Re: [asterisk-users] Asterisk & Vitelity Invite issues
Could you please write the problem your having and not the reason to the problem Maybe the reason is something else בתאריך 8 באוג׳ 2016 17:25, "Tammy Firefly" כתב: 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. There is no firewall on this machine at all. Thanks --Tammy -- _ -- 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
Re: [asterisk-users] Asterisk & Vitelity Invite issues
On Thu, Aug 11, 2016 at 9:04 AM, Tammy Firefly wrote: > 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. > The 'directmedia' or 'canreinvite' settings only apply to Asterisk generating a re-INVITE to initiate remote packet bridging. Setting that to 'no' will only prevent Asterisk from initiating a re-INVITE to perform said bridging; it won't apply to anything else. There's a whole host of reasons why Asterisk would generate a re-INVITE. That could be due to SIP session timers, or because a change occurred in the party identification via a connected line update. Asterisk will generate re-INVITEs when that happens, and there isn't a setting that will prevent that from happening. Asterisk should have no problem accepting and handling a re-INVITE from a provider, so long as it is formed correctly. If your provider can't accept a re-INVITE being sent to them, there's something seriously wrong with that provider. This is pretty core functionality in any SIP stack. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk & Vitelity Invite issues
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 > wrote: >> >> >> On 8/9/16 12:40 PM, Matt Fredrickson wrote: >>> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly >>> 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
Re: [asterisk-users] Asterisk & Vitelity Invite issues
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 wrote: > > > On 8/9/16 12:40 PM, Matt Fredrickson wrote: >> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly >> 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 -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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
Re: [asterisk-users] Asterisk & Vitelity Invite issues
On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly 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
Re: [asterisk-users] Asterisk & Vitelity Invite issues
On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly 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? -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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
