p800aul Posted January 24, 2011 Report Share Posted January 24, 2011 Hi Any help with this would be great. I have a patton 4112 2 x fxo gateway on Snom One. When making a call via this gateway the call will either connect or give a busy tone alternately, this behaviour is consistent, i.e. call to 01246123123 call rings and works fine, hang up, call 01246123123 line busy, hang up, call 01246123123 call rings and works fine and so on..... I have had the config of the patton checked by patton and they don’t see any issues which could cause this behaviour, we tried changing a few things on the patton with no effect. The logs etc are below along with the trunk set up. Thanks for any help regards Paul the trunk is setup: # Trunk 5 in domain localhost Name: Patton Type: register To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: RegRegistrar: 192.168.1.200 RegKeep: RegUser: Icid: Require: OutboundProxy: 192.168.1.200 Ani: DialExtension: 44 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: only_5xx Privacy: pai Glob: RequestTimeout: Codecs: CodecLock: true Expires: 3600 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: Colines: 2 DialogPermission: Log from a succesful call [9] 2011/01/23 22:32:47: [7] 2011/01/23 22:32:50: INVITE sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone> Call-ID: m6j5w1sdsk CSeq: 22153 INVITE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub User-Agent: snom-m9/9.2.42-a Content-Type: application/sdp Content-Length: 398 v=0 o=root 1565728340 1565728341 IN IP4 192.168.1.2 s=- c=IN IP4 192.168.1.2 t=0 0 m=audio 52578 RTP/AVP 0 8 18 3 9 2 10 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:VUyNQYRiVjnhewe7vV1qF+eJ9VtmWTiW1RZOyQT6|2^31 a=sendrecv [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22153 INVITE Content-Length: 0 [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22153 INVITE User-Agent: snom-PBX/4.2.0.3974 WWW-Authenticate: Digest realm="192.168.1.13",nonce="380850f72b1790232735c52375b1d44a",domain="sip:263016@192.168.1.13;user=phone",algorithm=MD5 Content-Length: 0 [7] 2011/01/23 22:32:50: ACK sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22153 ACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Content-Length: 0 [7] 2011/01/23 22:32:50: INVITE sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone> Call-ID: m6j5w1sdsk CSeq: 22154 INVITE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub User-Agent: snom-m9/9.2.42-a Authorization: Digest realm="192.168.1.13",nonce="380850f72b1790232735c52375b1d44a",response="6929aee06e8ccb51bbe0ab176106929d",username="45",uri="sip:263016@192.168.1.13;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 398 v=0 o=root 1565728340 1565728341 IN IP4 192.168.1.2 s=- c=IN IP4 192.168.1.2 t=0 0 m=audio 52578 RTP/AVP 0 8 18 3 9 2 10 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:VUyNQYRiVjnhewe7vV1qF+eJ9VtmWTiW1RZOyQT6|2^31 a=sendrecv [8] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22154 INVITE Content-Length: 0 [8] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: INVITE sip:263016@192.168.1.200;user=phone SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone> Call-ID: f5058ca5@pbx CSeq: 8184 INVITE Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Type: application/sdp Content-Length: 327 v=0 o=- 58077 58077 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 63112 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22154 INVITE Contact: <sip:45@xx.xx.xx.xx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 324 v=0 o=- 43827 43827 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 54492 RTP/AVP 0 8 9 2 3 96 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2011/01/23 22:32:50: PRACK sip:45@xx.xx.xx.xx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-0tvq9n;rport From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22155 PRACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" RAck: 1 22154 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Content-Length: 0 [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-0tvq9n;rport=4043;received=192.168.1.1 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22155 PRACK Contact: <sip:45@xx.xx.xx.xx:5060> User-Agent: snom-PBX/4.2.0.3974 Content-Length: 0 [7] 2011/01/23 22:32:51: SIP/2.0 100 Trying Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport=5060;received=192.168.1.13 From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone> Call-ID: f5058ca5@pbx CSeq: 8184 INVITE Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26 Content-Length: 0 [9] 2011/01/23 22:32:51: [7] 2011/01/23 22:32:54: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport=5060;received=192.168.1.13 From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773 Call-ID: f5058ca5@pbx CSeq: 8184 INVITE Contact: <sip:263016@192.168.1.200:5060> Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26 Supported: replaces Content-Type: application/sdp Content-Length: 221 v=0 o=MxSIP 0 57 IN IP4 192.168.1.200 s=SIP Call c=IN IP4 192.168.1.200 t=0 0 m=audio 4976 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2011/01/23 22:32:54: [9] 2011/01/23 22:32:54: [9] 2011/01/23 22:32:54: [7] 2011/01/23 22:32:54: ACK sip:263016@192.168.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-291e98728a7e6723bd58601166585878;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773 Call-ID: f5058ca5@pbx CSeq: 8184 ACK Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Length: 0 [9] 2011/01/23 22:32:54: [9] 2011/01/23 22:32:54: [9] 2011/01/23 22:32:54: [7] 2011/01/23 22:32:54: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22154 INVITE Contact: <sip:45@xx.xx.xx.xx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 Content-Type: application/sdp Content-Length: 324 v=0 o=- 43827 43827 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 54492 RTP/AVP 0 8 9 2 3 96 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2011/01/23 22:32:54: ACK sip:45@xx.xx.xx.xx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-zaqp37;rport From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22154 ACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Content-Length: 0 [8] 2011/01/23 22:32:59: [9] 2011/01/23 22:32:59: [7] 2011/01/23 22:33:00: BYE sip:45@xx.xx.xx.xx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-ce9s8k;rport From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22156 BYE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub Content-Length: 0 [9] 2011/01/23 22:33:00: [7] 2011/01/23 22:33:00: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-ce9s8k;rport=4043;received=192.168.1.1 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22156 BYE Contact: <sip:45@xx.xx.xx.xx:5060> User-Agent: snom-PBX/4.2.0.3974 Content-Length: 0 [9] 2011/01/23 22:33:00: [9] 2011/01/23 22:33:00: [7] 2011/01/23 22:33:00: BYE sip:263016@192.168.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-ac3b9d3338c98fa09f1c607eeeca5213;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773 Call-ID: f5058ca5@pbx CSeq: 8185 BYE Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Length: 0 And the log from a failed call made stright after the above. INVITE sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone> Call-ID: o77i5idc96 CSeq: 12645 INVITE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub User-Agent: snom-m9/9.2.42-a Content-Type: application/sdp Content-Length: 398 v=0 o=root 1833475499 1833475500 IN IP4 192.168.1.2 s=- c=IN IP4 192.168.1.2 t=0 0 m=audio 56684 RTP/AVP 0 8 18 3 9 2 10 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZT6rmB8cjbvcKGdGKF5E1VXrx1ZBnP7/04nGSRA7|2^31 a=sendrecv [9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:64380 [9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:64381 [9] 2011/01/23 22:31:51: Resolve 7529: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7529: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7529: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12645 INVITE Content-Length: 0 [9] 2011/01/23 22:31:51: Resolve 7530: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7530: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7530: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12645 INVITE User-Agent: snom-PBX/4.2.0.3974 WWW-Authenticate: Digest realm="192.168.1.13",nonce="c9244e6dab13e1f55b84ee9830031c0f",domain="sip:263016@192.168.1.13;user=phone",algorithm=MD5 Content-Length: 0 [7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.2:4043: ACK sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12645 ACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Content-Length: 0 [7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.2:4043: INVITE sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone> Call-ID: o77i5idc96 CSeq: 12646 INVITE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub User-Agent: snom-m9/9.2.42-a Authorization: Digest realm="192.168.1.13",nonce="c9244e6dab13e1f55b84ee9830031c0f",response="70b900d2b3d9130bd5d678d3f7985945",username="45",uri="sip:263016@192.168.1.13;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 398 v=0 o=root 1833475499 1833475500 IN IP4 192.168.1.2 s=- c=IN IP4 192.168.1.2 t=0 0 m=audio 56684 RTP/AVP 0 8 18 3 9 2 10 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZT6rmB8cjbvcKGdGKF5E1VXrx1ZBnP7/04nGSRA7|2^31 a=sendrecv [8] 2011/01/23 22:31:51: Tagging request with existing tag [9] 2011/01/23 22:31:51: Resolve 7531: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7531: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7531: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12646 INVITE Content-Length: 0 [8] 2011/01/23 22:31:51: Set the To domain based on From user 45@localhost [9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:53478 [9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:53479 [9] 2011/01/23 22:31:51: Resolve 7532: url sip:192.168.1.200 [9] 2011/01/23 22:31:51: Resolve 7532: udp 192.168.1.200 5060 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.200:5060: INVITE sip:263016@192.168.1.200;user=phone SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580 To: <sip:263016@192.168.1.200;user=phone> Call-ID: ed6c53a8@pbx CSeq: 27475 INVITE Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Type: application/sdp Content-Length: 327 v=0 o=- 28432 28432 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 53478 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [9] 2011/01/23 22:31:51: Resolve 7533: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7533: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7533: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12646 INVITE Contact: <sip:45@xx.xx.xx.xx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 324 v=0 o=- 16374 16374 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 64380 RTP/AVP 0 8 9 2 3 96 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.1:4043: PRACK sip:45@xx.xx.xx.xx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-4jmwo3;rport From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12647 PRACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" RAck: 1 12646 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Content-Length: 0 [9] 2011/01/23 22:31:51: Resolve 7534: udp 192.168.1.1 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.1:4043: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-4jmwo3;rport=4043;received=192.168.1.1 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12647 PRACK Contact: <sip:45@xx.xx.xx.xx:5060> User-Agent: snom-PBX/4.2.0.3974 Content-Length: 0 [7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.200:5060: SIP/2.0 502 Bad Gateway Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport=5060;received=192.168.1.13 From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580 To: <sip:263016@192.168.1.200;user=phone>;tag=3034763092 Call-ID: ed6c53a8@pbx CSeq: 27475 INVITE Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26 Content-Length: 0 [7] 2011/01/23 22:31:51: Call ed6c53a8@pbx: Clear last INVITE [9] 2011/01/23 22:31:51: Resolve 7535: url sip:192.168.1.200 [9] 2011/01/23 22:31:51: Resolve 7535: udp 192.168.1.200 5060 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.200:5060: ACK sip:263016@192.168.1.200;user=phone SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580 To: <sip:263016@192.168.1.200;user=phone>;tag=3034763092 Call-ID: ed6c53a8@pbx CSeq: 27475 ACK Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Length: 0 [5] 2011/01/23 22:31:51: INVITE Response 502 Bad Gateway: Terminate ed6c53a8@pbx [9] 2011/01/23 22:31:51: Resolve 7536: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7536: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7536: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:52: SIP Tx udp:192.168.1.2:4043: SIP/2.0 502 Bad Gateway Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12646 INVITE Contact: <sip:45@xx.xx.xx.xx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 Content-Length: 0 [8] 2011/01/23 22:31:52: Hangup: Call 101 not found [7] 2011/01/23 22:31:52: SIP Rx udp:192.168.1.2:4043: ACK sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12646 ACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Content-Length: 0 [7] 2011/01/23 22:31:57: SIP Rx udp:192.168.1.2:4043: REGISTER sip:192.168.1.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-90ywmz;rport From: "Paul Stead" <sip:44@192.168.1.13>;tag=s8ozvp To: "Paul Stead" <sip:44@192.168.1.13> Call-ID: ulydh2y8@snom CSeq: 3435 REGISTER Max-Forwards: 70 Contact: <sip:44@192.168.1.2:4043;transport=udp;line=j5sjvx>;reg-id=1;+sip.instance="<urn:uuid:249f54b0-67ba-445c-8433-55ee8f3a7b1a>" Supported: path, outbound, gruu User-Agent: snom-m9/9.2.42-a Authorization: Digest realm="192.168.1.13",nonce="10ceb016de3d4209ddadda412473a800",response="9afc717ab4f9d532a315f8378102e9f7",username="44",uri="sip:192.168.1.13",algorithm=MD5 Expires: 354 Content-Length: 0 [9] 2011/01/23 22:31:57: Resolve 7537: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:57: Resolve 7537: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:57: Resolve 7537: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:57: SIP Tx udp:192.168.1.2:4043: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-90ywmz;rport=4043 From: "Paul Stead" <sip:44@192.168.1.13>;tag=s8ozvp To: "Paul Stead" <sip:44@192.168.1.13>;tag=49ef1d8f34 Call-ID: ulydh2y8@snom CSeq: 3435 REGISTER Contact: <sip:44@192.168.1.2:4043;transport=udp;line=j5sjvx>;expires=352 Require: outbound Supported: path Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 24, 2011 Report Share Posted January 24, 2011 Is xx.xx.xx.xx the public IP address of the device? I see the private address is 192.168.1.13. The PBX should put the private IP address here I guess? What is the routing table of the system looking like (route print)? Quote Link to comment Share on other sites More sharing options...
p800aul Posted January 25, 2011 Author Report Share Posted January 25, 2011 Hi pbxnsip Thanks for your reply. I've been back on to patton as the issue seems to be a 502 from that trunk SIP/2.0 502 Bad GatewayVia: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043 I've run a debug on the patton and this hopefully will give them a clue as to why this is happening. It only happens on alternate calls i.e. a call to 123321 goes through, hang up, call to 123321 busy tone (502), hang up, call to 123321 goes through, this is regardless of time between the calls. The rest of the system works i have the pbx on the dmz and as it's a simple system i set it up using this from Matt post xx.xx.xx.xx is the public when i get a solution or not i'll come back Regards Paul Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 25, 2011 Report Share Posted January 25, 2011 If the PBX puts the public address in the SIP packet although it is sent in the LAN, then that is not okay. You will have the "hairpinning NAT" problem. You can fix this by changing the route on the server, that's why I was asking what the route on the server looks like. Quote Link to comment Share on other sites More sharing options...
p800aul Posted January 25, 2011 Author Report Share Posted January 25, 2011 If the PBX puts the public address in the SIP packet although it is sent in the LAN, then that is not okay. You will have the "hairpinning NAT" problem. You can fix this by changing the route on the server, that's why I was asking what the route on the server looks like. So is the routing in Matt's post ok for me? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 25, 2011 Report Share Posted January 25, 2011 So is the routing in Matt's post ok for me? Hard to say. Can you just enter "route print" in the CMD sell (Windows) and we can see if the routing is okay? If you have public IP addresses there which you don't want to expose, just replace them with 23.34.45.56 or something like that... Quote Link to comment Share on other sites More sharing options...
p800aul Posted January 25, 2011 Author Report Share Posted January 25, 2011 here you go by the way Patton fixed it Microsoft Windows XP [Version 5.1.2600] © Copyright 1985-2001 Microsoft Corp. C:\Documents and Settings\paul>route print =========================================================================== Interface List 0x1 ........................... MS TCP Loopback interface 0x2 ...00 24 1d a0 7f 65 ...... Realtek PCIe FE Family Controller - Packet Sched uler Miniport =========================================================================== =========================================================================== Active Routes: Network Destination Netmask Gateway Interface Metric 0.0.0.0 0.0.0.0 192.168.1.1 192.168.1.13 20 127.0.0.0 255.0.0.0 127.0.0.1 127.0.0.1 1 192.168.1.0 255.255.255.0 192.168.1.13 192.168.1.13 20 192.168.1.13 255.255.255.255 127.0.0.1 127.0.0.1 20 192.168.1.255 255.255.255.255 192.168.1.13 192.168.1.13 20 224.0.0.0 240.0.0.0 192.168.1.13 192.168.1.13 20 255.255.255.255 255.255.255.255 192.168.1.13 192.168.1.13 1 Default Gateway: 192.168.1.1 =========================================================================== Persistent Routes: None Regards Paul Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 26, 2011 Report Share Posted January 26, 2011 by the way Patton fixed it Whow, so that means you need a new firmware for the gateway? Or was is a Patton config option? Quote Link to comment Share on other sites More sharing options...
p800aul Posted January 26, 2011 Author Report Share Posted January 26, 2011 Whow, so that means you need a new firmware for the gateway? Or was is a Patton config option? Config, i was told originally that not having a line attached to the second FXO port wouldn't matter, it seems it does! They made a couple of changes to my original config the settings on the FXO interface so it will recognize a longer or shorter tone break and see this as a disconnect and go back on-hook to be ready to accept another call. Secondly, disabled cyclic routing in your hunt group. Since I only have one interface working at the moment, the hunt will now try the first interface over and over. Apparently this is easy change back when I need that 2nd interface in routing. Out of interest for everyone, here is my running config for a Patton 4112 with only one pstn line attached, I'm in the UK. Regards Paul #----------------------------------------------------------------# # # # SN4112/JO/EUI # # R5.2 2009-01-14 H323 SIP FXS FXO # # 2011-01-26T07:26:33 # # SN/00A0BA0609C0 # # Generated configuration file # # # #----------------------------------------------------------------# cli version 3.20 webserver port 80 language en sntp-client sntp-client server primary 194.35.252.7 port 123 version 4 sntp-client server secondary 194.164.127.5 port 123 version 4 sntp-client local-clock-offset system ic voice 0 low-bitrate-codec g729 profile ppp default profile call-progress-tone defaultDialtone play 1 1000 450 -6 profile call-progress-tone defaultAlertingtone play 1 1000 450 -13 pause 2 5000 profile call-progress-tone defaultBusytone play 1 300 450 -7 pause 2 300 profile call-progress-tone defaultReleasetone play 1 300 450 -7 pause 2 300 profile call-progress-tone defaultCongestiontone play 1 300 450 -7 pause 2 300 profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax transmission 2 bypass g711alaw64k profile pstn default profile sip default profile aaa default method 1 local method 2 none context ip router interface IF_IP_LAN ipaddress dhcp tcp adjust-mss rx mtu tcp adjust-mss tx mtu interface IF_IP_WAN ipaddress dhcp tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 192.168.1.1 0 context cs switch digit-collection timeout 2 interface sip IF_SIP_1 bind context sip-gateway GW_SIP_ALL_LINES route call dest-service HUNT_FXO remote 192.168.1.13 5060 early-connect early-disconnect address-translation outgoing-call request-uri user-part fix 10015 host-part to-header target-param none interface sip IF_SIP_2 bind context sip-gateway GW_SIP_ALL_LINES route call dest-service HUNT_FXO remote 192.168.1.13 5060 early-connect early-disconnect address-translation outgoing-call request-uri user-part fix 10016 host-part to-header target-param none interface fxo IF_FXO_1 route call dest-interface IF_SIP_1 loop-break-duration min 60 max 5000 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id dial-after timeout 1 mute-dialing interface fxo IF_FXO_2 route call dest-interface IF_SIP_2 loop-break-duration min 100 max 500 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id dial-after timeout 1 mute-dialing service hunt-group HUNT_FXO drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_FXO_1 route call 2 dest-interface IF_FXO_2 context cs switch no shutdown authentication-service AS_ALL_LINES username 10015 password Z+ApY8PXmFjMRxFr04ls2w== encrypted username 10016 password c7k7vrPq2MMY+mdxPJS6aQ== encrypted location-service LS_ALL_LINES identity 10015 identity 10016 context sip-gateway GW_SIP_ALL_LINES interface LAN bind interface IF_IP_LAN context router port 5060 context sip-gateway GW_SIP_ALL_LINES no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface IF_IP_LAN router no shutdown port fxo 0 0 use profile fxo gb encapsulation cc-fxo bind interface IF_FXO_1 switch no shutdown port fxo 0 1 use profile fxo gb encapsulation cc-fxo bind interface IF_FXO_2 switch shutdown Quote Link to comment Share on other sites More sharing options...
Recommended Posts
Join the conversation
You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.