[Freeswitch-users] Extension: No audio
João Mesquita
jmesquita at freeswitch.org
Sun Nov 8 18:39:21 PST 2009
It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise,
no go....
Have you changed the ext-sip-ip too?
Regards,
JM
On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:
> Hi again,
>
> Actually, changing the <param name="ext-rtp-ip" value="auto-nat"/> to
> <param name="ext-rtp-ip" value="$${external_sip_ip}"/> means that I
> now see the IP address in the INVITE message:
>
> v=0
> o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx
> s=FreeSWITCH
> c=IN IP4 124.xxx.xxx.xxx
> t=0 0
> m=audio 21234 RTP/AVP 0 2 9 8 101 13
>
> Why would this be? I thought auto-nat was meant to solve these issues?
>
> However, I still do not see the TRYING or RINGING messages.... ideas
> appreciated.
>
> Thanks!
>
> On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
> > OK.. thanks Mike.
> >
> > I assume I am using the Internal profile. I have defined user 2000
> > in the 'directory' using a context called family: switch_ivr.c:1367
> > Transfer sofia/internal/1000 at 192.168.1.120 to XML[2000 at family]
> >
> > This is an extract from sofia:
> >
> > sofia status profile internal
> >
> =================================================================================================
> > Name internal
> > Domain Name N/A
> > DBName sofia_reg_internal
> > Pres Hosts
> > Dialplan XML
> > Context public
> > Challenge Realm auto_from
> > RTP-IP 192.168.1.120
> > Ext-RTP-IP 124.xxx.xxx.xxx
> > SIP-IP 192.168.1.120
> > Ext-SIP-IP 124.xxx.xxx.xxx
> > URL sip:mod_sofia at 192.168.1.120:5060
> > BIND-URL sip:mod_sofia at 192.168.1.120:5060
> > HOLD-MUSIC silence
> > OUTBOUND-PROXY N/A
> > CODECS G726-32,G722,PCMU,PCMA
> > TEL-EVENT 101
> > DTMF-MODE rfc2833
> > CNG 13
> > SESSION-TO 0
> > MAX-DIALOG 0
> > NOMEDIA false
> > LATE-NEG false
> > PROXY-MEDIA false
> > AGGRESSIVENAT true
> > STUN-ENABLED true
> > STUN-AUTO-DISABLE false
> > CALLS-IN 100
> > FAILED-CALLS-IN 25
> > CALLS-OUT 38
> > FAILED-CALLS-OUT 31
> >
> > Registrations:
> >
> =================================================================================================
> > Call-ID: 68534BBA9B461526 at 58.169.138.53
> > User: 2000 at 192.168.1.120
> > Contact: "user" <sip:2000 at 58.xxx.xxx.xxx:5060>
> > Agent: dunno
> > Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30)
> > Host: freeswitch
> > IP: 58.xxx.xxx.xxx
> > Port: 5060
> > Auth-User: 2000
> > Auth-Realm: markcs.dyndns.org
> > MWI-Account: 2000 at 192.168.1.120
> >
> > The internal.xml file has a lot in it, but I guess these are the
> > important things for this profile:
> >
> > <param name="ext-rtp-ip" value="auto-nat"/>
> > <param name="ext-sip-ip" value="auto-nat"/>
> >
> > <param name="sip-port" value="$${internal_sip_port}"/>
> > <param name="rtp-ip" value="auto"/>
> >
> > I will try to change auto-nat to be $${external_sip_ip}
> >
> > One question though: Any idea why I never see the TRYING or RINGING
> > messages? Are tehse related to the RTP IP address or not? Without
> > these I assume something is incorrect and I do not hear ringback....
> >
> > Thanks!
> >
> > On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris <mike at jerris.com> wrote:
> >> Your packet traces would disagree with the statements below. It is
> >> sending your internal address in rtp, so its not set correctly on
> >> whatever profile your using to call out,
> >>
> >> MIke
> >>
> >> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:
> >>
> >>> Hi Mike,
> >>>
> >>> I should have put that in also.
> >>>
> >>> I do have external_rtp_ip set in my config. I have it set to my
> >>> domain name:
> >>> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomainname"/>
> >>>
> >>> I should also mention that if I use flaphone.com (which registers with
> >>> an external IP address), then I get audio. In sofia, I see my IP
> >>> addresses:
> >>>
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> =
> >>> ======================================================================
> >>> Name internal
> >>> Domain Name N/A
> >>> DBName sofia_reg_internal
> >>> Pres Hosts
> >>> Dialplan XML
> >>> Context public
> >>> Challenge Realm auto_from
> >>> RTP-IP 192.168.1.120
> >>> Ext-RTP-IP 124.xxx.xxx.xxx
> >>> SIP-IP 192.168.1.120
> >>> Ext-SIP-IP 124.xxx.xxx.x
> >>> URL sip:mod_sofia at 192.168.1.120:5060
> >>> BIND-URL sip:mod_sofia at 192.168.1.120:5060
> >>> HOLD-MUSIC silence
> >>> OUTBOUND-PROXY N/A
> >>> CODECS G726-32,G722,PCMU,PCMA
> >>> TEL-EVENT 101
> >>> DTMF-MODE rfc2833
> >>> CNG 13
> >>> SESSION-TO 0
> >>> MAX-DIALOG 0
> >>> NOMEDIA false
> >>> LATE-NEG false
> >>> PROXY-MEDIA false
> >>> AGGRESSIVENAT true
> >>> STUN-ENABLED true
> >>> STUN-AUTO-DISABLE false
> >>>
> >>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris <mike at jerris.com>
> >>> wrote:
> >>>> You don't have ext-rtp-ip set in your config.
> >>>>
> >>>> Mike
> >>>>
> >>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote:
> >>>>
> >>>>> Hi!
> >>>>>
> >>>>> I have FS natted and am connecting with an 'external' extension that
> >>>>> is registered to FS. ie the extension 2000 is registered on the
> >>>>> internet with a public IP through my router to FS (192.168.1.120 IP
> >>>>> address). uPnP works and I see that the extension is registered
> >>>>> successfully.
> >>>>>
> >>>>> The problem is that I do not get any audio
> >>>>>
> >>>>> When looking at the SIP trace, I see the INVITE but do not see a
> >>>>> TRYING or RINGING message. The extension is actually ringing. I
> >>>>> modified the RTP port range on the remote end to match the RTP ports
> >>>>> of freeswitch.
> >>>>>
> >>>>> I have put a sip trace in the pastebin at
> http://pastebin.freeswitch.org/11035
> >>>>>
> >>>>> If anyone has an idea what needs to be set to get audio, help
> >>>>> appreciated.
> >>>>>
> >>>>> Thanks!
> >>>>
> >>>>
> >>>> _______________________________________________
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> >>>> FreeSWITCH-users at lists.freeswitch.org
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> >>>> http://www.freeswitch.org
> >>>>
> >>>
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> >>
> >>
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