How many different DIDs do you have for this user? Just one? If so can you not map the user to a specific DID? In any case, throw a "info" app in your public dialplan and call the DID. You'll see there's all sorts of variables you can use for routing if you need to. -MC On Mon, Jan 10, 2011 at 11:30 PM, Darren Wiebe <darren@aleph-com.net> wrote:
Yeah, I bet. :) The outgoing call problem was my fault, I had an incorrect piece of dialplan. Here's the trace on an incoming call. I'm trying to get it to come to a particular DID in the public context instead of this.
2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331 Processing CLID NAME <7806283672>->sipuser in context public
What am I missing?
Here's the relevant provider entry from the external sip profile
<include> <gateway name="link2voip1"> <param name="username" value="sipuser"/> <param name="password" value="password"/> <param name="proxy" value="sip.ca1.link2voip.com"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry-seconds" value="30"/> </gateway> </include>
Sip Trace:
From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as01e3f5de Call-ID: 0472a7de73e00fc578b12f6679b6dd9d@CUSTOMERIP To: <sip:14034883602@sip.ca2.link2voip.com>;tag=m0Ur3NvDZma5H CSeq: 103 ACK Content-Length: 0
------------------------------------------------------------------------ recv 942 bytes from udp/[66.51.110.210]:5060 at 23:22:36.415027: ------------------------------------------------------------------------ INVITE sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2SIP/2. 0 Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd> Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0 Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060 From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as368d01dd To: <sip:14034883602@sip.ca2.link2voip.com> Contact: <sip:7806283672@CUSTOMERIP> Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Mon, 10 Jan 2011 23:22:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 235
v=0 o=root 12790 12791 IN IP4 CUSTOMERIP s=session c=IN IP4 66.51.110.210 t=0 0 m=audio 14648 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=nortpproxy:yes ------------------------------------------------------------------------ send 503 bytes to udp/[66.51.110.210]:5060 at 23:22:36.415027: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0 Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060 Record-Route: <sip:66.51.110.210;lr;ftag=as368d01dd> From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as368d01dd To: <sip:14034883602@sip.ca2.link2voip.com> Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP CSeq: 103 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06 00 Content-Length: 0
------------------------------------------------------------------------ 2011-01-10 16:22:36.415027 [NOTICE] switch_channel.c:784 New Channel sofia/exter nal/7806283672@CUSTOMERIP [c58ce64a-c99e-4a20-94a9-680e488f8fab] 2011-01-10 16:22:36.430652 [INFO] mod_dialplan_xml.c:331 Processing CLID NAME <7806283672>->sipuser in context public 2011-01-10 16:22:36.430652 [NOTICE] switch_core_state_machine.c:189 sofia/extern al/7806283672@CUSTOMERIP has executed the last dialplan instruction, hangin g up. 2011-01-10 16:22:36.430652 [NOTICE] switch_core_state_machine.c:191 Hangup sofia /external/7806283672@CUSTOMERIP [CS_EXECUTE] [NORMAL_CLEARING] send 818 bytes to udp/[66.51.110.210]:5060 at 23:22:36.430652: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0 Via: SIP/2.0/UDP CUSTOMERIP:5060;branch=z9hG4bK0bdd0021;rport=5060 From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as368d01dd To: <sip:14034883602@sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP CSeq: 103 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06 00 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE FER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "sipuser" <sip:sipuser@192.168.35.1> ;party=calling;privacy=o ff;screen=no
------------------------------------------------------------------------ 2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1273 Session 53 (sofia /external/7806283672@CUSTOMERIP) Ended 2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1275 Close Channel sof ia/external/7806283672@CUSTOMERIP [CS_DESTROY] recv 367 bytes from udp/[66.51.110.210]:5060 at 23:22:36.696268: ------------------------------------------------------------------------ ACK sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2SIP/2.0 Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bK05fe.c1fce6a3.0 From: "CLID NAME" <sip:7806283672@CUSTOMERIP>;tag=as368d01dd Call-ID: 2306e7456006e56f62f3e67673965635@CUSTOMERIP To: <sip:14034883602@sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD CSeq: 103 ACK Content-Length: 0
------------------------------------------------------------------------ recv 1230 bytes from udp/[66.51.110.210]:5060 at 23:22:37.789983: ------------------------------------------------------------------------ INVITE sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2SIP/2. 0 Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e> Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0 Via: SIP/2.0/UDP 66.51.127.163:5080 ;rport=5080;branch=z9hG4bK0D66gH7m2614F Max-Forwards: 66 From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=ZNcB4yyH3SD3e To: <sip:14034883602@66.51.110.210> Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad CSeq: 7014814 INVITE Contact: <sip:ciscosip@66.51.127.163:5080> User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 323 Remote-Party-ID: "CLID NAME" <sip:7806283672@66.51.127.163> ;screen=yes;pri vacy=off
v=0 o=CiscoSystemsSIP-GW-UserAgent 7458343035650957072 338127983379325658 IN IP4 66.51.127.163 s=SIP Call c=IN IP4 66.51.110.210 t=0 0 m=audio 14650 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=nortpproxy:yes ------------------------------------------------------------------------ send 494 bytes to udp/[66.51.110.210]:5060 at 23:22:37.789983: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0 Via: SIP/2.0/UDP 66.51.127.163:5080 ;rport=5080;branch=z9hG4bK0D66gH7m2614F Record-Route: <sip:66.51.110.210;lr;ftag=ZNcB4yyH3SD3e> From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=ZNcB4yyH3SD3e To: <sip:14034883602@66.51.110.210> Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad CSeq: 7014814 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06 00 Content-Length: 0
------------------------------------------------------------------------ 2011-01-10 16:22:37.789983 [NOTICE] switch_channel.c:784 New Channel sofia/exter nal/7806283672@66.51.127.163 [86b78ecd-469f-4a1c-9fe5-692a5941ff37] 2011-01-10 16:22:37.805608 [INFO] mod_dialplan_xml.c:331 Processing CLID NAME <7806283672>->sipuser in context public 2011-01-10 16:22:37.805608 [NOTICE] switch_core_state_machine.c:189 sofia/extern al/7806283672@66.51.127.163 has executed the last dialplan instruction, hanging up. 2011-01-10 16:22:37.805608 [NOTICE] switch_core_state_machine.c:191 Hangup sofia /external/7806283672@66.51.127.163 [CS_EXECUTE] [NORMAL_CLEARING] send 806 bytes to udp/[66.51.110.210]:5060 at 23:22:37.805608: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0 Via: SIP/2.0/UDP 66.51.127.163:5080 ;rport=5080;branch=z9hG4bK0D66gH7m2614F From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=ZNcB4yyH3SD3e To: <sip:14034883602@66.51.110.210>;tag=pjea7BymS6paS Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad CSeq: 7014814 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06 00 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE FER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "sipuser" <sip:sipuser@192.168.35.1> ;party=calling;privacy=o ff;screen=no
------------------------------------------------------------------------ 2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1273 Session 54 (sofia /external/7806283672@66.51.127.163) Ended 2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1275 Close Channel sof ia/external/7806283672@66.51.127.163 [CS_DESTROY] recv 352 bytes from udp/[66.51.110.210]:5060 at 23:22:37.914979: ------------------------------------------------------------------------ ACK sip:gw+link2voip2@192.168.35.1:5080;transport=udp;gw=link2voip2SIP/2.0 Via: SIP/2.0/UDP 66.51.110.210;branch=z9hG4bKfc48.d0a188f4.0 From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=ZNcB4yyH3SD3e Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad To: <sip:14034883602@66.51.110.210>;tag=pjea7BymS6paS CSeq: 7014814 ACK Content-Length: 0
------------------------------------------------------------------------ recv 1230 bytes from udp/[66.51.127.173]:5060 at 23:22:37.977477: ------------------------------------------------------------------------ INVITE sip:gw+link2voip1@192.168.35.1:5080;transport=udp;gw=link2voip1SIP/2. 0 Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N> Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0 Via: SIP/2.0/UDP 66.51.127.163:5080 ;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ Max-Forwards: 66 From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=17yv7m0rXBt8N To: <sip:14034883602@66.51.127.173> Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad CSeq: 7014814 INVITE Contact: <sip:ciscosip@66.51.127.163:5080> User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 323 Remote-Party-ID: "CLID NAME" <sip:7806283672@66.51.127.163> ;screen=yes;pri vacy=off
v=0 o=CiscoSystemsSIP-GW-UserAgent 4919904105816548778 787843793424096957 IN IP4 66.51.127.163 s=SIP Call c=IN IP4 66.51.127.173 t=0 0 m=audio 15488 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=nortpproxy:yes ------------------------------------------------------------------------ send 494 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0 Via: SIP/2.0/UDP 66.51.127.163:5080 ;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ Record-Route: <sip:66.51.127.173;lr;ftag=17yv7m0rXBt8N> From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=17yv7m0rXBt8N To: <sip:14034883602@66.51.127.173> Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad CSeq: 7014814 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06 00 Content-Length: 0
------------------------------------------------------------------------ 2011-01-10 16:22:37.977477 [NOTICE] switch_channel.c:784 New Channel sofia/exter nal/7806283672@66.51.127.163 [86c939d1-4de0-4a46-9203-518e0d6f7bc5] 2011-01-10 16:22:37.977477 [INFO] mod_dialplan_xml.c:331 Processing CLID NAME <7806283672>->sipuser in context public 2011-01-10 16:22:37.977477 [NOTICE] switch_core_state_machine.c:189 sofia/extern al/7806283672@66.51.127.163 has executed the last dialplan instruction, hanging up. 2011-01-10 16:22:37.977477 [NOTICE] switch_core_state_machine.c:191 Hangup sofia /external/7806283672@66.51.127.163 [CS_EXECUTE] [NORMAL_CLEARING] send 806 bytes to udp/[66.51.127.173]:5060 at 23:22:37.977477: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0 Via: SIP/2.0/UDP 66.51.127.163:5080 ;rport=5080;branch=z9hG4bK2Zrrm78UvreaQ From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=17yv7m0rXBt8N To: <sip:14034883602@66.51.127.173>;tag=QU7286erpFDXm Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad CSeq: 7014814 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-650393f 2010-12-29 13-15-14 -06 00 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE FER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "sipuser" <sip:sipuser@192.168.35.1> ;party=calling;privacy=o ff;screen=no
------------------------------------------------------------------------ 2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1273 Session 55 (sofia /external/7806283672@66.51.127.163) Ended 2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1275 Close Channel sof ia/external/7806283672@66.51.127.163 [CS_DESTROY] recv 352 bytes from udp/[66.51.127.173]:5060 at 23:22:38.071224: ------------------------------------------------------------------------ ACK sip:gw+link2voip1@192.168.35.1:5080;transport=udp;gw=link2voip1SIP/2.0 Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bK00fe.183ad5e2.0 From: "CLID NAME" <sip:7806283672@66.51.127.163>;tag=17yv7m0rXBt8N Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad To: <sip:14034883602@66.51.127.173>;tag=QU7286erpFDXm CSeq: 7014814 ACK Content-Length: 0
------------------------------------------------------------------------
On 10/01/2011 2:52 PM, Michael Collins wrote:
Or just give us your credentials and we'll "test it thoroughly" for you. :)
-MC
On Mon, Jan 10, 2011 at 1:24 PM, Brian West <brian@freeswitch.org> wrote:
can you put up a sip trace or something so we can help guide you?
/b
On Jan 10, 2011, at 2:26 PM, Darren Wiebe wrote:
Good Afternoon,
I'm trying to get my freeswitch box talking to Link2voip. Does anybody have sample XML files for them?
-- Darren Wiebe Aleph Communications -------------------- Phone: 1-877-702-2900 Fax: 1-866-274-4506 Email: darren@aleph-com.net
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-- Darren Wiebe Aleph Communications -------------------- Phone: 1-877-702-2900 Fax: 1-866-274-4506 Email: darren@aleph-com.net
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