[Freeswitch-users] Link2voip

Darren Wiebe darren at aleph-com.net
Tue Jan 11 04:42:38 MSK 2011


this user we've only got 6.  Thanks, that's a good idea.  I'll try it.

Darren Wiebe

On 11-01-10 06:12 PM, Michael Collins wrote:
> 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 
> <mailto: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
>     <http://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 <mailto: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 <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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 <mailto: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 <mailto: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
>     <mailto: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
>     <mailto: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
>     <mailto: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>
>>         >
>>         >
>>         > _______________________________________________
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>>         >
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>>
>>
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>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>>
>>     _______________________________________________
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>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     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>
>
>
>     _______________________________________________
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>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
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