[Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
k xd
kouxiaodong at gmail.com
Thu Jul 8 22:11:51 PDT 2010
I ever met same issue in EC2.
Modify the sip_profile configuration file like "internal.xml"
Replace the below item with actual ip address:
<param name="ext-rtp-ip" value="xxx.xxx.xxx.xxx"/>
Thanks,
Will
On Fri, Jul 9, 2010 at 7:31 AM, paul gore <paul.gore.j at gmail.com> wrote:
> I got ngrep trace for port 5060 while making a call to a US number via
> siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I
> heard no audio not even ringing.
> Is there anything in this trace which can help identify the problem?
>
> 10.194.206.102:5060 - is my local EC2 IP
> 184.72.206.204:5060 - is my public EC2 IP
> 77.72.169.128:5060 - siptraffic.com proxy IP
>
> Thanks!
>
>
>
> 67.33.160.119:18294 -> 10.194.206.102:5060
> INVITE sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486
> ;branch=z9hG4bK-d87543-f
> 524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: <
> sip:4000002 at 67.33.160.119:18027>..To: "45517
> 709248570"<sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>>..From:
> "4000002"<sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>
> >;tag=5f1ec15f..Call-
> ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow:
> INVITE, ACK, CANCEL, OPTIONS, BYE
> , REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type:
> application/sdp..Proxy-Authorization: Digest user
> name="4000002",realm="myserver.com
> ",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570 at v
> ersafon.com
> ",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000
> 0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp
> 41150..Content-Length: 417....v=0..o=-
> 8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4
> 192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107
> 119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8
> 46298..a=fmtp:101 0-15..a=rtpmap:107
> BV32/16000..a=rtpmap:119 BV32-FEC/16000..a=rtpmap:100
> SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap
> :105 SPEEX-FEC/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:101
> telephone-event/8000..a=sendrecv..
> #
> U 10.194.206.102:5060 -> 67.33.160.119:18294
> SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486
> ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r
> port=18294;received=67.33.160.119..From: "4000002" <
> sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>>;tag=5f1ec15f..To:
> "455177092
> 48570" <sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>>..Call-ID:
> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I
> NVITE..User-Agent: myserver..Content-Length: 0....
> #
> U 10.194.206.102:5080 -> 77.72.169.128:5060
> INVITE sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080
> ;rport;branch=z9h
> G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" <
> sip:0014444295793 at 184.72.206.204 <sip%3A0014444295793 at 184.72.206.204>
> >;tag=18853e82KDe7j.
> .To: <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>>..Call-ID:
> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647
> 2 INVITE..Contact: <sip:gw+voicetrading.com at 184.72.206.204:5080
> ;transport=udp;gw=voicetrading.com>..User-A
> gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE,
> UPDATE, INFO, REGISTER, REFER, NOTIFY..
> Supported: timer, precondition, path, replaces..Allow-Events: talk,
> refer..Content-Type: application/sdp..
> Content-Disposition: session..Content-Length: 295..X-FS-Support:
> update_display..Remote-Party-ID: "4000002
> " <sip:0014444295793 at 184.72.206.204 <sip%3A0014444295793 at 184.72.206.204>>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
> 1278518039
> 1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4
> 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8
> 3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3
> GSM/8000..a=rtpmap:101 telephone-event/80
> 00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20..
> #
> U 77.72.169.128:5060 -> 10.194.206.102:5080
> SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080
> ;rport;branch=z9hG4bKBU626KBp16t5Q..From
> : "4000002" <sip:0014444295793 at 184.72.206.204<sip%3A0014444295793 at 184.72.206.204>>;tag=18853e82KDe7j..To:
> <sip:0017705678570 at sip.siptraffic.co<sip%3A0017705678570 at sip.siptraffic.co>
> m>;tag=20113ac4c230cd6412168..Contact:
> sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c
> 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip
> Registrar/Proxy Server)..Allow: ACK,BYE,C
> ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type:
> application/sdp..Content-Length: 198....v=0..o=C
> ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4
> 77.72.168.40..t=0 0..m=audio 57672
> RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
> telephone-event/8000..a=ptime:20..
> #
> U 10.194.206.102:5060 -> 67.33.160.119:18294
> SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486
> ;branch=z9hG4bK-d87543-f524431af92cef56-1-
> -d87543-;rport=18294;received=67.33.160.119..From: "4000002" <
> sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>>;tag=5f1ec15f..To:
> "45517705678570" <sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID:
> ZDAzODE0Y2JkZjYzODE5NmVmNjk
> zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact:
> <sip:45517705678570 at 184.72.206.204:5060;transport=udp>..User-A
> gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL,
> OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer,
> precondition, path, replaces..Allow-Events:
> talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-
> summary, refer..Content-Type: application/sdp..Content-Disposition:
> session..Content-Length: 251..Remote-P
> arty-ID: "45517705678570" <sip:45517705678570 at 10.194.206.102<sip%3A45517705678570 at 10.194.206.102>
> >;party=calling;privacy=off;screen=no....v=0..
> o=FreeSWITCH 1278530815 1278530816 IN IP4
> 184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=
> audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
> telephone-event/8000..a=fmtp:101 0-16..a=sil
> enceSupp:off - - - -..a=ptime:20..
> #
>
>
>
> U 77.72.169.128:5060 -> 10.194.206.102:5080
> SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080
> ;rport;branch=z9hG4bKBU626KBp16t5Q..From
> : "4000002" <sip:0014444295793 at 184.72.206.204<sip%3A0014444295793 at 184.72.206.204>>;tag=18853e82KDe7j..To:
> <sip:0017705678570 at sip.siptraffic.co<sip%3A0017705678570 at sip.siptraffic.co>
> m>;tag=20113ac4c230cd6412168..Contact:
> sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c
> 381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip
> Registrar/Proxy Server)..Allow: ACK,BYE,C
> ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type:
> application/sdp..Content-Length: 204....v=0..o=C
> ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4
> 208.167.230.118..t=0 0..m=audio
> 57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
> telephone-event/8000..a=ptime:20..
> #
>
> U 67.33.160.119:18294 -> 10.194.206.102:5060
> ....
> #
> U 67.33.160.119:18294 -> 10.194.206.102:5060
> CANCEL sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486
> ;branch=z9hG4bK-d87543-f
> 524431af92cef56-1--d87543-;rport..To: "45517705678570"<
> sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com>>..From:
> "4000002"<s
> ip:4000002 at myserver.com <ip%3A4000002 at myserver.com>>;tag=5f1ec15f..Call-ID:
> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC
> EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com
> ",nonce="cf9019cc-f44a-4568-97d1-e98
> 83fb1821f",uri="sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>
> ",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c
>
> 55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent:
> X-Lite release 1011s stamp 41
> 150..Content-Length: 0....
> #
> U 10.194.206.102:5060 -> 67.33.160.119:18294
> SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486
> ;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport
> =18294;received=67.33.160.119..From: "4000002" <sip:4000002 at myserver.com<sip%3A4000002 at myserver.com>>;tag=5f1ec15f..To:
> "4551770567857
> 0" <sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID:
> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY
> mY...CSeq: 2 CANCEL..Content-Length: 0....
> #
> U 10.194.206.102:5060 -> 67.33.160.119:18294
> SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486
> ;branch=z9hG4bK-d87543-f524431af92cef56-
> 1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" <
> sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>>;tag=5f1ec15f..To
> : "45517705678570" <sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID:
> ZDAzODE0Y2JkZjYzODE5NmVmN
> jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE,
> ACK, BYE, CANCEL, OPTIONS, MESSA
> GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported:
> timer, precondition, path, repla
> ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presen
> ce.winfo, message-summary, refer..Content-Length: 0....
> #
> U 10.194.206.102:5080 -> 77.72.169.128:5060
> CANCEL sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080
> ;rport;branch=z9h
> G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" <
> sip:0014444295793 at 184.72.206.204 <sip%3A0014444295793 at 184.72.206.204>
> >;tag=18853e82KDe7j.
> .To: <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>>..Call-ID:
> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647
> 2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length:
> 0....
> #
>
> U 67.33.160.119:18294 -> 10.194.206.102:5060
> ACK sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com>SIP/2.0..Via: SIP/2.0/UDP 192.168.0.8:29486
> ;branch=z9hG4bK-d87543-f524
> 431af92cef56-1--d87543-;rport..To: "45517705678570" <
> sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com>
> >;tag=BXgB1FZBUZ3Da..F
> rom: "4000002"<sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>>;tag=5f1ec15f..Call-ID:
> ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm
> Y...CSeq: 2 ACK..Content-Length: 0....
> #
>
> U 77.72.169.128:5060 -> 10.194.206.102:5080
> SIP/2.0 200 Ok..Via: SIP/2.0/UDP 184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From:
> "4000002" <s
> ip:0014444295793 at 184.72.206.204 <ip%3A0014444295793 at 184.72.206.204>>;tag=18853e82KDe7j..To:
> <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>>..Contact:
> s
> ip:0017705678570 at 77.72.169.128:5060..Call-ID:
> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL
> ..Server: (Very nice Sip Registrar/Proxy Server)..Allow:
> ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA
> GE..Content-Length: 0....
> #
> U 77.72.169.128:5060 -> 10.194.206.102:5080
> SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080
> ;rport;branch=z9hG4bKBU626KBp16t5Q..Fr
> om: "4000002" <sip:0014444295793 at 184.72.206.204<sip%3A0014444295793 at 184.72.206.204>>;tag=18853e82KDe7j..To:
> <sip:0017705678570 at sip.siptraffic.
> com>..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID:
> 37f59333-04cc-122e-c381-12313b06cd32..CSeq:
> 133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow:
> ACK,BYE,CANCEL,INVITE,REGISTER,OP
> TIONS,INFO,MESSAGE..Content-Length: 0....
> #
> U 10.194.206.102:5080 -> 77.72.169.128:5060
> ACK sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>SIP/2.0..Via: SIP/2.0/UDP 184.72.206.204:5080
> ;rport;branch=z9hG4b
> KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" <
> sip:0014444295793 at 184.72.206.204 <sip%3A0014444295793 at 184.72.206.204>
> >;tag=18853e82KDe7j..To
> : <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>>..Call-ID:
> 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A
> CK..Content-Length: 0....
> #
>
>
>
>
>
>
>
>
> On Wed, Jul 7, 2010 at 5:50 PM, paul gore <paul.gore.j at gmail.com> wrote:
>
>> Seems like siptraffic uses 6 ip addresses for media, can that be the
>> problem? Is there any setting in a gateway config xml which helps with
>> that?
>> I will do ngrep thing and update.
>>
>> On 7/7/10, paul gore <paul.gore.j at gmail.com> wrote:
>> > This provider does work on another box which is not natted as ec2.
>> > Most puzzling here though is why call originaion via api even not
>> > going via siptraffic still gets no audio.
>> >
>> > On 7/7/10, Tony Graziano <tgraziano at myitdepartment.net> wrote:
>> >> You should try from a standalone or local installation to ensure it
>> works
>> >> with this provider and your account before you attempt to run it on ec2
>> >> (imo).
>> >>
>> >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin
>> >> <sos at sokhapkin.dyndns.org>wrote:
>> >>
>> >>> What "doesn't work" means? It could be (and most likely is not)
>> >>> FS-related
>> >>> problem
>> >>>
>> >>> On Wednesday 07 July 2010, Madovsky wrote:
>> >>> > I had same problem from this provider without to explain why.
>> >>> > One day it works, another day it doesn't, their support is crap...
>> >>> >
>> >>> > ----- Original Message -----
>> >>> > From: Anthony Minessale
>> >>> > To: freeswitch-users at lists.freeswitch.org
>> >>> > Sent: Wednesday, July 07, 2010 2:37 PM
>> >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on
>> >>> > outgoing
>> >>> > calls
>> >>> >
>> >>> >
>> >>> > not really, not with so little information.
>> >>> >
>> >>> >
>> >>> >
>> >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j at gmail.com>
>> >>> wrote:
>> >>> >
>> >>> > Firewall is configured according to the wiki, I also tried to
>> open
>> >>> all
>> >>> > udp ports, issue persists.
>> >>> > Actually the problem became more complex - outgoing calls don't
>> >>> > work
>> >>> > with one particular termination provider, siptraffic.com , any
>> >>> > ideas
>> >>> > why?
>> >>> > Outgoing calls also don't work when originating a call via js
>> >>> > script
>> >>> > or via FS api. Any clues on that one?
>> >>> >
>> >>> > On 7/6/10, paul gore <paul.gore.j at gmail.com> wrote:
>> >>> > > Hi there,
>> >>> > > I am experimenting with FS on EC2, I like results, but stuck
>> on
>> >>> weird
>> >>> > > audio issue - I followed FreeSwitch EC2 wiki article and
>> >>> > modified
>> >>> > > internal profile
>> >>> > > and vars.xml accordingly, but unfortunately still cannot get
>> it
>> >>> > > working. Incoming and outgoing calls made using a SIP phone to
>> >>> > FS
>> >>> > > extensions work just fine. As well as calls to FS from PSTN.
>> But
>> >>> > > calls to PSTN via gateways result in no audio at all, no ring,
>> >>> > > nothing, SIP signaling goes through OK. Sofia status profile
>> >>> > shows
>> >>> > > correct values for Ext-RTP-IP for both profiles -
>> >>> > > my static public IP, RTP-IP shows local IP.
>> >>> > > Any thoughts on that? Anybody can share working profile
>> >>> configuration
>> >>> > > may be?
>> >>> > > Please help, I really need to get this going.
>> >>> > >
>> >>> > > Thanks.
>>
>> >>> >
>> >>> > _______________________________________________
>> >>> > 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
>> >>> >
>> >>> >
>> >>> >
>> >>> >
>> >>> >
>> >>> > FreeSWITCH http://www.freeswitch.org/
>> >>> > ClueCon http://www.cluecon.com/
>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >
>> >>> > AIM: anthm
>> >>> >
>> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> <MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>> >
>> >>> >
>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>> >
>> >>> > IRC: irc.freenode.net #freeswitch
>> >>> >
>> >>> > FreeSWITCH Developer Conference
>> >>> >
>> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> <sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>> >
>> >>> >
>> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>> >
>> >>> > pstn:+19193869900
>> >>> >
>> >>> >
>> >>> >
>> >>> >
>> >>>
>> ---------------------------------------------------------------------------
>> >>> > ---
>>
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > 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
>> >>> >
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>>
>> >>
>> >>
>> >>
>> >> --
>> >> ======================
>> >> Tony Graziano, Manager
>> >> Telephone: 434.984.8430
>> >> sip: tgraziano at voice.myitdepartment.net
>> >> Fax: 434.984.8431
>> >>
>> >> Email: tgraziano at myitdepartment.net
>> >>
>> >> LAN/Telephony/Security and Control Systems Helpdesk:
>> >> Telephone: 434.984.8426
>> >> sip: helpdesk at voice.myitdepartment.net
>> >> Fax: 434.984.8427
>> >>
>> >> Helpdesk Contract Customers:
>> >> http://www.myitdepartment.net/gethelp/
>> >>
>> >> Why do mathematicians always confuse Halloween and Christmas?
>> >> Because 31 Oct = 25 Dec.
>> >>
>> >
>>
>
>
> _______________________________________________
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