[Freeswitch-users] Extension: No audio
Mark Campbell-Smith
mcampbellsmith at gmail.com
Sun Nov 8 18:32:04 PST 2009
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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>>>
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>>
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