[asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds

[Joshua Colp]

Andrew Martin wrote:
> ----- Original Message -----
>> From: "Joshua Colp"<jcolp at digium.com>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com>
>> Sent: Monday, May 11, 2015 12:32:06 PM
>> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls	after 32 seconds
>>
>> Andrew Martin wrote:
>>> ----- Original Message -----
>> <snip>
>>
>>> By doing a number of test calls today, I have managed to reproduce this
>>> while
>>> sip debugging was on, so I have that information available now as well:
>>> http://pastebin.com/ZJqzdvY3
>>>
>>> This was a call from 113 to 146 via a queue. Note that the asterisk server
>>> is
>>> at 10.10.32.251. I see the following:
>>> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
>>> SIP/2.0 180 Ringing
>>> SIP/2.0 180 Ringing
>>> SIP/2.0 200 OK
>>> ACK sip:146 at 10.10.32.96:5062 SIP/2.0
>>> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
>>> SIP/2.0 200 OK
>>> ACK sip:146 at 10.10.32.96:5062 SIP/2.0
>>> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
>>> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
>>> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
>>> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
>>> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0
>>>
>>> This appears to start out with a successful SIP conversation (ending with
>>> the
>>> first ACK), so it is unclear to me why we have two new sets of INVITEs sent
>>> afterwards.
>> Asterisk has sent a re-INVITE to have the media flow directly. The
>> device (seems) to respond with the 200 OK (you can tell based on the
>> CSeq) for the initial INVITE, and not for the re-INVITE. As Asterisk
>> gets no response to its re-INVITE it gives up and terminates the dialog.
>>
>
> Could this perhaps be because the phone doesn't support "bypass" or re-INVITEs?
> Is there a way to disable this functionality and instruct asterisk to just
> stay in the middle of the conversation (bridging or native-bridging) for the
> duration of the call? I thought that setting directmedia=no and
> directrtpsetup=no would disable re-INVITEs and force asterisk to use bridging
> mode, but perhaps something else is required?

That should be all that is required. If that were broken I'd expect 
issue reports to implode - what's the configuration?

-- 
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