<br><br><div class="gmail_quote">---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Salvatore Frandina</b> <span dir="ltr"><<a href="mailto:salvatore.frandina@gmail.com">salvatore.frandina@gmail.com</a>></span><br>
Date: 2010/1/21<br>Subject: App_Conference: SIPp SDP and DTMF mode<br>To: Neil Stratford <<a href="mailto:neils@vipadia.com">neils@vipadia.com</a>>, Mihai Balea <<a href="mailto:mihai@hates.ms">mihai@hates.ms</a>>, <a href="mailto:kapejod@ns1.jnetdns.de">kapejod@ns1.jnetdns.de</a><br>
<br><br><br clear="all">Hi,<br><br><div>I'm using SIPp application <a href="http://sipp.sourceforge.net/" target="_blank">http://sipp.sourceforge.net/</a> to generate a SIP call to open source PBX Asterisk. <br></div>
<div><br></div>Command to call the extension (extension) where the IP is IP address of Asterisk<br>
<div>[code]sipp -m 1 -d 36000000 -s extension -sf uac_modified.xml IP [/code]</div><div><br></div><div>In the configuration file uac_modified.xml there are the following lines<br></div><br><div>[code]</div><div>INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br>
</div> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br> From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]<br> To: sut <sip:[service]@[remote_ip]:[remote_port]><br>
Call-ID: [call_id]<br> CSeq: 1 INVITE<br> Contact: sip:sipp@[local_ip]:[local_port]<br> Max-Forwards: 70<br> Subject: Dummy User<br> User-Agent: sipp<br> Content-Type: application/sdp<br>
Content-Length: [len]<br><br> v=0<br> o=sipp 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br> s=-<br> c=IN IP[local_ip_type] [local_ip]<br> t=0 0<br> m=audio [auto_media_port] RTP/AVP 0 97 8 18 3 101 <br>
a=fmtp:18 annexb=yes<br> a=fmtp:101 0-11,16 <br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:97 SPEEX/8000<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:18 G729/8000<br> a=rtpmap:101 telephone-event/8000 <br>
a=sendrecv <br> m=video [media_port] RTP/AVP 115<br> a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520<br> a=rtpmap:115 H263-1998/90000<br> a=sendrecv <br> [/code]<br> <br>The call work well between SIPp and softphone (Eyebeam, X-lite), i can see all the messages in the Asterisk CLI. <br>
If i try to use App_conference application the SIPp user work only without video support. <br>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<br>
<br><div>[code]</div><div>sipp: The following events occured:<br></div>2010-01-21 16:07:10:392 1264086430.392045: Aborting call on unexpected message for Call-Id '<a href="mailto:1-3005@127.0.1.1" target="_blank">1-3005@127.0.1.1</a>': while pausing (index 5), received 'INFO <a href="http://sip:sipp@127.0.1.1:5061" target="_blank">sip:sipp@127.0.1.1:5061</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK11e48a8e;rport<br>Max-Forwards: 70<br>From: sut <<a href="http://sip:9999@192.168.0.4:5060" target="_blank">sip:9999@192.168.0.4:5060</a>>;tag=as47ad7951<br>To: sipp <<a href="http://sip:sipp@127.0.1.1:5061" target="_blank">sip:sipp@127.0.1.1:5061</a>>;tag=1<br>
Contact: <<a href="mailto:sip%3A9999@192.168.0.4" target="_blank">sip:9999@192.168.0.4</a>><br>Call-ID: <a href="mailto:1-3005@127.0.1.1" target="_blank">1-3005@127.0.1.1</a><br>CSeq: 102 INFO<br>User-Agent: Asterisk PBX 1.6.2.0<br>
Content-Type: application/media_control+xml<br>
Content-Length: 205<br><br><?xml version="1.0" encoding="utf-8" ?><br><media_control><br><vc_primitive><br><to_encoder><br><picture_fast_update><br></picture_fast_update><br>
</to_encoder><br></vc_primitive><br></media_control><br><div>'.</div><div>[/code]<br></div><br>If i disable the video support without the following lines<br>[code] m=video [media_port] RTP/AVP 115<br>
a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520<br> a=rtpmap:115 H263-1998/90000<br> a=sendrecv [/code]<br> the DTMF mode works well and there is no error.<br><br><div>The problem it's difficult can you help me?<br>
</div><div><br></div><div>Thank you very much</div><br><font color="#888888">-- <br>_______________________________________<br>Salvatore Frandina<br>website: <a href="http://frandinas.altervista.org" target="_blank">http://frandinas.altervista.org</a><br>
mail: <a href="mailto:salvatore.frandina@gmail.com" target="_blank">salvatore.frandina@gmail.com</a><br>
<br>_______________________________________<br><br>
</font></div><br><br clear="all"><br>-- <br>_______________________________________<br>Salvatore Frandina<br>website: <a href="http://frandinas.altervista.org">http://frandinas.altervista.org</a><br>mail: <a href="mailto:salvatore.frandina@gmail.com">salvatore.frandina@gmail.com</a><br>
<br>_______________________________________<br><br>