[asterisk-users] Transfer on RTP timeout?

[Dinesh Nair]

On 01/28/07 18:52 Florian Overkamp said the following:
> Nokia seems to have done something like this in their E-series (E60 etc) 
> with Avaya and Cisco. Anyone have a lowdown on the technical stuff there ?

i think that's a FMC (fixed mobile convergence) client which both avaya and 
cisco wrote for the E series platform. my stock E61 doesn't have such a 
client, though it has the SIP 2.0 symbian client.

as for the original poster, what you can probably do is to trap the hangup, 
and perhaps modify app_dial.c to set the hangup cause in DIALSTATUS for RTP 
timeouts, then take appropriate redialling action as part of the h extension.

do note that this is off the cuff, and i'm not sure how difficult it'd be 
to do this.

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh at alphaque.com                (0 0)   http://www.openmalaysiablog.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do                                        |
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done                                                              |
+=========================================================================+