Hi Anthony (McGarry),If possible, could you spend a few minutes summarizing this thread and create a JIRA ticket? That way it won't get forgotten about, and helps us keep track of things better. [1]Also for future ref, try and use the FS pastebin for pasting logs otherwise it makes threads a bit hard to read [2]Thank you!
Cal_________________________________________________________________________On Thu, Jul 25, 2013 at 12:20 AM, Anthony McGarry <agtmcgarry@gmail.com> wrote:
Still the same with auth-calls off. Appreciate the time you've spent on this.If you have time to put it in the lab then sure, if not no worries.On 24 Jul 2013, at 23:50, Anthony Minessale <anthony.minessale@gmail.com> wrote:is it any different if you turn off auth-calls?There must be something subtle about it that is not obvious.I may need to have it labbed up where i can actually mess with the code to figure it out._________________________________________________________________________On Wed, Jul 24, 2013 at 5:29 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote:
http://pastebin.freeswitch.org/21235On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale@gmail.com> wrote:can you do that again with this extra command right before recover is executed?sofia loglevel all 9maybe its failing on tcp and reverting to udp_________________________________________________________________________On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry@gmail.com> wrote:ok so I added INVITE sip:+353877857933@voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.Uploaded logs from both serversbefore crashafter crashOn 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale@gmail.com> wrote:I think I would need to see the whole sip trace of the original invite and the recover leg.The only diff I see between mine and yours is mine is using FS on both sides and its including transport=tcp_________________________________________________________________________On Wed, Jul 24, 2013 at 4:07 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
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@gmail.com> wrote:
So this would make more sense. was scratching my head earlier… didn't make senseHere's what fs getsINVITE sip:+353877857933@voice.plannet21.ie:5060 SIP/2.0Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK42DC2424
Here's whats sent from recovering fsINVITE sip:+35319032109@198.19.255.1:5060 SIP/2.0Via: SIP/2.0/UDP 78.158.110.24;rport;branch=z9hG4bKXggpr14eHNNKHLooking 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 inINVITE sip:+353877857933@voice.plannet21.ie:5060 SIP/2.0Via: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK4306116E;rport=33333SIP/2.0 100 TryingVia: SIP/2.0/TCP 198.19.255.1:5060;branch=z9hG4bK4306116E;rport=13129FS RecoverINVITE sip:+35319032109@198.19.255.1:5060 SIP/2.0Via: SIP/2.0/UDP 78.158.110.24;rport;branch=z9hG4bKKB0t2By0am6HBOn 24 Jul 2013, at 18:18, Anthony Minessale <anthony.minessale@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@1.x.x.x:5060 SIP/2.0Via: SIP/2.0/TCP 1.x.x.x;branch=z9hG4bKS0Q3SKXvB2Q9r;rport=50938Route: <sip:1004@1.x.x.x:50938;transport=tcp>_________________________________________________________________________On Wed, Jul 24, 2013 at 5:34 AM, Anthony McGarry <agtmcgarry@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.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@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@gmail.com> wrote:
I have pasted up the logs from a test call A leg UDP, B leg TCP that recovered okinitial call - fs crashedrecovered callOn 23 Jul 2013, at 16:39, Steven Ayre <steveayre@gmail.com> wrote:Is the B-leg the same call, or a new call?_________________________________________________________________________On 23 July 2013 16:31, Anthony McGarry <agtmcgarry@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@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@10.1
2013-07-23 16:19:46.439717 [NOTICE] switch_channel.c:1028 Rename Channel sofia/internal/0877857933@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@10.101.23.203:5060 SIP/2.0
Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK3tjcc8KU636FS
Route: <sip:0877857933@10.101.23.203:5060;lr>
Max-Forwards: 70
From: <sip:ae019032109@10.101.23.110:5060>;tag=B8BUB47FmmerF
To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
CSeq: 46960329 INVITE
Contact: <sip:0877857933@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@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@10.101.23.110:5060>;tag=B8BUB47FmmerF
To: <sip:0877857933@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@10.101.23.110:5060>;tag=B8BUB47FmmerF
To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
Contact: <sip:0877857933@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@10.101.23.110:5060>;tag=B8BUB47FmmerF
To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
Contact: <sip:0877857933@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@10.101.23.110:5060>;tag=B8BUB47FmmerF
To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
Contact: <sip:0877857933@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@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@78.158.110.24>;tag=BK8QvjBeDHXtN
To: <sip:+35319032109@198.19.255.1;transport=tcp>;tag=397C7E04-26CA
Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
CSeq: 46960329 INVITE
Contact: <sip:+35319032109@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@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@78.158.110.24>;tag=BK8QvjBeDHXtN
To: <sip:+35319032109@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@78.158.110.24>;tag=BK8QvjBeDHXtN
To: <sip:+35319032109@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@198.19.255.1>;party=called;screen=yes;privacy=off
Contact: <sip:+35319032109@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@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@78.158.110.24>;tag=BK8QvjBeDHXtN
To: <sip:+35319032109@198.19.255.1;transport=tcp>;tag=397C7E04-26CA
Call-ID: f9547f1f-6e4d-1231-028b-001f290685a4
CSeq: 46960329 ACK
Contact: <sip:+35319032109@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@10.101.23.203:5060 SIP/2.0
Via: SIP/2.0/UDP 10.101.23.110;rport;branch=z9hG4bK43B5D34y3cX2m
Route: <sip:0877857933@10.101.23.203:5060;lr>
Max-Forwards: 70
From: <sip:ae019032109@10.101.23.110:5060>;tag=B8BUB47FmmerF
To: <sip:0877857933@10.101.23.203>;tag=95ffcd055e0f78f7d5d397020e89288dba056a96
Call-ID: 1054-453-6232013151830-IMG2_DEG-2-10.101.23.203
CSeq: 46960329 ACK
Contact: <sip:0877857933@10.101.23.110:5060>
Content-Length: 0
------------------------------------------------------------------------
On 23 Jul 2013, at 14:47, Brian West <brian@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@gmail.com> escreveu:
>
>> Anyone using TCP in this scenario? Cannot find what I'm missing.
>
>
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IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@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@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@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@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@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@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@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@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@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@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@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@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@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@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@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@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org