[Freeswitch-users] Call Recovery when using TCP

Anthony Minessale anthony.minessale at gmail.com
Wed Jul 24 21:18:17 MSD 2013


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.
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
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>>> >
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>>>
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>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>
>
> --
> Anthony Minessale II
>
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>
> 
> 
>
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>
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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-- 
Anthony Minessale II

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AIM: anthm
MSN:anthony_minessale at hotmail.com
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