[asterisk-users] Call drop on Aastra SIP phones

[Joshua Colp]

Bruno Rocha wrote:
> Hello everybody,

Hola,

> I'm having issues with calls being dropped on Aastra phones, when the
> call is on hold. Tested with models 6863i and 6867i.
> I've figured that the call is dropped by Asterisk when it reaches the
> rtpholdtimeout limit.
>
> I've reported the issue to Aastra, asking them to implement some kind of
> "RTP keep-alive" feature on their phones. Maybe the phone could send
> some RTCP frame (or an empty RTP frame) just to prove it is alive.
> Unfortunately Aastra said the hold behaviour on the phone is correct, as
> per RFC 3264, section 8.4, 4th paragraph:
>
> Typically, when a user "presses" hold, the agent will generate an
> offer with all streams in the SDP indicating a direction of sendonly,
> and it will also locally mute, so that no media is sent to the far
> end, and no media is played out.

They are correct. The "rtpholdtimeout" option stems from a time when it 
was not possible to monitor the signaling of the call and is an 
Asterisk-ism. You've got a few options, though:

1. Increase the rtpholdtimeout
2. Don't use rtpholdtimeout and use SIP session timers instead (check 
the SIP Session-Timers section in sip.conf.sample)

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org