May be it's just a FAS? Are you able to call that number with a different SIP client using the same call termination provider? On Wednesday 07 July 2010, Vitalii Colosov wrote:
So "don't work" still means "no audio" and "sip is ok", clear...
RTP packets are lost somewhere on the way:
You can check where RTP packets are lost - between FS and siptraffic, or between FS and your SIP Client.
1.Install "ngrep" if you don't have it yet (on Ubuntu: apt-get install ngrep) 2.Run it: "ngrep port 5080" 3.Start a call and after few ngrep's messages, stop ngrep using Ctrl+C 4.Find the first INVITE line from your FS to siptraffic, it will contain your IP and port at the following lines: c=IN IP4 ****!!!YOUR-EC-IP!!!*****..t=0 0..m=audio ****!!!YOUR-RTP-PORT!!!**** RTP/AVP 5.Open new terminal window, and run "ngrep port ****!!!YOUR-RTP-PORT!!!****" 6. Wait 5 seconds 7. Stop ngrep using Ctrl+C 8. Hangup
Now on the second terminal you should see a lot of line pairs like: YOUR-EC-IP -> SIPTRAFFIC-IP SIPTRAFFIC-IP -> YOUR-EC-IP
If you see only one of the directions (e.g. only YOUR-EC-IP -> SIPTRAFFIC-IP), then some problem is between FS and Siptraffic.
If you see both directions then problem is not here and most probably on the way from FS to your SIP Client or somewhere else (inside FS?) If so, try to investigate this part using port 5060 (same way as 5080).
This analysis will narrow the problem a bit...
Regads, Vitalie
2010/7/7 paul gore <paul.gore.j@gmail.com>
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.
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-- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
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