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...t at > asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. PJSIP - sessions-timers support not working on 13.X (Gosmac) > 2. Re: PJSIP - sessions-timers support not working on 13.X > (Joshua Colp) > 3. Re: adding area code (Chad Wallace) > 4. Re: adding area code (Motty Cruz) > 5. Asterisk 13/PJSIP + registration (Jeremy Kister) > 6. Asterisk 1.8.32.3 chan_sip deadlock (Ishfaq Malik...

...ing and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint. I wonder to force asterisk to refresh the session in some cases when is needed . pjsip is able to refresh the session ? Cheers > On Apr 29, 2015, at 1:50 PM, Gosmac <goseeped at gmail.com> wrote: > > Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers > > I was getting this : > > [A...

...e person managing the list at > asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. PJSIP Video on WebRTC Ast 13 (Gosmac) > 2. Re: res_xmpp.c:3468 xmpp_client_reconnect: (ricky gutierrez) > 3. Re: Asterisk 13 : SILK codec ? (Steve Murphy) > 4. Re: Asterisk switching bridge to native_rtp even with > direct_media=no (Matthew Jordan) > 5. Re: Asterisk 13 : SILK codec ? (Matthew Jord...

Hi guys i was trying to get working sessions-timer over PJSIP channel i was trying to see what is supported or not about this features on the new pjsip channel since chan_sip was kind of flexible on this , at the moment since wiki says pjsip support 4 modes of operation (forced, no, required, yes) but if i try to change any of the timers parameters (timers, timers_min_se or timers_sess_expiries)

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine

The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

Thanks for replying Joshua. Cheers, Javier Riveros