I keep getting the same answer from people "Well the SIP implementation is fine if you use XXX IP Phone"
so obviously Asterisk was never designed to be used as a TDM gateway but merely as a PBX server only. On 4/17/05, Cameron Beattie <[EMAIL PROTECTED]> wrote: > This is very interesting to me since I am in the process of setting up SER > to Asterisk in a similar scenario. I'm surprised there haven't been more > posts. Maybe include SER <-> Asterisk in the title. There are other posters > on the list who use SER and Asterisk together who surely must have > encountered (overcome?) this problem since it is so fundamental. Perhaps a > bug should be raised? > > Regards > > Cameron > ----- Original Message ----- > From: "Daniel Corbe" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Thursday, April 14, 2005 7:29 AM > Subject: [Asterisk-Users] Loop Detection > > Hello, > > Is there any way to turn Loop Detection off or tune the params a bit? > I am having an issue with Call Forwarding on my SIP Proxy Server which > is causing me great pains. > > Here is the issue: > > 1) I have a SIP UA which registers with a SER proxy server. > 2) I have an Asterisk TDM gateway in my network, also which registers with > SER > 3) A call comes in through the PSTN to the Asterisk Gateway. The > Asterisk gateway sends the call to SER destined for my SIP UA > 4) SER sees that the SIP UA has call forwarding enabled so it creates > a new outbound call with the same Call ID but it has a different TAG= > line and Max-Forwards is set to 70. > 5) Since the fowarding number is out on the PSTN, SER routes the call > back through the same * gateway. > 6) Asterisk rejects the phone call with "Loop Detected" > > According to my interpretation of the RFC, it is more correct to base > loop detection off of the TAG= than it is off of the Call ID. Having > said that, SER also sets the Max Forwards on the call. > > Is there any way at all to get Asterisk to either base its loop > detection off the TAG= or respect the Max-Forwards setting? > > I've also attached a libpcap packet dump of a phone call. > > 389.764074 62.25.108.211 -> 66.165.175.44 SIP/SDP Request: INVITE > sip:[EMAIL PROTECTED], with session description > 389.885825 66.165.175.44 -> 62.25.108.211 SIP Status: 401 Unauthorized > 389.885999 62.25.108.211 -> 66.165.175.44 SIP Request: ACK > sip:[EMAIL PROTECTED] > 389.886104 62.25.108.211 -> 66.165.175.44 SIP/SDP Request: INVITE > sip:[EMAIL PROTECTED], with session description > 390.145261 66.165.175.44 -> 62.25.108.211 SIP Status: 100 trying -- > your call is important to us > 390.257658 66.165.175.44 -> 62.25.108.211 SIP/SDP Request: INVITE > sip:[EMAIL PROTECTED]:5060, with session description > 390.257706 62.25.108.211 -> 66.165.175.44 SIP Status: 482 Loop Detected > 390.801964 66.165.175.44 -> 62.25.108.211 SIP/SDP Request: INVITE > sip:[EMAIL PROTECTED]:5060, with session description > 390.802007 62.25.108.211 -> 66.165.175.44 SIP Status: 482 Loop Detected > 391.901785 66.165.175.44 -> 62.25.108.211 SIP/SDP Request: INVITE > sip:[EMAIL PROTECTED]:5060, with session description > 391.901829 62.25.108.211 -> 66.165.175.44 SIP Status: 482 Loop Detected > 393.991808 66.165.175.44 -> 62.25.108.211 SIP/SDP Request: INVITE > sip:[EMAIL PROTECTED]:5060, with session description > 393.991851 62.25.108.211 -> 66.165.175.44 SIP Status: 482 Loop Detected > 401.223872 62.25.108.211 -> 66.165.175.44 SIP Request: CANCEL > sip:[EMAIL PROTECTED] > > -------------------------------------------------------------------------------- > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users