[asterisk-users] OT: Which SIP method to use for this specificbehaviour ?

Olivier oza-4h07 at myamail.com
Mon Nov 5 10:42:41 CST 2007


Thanks for the tip.
If I may ask, do you if this signaling is support in Asterisk 1.4 ?

2007/11/5, Steve Langstaff <steve.langstaff at citel.com>:
>
>  Search for:
>
>      Reason: SIP ;cause=200 ;text="Call completed elsewhere"
>
>
>  ------------------------------
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Olivier
> *Sent:* 05 November 2007 15:22
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] OT: Which SIP method to use for this
> specificbehaviour ?
>
> Hello,
>
> Let SIP extensions 1001 and 1002 belong to an Asterisk calling group :
> whenever an coming call reaches this calling group, both extensions 1001 and
> 1002 receive a SIP INVITE message which makes these 2 phones starting to
> ring.
>
> When a callee picks up his phone, the other extension receives a CANCEL or
> BYE message which stops ringing.
>
> Is there any option you can include in CANCEL or BYE messages so that the
> SIP hardphones would understand it shouldn't have to log this call as it has
> been replied by someone else ?
>
> In other words, is there any Alert-info option which can be used to pilot
> phones call history logs  ?
> I didn't dare to search myself in IETF archives, given the number of
> standards, SIP is now including.
>
> Regards
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071105/c9a8bf21/attachment.htm 


More information about the asterisk-users mailing list