[Freeswitch-users] Call Recovery when using TCP

Anthony McGarry agtmcgarry at gmail.com
Thu Jul 25 02:29:35 MSD 2013 

http://pastebin.freeswitch.org/21235

On 24 Jul 2013, at 23:17, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> can you do that again with this extra command right before recover is executed?
> 
> sofia loglevel all 9 
> 
> maybe its failing on tcp and reverting to udp
> 
> 
> On Wed, Jul 24, 2013 at 5:01 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
> ok so I added INVITE sip:+353877857933 at voice.plannet21.ie:5060;transport=tcp SIP/2.0 to all outgoing INVITES from my client.
> 
> Uploaded logs from both servers
> 
> before crash
> http://pastebin.freeswitch.org/21233
> 
> after crash
> http://pastebin.freeswitch.org/21234
> 
> 
> On 24 Jul 2013, at 22:13, Anthony Minessale <anthony.minessale at 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 at 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 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.
>>>>> >
>>>>> >
>>>>> > _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > 
>>>>> > 
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
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>>>>> >
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>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
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>>>>> 
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>>>>> 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
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> 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
>>> _________________________________________________________________________
>>> 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
>> 
>> 
>> 
>> -- 
>> 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
> _________________________________________________________________________
> 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

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