---------- Forwarded message ---------- From: Salvatore Frandina <[email protected]> Date: 2010/1/21 Subject: App_Conference: SIPp SDP and DTMF mode To: Neil Stratford <[email protected]>, Mihai Balea <[email protected]>, [email protected]
Hi, I'm using SIPp application http://sipp.sourceforge.net/ to generate a SIP call to open source PBX Asterisk. Command to call the extension (extension) where the IP is IP address of Asterisk [code]sipp -m 1 -d 36000000 -s extension -sf uac_modified.xml IP [/code] In the configuration file uac_modified.xml there are the following lines [code] INVITE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[servi...@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:s...@[local_ip]:[local_port] Max-Forwards: 70 Subject: Dummy User User-Agent: sipp Content-Type: application/sdp Content-Length: [len] v=0 o=sipp 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[local_ip_type] [local_ip] t=0 0 m=audio [auto_media_port] RTP/AVP 0 97 8 18 3 101 a=fmtp:18 annexb=yes a=fmtp:101 0-11,16 a=rtpmap:0 PCMU/8000 a=rtpmap:97 SPEEX/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv m=video [media_port] RTP/AVP 115 a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520 a=rtpmap:115 H263-1998/90000 a=sendrecv [/code] The call work well between SIPp and softphone (Eyebeam, X-lite), i can see all the messages in the Asterisk CLI. If i try to use App_conference application the SIPp user work only without video support. Scenarios: there is a conference (DTMF mode enabled) where there are two or more users when i press a digit to see a generic user, the SIPp user returns the following error [code] sipp: The following events occured: 2010-01-21 16:07:10:392 1264086430.392045: Aborting call on unexpected message for Call-Id '[email protected]': while pausing (index 5), received 'INFO sip:[email protected]:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK11e48a8e;rport Max-Forwards: 70 From: sut <sip:[email protected]:5060>;tag=as47ad7951 To: sipp <sip:[email protected]:5061>;tag=1 Contact: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] CSeq: 102 INFO User-Agent: Asterisk PBX 1.6.2.0 Content-Type: application/media_control+xml Content-Length: 205 <?xml version="1.0" encoding="utf-8" ?> <media_control> <vc_primitive> <to_encoder> <picture_fast_update> </picture_fast_update> </to_encoder> </vc_primitive> </media_control> '. [/code] If i disable the video support without the following lines [code] m=video [media_port] RTP/AVP 115 a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520 a=rtpmap:115 H263-1998/90000 a=sendrecv [/code] the DTMF mode works well and there is no error. The problem it's difficult can you help me? Thank you very much -- _______________________________________ Salvatore Frandina website: http://frandinas.altervista.org mail: [email protected] _______________________________________ -- _______________________________________ Salvatore Frandina website: http://frandinas.altervista.org mail: [email protected] _______________________________________
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