)”) in new stack
[Sep 14 17:03:36] – Executing [start@subDialAndVoicemail:1] Dial(“SIP/0004133079F3-00000099”, “SIP/4437e64cad4e&SIP/sofusIpad&SIP/40878701mobil,10,TtWwXx”) in new stack
[Sep 14 17:03:36] WARNING[5891]: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
[Sep 14 17:03:36] WARNING[5891]: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
[Sep 14 17:03:36] == Using SIP RTP CoS mark 5
[Sep 14 17:03:36] – Called SIP/40878701mobil
[Sep 14 17:03:36] – SIP/40878701mobil-0000009a is ringing
[Sep 14 17:03:36] – SIP/40878701mobil-0000009a is ringing
[Sep 14 17:03:40] – SIP/40878701mobil-0000009a answered SIP/0004133079F3-00000099
[Sep 14 17:03:46] == Spawn extension (subDialAndVoicemail, start, 1) exited non-zero on ‘SIP/0004133079F3-00000099’
Hope someone is able to help
david55
September 14, 2012, 2:14pm
2
Set directmedia=no.
Some option for your normal call is inhibiting direct media, and something about the phone or the network is incompatible with it.
system
September 19, 2012, 10:12am
3
[quote=“david55”]Set directmedia=no.
Some option for your normal call is inhibiting direct media, and something about the phone or the network is incompatible with it.[/quote]
Thanks a lot!
It works very well now:)