mathy Posted January 20, 2009 Report Share Posted January 20, 2009 Hi everyone, I have a problem with my ISDN gateway (Patton SmartNode 4638-5BRI) sinds 1 week ... ,when i try to make a call from that gateway , i have always the error "forbidden ", but i can receive always call from that gateway and all the other trunks work perfectly for the outbound call... anyone have a idea of this problem ? I don't have make change before 1 month on the voip system (server,gateway,ip phone) ... here is a copy of my log if can help someone to solve the problem and i use pbxnsip 3.0.0.2998 (Win32) : INVITE sip:9069665262@bizzvoice.bizzdev.net SIP/2.0 Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-cf794530 From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net> Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:526@172.16.1.104:5060> Expires: 240 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 68660036 68660036 IN IP4 172.16.1.104 s=- c=IN IP4 172.16.1.104 t=0 0 m=audio 16410 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv [7] 2009/01/20 14:21:42: UDP: Opening socket on port 61468 [7] 2009/01/20 14:21:42: UDP: Opening socket on port 61469 [5] 2009/01/20 14:21:42: Identify trunk (domain name match) 13 [9] 2009/01/20 14:21:42: Resolve 101602: aaaa udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101602: a udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101602: udp 172.16.1.104 5060 [0] 2009/01/20 14:21:42: SIP Tx udp:172.16.1.104:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-cf794530 From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net>;tag=fcd35cf36c Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 101 INVITE Content-Length: 0 [9] 2009/01/20 14:21:42: Resolve 101603: aaaa udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101603: a udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101603: udp 172.16.1.104 5060 [0] 2009/01/20 14:21:42: SIP Tx udp:172.16.1.104:5060: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-cf794530 From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net>;tag=fcd35cf36c Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 101 INVITE User-Agent: pbxnsip-PBX/3.0.0.2998 WWW-Authenticate: Digest realm="bizzvoice.bizzdev.net",nonce="e77429a4a3af6a1ecbec81abaebe9ca7",domain="sip:9069665262@bizzvoice.bizzdev.net",algorithm=MD5 Content-Length: 0 [0] 2009/01/20 14:21:42: SIP Rx udp:172.16.1.104:5060: ACK sip:9069665262@bizzvoice.bizzdev.net SIP/2.0 Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-cf794530 From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net>;tag=fcd35cf36c Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 101 ACK Max-Forwards: 70 Contact: <sip:526@172.16.1.104:5060> User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 [0] 2009/01/20 14:21:42: SIP Rx udp:172.16.1.104:5060: INVITE sip:9069665262@bizzvoice.bizzdev.net SIP/2.0 Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-d09a6f5c From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net> Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="526",realm="bizzvoice.bizzdev.net",nonce="e77429a4a3af6a1ecbec81abaebe9ca7",uri="sip:9069665262@bizzvoice.bizzdev.net",algorithm=MD5,response="618292a3477d863f9ccbe61a6a9c0a5e" Contact: <sip:526@172.16.1.104:5060> Expires: 240 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 68660036 68660036 IN IP4 172.16.1.104 s=- c=IN IP4 172.16.1.104 t=0 0 m=audio 16410 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv [8] 2009/01/20 14:21:42: Tagging request with existing tag [9] 2009/01/20 14:21:42: Resolve 101604: aaaa udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101604: a udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101604: udp 172.16.1.104 5060 [0] 2009/01/20 14:21:42: SIP Tx udp:172.16.1.104:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-d09a6f5c From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net>;tag=fcd35cf36c Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 102 INVITE Content-Length: 0 [7] 2009/01/20 14:21:42: UDP: Opening socket on port 51300 [7] 2009/01/20 14:21:42: UDP: Opening socket on port 51301 [9] 2009/01/20 14:21:42: Resolve 101605: url sip:172.16.1.190 [9] 2009/01/20 14:21:42: Resolve 101605: udp 172.16.1.190 5060 [0] 2009/01/20 14:21:42: SIP Tx udp:172.16.1.190:5060: INVITE sip:069665262@172.16.1.190;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.1.243:5060;branch=z9hG4bK-8beb0a5b78e2e117595d814b20ad0992;rport From: "Mathy Jean-Charles" <sip:069669526@172.16.1.190;user=phone>;tag=57994 To: <sip:069665262@172.16.1.190;user=phone> Call-ID: 6ab844e3@pbx CSeq: 7236 INVITE Max-Forwards: 70 Contact: <sip:069669526@172.16.1.243:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2998 Content-Type: application/sdp Content-Length: 290 v=0 o=- 56814 56814 IN IP4 172.16.1.243 s=- c=IN IP4 172.16.1.243 t=0 0 m=audio 51300 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=sendrecv [9] 2009/01/20 14:21:42: Resolve 101606: aaaa udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101606: a udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101606: udp 172.16.1.104 5060 [0] 2009/01/20 14:21:42: SIP Tx udp:172.16.1.104:5060: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-d09a6f5c From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net>;tag=fcd35cf36c Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 102 INVITE Contact: <sip:526@172.16.1.243:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2998 Content-Type: application/sdp Content-Length: 253 v=0 o=- 9796 9796 IN IP4 172.16.1.243 s=- c=IN IP4 172.16.1.243 t=0 0 m=audio 61468 RTP/AVP 0 8 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=sendrecv [0] 2009/01/20 14:21:42: SIP Rx udp:172.16.1.190:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.1.243:5060;branch=z9hG4bK-8beb0a5b78e2e117595d814b20ad0992;rport=5060;received=172.16.1.243 From: "Mathy Jean-Charles" <sip:069669526@172.16.1.190;user=phone>;tag=57994 To: <sip:069665262@172.16.1.190;user=phone> Call-ID: 6ab844e3@pbx CSeq: 7236 INVITE Server: Patton SN4638 5BIS UI 00A0BA03938F R5.1 2008-01-18 H323 SIP BRI M5T SIP Stack/4.0.23.23 Content-Length: 0 [0] 2009/01/20 14:21:42: SIP Rx udp:172.16.1.190:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 172.16.1.243:5060;branch=z9hG4bK-8beb0a5b78e2e117595d814b20ad0992;rport=5060;received=172.16.1.243 From: "Mathy Jean-Charles" <sip:069669526@172.16.1.190;user=phone>;tag=57994 To: <sip:069665262@172.16.1.190;user=phone>;tag=2845360344 Call-ID: 6ab844e3@pbx CSeq: 7236 INVITE Contact: <sip:069665262@172.16.1.190:5060> Server: Patton SN4638 5BIS UI 00A0BA03938F R5.1 2008-01-18 H323 SIP BRI M5T SIP Stack/4.0.23.23 Content-Length: 0 [0] 2009/01/20 14:21:42: SIP Rx udp:172.16.1.190:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.16.1.243:5060;branch=z9hG4bK-8beb0a5b78e2e117595d814b20ad0992;rport=5060;received=172.16.1.243 From: "Mathy Jean-Charles" <sip:069669526@172.16.1.190;user=phone>;tag=57994 To: <sip:069665262@172.16.1.190;user=phone>;tag=2845360344 Call-ID: 6ab844e3@pbx CSeq: 7236 INVITE Server: Patton SN4638 5BIS UI 00A0BA03938F R5.1 2008-01-18 H323 SIP BRI M5T SIP Stack/4.0.23.23 Content-Length: 0 [7] 2009/01/20 14:21:42: Call 6ab844e3@pbx#57994: Clear last INVITE [9] 2009/01/20 14:21:42: Resolve 101607: url sip:069665262@172.16.1.190:5060 [9] 2009/01/20 14:21:42: Resolve 101607: udp 172.16.1.190 5060 [0] 2009/01/20 14:21:42: SIP Tx udp:172.16.1.190:5060: ACK sip:069665262@172.16.1.190:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.243:5060;branch=z9hG4bK-8beb0a5b78e2e117595d814b20ad0992;rport From: "Mathy Jean-Charles" <sip:069669526@172.16.1.190;user=phone>;tag=57994 To: <sip:069665262@172.16.1.190;user=phone>;tag=2845360344 Call-ID: 6ab844e3@pbx CSeq: 7236 ACK Max-Forwards: 70 Contact: <sip:069669526@172.16.1.243:5060;transport=udp> Content-Length: 0 [5] 2009/01/20 14:21:42: INVITE Response: Terminate 6ab844e3@pbx [7] 2009/01/20 14:21:42: Other Ports: 1 [7] 2009/01/20 14:21:42: Call Port: ebf34043-c02b625c@172.16.1.104#fcd35cf36c [0] 2009/01/20 14:21:42: SIP Tr udp:172.16.1.104:5060: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-d09a6f5c From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net>;tag=fcd35cf36c Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 102 INVITE Contact: <sip:526@172.16.1.243:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2998 Content-Type: application/sdp Content-Length: 253 v=0 o=- 9796 9796 IN IP4 172.16.1.243 s=- c=IN IP4 172.16.1.243 t=0 0 m=audio 61468 RTP/AVP 0 8 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:30 a=sendrecv [9] 2009/01/20 14:21:42: Resolve 101608: aaaa udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101608: a udp 172.16.1.104 5060 [9] 2009/01/20 14:21:42: Resolve 101608: udp 172.16.1.104 5060 [0] 2009/01/20 14:21:42: SIP Tx udp:172.16.1.104:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.16.1.104:5060;branch=z9hG4bK-d09a6f5c From: <sip:526@bizzvoice.bizzdev.net>;tag=df437f177e021198o0 To: <sip:9069665262@bizzvoice.bizzdev.net>;tag=fcd35cf36c Call-ID: ebf34043-c02b625c@172.16.1.104 CSeq: 102 INVITE Contact: <sip:526@172.16.1.243:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.0.2998 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
pbx support Posted January 20, 2009 Report Share Posted January 20, 2009 Looks like you have a dial plan that strips '9' and sends rest of the digits. Quote Link to comment Share on other sites More sharing options...
mathy Posted January 20, 2009 Author Report Share Posted January 20, 2009 Looks like you have a dial plan that strips '9' and sends rest of the digits. yes i use in my dial plan the 9 to make a call to my isdn gateway but it's just a digits that is not use for the number , in the exemple that i use i want to call to 069 665262 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 20, 2009 Report Share Posted January 20, 2009 yes i use in my dial plan the 9 to make a call to my isdn gateway but it's just a digits that is not use for the number , in the exemple that i use i want to call to 069 665262 Does the gateway give you any log? Maybe there is some dial plan missing on the gateway. Or it does not trust the IP address of the PBX. On the PBX the setup seems to be fine. Quote Link to comment Share on other sites More sharing options...
mathy Posted January 20, 2009 Author Report Share Posted January 20, 2009 Does the gateway give you any log? Maybe there is some dial plan missing on the gateway. Or it does not trust the IP address of the PBX. On the PBX the setup seems to be fine. i have try to reloade an old configuration that working for my gateway and i have always the problem only inbound call work ... every time i try to make a outbound call from that gateway i receive the error forbidden for what cause can i receive a forbidden call ? Quote Link to comment Share on other sites More sharing options...
pbx support Posted January 20, 2009 Report Share Posted January 20, 2009 yes i use in my dial plan the 9 to make a call to my isdn gateway but it's just a digits that is not use for the number , in the exemple that i use i want to call to 069 665262 Let me ask this way.. is 069 665262 a valid number that your Patton gateway can reach? I mean, is it connected to some public network or a test setup? Quote Link to comment Share on other sites More sharing options...
mathy Posted January 21, 2009 Author Report Share Posted January 21, 2009 Let me ask this way.. is 069 665262 a valid number that your Patton gateway can reach? I mean, is it connected to some public network or a test setup? yes 069 665262 is a valid number that my isdn gateway can reach normaly ... this number is a common number in belgium and my gateway is connected on classic isdn network on belgium so , and bbefore one week , everythink worked fine on this number .... Quote Link to comment Share on other sites More sharing options...
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