I'll change this immediately thanks,
mjr
On Jul 13, 2011, at 11:08 AM, Eric Wieling wrote:
>
>
> Sent from a computer
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>> Mark Rosedale
>> Sent: Tuesday, July 12, 2011 4:33 PM
>> To: Asterisk Users Mailing List - Non-Commercial
So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting mysterious
dropped calls. This only happens on calls that are outbound on Dahdi and mostly
happens in conference calls particularly 8xx-xxx-
This is the output of the hangup.
[Ksebpbx1*CLI>
[0KPRI Span: 1 q931_hangup: oth
s wrote:
> On 07/08/2011 06:07 PM, Mark Rosedale wrote:
>> Looks like the patch is in 1.8.5-rc1...I may just roll with that version. If
>> that doesn't work then I may patch it manually like you suggest.
>
> Not sure where you looked but afaict that patch has not been appl
Looks like the patch is in 1.8.5-rc1...I may just roll with that version. If
that doesn't work then I may patch it manually like you suggest.
Thanks,
mjr
On Jul 8, 2011, at 12:00 PM, Patrick Lists wrote:
> On 07/08/2011 05:01 PM, Mark Rosedale wrote:
>
>> This is not working t
So in troubleshooting a different issue I'm having I decided to upgrade one of
my backup servers to 1.8.4.4 instead of running on the svn branch. Ultimately
this is where I'd like to be in the end.
However, I'm having one issue that seems to be related to this ticket
(https://issues.asterisk.o
iously increase the size of your logging – but should provide
> you with a very thorough trace of the call as its in flight, including the
> SIP dialog between the phone/server.
> Perhaps you can enable the above, place a call that drops, then snip that
> section of the full log and se
fuse
> Resolved the issue by stopping Asterisk from sending these re-invites during
> a live call.
>
> Hope that helps! I have more SIP debugs/logs if they're useful to ya.
>
> JT
>
>
> -Original Message-
> From: Mark Rosedale [mailto:mrosed...@o
What would I be looking for in the logs to indicate that time?
I'm looking into the sip session timers. I believe the issue lies there, but
haven't confirmed that just yet.
On Jul 1, 2011, at 10:31 AM, Jonathan Thomas wrote:
> 900ms?
--
___
Hello,
I'm using Asterisk 1.8-svn branch. I'm having an issue with dropped outbound
calls, particularly outbound conference calls (conference calls are the only
confirmed dropped calls).
The issue is that on my end people will randomly be hung up from the conference
call, upon redialing they
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