4 Jun
2013
4 Jun
'13
1:07 p.m.
At some point i got session progress and then loads of OKs back from
freeswitch, but either the phone didn't receive or didn't accept it.
the ringing tone would keep on playing in the phone instead. Randomly
the session progress seems to still get processed by the phone, i
would hear freeswitch rtp, but again the phone would often ignore all
the OKs afterwards...
Sometimes it also seemed that kamailio was sending the INVITE to the
phone instead of to freeswitch, or when i played around between
changing $du or $ru the INVITE gets sends to freeswitch but with the
wrong URI pointing to the phone instead of 127.0.0.1:5070 which is
where freeswitch is listening.
I guess it would be easier to reproduce if that random factor wasn't
there, but at least it's failing most of the time, only in different
ways.
I had hoped I could get at least confirmation that it "works here" to
keep me going :P
I will test with xlog when I can test at home again which would at
least exclude NAT issues.
On 6/4/13, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:
Hello,
On 6/2/13 10:57 PM, hiro wrote:
I'm still thinking about this issue and
wondering:
is it even compliant to the RFC to go directly from ringing to session
progress and then OK?
if I understand correctly what you mean, then it is ok from
RFC point
of view. Why you think would be a problem?
Cheers,
Daniel
Because that's what freeswitch is answering
with
when I try to relay the call to it's voicemail when user is busy in
kamailio.
On 6/1/13, hiro <23hiro(a)gmail.com> wrote:
> I tried for multiple hours to operate the debugger, also looking at
> -ddd output at stdout for many days.
> But I'm none the wiser.
> voicemail works from route[location] (e.g. if extension is not
> registered), but not after late errors like busy while ringing.
>
> On 5/23/13, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:
>> On 5/19/13 2:05 PM, hiro wrote:
>>> i'm trying to use the example kamailio.cfg to route to voicemail
>>> server on busy or decline.
>>> Only thing I did was adding decline code to
t_check_status("486|408"),
>>> enabling the preprocessor variable for voicemail and changing the
>>> voicemail host and port to my voicemail server.
>>> No requests arrive on my voicemail server and the dial tone keeps
>>> ringing even if phone is busy and when I decline on the receiving
>>> phone there's a new INVITE sent to the phone directly afterwards.
>>> Am I doing something wrong here?
>> enable debugger module with cfgtrace option and see if the right
>> actions
>> are executed in the configuration file. Maybe there is a
>> misconfiguration or wrong condition somewhere.
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
>> * http://asipto.com/u/katu *
>>
>>
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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *