[Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
Vitalii Colosov
vetali100 at gmail.com
Fri Jul 9 00:57:40 PDT 2010
You have provided only SIP log. Where is the media analysis?
You should have run another ngrep to see if media traffic is OK between FS
and siptraffic:
"ngrep port 31564"
I took the port value from the following line:
c=IN IP4 184.72.206.204..t=0 0..m=audio 31564 RTP/AVP
>From the following SIP request:
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..
#
2010/7/9 k xd <kouxiaodong at gmail.com>
> 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.
>>> >>
>>> >
>>>
>>
>>
>> _______________________________________________
>> 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
>
>
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