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

Sergey Okhapkin sos at sokhapkin.dyndns.org
Wed Jul 7 14:24:39 PDT 2010 

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 at 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 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.
> > >> >
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> > >> >
> > >> >
> > >> >
> > >> >
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> > >> >essale at gmail.com>
> >
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> >com>
> >
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> > >> >
> > >> >   FreeSWITCH Developer Conference
> > >> >  
> > >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.or
> > >> >g>
> >
> > <sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.or
> >g>
> >
> > >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B88
> > >> >8 at conference.freeswitch.org>
> >
> > <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%25
> >2B888 at conference.freeswitch.org>
> >
> > >> >   pstn:+19193869900
> >
> > -------------------------------------------------------------------------
> >--
> >
> > >> > ---
> > >> >
> > >> >
> > >> >   _______________________________________________
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> > > --
> > > ======================
> > > 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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