[Freeswitch-users] Extension: No audio
Mark Campbell-Smith
mcampbellsmith at gmail.com
Sun Nov 8 18:14:48 PST 2009
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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