[Freeswitch-users] Extension: No audio

Mark Campbell-Smith mcampbellsmith at gmail.com
Sun Nov 8 20:18:14 PST 2009


I think I've fixed it, but I had to change a few things...

I had a host name set in vars.xml for external_rtp_ip and for
external_sip_ip.  Having the external_rtp_ip set to a hostname, sofia
showed the

RTP-IP                  192.168.1.120
Ext-RTP-IP              host:myhostname
SIP-IP                  192.168.1.120
Ext-SIP-IP              124.190.249.9

I think this caused some problems.  Once this was changed back to
stun, I now get RINGING messages and I get audio.

I still have ext-rtp-ip and ext-sip-ip set to auto-nat in internal.xml.

Could this be the cause or is there something else that caused this
issue?  I am using FreeSWITCH Version 1.0.trunk (15126)



On Mon, Nov 9, 2009 at 2:40 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> 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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>>
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