[Asterisk-Users] Loop Detection

[Daniel Corbe]

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 <kjcsb at orcon.net.nz> 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" <daniel.junkmail at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at 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:448701419604 at 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 at 66.165.175.44
> 389.886104 62.25.108.211 -> 66.165.175.44 SIP/SDP Request: INVITE
> sip:448701419604 at 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 at 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 at 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 at 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 at 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 at 66.165.175.44
> 
> --------------------------------------------------------------------------------
> 
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