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@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@67.33.160.119:18027>..To: "45517
709248570"<sip:45517705678570@myserver.com>..From: "4000002"<sip:4000002@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@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@myserver.com>;tag=5f1ec15f..To: "455177092
48570" <sip:45517705678570@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@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@184.72.206.204>;tag=18853e82KDe7j.
.To: <sip:0017705678570@sip.siptraffic.com>..Call-ID: 37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647
2 INVITE..Contact: <sip:gw+voicetrading.com@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@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@184.72.206.204>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co
m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@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@myserver.com>;tag=5f1ec15f..To:
"45517705678570" <sip:45517705678570@myserver.com>;tag=BXgB1FZBUZ3Da..Call-ID: ZDAzODE0Y2JkZjYzODE5NmVmNjk
zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact: <sip:45517705678570@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@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@184.72.206.204>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.co
m>;tag=20113ac4c230cd6412168..Contact: sip:0017705678570@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@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@myserver.com>..From: "4000002"<s
ip:4000002@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@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@myserver.com>;tag=5f1ec15f..To: "4551770567857
0" <sip:45517705678570@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@myserver.com>;tag=5f1ec15f..To
: "45517705678570" <sip:45517705678570@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@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@184.72.206.204>;tag=18853e82KDe7j.
.To: <sip:0017705678570@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@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@myserver.com>;tag=BXgB1FZBUZ3Da..F
rom: "4000002"<sip:4000002@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@184.72.206.204>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.com>..Contact: s
ip:0017705678570@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@184.72.206.204>;tag=18853e82KDe7j..To: <sip:0017705678570@sip.siptraffic.
com>..Contact: sip:0017705678570@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@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@184.72.206.204>;tag=18853e82KDe7j..To
: <sip:0017705678570@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@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@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@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@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@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@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@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 http://www.freeswitch.org/
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users@lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> >
>>> > MSN:anthony_minessale@hotmail.com<MSN%3Aanthony_minessale@hotmail.com>
>>> >
>>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com>
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> >
>>> > sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org>
>>> >
>>> > googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org>
>>> > pstn:+19193869900
>>> >
>>> >
>>> >
>>> >
>>> ---------------------------------------------------------------------------
>>> > --->> --
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users@lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> ======================
>> Tony Graziano, Manager
>> Telephone: 434.984.8430
>> sip: tgraziano@voice.myitdepartment.net
>> Fax: 434.984.8431
>>
>> Email: tgraziano@myitdepartment.net
>>
>> LAN/Telephony/Security and Control Systems Helpdesk:
>> Telephone: 434.984.8426
>> sip: helpdesk@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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