[Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls

Anthony Minessale anthony.minessale at gmail.com
Fri Jul 9 07:57:14 PDT 2010


in the future post logs to pastebin,
notice how big and ugly this thread is now quoting your log and toting it
around on each reply.
Also consider coming to IRC for real-time interaction and post the
conclusion here once you figure it out,


On Fri, Jul 9, 2010 at 9:20 AM, paul gore <paul.gore.j at gmail.com> wrote:

> As I mentioned in my first post I did that, I did everything as per
> ec2 wiki, incuding setting ext-ip in profiles.
> I have everything working fine, except no audio via siptraffic.com.
> I will do pcap and rtp trace via ngrep over weekend and update.
>
> On 7/9/10, k xd <kouxiaodong at gmail.com> wrote:
> > 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%3A45517705678570 at myserver.com <sip%253A45517705678570 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>
> <sip%3A45517705678570 at myserver.com <sip%253A45517705678570 at myserver.com>
> >>..From:
> >> "4000002"<sip:4000002 at myserver.com <sip%3A4000002 at myserver.com> <
> sip%3A4000002 at myserver.com <sip%253A4000002 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> <
> sip%3A4000002 at myserver.com <sip%253A4000002 at myserver.com>
> >>;tag=5f1ec15f..To:
> >> "455177092
> >>   48570"
> >> <sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com><
> sip%3A45517705678570 at myserver.com <sip%253A45517705678570 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%3A0017705678570 at sip.siptraffic.com<sip%253A0017705678570 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> <
> sip%3A0014444295793 at 184.72.206.204 <sip%253A0014444295793 at 184.72.206.204>>
> >> >;tag=18853e82KDe7j.
> >>   .To:
> >> <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>
> <sip%3A0017705678570 at sip.siptraffic.com<sip%253A0017705678570 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>
> >> <sip%3A0014444295793 at 184.72.206.204<sip%253A0014444295793 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><
> sip%3A0014444295793 at 184.72.206.204 <sip%253A0014444295793 at 184.72.206.204>
> >>;tag=18853e82KDe7j..To:
> >> <sip:0017705678570 at sip.siptraffic.co<sip%3A0017705678570 at sip.siptraffic.co>
> <sip%3A0017705678570 at sip.siptraffic.co<sip%253A0017705678570 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> <
> sip%3A4000002 at myserver.com <sip%253A4000002 at myserver.com>
> >>;tag=5f1ec15f..To:
> >>   "45517705678570"
> >> <sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com><
> sip%3A45517705678570 at myserver.com <sip%253A45517705678570 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>
> <sip%3A45517705678570 at 10.194.206.102<sip%253A45517705678570 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><
> sip%3A0014444295793 at 184.72.206.204 <sip%253A0014444295793 at 184.72.206.204>
> >>;tag=18853e82KDe7j..To:
> >> <sip:0017705678570 at sip.siptraffic.co<sip%3A0017705678570 at sip.siptraffic.co>
> <sip%3A0017705678570 at sip.siptraffic.co<sip%253A0017705678570 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%3A45517705678570 at myserver.com <sip%253A45517705678570 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>
> >> <sip%3A45517705678570 at myserver.com<sip%253A45517705678570 at myserver.com>
> >>..From:
> >> "4000002"<s
> >>   ip:4000002 at myserver.com <ip%3A4000002 at myserver.com>
> >> <ip%3A4000002 at myserver.com <ip%253A4000002 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>
> <sip%3A45517705678570 at myserver.com <sip%253A45517705678570 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><
> sip%3A4000002 at myserver.com <sip%253A4000002 at myserver.com>
> >>;tag=5f1ec15f..To:
> >> "4551770567857
> >>   0" <sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>
> >> <sip%3A45517705678570 at myserver.com<sip%253A45517705678570 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> <
> sip%3A4000002 at myserver.com <sip%253A4000002 at myserver.com>
> >>;tag=5f1ec15f..To
> >>   : "45517705678570"
> >> <sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com><
> sip%3A45517705678570 at myserver.com <sip%253A45517705678570 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%3A0017705678570 at sip.siptraffic.com<sip%253A0017705678570 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> <
> sip%3A0014444295793 at 184.72.206.204 <sip%253A0014444295793 at 184.72.206.204>>
> >> >;tag=18853e82KDe7j.
> >>   .To:
> >> <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>
> <sip%3A0017705678570 at sip.siptraffic.com<sip%253A0017705678570 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%3A45517705678570 at myserver.com<sip%253A45517705678570 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> <
> sip%3A45517705678570 at myserver.com <sip%253A45517705678570 at myserver.com>>
> >> >;tag=BXgB1FZBUZ3Da..F
> >>   rom: "4000002"<sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>
> >> <sip%3A4000002 at myserver.com <sip%253A4000002 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>
> >> <ip%3A0014444295793 at 184.72.206.204<ip%253A0014444295793 at 184.72.206.204>
> >>;tag=18853e82KDe7j..To:
> >> <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>
> <sip%3A0017705678570 at sip.siptraffic.com<sip%253A0017705678570 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><
> sip%3A0014444295793 at 184.72.206.204 <sip%253A0014444295793 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%3A0017705678570 at sip.siptraffic.com<sip%253A0017705678570 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> <
> sip%3A0014444295793 at 184.72.206.204 <sip%253A0014444295793 at 184.72.206.204>>
> >> >;tag=18853e82KDe7j..To
> >>   :
> >> <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>
> <sip%3A0017705678570 at sip.siptraffic.com<sip%253A0017705678570 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>
> >
> >>> <MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> <MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_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>
> >
> >>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> <PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.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>
> >
> >>> <sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> <sip%253A888 at conference.freeswitch.org<sip%25253A888 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>
> >
> >>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> <googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 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
>



-- 
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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
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