[Freeswitch-users] Call Recovery when using TCP
Anthony Minessale
anthony.minessale at gmail.com
Thu Jul 25 01:07:13 MSD 2013
maybe its because transport=tcp is not there in the req invite, I'll see if
I can force that into the url when applicable.
The via from the 200ok is the one that is preserved for the recover invite.
This is all a bit of trickery cos the sofia stack has no idea its doing a
recovery it thinks its a new outbound uac call.
We someday need to change the code to allow us to create a uas instance on
demand from the dialog data we have.
On Wed, Jul 24, 2013 at 3:49 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
> So this would make more sense. was scratching my head earlier… didn't make
> sense
>
> Here's what fs gets
>
> INVITE sip:+353877857933 at voice.plannet21.ie:5060 SIP/2.0
> Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK42DC2424
>
> Here's whats sent from recovering fs
>
> INVITE sip:+35319032109 at 198.19.255.1:5060 SIP/2.0
> Via: SIP/2.0/UDP 78.158.110.24;rport;branch=z9hG4bKXggpr14eHNNKH
> Route: <sip:+35319032109 at 198.19.255.1:35556;transport=tcp>
>
> Looking for differences the only one I can see is rport. I see on fisheye,
> for the update, I can see reference to sip_network_port, is it related, I
> have no experience with C. Does the initial invite need to have rport in
> the via?
> As a test I added rport but still the same behaviour, although 33333 is
> not real and I just statically set it outgoing from uac, but this causes
> subsequent packets to update and have the rport present
>
> FS in
>
> INVITE sip:+353877857933 at voice.plannet21.ie:5060 SIP/2.0
> Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK4306116E;rport=33333
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK4306116E;rport=13129
>
> FS Recover
>
> INVITE sip:+35319032109 at 198.19.255.1:5060 SIP/2.0
> Via: SIP/2.0/UDP 78.158.110.24;rport;branch=z9hG4bKKB0t2By0am6HB
> Route: <sip:+35319032109 at 198.19.255.1:13129;transport=tcp>
>
>
> On 24 Jul 2013, at 18:18, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
> Interesting. In the test I did when making the patch, the recover INVITE
> was tcp, it depends heavily on the VIA header on the original invite having
> TCP present in it.
>
> send 1281 bytes to tcp/[1.x.x.x]:5060 at 17:14:42.035251:
> ------------------------------------------------------------------------
> INVITE sip:mod_sofia at 1.x.x.x:5060 SIP/2.0
> Via: SIP/2.0/TCP 1.x.x.x;branch=z9hG4bKS0Q3SKXvB2Q9r;rport=50938
> Route: <sip:1004 at 1.x.x.x:50938;transport=tcp>
>
>
>
> On Wed, Jul 24, 2013 at 5:34 AM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>
>> Thanks Anthony, yes latest head worked.
>>
>> A leg still sends the recovery INVITE as UDP however now it has the route
>> header with no loose routing.
>>
>> Route: <sip:+35314611947 at 198.19.255.1:61767;transport=tcp>
>>
>> So I'm assuming the UAC uses this info to find session in UAC Table and
>> reestablishes the session, but as UDP, even though UAC specifically is told
>> to only use TCP.
>>
>> To test I put a firewall in the path and blocked UDP 5060 and the call
>> failed to recover, as expected.
>>
>> So I think for this to work UDP will still need to be open on the UAC to
>> recover the call.
>>
>> As you said it won't work for all UACs or all situations but its a
>> workable solution.
>>
>> For reference this works when UAC is either Cisco or Dialogic.
>>
>>
>> On 23 Jul 2013, at 19:53, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>>
>> Try latest head, no promises on every endpoint.
>> P.S. try Jira next time.
>>
>>
>>
>> On Tue, Jul 23, 2013 at 11:24 AM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>
>>> I have pasted up the logs from a test call A leg UDP, B leg TCP that
>>> recovered ok
>>>
>>> initial call - fs crashed
>>> http://pastebin.com/0xe0QyFC
>>>
>>> recovered call
>>> http://pastebin.com/ByjJ4nhf
>>>
>>>
>>> On 23 Jul 2013, at 16:39, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Is the B-leg the same call, or a new call?
>>>
>>>
>>>
>>> On 23 July 2013 16:31, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>>>
>>>> Thanks Brian,
>>>>
>>>> Initially I though the same and looked for something to migrate the tcp
>>>> session, tcpcp and sockmi, but no joy.
>>>>
>>>> I was doing some more testing and noticed that if the B leg was TCP and
>>>> the A leg UDP the call recovered.
>>>>
>>>> So I though my earlier assumption about TCP connection dropping was
>>>> wrong as it seems the B leg reestablishes the session on the recovering
>>>> server.
>>>> The issue just seems to be the recovery of the A leg. FS always sends
>>>> the A leg recovery INVITE as UDP. Even if original call was TCP. If I could
>>>> force it to use TCP I believe it would recover the call.
>>>>
>>>> Below is a call with A leg as UDP and B leg as TCP thats recovers fine.
>>>>
>>>> 2013-07-23 16:19:46.419717 [NOTICE] switch_channel.c:1030 New Channel
>>>> sofia/private/+35319032109 at 198
>>>> 2013-07-23 16:19:46.419717 [NOTICE] switch_channel.c:1028 Rename
>>>> Channel sofia/private/+35319032109@
>>>> 2013-07-23 16:19:46.419717 [NOTICE] switch_core_sqldb.c:2744
>>>> Resurrecting fallen channel sofia/priva
>>>> 2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1030 New Channel
>>>> sofia/internal/0877857933 at 10.1
>>>> 2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1028 Rename
>>>> Channel sofia/internal/0877857933 at 1
>>>> 2013-07-23 16:19:46.439717 [NOTICE] switch_core_sqldb.c:2744
>>>> Resurrecting fallen channel sofia/inter
>>>> send 1110 bytes to udp/[10.101.23.203]:5060 at 15:19:46.459562:
>>>>
>>>> ------------------------------------------------------------------------
>>>> INVITE sip:0877857933 at 10.101.23.203:5060 SIP/2.0
>>>> Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS
>>>> Route: <sip:0877857933 at 10.101.23.203:5060;lr>
>>>> Max-Forwards: 70
>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>> To: <sip:0877857933 at 10.101.23.203
>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>> CSeq: 46960329 INVITE
>>>> Contact: <sip:0877857933 at 10.101.23.110:5060>
>>>> User-Agent: LAB - SBC
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, precondition, path, replaces
>>>> Allow-Events: talk, hold, conference, refer
>>>> Privacy: none
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 246
>>>> X-FS-Support: update_display,send_info
>>>> P-Asserted-Identity: "ae019032109" <sip:ae019032109 at 10.101.23.203>
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1374564092 1374564094 IN IP4 10.101.24.110
>>>> s=FreeSWITCH
>>>> c=IN IP4 10.101.24.110
>>>> t=0 0
>>>> m=audio 28694 RTP/AVP 8 101 13
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=rtpmap:13 CN/8000
>>>> a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 398 bytes from udp/[10.101.23.203]:5060 at 15:19:46.463116:
>>>>
>>>> ------------------------------------------------------------------------
>>>> SIP/2.0 100 Trying
>>>> Via: SIP/2.0/UDP
>>>> 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>> To: <sip:0877857933 at 10.101.23.203
>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>> CSeq: 46960329 INVITE
>>>> Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>>>> Content-Length: 0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:46.463966:
>>>>
>>>> ------------------------------------------------------------------------
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>> 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>> To: <sip:0877857933 at 10.101.23.203
>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>> Contact: <sip:0877857933 at 10.101.23.203:5060>
>>>> CSeq: 46960329 INVITE
>>>> Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>>>> Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE,
>>>> NOTIFY, INFO, REFER, UPDATE
>>>> Supported: path, replaces, timer, tdialog
>>>> Require: timer
>>>> Session-Expires: 1800;refresher=uas
>>>> Accept: application/sdp, application/dtmf-relay, text/plain
>>>> Content-Type: application/sdp
>>>> Content-Length: 239
>>>>
>>>> v=0
>>>> o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>>>> s=Dialogic-SIP
>>>> c=IN IP4 10.101.24.203
>>>> t=0 0
>>>> m=audio 8332 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:47.101507:
>>>>
>>>> ------------------------------------------------------------------------
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>> 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>> To: <sip:0877857933 at 10.101.23.203
>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>> Contact: <sip:0877857933 at 10.101.23.203:5060>
>>>> CSeq: 46960329 INVITE
>>>> Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>>>> Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE,
>>>> NOTIFY, INFO, REFER, UPDATE
>>>> Supported: path, replaces, timer, tdialog
>>>> Require: timer
>>>> Session-Expires: 1800;refresher=uas
>>>> Accept: application/sdp, application/dtmf-relay, text/plain
>>>> Content-Type: application/sdp
>>>> Content-Length: 239
>>>>
>>>> v=0
>>>> o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>>>> s=Dialogic-SIP
>>>> c=IN IP4 10.101.24.203
>>>> t=0 0
>>>> m=audio 8332 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 961 bytes from udp/[10.101.23.203]:5060 at 15:19:48.070746:
>>>>
>>>> ------------------------------------------------------------------------
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/UDP
>>>> 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS;received=10.101.23.110
>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>> To: <sip:0877857933 at 10.101.23.203
>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>> Contact: <sip:0877857933 at 10.101.23.203:5060>
>>>> CSeq: 46960329 INVITE
>>>> Server: Dialogic-SIP/10.5.3.372 IMG2_DEG 2
>>>> Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE,
>>>> NOTIFY, INFO, REFER, UPDATE
>>>> Supported: path, replaces, timer, tdialog
>>>> Require: timer
>>>> Session-Expires: 1800;refresher=uas
>>>> Accept: application/sdp, application/dtmf-relay, text/plain
>>>> Content-Type: application/sdp
>>>> Content-Length: 239
>>>>
>>>> v=0
>>>> o=Dialogic_SDP 963778 1 IN IP4 10.101.23.203
>>>> s=Dialogic-SIP
>>>> c=IN IP4 10.101.24.203
>>>> t=0 0
>>>> m=audio 8332 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=silenceSupp:off - - - -
>>>> a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>> send 1175 bytes to tcp/[198.19.255.1]:5060 at 15:19:49.473346:
>>>>
>>>> ------------------------------------------------------------------------
>>>> INVITE sip:+35319032109 at 198.19.255.1:5060;transport=tcp SIP/2.0
>>>> Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q
>>>> Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKZ21tSFUQvy38c
>>>> Max-Forwards: 69
>>>> From: "+353877857933" <sip:+353877857933 at 78.158.110.24
>>>> >;tag=BK8QvjBeDHXtN
>>>> To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>>>> Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>>>> CSeq: 46960329 INVITE
>>>> Contact: <sip:+35319032109 at 78.158.110.24:5060;transport=tcp>
>>>> User-Agent: PlanNet21 Communications - SBC
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY
>>>> Supported: timer, precondition, path, replaces
>>>> Allow-Events: talk, hold, conference, refer
>>>> Privacy: none
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session
>>>> Content-Length: 246
>>>> X-FS-Support: update_display,send_info
>>>> P-Asserted-Identity: "+353877857933" <
>>>> sip:+353877857933 at 78.158.110.24>
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1374569678 1374569680 IN IP4 78.158.110.24
>>>> s=FreeSWITCH
>>>> c=IN IP4 78.158.110.24
>>>> t=0 0
>>>> m=audio 23108 RTP/AVP 8 101 13
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=rtpmap:13 CN/8000
>>>> a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 484 bytes from tcp/[198.19.255.1]:5060 at 15:19:49.536636:
>>>>
>>>> ------------------------------------------------------------------------
>>>> SIP/2.0 100 Trying
>>>> Via: SIP/2.0/TCP
>>>> 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q,SIP/2.0/TCP
>>>> 78.158.110.24;rport;
>>>> From: "+353877857933" <sip:+353877857933 at 78.158.110.24
>>>> >;tag=BK8QvjBeDHXtN
>>>> To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>>>> Date: Tue, 23 Jul 2013 15:19:49 GMT
>>>> Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>>>> CSeq: 46960329 INVITE
>>>> Allow-Events: telephone-event
>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>> Content-Length: 0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> recv 1073 bytes from tcp/[198.19.255.1]:5060 at 15:19:49.601460:
>>>>
>>>> ------------------------------------------------------------------------
>>>> SIP/2.0 200 OK
>>>> Via: SIP/2.0/TCP
>>>> 78.158.110.24;rport;branch=z9hG4bKBNF06yNpU187Q,SIP/2.0/TCP
>>>> 78.158.110.24;rport;
>>>> From: "+353877857933" <sip:+353877857933 at 78.158.110.24
>>>> >;tag=BK8QvjBeDHXtN
>>>> To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>>>> Date: Tue, 23 Jul 2013 15:19:49 GMT
>>>> Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>>>> CSeq: 46960329 INVITE
>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>> Allow-Events: telephone-event
>>>> Remote-Party-ID: "Ian McGrath" <sip:+1947 at 198.19.255.1
>>>> >;party=called;screen=yes;privacy=off
>>>> Contact: <sip:+35319032109 at 198.19.255.1:5060;transport=tcp>
>>>> Supported: replaces
>>>> Supported: sdp-anat
>>>> Server: Cisco-SIPGateway/IOS-12.x
>>>> Supported: timer
>>>> Content-Type: application/sdp
>>>> Content-Length: 247
>>>>
>>>> v=0
>>>> o=CiscoSystemsSIP-GW-UserAgent 3902 8614 IN IP4 198.19.255.1
>>>> s=SIP Call
>>>> c=IN IP4 198.19.255.1
>>>> t=0 0
>>>> m=audio 19050 RTP/AVP 8 101
>>>> c=IN IP4 198.19.255.1
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> ------------------------------------------------------------------------
>>>> send 440 bytes to tcp/[198.19.255.1]:5060 at 15:19:49.603288:
>>>>
>>>> ------------------------------------------------------------------------
>>>> ACK sip:+35319032109 at 198.19.255.1:5060;transport=tcp SIP/2.0
>>>> Via: SIP/2.0/TCP 78.158.110.24;rport;branch=z9hG4bKcy8r8S6SraZtK
>>>> Max-Forwards: 70
>>>> From: "+353877857933" <sip:+353877857933 at 78.158.110.24
>>>> >;tag=BK8QvjBeDHXtN
>>>> To: <sip:+35319032109 at 198.19.255.1;transport=tcp>;tag=397C7E04-26CA
>>>> Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
>>>> CSeq: 46960329 ACK
>>>> Contact: <sip:+35319032109 at 78.158.110.24:5060;transport=tcp>
>>>> Content-Length: 0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>> send 466 bytes to udp/[10.101.23.203]:5060 at 15:19:49.606747:
>>>>
>>>> ------------------------------------------------------------------------
>>>> ACK sip:0877857933 at 10.101.23.203:5060 SIP/2.0
>>>> Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK43B5D34y3cX2m
>>>> Route: <sip:0877857933 at 10.101.23.203:5060;lr>
>>>> Max-Forwards: 70
>>>> From: <sip:ae019032109 at 10.101.23.110:5060>;tag=B8BUB47FmmerF
>>>> To: <sip:0877857933 at 10.101.23.203
>>>> >;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
>>>> Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
>>>> CSeq: 46960329 ACK
>>>> Contact: <sip:0877857933 at 10.101.23.110:5060>
>>>> Content-Length: 0
>>>>
>>>>
>>>> ------------------------------------------------------------------------
>>>>
>>>>
>>>>
>>>>
>>>> On 23 Jul 2013, at 14:47, Brian West <brian at freeswitch.org> wrote:
>>>>
>>>> > You can't do call recovery on TCP at the moment, You have no way to
>>>> re-establish the TCP connections once FreeSWITCH goes down.
>>>> >
>>>> > /b
>>>> >
>>>> > Em Jul 23, 2013, às 8:01 AM, Anthony McGarry <agtmcgarry at gmail.com>
>>>> escreveu:
>>>> >
>>>> >> Anyone using TCP in this scenario? Cannot find what I'm missing.
>>>> >
>>>> >
>>>> >
>>>> _________________________________________________________________________
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>>>> >
>>>> >
>>>> >
>>>> >
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>>>>
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>>>
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>>>
>>>
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>>>
>>>
>>>
>>>
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>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> 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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> 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
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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