On 06.06.2013, at 15:05, Jonas Kellens <jonas.kell...@telenet.be> wrote:
> Hello, > > when picking up an incoming call from one ip phone on another ip phone, the > call terminates after about 5 to 10 seconds. > > When reading out the hangup cause variable in the h-extention of the > dialplan, the hangup cause seems to be 111. > > > In the dialplan output, you can see that SIP-peer sipacc3 picks up the > incoming channel SipAgenT01-00001454, and the call is answered. After 7 > seconds, the conversation is terminated. > > [Jun 6 10:13:15] VERBOSE[21118] pbx.c: [Jun 6 10:13:15] -- Executing > [120@sub-pickup:25] Pickup("SIP/sipacc3-0000147c", > "SIP/SipAgenT01-00001454@PICKUPMARK") in new stack > [Jun 6 10:13:15] VERBOSE[20788] app_queue.c: [Jun 6 10:13:15] -- > SIP/sipacc3-0000147c answered SIP/SipAgenT01-00001454 > > [Jun 6 10:13:22] VERBOSE[20788] pbx.c: [Jun 6 10:13:22] -- Executing > [h@pbx-routing:3] NoOp("SIP/SipAgenT01-00001454", "hangup cause = 111") in > new stack > > > > Questions : > > 1. what can cause a hangup cause 111 ? What is the meaning of hangup cause > 111 ? > > 2. on voip-info.org I read "111 protocol error 500 Server internal error". > How can I fix this ?? Using Asterisk 1.8.12.2 on CentOS. Hi Jonas, when the calls is answered, do you have correct both-way audio as well? Please enter "sip set debug on" on the Asterisk console and paste the output. It could also be helpful if you could paste your dialplan. -- marie -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users