<DIV>Date: Thu, 10 May 2007 08:44:08 +0200<BR>From: Olle E Johansson <<A href="http://mail01.mail.com/scripts/mail/compose.mail?compose=1&.ob=ab9262bafcb62299c85f2b171a1423b4943f821d&composeto=olle%40voop.com"><FONT face=Arial>olle@voop.com</FONT></A>><BR>Subject: Re: [asterisk-dev] Response to Subscribe with Expire set to<BR> 0.<BR>To: Asterisk Developers Mailing List <<A href="http://mail01.mail.com/scripts/mail/compose.mail?compose=1&.ob=ab9262bafcb62299c85f2b171a1423b4943f821d&composeto=asterisk-dev%40lists.digium.com"><FONT face=Arial>asterisk-dev@lists.digium.com</FONT></A>><BR>Message-ID: <<A href="http://mail01.mail.com/scripts/mail/compose.mail?compose=1&.ob=ab9262bafcb62299c85f2b171a1423b4943f821d&composeto=DC3025C1-E9F3-4079-BD76-F6B4D70968D5%40voop.com"><FONT face=Arial>DC3025C1-E9F3-4079-BD76-F6B4D70968D5@voop.com</FONT></A>><BR>Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed<BR></DIV>
<DIV> </DIV>
<DIV>>Which version are you using?<BR>>Can you provide me with a SIP debug?<BR></DIV>
<DIV> </DIV>
<DIV>Hi Olle,</DIV>
<DIV> </DIV>
<DIV>I am using version 1.2.17. The show version CLI gives:</DIV>
<DIV>"Asterisk SVN --r127M built by root @ Ray_dev_VM on a i686 running Linux on 2007-05-10 08:35:40 UTC"</DIV>
<DIV> </DIV>
<DIV>The following is the sip debug msg I caught:</DIV>
<DIV><FONT size=2>
<P><-- SIP read from 10.10.150.164:5060: </P>
<P>SUBSCRIBE sip:10.10.150.154@10.10.150.154 SIP/2.0</P>
<P>Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKZgUGdQQPSIVL4XzkREhel0dT8t.C</P>
<P>Max-Forwards: 70</P>
<P>From: <sip:9002@10.10.150.154>;tag=zx0blk2XNgw</P>
<P>To: <sip:10.10.150.154@10.10.150.154></P>
<P>Call-ID: 87FL32ysiq6y7LBxVuoSI37@10.10.150.154</P>
<P>CSeq: 49310 SUBSCRIBE</P>
<P>Contact: <sip:9002@10.10.150.164></P>
<P>Accept: application/simple-message-summary</P>
<P>Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO</P>
<P>Allow-Events: dialog,message-summary</P>
<P>Content-Disposition: session</P>
<P>Event: message-summary</P>
<P>Expires: 3600</P>
<P>Supported: replaces</P>
<P>User-Agent: VoipLan Gateway/07.63 (MAC=00087B030A9F; SER= 02605; HW=12)</P>
<P>Content-Type: text/plain</P>
<P>Content-Length: 0</P>
<P> </P>
<P>--- (18 headers 0 lines) ---</P>
<P>May 13 03:38:29 DEBUG[28558]: chan_sip.c:3193 sip_alloc: Allocating new SIP dialog for 87FL32ysiq6y7LBxVuoSI37@10.10.150.154 - SUBSCRIBE (No RTP)</P>
<P>Using latest SUBSCRIBE request as basis request</P>
<P>Sending to 10.10.150.164 : 5060 (NAT)</P>
<P>Found peer '9002'</P>
<P>Looking for 10.10.150.154 in custom (domain 10.10.150.154)</P>
<P>Transmitting (NAT) to 10.10.150.164:5060:</P>
<P>SIP/2.0 200 OK</P>
<P>Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKZgUGdQQPSIVL4XzkREhel0dT8t.C;received=10.10.150.164</P>
<P>From: <sip:9002@10.10.150.154>;tag=zx0blk2XNgw</P>
<P>To: <sip:10.10.150.154@10.10.150.154>;tag=as6c1cb383</P>
<P>Call-ID: 87FL32ysiq6y7LBxVuoSI37@10.10.150.154</P>
<P>CSeq: 49310 SUBSCRIBE</P>
<P>User-Agent: Asterisk PBX</P>
<P>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</P>
<P>Expires: 0</P>
<P>Content-Length: 0</P>
<P> </P>
<P>---</P>
<P>Destroying call '87FL32ysiq6y7LBxVuoSI37@10.10.150.154'</P>
<P>w v</P>
<P><-- SIP read from 10.10.150.164:5060: </P>
<P>SUBSCRIBE sip:10.10.150.154@10.10.150.154 SIP/2.0</P>
<P>Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKvBdoBF4DKyUrazDA08E5</P>
<P>Max-Forwards: 70</P>
<P>From: <sip:9002@10.10.150.154>;tag=Hps1S2unWoEq</P>
<P>To: <sip:10.10.150.154@10.10.150.154></P>
<P>Call-ID: fQYOFvlv@10.10.150.154</P>
<P>CSeq: 9794 SUBSCRIBE</P>
<P>Contact: <sip:9002@10.10.150.164></P>
<P>Accept: application/simple-message-summary</P>
<P>Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO</P>
<P>Allow-Events: dialog,message-summary</P>
<P>Content-Disposition: session</P>
<P>Event: message-summary</P>
<P>Expires: 3600</P>
<P>Supported: replaces</P>
<P>User-Agent: VoipLan Gateway/07.63 (MAC=00087B030A9F; SER= 02605; HW=12)</P>
<P>Content-Type: text/plain</P>
<P>Content-Length: 0</P>
<P> </P>
<P>--- (18 headers 0 lines) ---</P>
<P>May 13 03:38:30 DEBUG[28558]: chan_sip.c:3193 sip_alloc: Allocating new SIP dialog for fQYOFvlv@10.10.150.154 - SUBSCRIBE (No RTP)</P>
<P>Using latest SUBSCRIBE request as basis request</P>
<P>Sending to 10.10.150.164 : 5060 (NAT)</P>
<P>Found peer '9002'</P>
<P>Looking for 10.10.150.154 in custom (domain 10.10.150.154)</P>
<P>Transmitting (NAT) to 10.10.150.164:5060:</P>
<P>SIP/2.0 200 OK</P>
<P>Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKvBdoBF4DKyUrazDA08E5;received=10.10.150.164</P>
<P>From: <sip:9002@10.10.150.154>;tag=Hps1S2unWoEq</P>
<P>To: <sip:10.10.150.154@10.10.150.154>;tag=as3efaa043</P>
<P>Call-ID: fQYOFvlv@10.10.150.154</P>
<P>CSeq: 9794 SUBSCRIBE</P>
<P>User-Agent: Asterisk PBX</P>
<P>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</P>
<P>Expires: 0</P>
<P>Content-Length: 0</P>
<P> </P>
<P>---</P>
<P>Destroying call 'fQYOFvlv@10.10.150.154'</P>
<P>ersion</P>
<P>Asterisk SVN--r127M built by root @ Ray_dev_VM on a i686 running Linux on 2007-05-13 08:35:40 UTC</P>
<P>*CLI> </P>
<P><-- SIP read from 10.10.150.164:5060: </P>
<P>SUBSCRIBE sip:10.10.150.154@10.10.150.154 SIP/2.0</P>
<P>Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKtCU8wBJeHio.QuanNy60lH6Ynj4UT14</P>
<P>Max-Forwards: 70</P>
<P>From: <sip:9002@10.10.150.154>;tag=0xhXUTUuC</P>
<P>To: <sip:10.10.150.154@10.10.150.154></P>
<P>Call-ID: GmqhBccGcOpMdxPDaNERAiYOy82@10.10.150.154</P>
<P>CSeq: 47467 SUBSCRIBE</P>
<P>Contact: <sip:9002@10.10.150.164></P>
<P>Accept: application/simple-message-summary</P>
<P>Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO</P>
<P>Allow-Events: dialog,message-summary</P>
<P>Content-Disposition: session</P>
<P>Event: message-summary</P>
<P>Expires: 3600</P>
<P>Supported: replaces</P>
<P>User-Agent: VoipLan Gateway/07.63 (MAC=00087B030A9F; SER= 02605; HW=12)</P>
<P>Content-Type: text/plain</P>
<P>Content-Length: 0</P>
<P> </P>
<P>--- (18 headers 0 lines) ---</P>
<P>May 13 03:38:31 DEBUG[28558]: chan_sip.c:3193 sip_alloc: Allocating new SIP dialog for GmqhBccGcOpMdxPDaNERAiYOy82@10.10.150.154 - SUBSCRIBE (No RTP)</P>
<P>Using latest SUBSCRIBE request as basis request</P>
<P>Sending to 10.10.150.164 : 5060 (NAT)</P>
<P>Found peer '9002'</P>
<P>Looking for 10.10.150.154 in custom (domain 10.10.150.154)</P>
<P>Transmitting (NAT) to 10.10.150.164:5060:</P>
<P>SIP/2.0 200 OK</P>
<P>Via: SIP/2.0/UDP 10.10.150.164;branch=z9hG4bKtCU8wBJeHio.QuanNy60lH6Ynj4UT14;received=10.10.150.164</P>
<P>From: <sip:9002@10.10.150.154>;tag=0xhXUTUuC</P>
<P>To: <sip:10.10.150.154@10.10.150.154>;tag=as0a0daba8</P>
<P>Call-ID: GmqhBccGcOpMdxPDaNERAiYOy82@10.10.150.154</P>
<P>CSeq: 47467 SUBSCRIBE</P>
<P>User-Agent: Asterisk PBX</P>
<P>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</P>
<P>Expires: 0</P>
<P>Content-Length: 0</P></FONT></DIV>
<DIV><BR> </DIV><BR>
--
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