My a2billing context [callingcard]
exten => _X.,1,DeadAGI(a2billing.php) Willian Mazzardo Depto TI - SYSSVOIP www.syssvoip.com.br 55 3537 2030 2013/7/17 Dani Popa <dani.p...@gmail.com> > what contex hit invite from opensips ? > > > On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazzardo - SYSSVOIP < > will...@syssvoip.com.br> wrote: > >> Hi Dani ... thanks ... i have for now insecure=very ... my asterisk >> version is 1.4... and this type of setting is for 1.6+ >> >> Willian Mazzardo >> Depto TI - SYSSVOIP >> www.syssvoip.com.br >> 55 3537 2030 >> >> >> 2013/7/17 Dani Popa <dani.p...@gmail.com> >> >>> set opensips peer to insecure=port,invite >>> >>> >>> On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP < >>> will...@syssvoip.com.br> wrote: >>> >>>> Hi Stephens... how do I do this? >>>> >>>> Willian Mazzardo >>>> Depto TI - SYSSVOIP >>>> www.syssvoip.com.br >>>> 55 3537 2030 >>>> >>>> >>>> 2013/7/17 Stephen Vigus <svi...@gmail.com> >>>> >>>>> Hi Willian >>>>> >>>>> You most likely need to configure Asterisk to not authenticate SIP >>>>> requests coming from Opensips. >>>>> >>>>> Regards >>>>> Stephen >>>>> >>>>> >>>>> >>>>> On Wed, Jul 17, 2013 at 3:32 AM, Willian Mazzardo - SYSSVOIP < >>>>> will...@syssvoip.com.br> wrote: >>>>> >>>>>> Hi all.. >>>>>> >>>>>> I know this is a very simple scenario, all PSTN calls be routed to >>>>>> asterisk to do the billing job, but im having some problems, this is my >>>>>> scenario: >>>>>> >>>>>> Sip Client (10.0.0.3) > Opensips (10.1.1.2) > Asterisk (10.1.1.247) >>>>>> ..... > PSTN >>>>>> >>>>>> Calls between sip clients on Opensips are working, but when I try to >>>>>> call over Asterisk, I have Proxy authentication problem. >>>>>> >>>>>> Here is my logs: >>>>>> >>>>>> Opensips: http://pastebin.com/SWpuRHku >>>>>> Asterisk: http://pastebin.com/6jp50LSS >>>>>> >>>>>> [opensips] >>>>>> host=10.1.1.2 >>>>>> type=friend >>>>>> context=callingcard >>>>>> qualify=no >>>>>> insecure=very >>>>>> fromdomain=10.1.1.2 >>>>>> >>>>>> >>>>>> Route: http://pastebin.com/mLgpXiNx >>>>>> >>>>>> Can someone help me on this? >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> Willian Mazzardo >>>>>> Depto TI - SYSSVOIP >>>>>> www.syssvoip.com.br >>>>>> 55 3537 2030 >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users@lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users@lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users@lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> -- >>> Dani Popa >>> >>> _______________________________________________ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > -- > Dani Popa > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
_______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
