Hi, The way youve described it seems like you are not routing anything at all to the asterisk. It all depends on your comfiguration on how you handled the call. Somehow Ive a feeling that asterisk is used only for voicemail and is called only once the B party is not found in lookup(location) function. Some cfg file snippet of your can help everyone understand the real cause.
Regards, Sammy On Oct 3, 2015 3:01 AM, "amjad ali" <amjadali...@hotmail.com> wrote: > Hi, > > I have kamailio and asterisk running on same machine. When I make an > internal call, it routes the call to other extension. However, if I stop > the asterisk service then the call still routes the same way to other > extension. Is there a chance that kamailio does the call routing it self. > > Also, I cannot see any sip extension registered at asterisk when I run a > command "Sip show peers" neither there is any activity in Asterisk's log. > > Regards, > Amjad > > _______________________________________________ > 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 > >
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