[asterisk-dev] SIP Missed Calls when calling SIP phones parallel

[Gunnar Schaller]

Hello,
Have a look to my posting in this mailing list of 6th June:
"Snom missed call / completed elsewhere"
Olle created a branch and as I wrote there is a bug in it, you have to
remove the check to p->owner. I testet it and it works good with this
little change.

Gunnar


Thursday, June 21, 2007, 12:54:56 PM, you wrote:

> Hi,

> i have just discovered that, when calling several phones parallel, all
> the telephones that could not answer the call, will record an missed
> call. E.g. Dial(SIP/100&SIP/101).

> While browsing the mailing list it seems, this is a common problem for
> many people.

> I have looked at the code of app_dial.c

> While parsing the data it tries to setup the call for each
> technology-number pair, and it remembers the called channels in a list.

> If one of the called channels answers, it issues an hangup to all other
> channels.

> Is it possible to tell the hangup that the reason the the hangup is that
> the call is answered elsewhere? So the channel could decide if this
> information could the used or not.

> In the case of SIP there could be included an Reson-Header in the
> CANCEL-Message:

> Reason: SIP;cause=200;text="Call completed elsewhere"

> See Posting here:
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg153846.html

> So, basically two changes would be needed.

> Extending the hangup, to include an reason code and in each channel
> deciding, if the information should be ignored or dealt with.

> I wanted to search bugs.digium.com if such a feature is planned, but i
> could't reach it. If there is something similar, I apologize ...

> If not, what do you think. Is this possible ?

> Kind regards,

> Tobias

> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --

> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev