to clarify, it doesn't normally happen, just in some rare non-reproducable cases after i played with the config in some very random ways. have been playing to corner that case for many hours, but going to sleep now.
On 6/2/13, hiro 23hiro@gmail.com 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? 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@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@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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users