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

[Tobias Wolf]

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