[asterisk-users] Call drop on Aastra SIP phones

Bruno Rocha bruno at 3gnt.net
Tue Jul 15 09:21:11 CDT 2014

Hi Joshua!

On 2014-07-15 13:39, Joshua Colp wrote:
> 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)

Thanks for the clarification! I will try the SIP Session-Timers.

Bruno Rocha

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