[Freeswitch-users] Extension: No audio
Mark Campbell-Smith
mcampbellsmith at gmail.com
Sun Nov 8 19:40:17 PST 2009
Is there a way to determine if FS has detected nat? I am behind UPnP
and I can see on the router the mappings for Freeswitch.
2009/11/9 João Mesquita <jmesquita at freeswitch.org>:
> 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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>> >
>>
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