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 TAGline 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:448701419604@66.165.175.44, 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:448701419604@66.165.175.44 389.886104 62.25.108.211 -> 66.165.175.44 SIP/SDP Request: INVITE sip:448701419604@66.165.175.44, 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:13012370123@62.25.108.211:5060, 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:13012370123@62.25.108.211:5060, 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:13012370123@62.25.108.211:5060, 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:13012370123@62.25.108.211:5060, 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:448701419604@66.165.175.44 -------------- next part -------------- A non-text attachment was scrubbed... Name: voip4u-2005040401.dump Type: application/octet-stream Size: 14356 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050413/9b6ef943/voip4u-2005040401.obj
Daniel Corbe <daniel.junkmail@gmail.com> wrote:>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.All I can do is sympathize. The same problem occurs when a call comes in through Asterisk, gets sent to SER, then comes back to Asterisk 20 seconds later for voicemail. I have contemplated just commenting out the check in chan_sip.c, but I haven't tried this. Not sure if this might cause other problems. Asterisk has many SIP deficiencies. Asterisk has been built as a monolithic PBX, and it seems to do okay using SIP phones as channels. If you want Asterisk to simply act as a SIP UA, you are going to run into a whole slew of problems. I'm not holding my breath waiting for this to change. Doug -- Doug Meredith (doug.meredith@systemguard.com) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com
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" <daniel.junkmail@gmail.com> 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 TAGline 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:448701419604@66.165.175.44, 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:448701419604@66.165.175.44 389.886104 62.25.108.211 -> 66.165.175.44 SIP/SDP Request: INVITE sip:448701419604@66.165.175.44, 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:13012370123@62.25.108.211:5060, 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:13012370123@62.25.108.211:5060, 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:13012370123@62.25.108.211:5060, 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:13012370123@62.25.108.211:5060, 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:448701419604@66.165.175.44 --------------------------------------------------------------------------------> _______________________________________________ > 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