[Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
Vitalii Colosov
vetali100 at gmail.com
Thu Jul 8 01:56:42 PDT 2010
There should not be any problem if they use multiple IP addresses for media
stream, as long as siptraffic's SDP contains correct IP that will
participate in RTPs exchange...
Do note that media IP can be different from sip IP, and it is OK, if
implemented properly...
2010/7/8 paul gore <paul.gore.j at gmail.com>
> 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
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> >>> >
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> >>> >
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
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> >>> >
> >>> > 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
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> >>> >
> >>>
> >>>
> >>> _______________________________________________
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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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