Nathan Dennis
2007-Jul-04 22:56 UTC
[asterisk-users] Problems with SIP Registration on VPN Link
Hi, We are having major problems with a remote site that links to the head office via a VPN tunnel. The phones will register fine and work for a few minutes to hours but then will drop their connection and will no register to asterisk even with a restart of the phone. We have 2 other remote sites that work exactly same and they are not having any issues so i believe it has to be be something to do with the network rather then asterisk but this is the sip debug for a phone trying to register. Any idea where i should start to look as this has me totally confused as obviously the phones can communicate with asterisk at all times just something is causing the registration to get screwed up. Jul 4 09:43:46 NOTICE[7320]: chan_sip.c:6599 check_auth: stale nonce received from '<sip:763 at 192.168.10.12;user=phone>' Transmitting (NAT) to 192.168.12.63:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63 From: "Edmonton Boardroom 1" <sip:763 at 192.168.10.12;user=phone>;tag=65cbed22c3593805 To: <sip:763 at 192.168.10.12;user=phone>;tag=as4d6893cc Call-ID: cb87be6d32f3f26e at 192.168.12.63 CSeq: 10005 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48f69f92", stale=true Content-Length: 0 --- Scheduling destruction of call 'cb87be6d32f3f26e at 192.168.12.63' in 15000 ms cnsmavs1*CLI> <-- SIP read from 192.168.12.63:5060: REGISTER sip:192.168.10.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15 From: "Edmonton Boardroom 1" <sip:763 at 192.168.10.12;user=phone>;tag=65cbed22c3593805 To: <sip:763 at 192.168.10.12;user=phone> Contact: <sip:763 at 192.168.12.63:5060;transport=udp;user=phone> Supported: path Authorization: Digest username="763", realm="asterisk", algorithm=MD5, uri="sip:192.168.10.12", nonce="587da437", response="4bd29b9213057e3e2f3a5270748fbe85" all-ID: cb87be6d32f3f26e at 192.168.12.63 CSeq: 10005 REGISTER Expires: 3600 User-Agent: Grandstream GXP2000 1.1.2.23 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,M ESSAGE Content-Length: 0 --- (14 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.12.63 : 5060 (NAT) Transmitting (NAT) to 192.168.12.63:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63 From: "Edmonton Boardroom 1" <sip:763 at 192.168.10.12;user=phone>;tag=65cbed22c3593805 To: <sip:763 at 192.168.10.12;user=phone> Call-ID: cb87be6d32f3f26e at 192.168.12.63 CSeq: 10005 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:763 at 192.168.10.12> Content-Length: 0 --- Jul 4 09:43:48 NOTICE[7320]: chan_sip.c:6599 check_auth: stale nonce received from '<sip:763 at 192.168.10.12;user=phone>' Transmitting (NAT) to 192.168.12.63:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.12.63:5060;branch=z9hG4bKfed637754cf22f15;received=192.168.12.63 From: "Edmonton Boardroom 1" <sip:763 at 192.168.10.12;user=phone>;tag=65cbed22c3593805 To: <sip:763 at 192.168.10.12;user=phone>;tag=as4d6893cc Call-ID: cb87be6d32f3f26e at 192.168.12.63 CSeq: 10005 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="750fc224", stale=true Content-Length: 0 Nathan Dennis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070705/bc144366/attachment.htm