when you send a call in asterisk, do you see in asterisj cli that call hit you callingcard context or it hit default context ?
On Wed, Jul 17, 2013 at 1:55 PM, Willian Mazzardo - SYSSVOIP < will...@syssvoip.com.br> wrote: > 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 > > -- Dani Popa
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