my bad, both sides are generating re-invites. Vitelity ignores any inbound invites to continue call flow. to keep the call going our pbx has to deal with their re-invites otherwise the call terminates at 30 minutes on the dot. Our side is ignoring the inbound invites from vitelity and that causes the call to be torn down.
On 8/10/16 4:21 PM, Matt Fredrickson wrote: > Wait a second, I thought in your original email that you said that > Asterisk was generating reinvites. It sounds now like you're saying > that the remote side is initiating reinvites instead. > > My understanding is that the canreinvite/directmedia option only > influences Asterisk's behavior with regards to generating reinivites. > If it receives a reinvite, I don't think these options will do > anything about that. In fact, I'd guess that not properly responding > to a received reinvite is going to potentially break things from the > SIP perspective. > > Matthew Fredrickson > > > On Wed, Aug 10, 2016 at 4:53 PM, Tammy Firefly <tammy-li...@wiztech.biz> > wrote: >> >> >> On 8/9/16 12:40 PM, Matt Fredrickson wrote: >>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-li...@wiztech.biz> >>> wrote: >>>> Hi All, >>>> >>>> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >>>> split off to where they need to go. We are having a problem getting >>>> chan_sip to quit ignoring re-invites from Vitelity. Our side ends up >>>> sending a reinvite which their side & they do not support us sending a >>>> reinvite. Ive tried: >>>> >>>> canreinvite=no which was supposedly replaced by: >>>> >>>> directmedia=no >>>> >>>> Can anyone shed any light on this matter? I'd love to get this fixed. >>>> >>> >>> Those options *should* influence chan_sip's reinvite behavior - at >>> least they have from my experiences working with chan_sip. Do you >>> know what is triggering the reinvite in the first place, or does it >>> look like a normal media reinvite? >>> >> >> >> every 15 minutes vitelity sends a re-invite to keep the call going. I >> have a packet capture from it if you'd like it feel free to email me off >> list @ tamara.wis...@wiztech.biz >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users