[asterisk-dev] Asterisk 1.8 and SIP Diversion: header
Pavel Troller
patrol at sinus.cz
Fri Feb 24 12:13:31 CST 2012
Hi Jonathan,
thanks for sending me this. I immediately tested it. The patch, even
it's for the trunk, still applies almost cleanly (just CHANGES file fails)
to the 1.8 SVN branch. And it does exactly what it's supposed to - without
the option in the config file, I have two Diversion: headers, with it I have
just the proper one.
So, I'm voting for putting this to the trunk, but to the 1.8 as well
(otherwise it seems it's a functionality regression from 1.6).
With regards,
Pavel
> Congratulations, you've been drafted to test this!
> https://reviewboard.asterisk.org/r/1769/
>
> ----- Original Message -----
> From: "Pavel Troller" <patrol at sinus.cz>
> To: asterisk-dev at lists.digium.com
> Sent: Friday, February 24, 2012 10:25:12 AM
> Subject: [asterisk-dev] Asterisk 1.8 and SIP Diversion: header
>
> Hi!
> Since 1.6 days, my dialplan is handling adding the Diversion: header to the
> set of SIP headers for a diverted calls. The diversion is handled internally,
> by the dialplan itself (using the database query) and I don't know a method,
> how the destination channel (in this case SIP) can get a diversion information.
> Now, in asterisk 1.8, during tracing SIP calls when searching for another
> problem, I found that now I have TWO Diversion headers. One is my own old one,
> and the second seems to be added by the SIP channel automatically (and I really
> don't know where the channel is getting this info).
> The problem is, that the header added by me is better: it contains a reason
> information (the second contains reason=unknown) and the screening, privacy and
> count information (the second one doesn't contain neither of them). I would
> like to solve this problem by:
> 1) either supplying the missing information (reason, screening, privacy) to
> the SIP channel to be able to build the header properly and stop adding my one,
> 2) or keeping to add my one in its full glory and tell the SIP channel not
> to add its one.
> However, I don't know, how to achieve neither 1) nor 2) (with other means
> than directly patching the source code). Maybe, it would help me to know, how
> the channels gets the info that the call is undergoing a diversion, to mask
> it out of the channel, thus to prevent it from adding the header.
>
> With regards, Pavel
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
More information about the asterisk-dev
mailing list