Hi guys,<br><br>I have a system which is disconnecting between 3 and 6 calls per day with a maximum retries exceeded on transmission error.<br><br>I have turned on SIP debug and found that it retries 6 times to send a 200 OK to the remote host and then terminates the call with a BYE. (See below for log excerpts.) The person ringing in just keeps on ringing until a timeout at the host is reached and the maximum retries are exceeded once the phone call has been answered.<br>
<br>Seeing as it is intermittent and fairly similar to this post <a href="http://www.mail-archive.com/asterisk-users@lists.digium.com/msg178034.html">http://www.mail-archive.com/asterisk-users@lists.digium.com/msg178034.html</a> I was wondering if I have the same issue and wanted to chang the setting in chan_sip.c. Would someone be able to confirm that the line I should adjust (so the call is no longer terminated if the other side of the conversation doesn't respect the 200 OK) is<br>
<br>res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);<br><br>and what values in the line I should change.<br><br>Whilst I believe all the NAT settings are correct, I am also going to change the network layout so the server has a direct pulic IP address on one interface and an internal IP address on the second interface so we can avoid any issues related to NAT.<br>
<br>Thanks for your assistance.<br><br><br>The log (trimmed but I think there might be a second call in here as well):<br><br>[Jun 23 15:45:22] VERBOSE[2687] logger.c:<br><--- SIP read from xxx.x.xx.xxx:5060 ---><br>
INVITE sip:0xxxxxxxxx@xxx.x.xx.xxx:5060 SIP/2.0<br>Via: SIP/2.0/UDP xxx.x.xx.xxx:5060;branch=z9hG4bK788e32928c747f8eac98e4e3d60214a4<br>Max-Forwards: 69<br>From: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx>;tag=1c1359910123<br>
To: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx;user=phone><br>Call-ID: 1359909400236200854521<br>CSeq: 1 INVITE<br>Supported: em,timer,replaces,path<br>Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE<br>
User-Agent: VIC-VOGW-01/v.4.80A.012.010<br>Content-Type: application/sdp<br>Content-Length: 304<br>Date: Mon, 23 Jun 2008 05:45:22 GMT<br>Contact: <sip:0xxxxxxxxx@xxx.x.xx.xxx:5060;transport=udp><br><br>v=0<br>o=AudiocodesGW 1359890123 1359889824 IN IP4 xxx.x.xx.xxx<br>
s=Phone-Call<br>c=IN IP4 xxx.x.xx.xxx<br>t=0 0<br>m=audio 14988 RTP/AVP 18 8 101<br>c=IN IP4 xxx.x.xx.xxx<br>a=rtpmap:18 g729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:8 pcma/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>
a=ptime:20<br>a=sendrecv<br><br><-------------><br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: --- (14 headers 14 lines) ---<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Sending to xxx.x.xx.xxx : 5060 (NAT)<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Using INVITE request as basis request - 1359909400236200854521<br>
[Jun 23 15:45:22] VERBOSE[2687] logger.c: Found peer 'XxxXxxxx-2'<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Found RTP audio format 18<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Found RTP audio format 8<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Found RTP audio format 101<br>
[Jun 23 15:45:22] VERBOSE[2687] logger.c: Peer audio RTP is at port xxx.x.xx.xxx:14988<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Found audio description format g729 for ID 18<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Found audio description format pcma for ID 8<br>
[Jun 23 15:45:22] VERBOSE[2687] logger.c: Found audio description format telephone-event for ID 101<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)<br>
[Jun 23 15:45:22] VERBOSE[2687] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: Peer audio RTP is at port xxx.x.xx.xxx:14988<br>
[Jun 23 15:45:22] VERBOSE[2687] logger.c: Looking for 0xxxxxxxxx in custom-get-did-from-sip (domain xxx.x.xx.xxx)<br>[Jun 23 15:45:22] VERBOSE[2687] logger.c: list_route: hop: <sip:0xxxxxxxxx@xxx.x.xx.xxx:5060;transport=udp><br>
[Jun 23 15:45:22] VERBOSE[2687] logger.c:<br><--- Transmitting (NAT) to xxx.x.xx.xxx:5060 ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP xxx.x.xx.xxx:5060;branch=z9hG4bK788e32928c747f8eac98e4e3d60214a4;received=xxx.x.xx.xxx<br>
From: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx>;tag=1c1359910123<br>To: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx;user=phone><br>Call-ID: 1359909400236200854521<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Contact: <sip:0xxxxxxxxx@xxx.x.xx.xxx><br>Content-Length: 0<br><br>[Jun 23 15:45:22] DEBUG[4737] chan_sip.c: Call to peer '201' is 2 out of 50<br>[Jun 23 15:45:22] VERBOSE[4737] logger.c: Audio is at <a href="http://10.0.0.120">10.0.0.120</a> port 14000<br>
[Jun 23 15:45:22] VERBOSE[4737] logger.c: Adding codec 0x8 (alaw) to SDP<br>[Jun 23 15:45:22] VERBOSE[4737] logger.c: Adding codec 0x4 (ulaw) to SDP<br>[Jun 23 15:45:22] VERBOSE[4737] logger.c: Adding non-codec 0x1 (telephone-event) to SDP<br>
[Jun 23 15:45:22] VERBOSE[4737] logger.c: Reliably Transmitting (no NAT) to <a href="http://10.0.0.220:5060">10.0.0.220:5060</a>:<br>INVITE <a href="http://sip:201@10.0.0.220:5060">sip:201@10.0.0.220:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 10.0.0.120:5060;branch=z9hG4bK7bf65928;rport<br>
From: "0xxxxxxxxx" <<a href="mailto:sip%3A0xxxxxxxxx@10.0.0.120">sip:0xxxxxxxxx@10.0.0.120</a>>;tag=as6886acbe<br>To: <<a href="http://sip:201@10.0.0.220:5060">sip:201@10.0.0.220:5060</a>><br>Contact: <<a href="mailto:sip%3A0xxxxxxxxx@10.0.0.120">sip:0xxxxxxxxx@10.0.0.120</a>><br>
Call-ID: <a href="mailto:37a77d2663c1934c3a04957a59b4878e@10.0.0.120">37a77d2663c1934c3a04957a59b4878e@10.0.0.120</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Mon, 23 Jun 2008 05:45:22 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Content-Type: application/sdp<br>Content-Length: 258<br><br>v=0<br>o=root 2596 2596 IN IP4 <a href="http://10.0.0.120">10.0.0.120</a><br>
s=session<br>c=IN IP4 <a href="http://10.0.0.120">10.0.0.120</a><br>t=0 0<br>m=audio 14000 RTP/AVP 8 0 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>
a=ptime:20<br>a=sendrecv<br><br><br>[Jun 23 15:45:42] VERBOSE[2687] logger.c:<br><--- SIP read from <a href="http://10.0.0.220:5060">10.0.0.220:5060</a> ---><br>SIP/2.0 200 OK<br>To: <<a href="http://sip:201@10.0.0.220:5060">sip:201@10.0.0.220:5060</a>>;tag=605a42dafdadd833i0<br>
From: "0xxxxxxxxx" <<a href="mailto:sip%3A0xxxxxxxxx@10.0.0.120">sip:0xxxxxxxxx@10.0.0.120</a>>;tag=as6886acbe<br>Call-ID: <a href="mailto:37a77d2663c1934c3a04957a59b4878e@10.0.0.120">37a77d2663c1934c3a04957a59b4878e@10.0.0.120</a><br>
CSeq: 103 BYE<br>Via: SIP/2.0/UDP 10.0.0.120:5060;branch=z9hG4bK57799f65<br>Server: Linksys/SPA942-5.2.8<br>Content-Length: 0<br><br><br><-------------><br>[Jun 23 15:45:42] VERBOSE[2687] logger.c: --- (8 headers 0 lines) ---<br>
[Jun 23 15:45:42] VERBOSE[2687] logger.c: Really destroying SIP dialog '<a href="mailto:37a77d2663c1934c3a04957a59b4878e@10.0.0.120">37a77d2663c1934c3a04957a59b4878e@10.0.0.120</a>' Method: INVITE<br>[Jun 23 15:45:42] VERBOSE[2687] logger.c: Retransmitting #1 (NAT) to xxx.x.xx.xxx:5060:<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP xxx.x.xx.xxx:5060;branch=z9hG4bK788e32928c747f8eac98e4e3d60214a4;received=xxx.x.xx.xxx<br>From: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx>;tag=1c1359910123<br>To: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx;user=phone>;tag=as3c4944c2<br>
Call-ID: 1359909400236200854521<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:0xxxxxxxxx@xxx.x.xx.xxx><br>
Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 2596 2596 IN IP4 xxx.x.xx.xxx<br>s=session<br>c=IN IP4 xxx.x.xx.xxx<br>t=0 0<br>m=audio 12734 RTP/AVP 8 18 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br>[Jun 23 15:45:42] VERBOSE[2687] logger.c: Retransmitting #2 (NAT) to xxx.x.xx.xxx:5060:<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP xxx.x.xx.xxx:5060;branch=z9hG4bK788e32928c747f8eac98e4e3d60214a4;received=xxx.x.xx.xxx<br>From: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx>;tag=1c1359910123<br>To: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx;user=phone>;tag=as3c4944c2<br>
Call-ID: 1359909400236200854521<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:0xxxxxxxxx@xxx.x.xx.xxx><br>
Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 2596 2596 IN IP4 xxx.x.xx.xxx<br>s=session<br>c=IN IP4 xxx.x.xx.xxx<br>t=0 0<br>m=audio 12734 RTP/AVP 8 18 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br>[Jun 23 15:45:42] VERBOSE[2687] logger.c: Retransmitting #3 (NAT) to xxx.x.xx.xxx:5060:<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP xxx.x.xx.xxx:5060;branch=z9hG4bK788e32928c747f8eac98e4e3d60214a4;received=xxx.x.xx.xxx<br>From: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx>;tag=1c1359910123<br>To: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx;user=phone>;tag=as3c4944c2<br>
Call-ID: 1359909400236200854521<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:0xxxxxxxxx@xxx.x.xx.xxx><br>
Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 2596 2596 IN IP4 xxx.x.xx.xxx<br>s=session<br>c=IN IP4 xxx.x.xx.xxx<br>t=0 0<br>m=audio 12734 RTP/AVP 8 18 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br>[Jun 23 15:45:43] VERBOSE[2687] logger.c: Retransmitting #4 (NAT) to xxx.x.xx.xxx:5060:<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP xxx.x.xx.xxx:5060;branch=z9hG4bK788e32928c747f8eac98e4e3d60214a4;received=xxx.x.xx.xxx<br>From: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx>;tag=1c1359910123<br>To: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx;user=phone>;tag=as3c4944c2<br>
Call-ID: 1359909400236200854521<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:0xxxxxxxxx@xxx.x.xx.xxx><br>
Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 2596 2596 IN IP4 xxx.x.xx.xxx<br>s=session<br>c=IN IP4 xxx.x.xx.xxx<br>t=0 0<br>m=audio 12734 RTP/AVP 8 18 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br>[Jun 23 15:45:45] VERBOSE[2687] logger.c: Retransmitting #5 (NAT) to xxx.x.xx.xxx:5060:<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP xxx.x.xx.xxx:5060;branch=z9hG4bK788e32928c747f8eac98e4e3d60214a4;received=xxx.x.xx.xxx<br>From: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx>;tag=1c1359910123<br>To: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx;user=phone>;tag=as3c4944c2<br>
Call-ID: 1359909400236200854521<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:0xxxxxxxxx@xxx.x.xx.xxx><br>
Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 2596 2596 IN IP4 xxx.x.xx.xxx<br>s=session<br>c=IN IP4 xxx.x.xx.xxx<br>t=0 0<br>m=audio 12734 RTP/AVP 8 18 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br>[Jun 23 15:45:45] VERBOSE[2687] logger.c:<br><--- SIP read from <a href="http://10.0.0.220:5060">10.0.0.220:5060</a> ---><br>
BYE <a href="mailto:sip%3A0xxxxxxxxx@10.0.0.120">sip:0xxxxxxxxx@10.0.0.120</a> SIP/2.0<br>Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bK-6da3d2e2<br>From: <<a href="mailto:sip%3A201@10.0.0.220">sip:201@10.0.0.220</a>>;tag=7ae697d099518d81i0<br>
To: "0xxxxxxxxx" <<a href="mailto:sip%3A0xxxxxxxxx@10.0.0.120">sip:0xxxxxxxxx@10.0.0.120</a>>;tag=as2b6c1d69<br>Call-ID: <a href="mailto:2975319b72894a2c4151361567c5b360@10.0.0.120">2975319b72894a2c4151361567c5b360@10.0.0.120</a><br>
CSeq: 105 BYE<br>Max-Forwards: 70<br>User-Agent: Linksys/SPA942-5.2.8<br>Content-Length: 0<br><br><br><-------------><br>[Jun 23 15:45:45] VERBOSE[2687] logger.c: --- (9 headers 0 lines) ---<br>[Jun 23 15:45:45] VERBOSE[2687] logger.c: Sending to <a href="http://10.0.0.220">10.0.0.220</a> : 5060 (no NAT)<br>
[Jun 23 15:45:45] VERBOSE[2687] logger.c: Scheduling destruction of SIP dialog '<a href="mailto:2975319b72894a2c4151361567c5b360@10.0.0.120">2975319b72894a2c4151361567c5b360@10.0.0.120</a>' in 6400 ms (Method: BYE)<br>
[Jun 23 15:45:45] VERBOSE[2687] logger.c:<br><--- Transmitting (no NAT) to <a href="http://10.0.0.220:5060">10.0.0.220:5060</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 10.0.0.220:5060;branch=z9hG4bK-6da3d2e2;received=<a href="http://10.0.0.220">10.0.0.220</a><br>
From: <<a href="mailto:sip%3A201@10.0.0.220">sip:201@10.0.0.220</a>>;tag=7ae697d099518d81i0<br>To: "0xxxxxxxxx" <<a href="mailto:sip%3A0xxxxxxxxx@10.0.0.120">sip:0xxxxxxxxx@10.0.0.120</a>>;tag=as2b6c1d69<br>
Call-ID: <a href="mailto:2975319b72894a2c4151361567c5b360@10.0.0.120">2975319b72894a2c4151361567c5b360@10.0.0.120</a><br>CSeq: 105 BYE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Contact: <<a href="mailto:sip%3A0xxxxxxxxx@10.0.0.120">sip:0xxxxxxxxx@10.0.0.120</a>><br>Content-Length: 0<br><br><br><------------><br>[Jun 23 15:45:48] VERBOSE[2687] logger.c: Retransmitting #6 (NAT) to xxx.x.xx.xxx:5060:<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP xxx.x.xx.xxx:5060;branch=z9hG4bK788e32928c747f8eac98e4e3d60214a4;received=xxx.x.xx.xxx<br>From: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx>;tag=1c1359910123<br>To: <sip:0xxxxxxxxx@xxx.xxxxxxxx.xxx.xx;user=phone>;tag=as3c4944c2<br>
Call-ID: 1359909400236200854521<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <sip:0xxxxxxxxx@xxx.x.xx.xxx><br>
Content-Type: application/sdp<br>Content-Length: 285<br><br>v=0<br>o=root 2596 2596 IN IP4 xxx.x.xx.xxx<br>s=session<br>c=IN IP4 xxx.x.xx.xxx<br>t=0 0<br>m=audio 12734 RTP/AVP 8 18 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br><br>[Jun 23 15:45:51] VERBOSE[2687] logger.c: SIP Response message for INCOMING dialog BYE arrived<br>
[Jun 23 15:45:51] VERBOSE[2687] logger.c: Really destroying SIP dialog '<a href="mailto:2975319b72894a2c4151361567c5b360@10.0.0.120">2975319b72894a2c4151361567c5b360@10.0.0.120</a>' Method: BYE<br>[Jun 23 15:45:52] WARNING[2687] chan_sip.c: Maximum retries exceeded on transmission 1359909400236200854521 for seqno 1 (Critical Response)<br>
[Jun 23 15:45:52] VERBOSE[2687] logger.c: Really destroying SIP dialog '1359909400236200854521' Method: BYE<br><br>