6 thoughts on - Asterisk & Vitelity Invite Issues

• 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.wisdom@wiztech.biz

• 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

• 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

• 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