Fisher Networks Posted February 5, 2008 Report Share Posted February 5, 2008 Hey, I started our settings from scratch and appear to be having issues getting the ability to make outbound calls (and inbound, really). We use bandwidth.com SIP trunking and with their IP in the domain field, the same IP in the outbound gateway and our number as the DID in a SIP Gateway we can't get out. The only log I get is the following. [5] 2008/02/04 17:52:21: Identify trunk (domain name match) 1 [5] 2008/02/04 17:52:21: Dialplan External: Match 206xxxxxxx@domain.com to <sip:2067690931@216.82.x.x;user=phone> on trunk BWGW1 It does not appear to be getting out at all at this point. The phone says this: [5]4/2/2008 18:54:08: Dialog 7/6 going to trying [5]4/2/2008 18:54:08: Dialog 7/6 going to early [5]4/2/2008 18:54:08: Dialog 7/6 going to terminated [5]4/2/2008 18:54:08: timeout::callback: Registering with timeout of 0 ms [5]4/2/2008 18:54:08: timeout::callback: Registering with timeout of 0 ms [2]4/2/2008 18:54:36: Registered at registrar as 501@domain.com [0]4/2/2008 18:54:37: Webclient: Could not find host snom360.htm:80 [0]4/2/2008 18:54:37: Webclient: Could not find host snom360-00041323C1DB.htm:80 I think the [5]s are the only important logs there. Anyway, the dial plan is a *, supposedly to accept any input and pass it along. Any ideas what I am missing? I simplified the internal setup, so there are no hunt groups, attendants or anything but a single, registered extension. Also, what ports do I need to make sure are open? I know of 5060 and 5061. Thanks for your help! Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 5, 2008 Report Share Posted February 5, 2008 [5] 2008/02/04 17:52:21: Identify trunk (domain name match) 1 [5] 2008/02/04 17:52:21: Dialplan External: Match 206xxxxxxx@domain.com to <sip:2067690931@216.82.x.x;user=phone> on trunk BWGW1 I would set an outbound proxy on the trunk. Also, if you change the IP configuration, you should restart the service (just to be sure). If it does not help, turn SIP logging on and see what the PBX tries to send out. Quote Link to comment Share on other sites More sharing options...
Fisher Networks Posted February 5, 2008 Author Report Share Posted February 5, 2008 The outbound proxy should be the Trunking service IP or my gateway? "SIP Logging" was on, but under that I enabled every type of SIP logging. However, the same (lack of) errors appear. However the SIP trace on the phone yields more info: Sent to tls:10.0.1.3:5061 at 5/2/2008 01:00:28:050 (1254 bytes): INVITE sip:206xxxxxxx@domain.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-dktogirkbrph;rport From: "User" <sip:501@domain.com>;tag=9yc0cxo949 To: <sip:206xxxxxxx@domain.com;user=phone> Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:501@10.0.1.109:4967;transport=tls;line=uexh66e7>;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/6.2.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 471 v=0 o=root 1927523315 1927523315 IN IP4 10.0.1.109 s=call c=IN IP4 10.0.1.109 t=0 0 m=audio 51080 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XUR1mT6ftFtPm4go7v37e/vtfmrfHzWmQ4OxofLW 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:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv -------------------------------------------------------------------------------- Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:140 (323 bytes): SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-dktogirkbrph;rport=4967 From: "Tanya " <sip:501@domain.com>;tag=9yc0cxo949 To: <sip:206xxxxxxx@domain.com;user=phone>;tag=0e7ad1ccc9 Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB CSeq: 1 INVITE Content-Length: 0 -------------------------------------------------------------------------------- Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:150 (685 bytes): NOTIFY sip:501@10.0.1.109:4967;transport=tls;line=uexh66e7 SIP/2.0 Via: SIP/2.0/TLS 10.0.1.3:5061;branch=z9hG4bK-0d5b74cb3e5a5bdc447bbbe02c21b327;rport From: <sip:501@domain.com;user=phone>;tag=f7b6300e03 To: <sip:501@domain.com>;tag=oyj484iitx Call-ID: 3c267009c832-wjx5tt2ysc3u@snom360-00041323C1DB CSeq: 19600 NOTIFY Max-Forwards: 70 Contact: <sip:10.0.1.3:5061;transport=tls> Event: dialog Subscription-State: active;expires=187 Content-Type: application/dialog-info+xml Content-Length: 158 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="36" state="full" entity="sip:501@domain.com"></dialog-info> <snip> Sent to tls:10.0.1.3:5061 at 5/2/2008 01:00:28:380 (318 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.1.3:5061;branch=z9hG4bK-24329e1bc69a2121a722c9dd45bdb3e5;rport=5061 From: <sip:501@domain.com;user=phone>;tag=f7b6300e03 To: <sip:501@domain.com>;tag=oyj484iitx Call-ID: 3c267009c832-wjx5tt2ysc3u@snom360-00041323C1DB CSeq: 19601 NOTIFY Content-Length: 0 -------------------------------------------------------------------------------- Received from tls:10.0.1.3:5061 at 5/2/2008 01:00:28:390 (402 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.1.109:4967;branch=z9hG4bK-wweqrjre6ksm;rport=4967 From: "Tanya " <sip:501@domain.com>;tag=9yc0cxo949 To: <sip:206xxxxxxx@domain.com;user=phone>;tag=0e7ad1ccc9 Call-ID: 3c272286c0df-b8iwzmbz2a5g@snom360-00041323C1DB CSeq: 2 PRACK Contact: <sip:501@10.0.1.3:5061;transport=tls> User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Fisher Networks Posted February 5, 2008 Author Report Share Posted February 5, 2008 I set the outbound proxy to the server's internal NIC (that the clients all funnel through) and it said "Connection refused". Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 5, 2008 Report Share Posted February 5, 2008 I set the outbound proxy to the server's internal NIC (that the clients all funnel through) and it said "Connection refused". Set the outbound proxy of the trunk to your service providers address. The outbound proxy on the phone should point to the PBX address. And BTW you better upgrade those phones to 7.1.30, otherwise you will experience problems with SRTP. You better include the logs from the PBX, not from the phone - then it is easier to see the PBX perspective! Quote Link to comment Share on other sites More sharing options...
Fisher Networks Posted February 5, 2008 Author Report Share Posted February 5, 2008 This is all I get when I issue a call: [5] 2008/02/05 09:09:29: Dialplan External: Match 206xxxxxxx@domain.com to <sip:206xxxxxxx@216.82.xxx.xxx;user=phone> on trunk BWGW1 The phone says "Forbidden". It doesn't appear the PBX is doing anything. Quote Link to comment Share on other sites More sharing options...
Fisher Networks Posted February 7, 2008 Author Report Share Posted February 7, 2008 Now we're getting somewhere. The logs now appear to be dial plan issues? I'm not too sure what the error is here besides the e164 message. [7]2008/02/07 13:43:17: SIP Tx tls:10.0.1.109:3337:SIP/2.0 404 Number not in e164 format, example +12125551212 Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-d67yw065242a;rport=3337 From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96 Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB CSeq: 1 INVITE Contact: <sip:501@10.0.1.3:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 [7] 2008/02/07 13:43:17: SIP Rx tls:10.0.1.109:3337: PRACK sip:501@10.0.1.3:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-8mic0lb2vaj9;rport From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96 Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:501@10.0.1.109:3337;transport=tls;line=uexh66e7>;flow-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 [7] 2008/02/07 13:43:17: SIP Tx tls:10.0.1.109:3337: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-8mic0lb2vaj9;rport=3337 From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96 Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB CSeq: 2 PRACK Contact: <sip:501@10.0.1.3:5061;transport=tls> User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Length: 0 [7] 2008/02/07 13:43:18: SIP Rx tls:10.0.1.109:3337: ACK sip:1206xxxxxxx@domain.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.0.1.109:3337;branch=z9hG4bK-d67yw065242a;rport From: "Tanya " <sip:501@domain.com>;tag=jkywkd00oi To: <sip:1206xxxxxxx@domain.com;user=phone>;tag=947f934d96 Call-ID: 3c2a86c883d6-brzvy5qccwjp@snom360-00041323C1DB CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:501@10.0.1.109:3337;transport=tls;line=uexh66e7>;flow-id=1 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Fisher Networks Posted February 8, 2008 Author Report Share Posted February 8, 2008 I just tested outgoing and got another number to ring. I think this thread is done. Thanks! Quote Link to comment Share on other sites More sharing options...
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