Hi all,
I have ser operating with a combination of X-lite and grandstream phones. I am trying to get it it play nicely with Asterisk and have run into a strange situation. Calling from ser to asterisk works fine but when I try to call back from asterisk to ser I run into problems, the sip device (behind NAT) rings but when I answer the call ser tries to send the ACK back to the private IP and not the public IP (found using STUN) that it was using up until that point.
I have the following debug from ngrep to illustrate the problem. Offending SIP packet marked with ########s
Any ideas?
# U 222.152.11.236:36595 -> xx.xx.xx.52:5060 SIP/2.0 100 trying. Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0. Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b. From: "Hamish Archer" sip:100@xx.xx.xx.51:5080;tag=as12db357a. To: sip:17778134000@proxy02.blahblah.xxx. Call-ID: 1738a1ee3d2310af4082784e4d0a6d89@xx.xx.xx.51. CSeq: 102 INVITE. User-Agent: Grandstream BT100 1.0.4.54. Content-Length: 0. .
# U 222.152.11.236:36595 -> xx.xx.xx.52:5060 SIP/2.0 180 ringing. Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0. Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b. Record-Route: sip:xx.xx.xx.52;ftag=as12db357a;lr=on. From: "Hamish Archer" sip:100@xx.xx.xx.51:5080;tag=as12db357a. To: sip:17778134000@proxy02.blahblah.xxx;tag=760cdf6859149410. Call-ID: 1738a1ee3d2310af4082784e4d0a6d89@xx.xx.xx.51. CSeq: 102 INVITE. User-Agent: Grandstream BT100 1.0.4.54. Content-Length: 0. .
# U 222.152.11.236:36595 -> xx.xx.xx.52:5060 SIP/2.0 180 ringing. Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0. Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b. Record-Route: sip:xx.xx.xx.52;ftag=as12db357a;lr=on. From: "Hamish Archer" sip:100@xx.xx.xx.51:5080;tag=as12db357a. To: sip:17778134000@proxy02.blahblah.xxx;tag=efb3a3fe9b832ccc. Call-ID: 1738a1ee3d2310af4082784e4d0a6d89@xx.xx.xx.51. CSeq: 102 INVITE. User-Agent: Grandstream BT100 1.0.4.54. Content-Length: 0. .
# U 222.152.11.236:36595 -> xx.xx.xx.52:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0. Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b. Record-Route: sip:xx.xx.xx.52;ftag=as12db357a;lr=on. From: "Hamish Archer" sip:100@xx.xx.xx.51:5080;tag=as12db357a. To: sip:17778134000@proxy02.blahblah.xxx;tag=760cdf6859149410. Call-ID: 1738a1ee3d2310af4082784e4d0a6d89@xx.xx.xx.51. CSeq: 102 INVITE. User-Agent: Grandstream BT100 1.0.4.54. Contact: sip:17778134000@192.168.1.158. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 209. . v=0. o=17778134000 8000 8000 IN IP4 192.168.1.158. s=SIP Call. c=IN IP4 222.152.11.236. t=0 0. m=audio 5004 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
################################################# U xx.xx.xx.52:5060 -> 192.168.1.158:5060 ACK sip:17778134000@192.168.1.158 SIP/2.0. Max-Forwards: 10. Record-Route: sip:xx.xx.xx.52;ftag=as12db357a;lr=on. Via: SIP/2.0/UDP xx.xx.xx.52;branch=0. Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b. Route: sip:17778134000@192.168.1.158. From: "Hamish Archer" sip:100@xx.xx.xx.51:5080;tag=as12db357a. To: sip:17778134000@proxy02.blahblah.xxx;tag=760cdf6859149410. Contact: sip:100@xx.xx.xx.51:5080. Call-ID: 1738a1ee3d2310af4082784e4d0a6d89@xx.xx.xx.51. CSeq: 102 ACK. User-Agent: Asterisk PBX. Content-Length: 0. ###################################################
# U 222.152.11.236:36595 -> xx.xx.xx.52:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0. Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b. Record-Route: sip:xx.xx.xx.52;ftag=as12db357a;lr=on. From: "Hamish Archer" sip:100@xx.xx.xx.51:5080;tag=as12db357a. To: sip:17778134000@proxy02.blahblah.xxx;tag=760cdf6859149410. Call-ID: 1738a1ee3d2310af4082784e4d0a6d89@xx.xx.xx.51. CSeq: 102 INVITE. User-Agent: Grandstream BT100 1.0.4.54. Contact: sip:17778134000@192.168.1.158. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 210. . v=0. o=17778134000 8000 8000 IN IP4 192.168.1.158. s=SIP Call. c=IN IP4 222.152.11.236. t=0 0. m=audio 35337 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.