It is if FS was able to detect NAT. Are you behind PMP or UPnP? Otherwise, no go....<br><br>Have you changed the ext-sip-ip too?<br><br>Regards,<br><br>JM<br><br><br><div class="gmail_quote">On Mon, Nov 9, 2009 at 12:32 AM, Mark Campbell-Smith <span dir="ltr"><<a href="mailto:mcampbellsmith@gmail.com">mcampbellsmith@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi again,<br>
<br>
Actually, changing the <param name="ext-rtp-ip" value="auto-nat"/> to<br>
<param name="ext-rtp-ip" value="$${external_sip_ip}"/> means that I<br>
now see the IP address in the INVITE message:<br>
<br>
v=0<br>
o=FreeSWITCH 1257711702 1257711703 IN IP4 124.xxx.xxx.xxx<br>
s=FreeSWITCH<br>
c=IN IP4 124.xxx.xxx.xxx<br>
t=0 0<br>
m=audio 21234 RTP/AVP 0 2 9 8 101 13<br>
<br>
Why would this be? I thought auto-nat was meant to solve these issues?<br>
<br>
However, I still do not see the TRYING or RINGING messages.... ideas<br>
appreciated.<br>
<br>
Thanks!<br>
<div><div></div><div class="h5"><br>
On Mon, Nov 9, 2009 at 1:14 PM, Mark Campbell-Smith<br>
<<a href="mailto:mcampbellsmith@gmail.com">mcampbellsmith@gmail.com</a>> wrote:<br>
> OK.. thanks Mike.<br>
><br>
> I assume I am using the Internal profile. I have defined user 2000<br>
> in the 'directory' using a context called family: switch_ivr.c:1367<br>
> Transfer sofia/internal/<a href="mailto:1000@192.168.1.120">1000@192.168.1.120</a> to XML[2000@family]<br>
><br>
> This is an extract from sofia:<br>
><br>
> sofia status profile internal<br>
> =================================================================================================<br>
> Name internal<br>
> Domain Name N/A<br>
> DBName sofia_reg_internal<br>
> Pres Hosts<br>
> Dialplan XML<br>
> Context public<br>
> Challenge Realm auto_from<br>
> RTP-IP 192.168.1.120<br>
> Ext-RTP-IP 124.xxx.xxx.xxx<br>
> SIP-IP 192.168.1.120<br>
> Ext-SIP-IP 124.xxx.xxx.xxx<br>
> URL <a href="http://sip:mod_sofia@192.168.1.120:5060" target="_blank">sip:mod_sofia@192.168.1.120:5060</a><br>
> BIND-URL <a href="http://sip:mod_sofia@192.168.1.120:5060" target="_blank">sip:mod_sofia@192.168.1.120:5060</a><br>
> HOLD-MUSIC silence<br>
> OUTBOUND-PROXY N/A<br>
> CODECS G726-32,G722,PCMU,PCMA<br>
> TEL-EVENT 101<br>
> DTMF-MODE rfc2833<br>
> CNG 13<br>
> SESSION-TO 0<br>
> MAX-DIALOG 0<br>
> NOMEDIA false<br>
> LATE-NEG false<br>
> PROXY-MEDIA false<br>
> AGGRESSIVENAT true<br>
> STUN-ENABLED true<br>
> STUN-AUTO-DISABLE false<br>
> CALLS-IN 100<br>
> FAILED-CALLS-IN 25<br>
> CALLS-OUT 38<br>
> FAILED-CALLS-OUT 31<br>
><br>
> Registrations:<br>
> =================================================================================================<br>
> Call-ID: <a href="mailto:68534BBA9B461526@58.169.138.53">68534BBA9B461526@58.169.138.53</a><br>
> User: <a href="mailto:2000@192.168.1.120">2000@192.168.1.120</a><br>
> Contact: "user" <sip:2000@58.xxx.xxx.xxx:5060><br>
> Agent: dunno<br>
> Status: Registered(UDP)(unknown) EXP(2009-11-09 14:58:30)<br>
> Host: freeswitch<br>
> IP: 58.xxx.xxx.xxx<br>
> Port: 5060<br>
> Auth-User: 2000<br>
> Auth-Realm: <a href="http://markcs.dyndns.org" target="_blank">markcs.dyndns.org</a><br>
> MWI-Account: <a href="mailto:2000@192.168.1.120">2000@192.168.1.120</a><br>
><br>
> The internal.xml file has a lot in it, but I guess these are the<br>
> important things for this profile:<br>
><br>
> <param name="ext-rtp-ip" value="auto-nat"/><br>
> <param name="ext-sip-ip" value="auto-nat"/><br>
><br>
> <param name="sip-port" value="$${internal_sip_port}"/><br>
> <param name="rtp-ip" value="auto"/><br>
><br>
> I will try to change auto-nat to be $${external_sip_ip}<br>
><br>
> One question though: Any idea why I never see the TRYING or RINGING<br>
> messages? Are tehse related to the RTP IP address or not? Without<br>
> these I assume something is incorrect and I do not hear ringback....<br>
><br>
> Thanks!<br>
><br>
> On Mon, Nov 9, 2009 at 11:41 AM, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>> wrote:<br>
>> Your packet traces would disagree with the statements below. It is<br>
>> sending your internal address in rtp, so its not set correctly on<br>
>> whatever profile your using to call out,<br>
>><br>
>> MIke<br>
>><br>
>> On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote:<br>
>><br>
>>> Hi Mike,<br>
>>><br>
>>> I should have put that in also.<br>
>>><br>
>>> I do have external_rtp_ip set in my config. I have it set to my<br>
>>> domain name:<br>
>>> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomainname"/><br>
>>><br>
>>> I should also mention that if I use <a href="http://flaphone.com" target="_blank">flaphone.com</a> (which registers with<br>
>>> an external IP address), then I get audio. In sofia, I see my IP<br>
>>> addresses:<br>
>>><br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> =<br>
>>> ======================================================================<br>
>>> Name internal<br>
>>> Domain Name N/A<br>
>>> DBName sofia_reg_internal<br>
>>> Pres Hosts<br>
>>> Dialplan XML<br>
>>> Context public<br>
>>> Challenge Realm auto_from<br>
>>> RTP-IP 192.168.1.120<br>
>>> Ext-RTP-IP 124.xxx.xxx.xxx<br>
>>> SIP-IP 192.168.1.120<br>
>>> Ext-SIP-IP 124.xxx.xxx.x<br>
>>> URL <a href="http://sip:mod_sofia@192.168.1.120:5060" target="_blank">sip:mod_sofia@192.168.1.120:5060</a><br>
>>> BIND-URL <a href="http://sip:mod_sofia@192.168.1.120:5060" target="_blank">sip:mod_sofia@192.168.1.120:5060</a><br>
>>> HOLD-MUSIC silence<br>
>>> OUTBOUND-PROXY N/A<br>
>>> CODECS G726-32,G722,PCMU,PCMA<br>
>>> TEL-EVENT 101<br>
>>> DTMF-MODE rfc2833<br>
>>> CNG 13<br>
>>> SESSION-TO 0<br>
>>> MAX-DIALOG 0<br>
>>> NOMEDIA false<br>
>>> LATE-NEG false<br>
>>> PROXY-MEDIA false<br>
>>> AGGRESSIVENAT true<br>
>>> STUN-ENABLED true<br>
>>> STUN-AUTO-DISABLE false<br>
>>><br>
>>> On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>
>>> wrote:<br>
>>>> You don't have ext-rtp-ip set in your config.<br>
>>>><br>
>>>> Mike<br>
>>>><br>
>>>> On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote:<br>
>>>><br>
>>>>> Hi!<br>
>>>>><br>
>>>>> I have FS natted and am connecting with an 'external' extension that<br>
>>>>> is registered to FS. ie the extension 2000 is registered on the<br>
>>>>> internet with a public IP through my router to FS (192.168.1.120 IP<br>
>>>>> address). uPnP works and I see that the extension is registered<br>
>>>>> successfully.<br>
>>>>><br>
>>>>> The problem is that I do not get any audio<br>
>>>>><br>
>>>>> When looking at the SIP trace, I see the INVITE but do not see a<br>
>>>>> TRYING or RINGING message. The extension is actually ringing. I<br>
>>>>> modified the RTP port range on the remote end to match the RTP ports<br>
>>>>> of freeswitch.<br>
>>>>><br>
>>>>> I have put a sip trace in the pastebin at <a href="http://pastebin.freeswitch.org/11035" target="_blank">http://pastebin.freeswitch.org/11035</a><br>
>>>>><br>
>>>>> If anyone has an idea what needs to be set to get audio, help<br>
>>>>> appreciated.<br>
>>>>><br>
>>>>> Thanks!<br>
>>>><br>
>>>><br>
>>>> _______________________________________________<br>
>>>> FreeSWITCH-users mailing list<br>
>>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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>>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>>><br>
>>><br>
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>><br>
>><br>
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</div></div></blockquote></div><br>