Derek Bolichowski wrote:
HI Michael,
You can set this in sip.conf:
session-timers=refuse
I know of this option - it doesn't help, because the provider ignores it
(on some calls) and the call is dropped anyway.
Normally, there is no problem with the timers. And the problem which
occurred he
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier
Sent: Wednesday, November 30, 2016 12:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped call
Hello all!
I can see a strange problem during invite in dialog in the context of
timer handling.
Given is the following incoming call from provider at 8.195.88.234 (2@2)
to my asterisk at 28.19.57.152 (1@1):
After 900s suddenly *asterisk* starts the timer reinvite - I would have
expected the rei
On 14-01-06 09:27 AM, Nick Cameo wrote:
Hello Everyone,
Just getting in a new cisco router, and would really like to get it up and
running as soon
as possible. Everything is configured from what we can see. This is a NAT
setup.
After 2 seconds on a successfully established call we reach retrans
Hello Eric, I knew this problem all so well however, never knew CISCO sip
alg was enabled by
default. The following settings got us up and going shortly after the email:
no ip nat service sip udp port 5060
ip nat inside source static udp 192.168.2.5 5060 interface Dialer0 5060
access-list 130 pe
, 2014 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped call on new CISCO router for no reason!
Hello Everyone,
Just getting in a new cisco router, and would really like to get it up and
running as soon as possible. Everything is configured from
Hello Everyone,
Just getting in a new cisco router, and would really like to get it up and
running as soon
as possible. Everything is configured from what we can see. This is a NAT
setup.
After 2 seconds on a successfully established call we reach retrans max,
and asterisk
disconnects the call. We
...@intuitiveengineering.com]
Sent: Friday, July 10, 2009 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dropped Call Problem -- Looking for ideas and a
consultant.
Steve,
Thanks for your thoughts. I am tearing out my last bit of hair on this
one.
We only use
l see what I can do to get you
more info.
Connor Spiess
Network Specialist
-Original Message-
From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com]
Sent: Friday, July 10, 2009 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dropped
Spiess
> Network Specialist
>
>
> -Original Message-
> From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com]
> Sent: Friday, July 10, 2009 10:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Dropped Call
> >
> >
> > -Original Message-
> > From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com]
> > Sent: Friday, July 10, 2009 10:58 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] Dropped Call Problem
al Message-
> From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com]
> Sent: Friday, July 10, 2009 10:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas
> and a consultant.
>
> H
ssage-
> From: Mark Engelhardt [mailto:ma...@intuitiveengineering.com]
> Sent: Friday, July 10, 2009 10:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas
> and a consultant.
>
> Hello Eve
: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a
> consultant.
>
> Hello Everyone.
>
> We have:
>
> Asterisk 1.4.21.2
> zaptel-1.4.11
> libpri-1.4.5
> Sangoma A101D Connected to a PRI
> C
to:ma...@intuitiveengineering.com]
Sent: Friday, July 10, 2009 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Dropped Call Problem -- Looking for ideas and a
consultant.
Hello Everyone.
We have:
Asterisk 1.4.21.2
zaptel-1.4.11
libpri-1.4.5
Sangoma
Hello Everyone.
We have:
Asterisk 1.4.21.2
zaptel-1.4.11
libpri-1.4.5
Sangoma A101D Connected to a PRI
Cicso 7960G phones (About 30 of them)
We have a problem with dropped calls that has going on for a long
time. We get up to 5 dropped calls on a bad day. They all seem to be
incoming calls.
Trixbox 2.2 Beta with freePBX 2.2.0rc1
I have a setup that looks something like this in ASCII art:
Teliax IAX Trunk --+
|
V
Embarq PRI > Tandem switch > Ottawa Office Server--+
+-->
Kohler, Jeffrey wrote:
I am encountering an intermittent issue where some of my calls are being
dropped. Most of the calls that are made are successful. However, some
calls will be dropped after having been connected for some time.
Each time a call gets dropped, I get output similar to the fol
I am encountering an intermittent issue where some of my calls are being
dropped. Most of the calls that are made are successful. However, some
calls will be dropped after having been connected for some time.
Each time a call gets dropped, I get output similar to the following in
the Asterisk c
do you check irq misses? what is your set up? os? ,kernel?, full
voip?, voip+tdm?
On Fri, 29 Oct 2004 08:54:42 +0200, Doug Reid -Stormcorp
<[EMAIL PROTECTED]> wrote:
> Hi all
>
> We have had Asterisk drop calls every now and then, does anyone know why
> this happens? It is seldom but does happe
Hi all
We have had Asterisk drop calls every now and then, does anyone know why
this happens? It is seldom but does happen. We have plenty of memory
in the server.
Regards
Doug Reid
Director
Stormcorp Network Solutions (Pty) Ltd
Tel:+27 11 807 1141
Fax:+27 11 807 3504
Mobile: +27 83 989
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