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<TITLE>RE: [asterisk-users] Asterisk not sending RTP</TITLE>
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<P><FONT SIZE=2>well, here is the full SIP debug:</FONT>
</P>
<P><FONT SIZE=2>Sep 3 10:05:59 DEBUG[6139] manager.c: Manager received command 'Originate'</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 DEBUG[6139] chan_sip.c: Setting NAT on RTP to 0</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 DEBUG[6139] chan_sip.c: Outgoing Call for 972544482826</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6139] logger.c: We're at 192.117.233.176 port 18372</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6139] logger.c: Adding codec 0x4 (ulaw) to SDP</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6139] logger.c: Adding codec 0x8 (alaw) to SDP</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6139] logger.c: 13 headers, 9 lines</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6139] logger.c: Reliably Transmitting (no NAT) to 62.219.61.73:5060:</FONT>
<BR><FONT SIZE=2>INVITE sip:972544482826@62.219.61.73 SIP/2.0</FONT>
<BR><FONT SIZE=2>Via: SIP/2.0/UDP 192.117.233.176:5060;branch=z9hG4bK33e91b50;rport</FONT>
<BR><FONT SIZE=2>From: "972544482826" <sip:972544482826@192.117.233.176>;tag=as5514479f</FONT>
<BR><FONT SIZE=2>To: <sip:972544482826@62.219.61.73></FONT>
<BR><FONT SIZE=2>Contact: <sip:972544482826@192.117.233.176></FONT>
<BR><FONT SIZE=2>Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176</FONT>
<BR><FONT SIZE=2>CSeq: 102 INVITE</FONT>
<BR><FONT SIZE=2>User-Agent: Asterisk PBX</FONT>
<BR><FONT SIZE=2>Max-Forwards: 70</FONT>
<BR><FONT SIZE=2>Date: Sun, 03 Sep 2006 09:05:59 GMT</FONT>
<BR><FONT SIZE=2>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</FONT>
<BR><FONT SIZE=2>Content-Type: application/sdp</FONT>
<BR><FONT SIZE=2>Content-Length: 188</FONT>
</P>
<P><FONT SIZE=2>v=0</FONT>
<BR><FONT SIZE=2>o=root 6093 6093 IN IP4 192.117.233.176</FONT>
<BR><FONT SIZE=2>s=session</FONT>
<BR><FONT SIZE=2>c=IN IP4 192.117.233.176</FONT>
<BR><FONT SIZE=2>t=0 0</FONT>
<BR><FONT SIZE=2>m=audio 18372 RTP/AVP 0 8</FONT>
<BR><FONT SIZE=2>a=rtpmap:0 PCMU/8000</FONT>
<BR><FONT SIZE=2>a=rtpmap:8 PCMA/8000</FONT>
<BR><FONT SIZE=2>a=silenceSupp:off - - - -</FONT>
</P>
<P><FONT SIZE=2>---</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6124] logger.c:</FONT>
<BR><FONT SIZE=2><-- SIP read from 62.219.61.73:5060:</FONT>
<BR><FONT SIZE=2>SIP/2.0 100 Trying</FONT>
<BR><FONT SIZE=2>Via: SIP/2.0/UDP 192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport=5060</FONT>
<BR><FONT SIZE=2>From: "972544482826" <sip:972544482826@192.117.233.176>;tag=as5514479f</FONT>
<BR><FONT SIZE=2>To: <sip:972544482826@62.219.61.73> </FONT>
<BR><FONT SIZE=2>Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176</FONT>
<BR><FONT SIZE=2>CSeq: 102 INVITE</FONT>
</P>
<BR>
<P><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (6 headers 0 lines)Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (6 headers 0 li</FONT></P>
<P><FONT SIZE=2>nes)---</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 DEBUG[6124] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a8dc38a2198b8fb0</FONT></P>
<P><FONT SIZE=2>f40b63f7680671e@192.117.233.176' Request 102: Found</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6124] logger.c:</FONT>
<BR><FONT SIZE=2><-- SIP read from 62.219.61.73:5060:</FONT>
<BR><FONT SIZE=2>SIP/2.0 183 Session Progress</FONT>
<BR><FONT SIZE=2>Via: SIP/2.0/UDP 192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport=5060</FONT>
<BR><FONT SIZE=2>From: "972544482826" <sip:972544482826@192.117.233.176>;tag=as5514479f</FONT>
<BR><FONT SIZE=2>To: <sip:972544482826@62.219.61.73>;tag=2607</FONT>
<BR><FONT SIZE=2>Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176</FONT>
<BR><FONT SIZE=2>CSeq: 102 INVITE</FONT>
<BR><FONT SIZE=2>Supported: timer,100rel</FONT>
<BR><FONT SIZE=2>Content-Length: 0</FONT>
</P>
<BR>
<P><FONT SIZE=2>Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (8 headers 0 lines)Sep 3 10:05:59 VERBOSE[6124] logger.c: --- (8 headers 0 li</FONT></P>
<P><FONT SIZE=2>nes)---</FONT>
<BR><FONT SIZE=2>Sep 3 10:05:59 DEBUG[6124] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a8dc38a2198b8fb0</FONT></P>
<P><FONT SIZE=2>f40b63f7680671e@192.117.233.176' Request 102: Found</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:00 VERBOSE[6124] logger.c:</FONT>
<BR><FONT SIZE=2><-- SIP read from 62.219.61.73:5060:</FONT>
<BR><FONT SIZE=2>SIP/2.0 183 Session Progress</FONT>
<BR><FONT SIZE=2>Via: SIP/2.0/UDP 192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport=5060</FONT>
<BR><FONT SIZE=2>From: "972544482826" <sip:972544482826@192.117.233.176>;tag=as5514479f</FONT>
<BR><FONT SIZE=2>To: <sip:972544482826@62.219.61.73>;tag=2607</FONT>
<BR><FONT SIZE=2>Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176</FONT>
<BR><FONT SIZE=2>CSeq: 102 INVITE</FONT>
<BR><FONT SIZE=2>Content-Type: application/sdp</FONT>
<BR><FONT SIZE=2>Supported: timer,100rel</FONT>
<BR><FONT SIZE=2>Content-Length: 123</FONT>
</P>
<P><FONT SIZE=2>v=0</FONT>
<BR><FONT SIZE=2>o=MG4000|2.0 14726 14726 IN IP4 62.219.61.73</FONT>
<BR><FONT SIZE=2>s=-</FONT>
<BR><FONT SIZE=2>c=IN IP4 62.219.61.73</FONT>
<BR><FONT SIZE=2>t=0 0</FONT>
<BR><FONT SIZE=2>m=audio 51644 RTP/AVP 0</FONT>
<BR><FONT SIZE=2>a=ptime:10</FONT>
</P>
<P><FONT SIZE=2>Sep 3 10:06:00 VERBOSE[6124] logger.c: --- (9 headers 7 lines)Sep 3 10:06:00 VERBOSE[6124] logger.c: --- (9 headers 7 li</FONT></P>
<P><FONT SIZE=2>nes)---</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:00 DEBUG[6124] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4a8dc38a2198b8fb0</FONT></P>
<P><FONT SIZE=2>f40b63f7680671e@192.117.233.176' Request 102: Found</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:00 VERBOSE[6124] logger.c: Found RTP audio format 0</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:00 VERBOSE[6124] logger.c: Peer audio RTP is at port 62.219.61.73:51644</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:00 DEBUG[6124] chan_sip.c: Peer audio RTP is at port 62.219.61.73:51644</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:00 VERBOSE[6124] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), c</FONT></P>
<P><FONT SIZE=2>ombined - 0x4 (ulaw)</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:00 VERBOSE[6124] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined</FONT></P>
<P><FONT SIZE=2> - 0x0 (nothing)</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c:</FONT>
<BR><FONT SIZE=2><-- SIP read from 62.219.61.73:5060:</FONT>
<BR><FONT SIZE=2>SIP/2.0 200 OK</FONT>
<BR><FONT SIZE=2>Via: SIP/2.0/UDP 192.117.233.176:5060;received=192.117.233.176;branch=z9hG4bK33e91b50;rport=5060</FONT>
<BR><FONT SIZE=2>From: "972544482826" <sip:972544482826@192.117.233.176>;tag=as5514479f</FONT>
<BR><FONT SIZE=2>To: <sip:972544482826@62.219.61.73>;tag=2607</FONT>
<BR><FONT SIZE=2>Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176</FONT>
<BR><FONT SIZE=2>CSeq: 102 INVITE</FONT>
<BR><FONT SIZE=2>Content-Type: application/sdp</FONT>
<BR><FONT SIZE=2>Contact: <sip:972544482826@62.219.61.73:5060;transport=udp>;user=phone</FONT>
<BR><FONT SIZE=2>Supported: timer,100rel</FONT>
<BR><FONT SIZE=2>Content-Length: 123</FONT>
</P>
<P><FONT SIZE=2>v=0</FONT>
<BR><FONT SIZE=2>o=MG4000|2.0 14726 14726 IN IP4 62.219.61.73</FONT>
<BR><FONT SIZE=2>s=-</FONT>
<BR><FONT SIZE=2>c=IN IP4 62.219.61.73</FONT>
<BR><FONT SIZE=2>t=0 0</FONT>
<BR><FONT SIZE=2>m=audio 51644 RTP/AVP 0</FONT>
<BR><FONT SIZE=2>a=ptime:10</FONT>
</P>
<P><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: --- (10 headers 7 lines)Sep 3 10:06:01 VERBOSE[6124] logger.c: --- (10 headers 7</FONT></P>
<P><FONT SIZE=2>lines)---</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 DEBUG[6124] chan_sip.c: Acked pending invite 102</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 DEBUG[6124] chan_sip.c: Stopping retransmission on '4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176' of R</FONT></P>
<P><FONT SIZE=2>equest 102: Match Found</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: Found RTP audio format 0</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: Peer audio RTP is at port 62.219.61.73:51644</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 DEBUG[6124] chan_sip.c: Peer audio RTP is at port 62.219.61.73:51644</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), c</FONT></P>
<P><FONT SIZE=2>ombined - 0x4 (ulaw)</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined</FONT></P>
<P><FONT SIZE=2> - 0x0 (nothing)</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 DEBUG[6124] chan_sip.c: build_route: Contact hop: <sip:972544482826@62.219.61.73:5060;transport=udp>;user=</FONT></P>
<P><FONT SIZE=2>phone</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: list_route: hop: <sip:972544482826@62.219.61.73:5060;transport=udp></FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: set_destination: Parsing <sip:972544482826@62.219.61.73:5060;transport=udp> for ad</FONT></P>
<P><FONT SIZE=2>dress/port to send to</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: set_destination: set destination to 62.219.61.73, port 5060</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6124] logger.c: Transmitting (no NAT) to 62.219.61.73:5060:</FONT>
<BR><FONT SIZE=2>ACK sip:972544482826@62.219.61.73:5060;transport=udp SIP/2.0</FONT>
<BR><FONT SIZE=2>Via: SIP/2.0/UDP 192.117.233.176:5060;branch=z9hG4bK052932bd;rport</FONT>
<BR><FONT SIZE=2>From: "972544482826" <sip:972544482826@192.117.233.176>;tag=as5514479f</FONT>
<BR><FONT SIZE=2>To: <sip:972544482826@62.219.61.73>;tag=2607</FONT>
<BR><FONT SIZE=2>Contact: <sip:972544482826@192.117.233.176></FONT>
<BR><FONT SIZE=2>Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176</FONT>
<BR><FONT SIZE=2>CSeq: 102 ACK</FONT>
<BR><FONT SIZE=2>User-Agent: Asterisk PBX</FONT>
<BR><FONT SIZE=2>Max-Forwards: 70</FONT>
<BR><FONT SIZE=2>Content-Length: 0</FONT>
</P>
<BR>
<P><FONT SIZE=2>---</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 DEBUG[6097] channel.c: Avoiding initial deadlock for 'SIP/BezeqInt-007c1920'</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6139] logger.c: > Channel SIP/BezeqInt-007c1920 was answered.</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6147] logger.c: -- Executing NoOp("SIP/BezeqInt-007c1920", "in MusicOnHoldTest") in new stack</FONT></P>
<P><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6147] logger.c: -- Executing Answer("SIP/BezeqInt-007c1920", "") in new stack</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6147] logger.c: -- Executing Playback("SIP/BezeqInt-007c1920", "AEN2") in new stack</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 DEBUG[6147] channel.c: Scheduling timer at 160 sample intervals</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:01 VERBOSE[6147] logger.c: -- Playing 'AEN2' (language 'en')</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 VERBOSE[6124] logger.c:</FONT>
<BR><FONT SIZE=2><-- SIP read from 62.219.61.73:5060:</FONT>
<BR><FONT SIZE=2>BYE sip:972544482826@192.117.233.176 SIP/2.0</FONT>
<BR><FONT SIZE=2>Via: SIP/2.0/UDP 62.219.61.73:5060;branch=z9hG4bK0g1mob302o71ekc331o0sd0000g00.1</FONT>
<BR><FONT SIZE=2>Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176</FONT>
<BR><FONT SIZE=2>From: <sip:972544482826@62.219.61.73>;tag=2607</FONT>
<BR><FONT SIZE=2>To: "972544482826" <sip:972544482826@192.117.233.176>;tag=as5514479f</FONT>
<BR><FONT SIZE=2>CSeq: 1 BYE</FONT>
<BR><FONT SIZE=2>Supported: timer,100rel</FONT>
<BR><FONT SIZE=2>Max-Forwards: 69</FONT>
<BR><FONT SIZE=2>Content-Length: 0</FONT>
</P>
<BR>
<P><FONT SIZE=2>Sep 3 10:06:38 VERBOSE[6124] logger.c: --- (9 headers 0 lines)Sep 3 10:06:38 VERBOSE[6124] logger.c: --- (9 headers 0 li</FONT></P>
<P><FONT SIZE=2>nes)---</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 VERBOSE[6124] logger.c: Sending to 62.219.61.73 : 5060 (non-NAT)</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 VERBOSE[6124] logger.c: Transmitting (no NAT) to 62.219.61.73:5060:</FONT>
<BR><FONT SIZE=2>SIP/2.0 200 OK</FONT>
<BR><FONT SIZE=2>Via: SIP/2.0/UDP 62.219.61.73:5060;branch=z9hG4bK0g1mob302o71ekc331o0sd0000g00.1;received=62.219.61.73</FONT>
<BR><FONT SIZE=2>From: <sip:972544482826@62.219.61.73>;tag=2607</FONT>
<BR><FONT SIZE=2>To: "972544482826" <sip:972544482826@192.117.233.176>;tag=as5514479f</FONT>
<BR><FONT SIZE=2>Call-ID: 4a8dc38a2198b8fb0f40b63f7680671e@192.117.233.176</FONT>
<BR><FONT SIZE=2>CSeq: 1 BYE</FONT>
<BR><FONT SIZE=2>User-Agent: Asterisk PBX</FONT>
<BR><FONT SIZE=2>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</FONT>
<BR><FONT SIZE=2>Contact: <sip:972544482826@192.117.233.176></FONT>
<BR><FONT SIZE=2>Content-Length: 0</FONT>
<BR><FONT SIZE=2>X-Asterisk-HangupCause: Normal Clearing</FONT>
</P>
<BR>
<P><FONT SIZE=2>---</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 DEBUG[6147] channel.c: Scheduling timer at 0 sample intervals</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 VERBOSE[6147] logger.c: == Spawn extension (MusicOnHoldTest, 972544482826, 3) exited non-zero on 'SIP/Be</FONT></P>
<P><FONT SIZE=2>zeqInt-007c1920'</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 DEBUG[6147] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 DEBUG[6147] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,d</FONT></P>
<P><FONT SIZE=2>context,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield) VALUES (</FONT></P>
<P><FONT SIZE=2>'2006-09-03 10:06:01','972544482826','972544482826','972544482826','MusicOnHoldTest', 'SIP/BezeqInt-007c1920','','Playback</FONT></P>
<P><FONT SIZE=2>','AEN2',37,37,'ANSWERED',3,'','1157274359.7','')</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 DEBUG[6147] pbx.c: Function result is '972544482826'</FONT>
<BR><FONT SIZE=2>Sep 3 10:06:38 DEBUG[6147] pbx.c: Function result is '972544482826'</FONT>
</P>
<P><FONT SIZE=2>Any idea would be highly appreciated...</FONT>
</P>
<P><FONT SIZE=2>Nir S</FONT>
</P>
<P><FONT SIZE=2>-----Original Message-----</FONT>
<BR><FONT SIZE=2>From: asterisk-users-bounces@lists.digium.com [<A HREF="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</A>] On Behalf Of Jeremy McNamara</FONT></P>
<P><FONT SIZE=2>Sent: Sunday, September 03, 2006 4:09 PM</FONT>
<BR><FONT SIZE=2>To: nirs@atelis.net; Asterisk Users Mailing List - Non-Commercial Discussion</FONT>
<BR><FONT SIZE=2>Subject: Re: [asterisk-users] Asterisk not sending RTP</FONT>
</P>
<P><FONT SIZE=2>Nir Simionovich wrote:</FONT>
<BR><FONT SIZE=2>> Any ideas anyone ?</FONT>
</P>
<BR>
<P><FONT SIZE=2>Do you have a compatible codec?</FONT>
<BR><FONT SIZE=2>What does the SDP show?</FONT>
<BR><FONT SIZE=2>Is sip.conf binding to a valid IP address?</FONT>
</P>
<BR>
<BR>
<P><FONT SIZE=2>Jeremy McNamara</FONT>
<BR><FONT SIZE=2>_______________________________________________</FONT>
<BR><FONT SIZE=2>--Bandwidth and Colocation provided by Easynews.com --</FONT>
</P>
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