I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doesn't know about. Surely, surely.... someone else must have seen this? hermes*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message xxx.187.128.20 3254104 06b217722a8 00102/00031 ulaw Yes Rx: REFER xxx.187.142.233 3254102 e1acc8e-eb8 00101/00002 ulaw No Rx: ACK 2 active SIP channels hermes*CLI> hermes*CLI> Mar 28 16:01:15 NOTICE[11656]: chan_sip.c:6734 get_refer_info: Supervised transfer requested, but unable to find callid 'bc743aeb-13b250a9-d8f2ccf4@xxx.187.128.20'. Both legs must reside on Asterisk box to transfer at this time. Looks the Asterisk doesn't know about the call-id. Why not? Doug.> -----Original Message----- > From: Douglas Garstang > Sent: Tuesday, March 28, 2006 8:30 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: NATted phones transferring calls - BUG0003710 > > > I made a call from 3254102 to 2944093. I then tried to do a > transfer to 3254107. > IP addresses have been changed to protect the innocent. > > It appears this related to bug 3710. It's unclear from the > bug if the problem has been fixed or not. If it hasn't, then > this seems pretty serious and would I guess affect any > NAT-ted phones ability to transfer calls. > > Here's the REFER that the phone at 2944093 sends directly to Asterisk: > > U 216.186.128.68:5060 -> 216.186.142.203:5060 > REFER sip:3254102@216.186.142.203 SIP/2.0. > Via: SIP/2.0/UDP 216.186.128.68;branch=z9hG4bKba3b074892377BD1. > From: <sip:2944093@216.186.128.68>;tag=C06397B-C3C1D97A. > To: "Test User" <sip:3254102@216.186.142.203>;tag=as33e7dd7c. > CSeq: 2 REFER. > Call-ID: 4053b9972e7851f455d9d16e7706d3f4@216.186.142.203. > Contact: <sip:2944093@216.186.128.68>. > User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067. > Refer-To: > <sip:3254107@ipt.oneeighty.com;user=phone?Replaces=a13c692-349 > c9c70-7bd27d33%40172.31.99.4%3Bto-tag%3Das2a8d818b%3Bfrom-tag% > 3D3DE1A6BE-7262B959>. > Referred-By: <sip:2944093@216.186.128.68>. > Max-Forwards: 70. > Content-Length: 0. > > Asterisk then goes and complains: > > Mar 27 16:25:57 NOTICE[20511]: chan_sip.c:6734 > get_refer_info: Supervised transfer requested, but unable to > find callid 'a13c692-349c9c70-7bd27d33@172.31.99.4'. Both > legs must reside on Asterisk box to transfer at this time. > > The phone's real IP address is 172.31.99.4. > I'm not really sure what the problem is except that it works > fine when there's no NAT involved. I can see the real IP > address in the dialog. I wonder if that's what is confusing Asterisk? > > Doug. >
29 mar 2006 kl. 01.03 skrev Douglas Garstang:> I made the post below earlier today. I'v since removed all NAT from > the equation and the problem still persists. Basically I am trying > to transfer a call. The transferring phone sends a REFER message to > asterisk with a call id that Asterisk doesn't know about. Surely, > surely.... someone else must have seen this? > > hermes*CLI> sip show channels > Peer User/ANR Call ID Seq (Tx/Rx) Form > Hold Last Message > xxx.187.128.20 3254104 06b217722a8 00102/00031 ulaw > Yes Rx: REFER > xxx.187.142.233 3254102 e1acc8e-eb8 00101/00002 ulaw > No Rx: ACK > 2 active SIP channels > hermes*CLI> > hermes*CLI> > Mar 28 16:01:15 NOTICE[11656]: chan_sip.c:6734 get_refer_info: > Supervised transfer requested, but unable to find callid > 'bc743aeb-13b250a9-d8f2ccf4@xxx.187.128.20'. Both legs must reside > on Asterisk box to transfer at this time. > > Looks the Asterisk doesn't know about the call-id. Why not?' That call may be on another server - your sip show channels confirm that it is not on this server. The error message clearly says that both legs must reside on Asterisk box... /O
Olle, xxx.187.142.20 is a phone (not sure if it's the caller or callee) xxx.187.142.203 is an OpenSER system, which sits in between the phones and Asterisk. I guess OpenSER must be changing the callid when it forwards SIP messages to Asterisk. It's surprising that no one else seems to have encountered this problem, as a lot of people use OpenSER, as we have done, to get around some pretty serious limitations with Asterisk. Why on earth is OpenSER's ip address in there anyway? It's not in the RTP path. Asterisk should have the phone's IP address in there. It's a little like SIP subscribe/notify messages. Asterisk sends notify messages back to OpenSER, where the SUBSCRIBE's came from, rather than sending them directly to the phone. Why is that? Arrgh. Doug -----Original Message----- From: Olle E Johansson [mailto:oej@edvina.net] Sent: Tue 3/28/2006 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Transferring calls - BUG0003710 29 mar 2006 kl. 01.03 skrev Douglas Garstang: > I made the post below earlier today. I'v since removed all NAT from > the equation and the problem still persists. Basically I am trying > to transfer a call. The transferring phone sends a REFER message to > asterisk with a call id that Asterisk doesn't know about. Surely, > surely.... someone else must have seen this? > > hermes*CLI> sip show channels > Peer User/ANR Call ID Seq (Tx/Rx) Form > Hold Last Message > xxx.187.128.20 3254104 06b217722a8 00102/00031 ulaw > Yes Rx: REFER > xxx.187.142.233 3254102 e1acc8e-eb8 00101/00002 ulaw > No Rx: ACK > 2 active SIP channels > hermes*CLI> > hermes*CLI> > Mar 28 16:01:15 NOTICE[11656]: chan_sip.c:6734 get_refer_info: > Supervised transfer requested, but unable to find callid > 'bc743aeb-13b250a9-d8f2ccf4@xxx.187.128.20'. Both legs must reside > on Asterisk box to transfer at this time. > > Looks the Asterisk doesn't know about the call-id. Why not? ' That call may be on another server - your sip show channels confirm that it is not on this server. The error message clearly says that both legs must reside on Asterisk box... /O _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users