[Freeswitch-users] Link2voip
Michael Collins
msc at freeswitch.org
Tue Jan 11 04:12:36 MSK 2011
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 at 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 at CUSTOMERIP>;tag=as01e3f5de
> Call-ID: 0472a7de73e00fc578b12f6679b6dd9d at CUSTOMERIP
> To: <sip:14034883602 at 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 at 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 at CUSTOMERIP>;tag=as368d01dd
> To: <sip:14034883602 at sip.ca2.link2voip.com>
> Contact: <sip:7806283672 at CUSTOMERIP>
> Call-ID: 2306e7456006e56f62f3e67673965635 at 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 at CUSTOMERIP>;tag=as368d01dd
> To: <sip:14034883602 at sip.ca2.link2voip.com>
> Call-ID: 2306e7456006e56f62f3e67673965635 at 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 at 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 at 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 at 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 at CUSTOMERIP>;tag=as368d01dd
> To: <sip:14034883602 at sip.ca2.link2voip.com>;tag=N9mH5gDHvX0QD
> Call-ID: 2306e7456006e56f62f3e67673965635 at 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 at 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 at CUSTOMERIP) Ended
> 2011-01-10 16:22:36.430652 [NOTICE] switch_core_session.c:1275 Close
> Channel sof
> ia/external/7806283672 at CUSTOMERIP [CS_DESTROY]
> recv 367 bytes from udp/[66.51.110.210]:5060 at 23:22:36.696268:
> ------------------------------------------------------------------------
> ACK sip:gw+link2voip2 at 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 at CUSTOMERIP>;tag=as368d01dd
> Call-ID: 2306e7456006e56f62f3e67673965635 at CUSTOMERIP
> To: <sip:14034883602 at 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 at 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 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
> To: <sip:14034883602 at 66.51.110.210>
> Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
> CSeq: 7014814 INVITE
> Contact: <sip:ciscosip at 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 at 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 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
> To: <sip:14034883602 at 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 at 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 at 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 at 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 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
> To: <sip:14034883602 at 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 at 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 at 66.51.127.163) Ended
> 2011-01-10 16:22:37.805608 [NOTICE] switch_core_session.c:1275 Close
> Channel sof
> ia/external/7806283672 at 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 at 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 at 66.51.127.163>;tag=ZNcB4yyH3SD3e
> Call-ID: 59d610df-97b3-122e-9ca3-b3e1326322ad
> To: <sip:14034883602 at 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 at 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 at 66.51.127.163>;tag=17yv7m0rXBt8N
> To: <sip:14034883602 at 66.51.127.173>
> Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
> CSeq: 7014814 INVITE
> Contact: <sip:ciscosip at 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 at 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 at 66.51.127.163>;tag=17yv7m0rXBt8N
> To: <sip:14034883602 at 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 at 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 at 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 at 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 at 66.51.127.163>;tag=17yv7m0rXBt8N
> To: <sip:14034883602 at 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 at 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 at 66.51.127.163) Ended
> 2011-01-10 16:22:37.977477 [NOTICE] switch_core_session.c:1275 Close
> Channel sof
> ia/external/7806283672 at 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 at 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 at 66.51.127.163>;tag=17yv7m0rXBt8N
> Call-ID: 59f5177c-97b3-122e-9ca3-b3e1326322ad
> To: <sip:14034883602 at 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 at 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 at aleph-com.net
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Darren Wiebe
> Aleph Communications
> --------------------
> Phone: 1-877-702-2900
> Fax: 1-866-274-4506
> Email: darren at aleph-com.net
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110111/8a58b3e8/attachment-0001.html
More information about the FreeSWITCH-users
mailing list