[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!
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>>> users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>> _______________________________________________
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>> >>
>> >>
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>> >
>>
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