[Freeswitch-users] Link2voip
Darren Wiebe
darren at aleph-com.net
Tue Jan 11 02:30:11 MSK 2011
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=link2voip2 SIP/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=link2voip2
SIP/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=link2voip2 SIP/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=link2voip2
SIP/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=link2voip1 SIP/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=link2voip1
SIP/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
> <mailto: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 <mailto:darren at aleph-com.net>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> <mailto: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
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
--
Darren Wiebe
Aleph Communications
--------------------
Phone: 1-877-702-2900
Fax: 1-866-274-4506
Email: darren at aleph-com.net
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