- Original Message -
> From: "John Novack SCII_U"
> To: "Asterisk Users Mailing List, Non-Commercial Discussion"
> , "Andrew Martin"
>
> Sent: Monday, October 8, 2018 4:29:41 PM
> Subject: Re: [asterisk-users] Dropped calls when all DAH
Hello,
I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x
analog
POTS lines coming into my Asterisk server from the phone company. Internally, I
have about 180 SIP clients defined in sip.conf. What appears to be happening is
that if existing calls are consuming all 8 exter
pouts, or what I should
look at next for additional debug information?
Thanks,
Andrew Martin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
- Original Message -
> From: "John Kiniston"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, July 29, 2015 11:53:13 AM
> Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are
> registered
>
> Wow, Looks like they have really
- Original Message -
> From: "John Kiniston"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, July 28, 2015 12:12:05 PM
> Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are
> registered
>
> In your queues.conf do you have a l
Hello,
I am running Asterisk 11 on CentOS 6.x. I have configured several queues as
follows in extensions.conf:
exten => s,1,Queue(myqueue,rtnC,18)
same => n,Background(user_unavail)
same => n,WaitExten(10)
exten => 1,1,Voicemail(@my-vm,s)
This rings the phones in the queue for 18 seconds. If
- Original Message -
> From: "Steve Davies"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, May 13, 2015 11:39:29 AM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls after 32 seconds
>
> Hi,
>
> In my experience,
- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, May 13, 2015 10:50:02 AM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, May 13, 2015 10:10:25 AM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, May 12, 2015 5:42:57 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Andrew Martin"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, May 11, 2015 4:18:58 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Andrew Martin"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, May 11, 2015 1:35:07 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, May 11, 2015 1:24:53 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls after 32 seconds
>
> > Could this perhaps be be
- Original Message -
> From: "Joshua Colp"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Monday, May 11, 2015 12:32:06 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped
> calls
- Original Message -
> From: "Andrew Martin"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, May 8, 2015 5:12:28 PM
> Subject: [asterisk-users] "Retransmission Timeout" results in dropped calls
>
;Retransmission timeout" problem?
Here is my sip.conf:
general]
directmedia=no
directrtpsetup=no
dtmfmode=rfc2833
context=internal
allowsubscribe=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
localnet=10.10.32.0/255.255.248.0
[123]
secret=11
host=dynamic
type=friend
Th
James,
The WaitExten()s just provide a pause between the two Queue() calls to
let the first group of phones finish ringing. In this example I am ringing
the same group (queue_level_1) twice, however in a real-world scenario I
would ring queue_level_1 and then ring queue_level_2 which each have a
P/265-2931'
This only happens occassionally; most of the time the phones will all
immediately stop ringing once one of them picks up. Do you have any ideas about
what could be wrong here or what else I can do to debug?
Thanks,
Andrew Martin
--
__
- Original Message -
> From: "Guenther Boelter"
> To: asterisk-users@lists.digium.com
> Sent: Tuesday, May 5, 2015 1:05:44 AM
> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
>
> > Looking into it further, in my case it does not appear to be a
> > NATing iss
- Original Message -
> From: "Administrator TOOTAI"
> To: asterisk-users@lists.digium.com
> Sent: Friday, May 1, 2015 6:42:38 AM
> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
>
> Le 01/05/2015 00:05, Andrew Martin a écrit
- Original Message -
> From: "Administrator TOOTAI"
> To: asterisk-users@lists.digium.com
> Sent: Thursday, April 30, 2015 4:43:33 PM
> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
>
> > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and
>
ebin.com/uZSMKczk
What else can I try to debug these problems? Since it is intermittent, I am not
always able to reproduce (sometimes the calls work just fine).
Thanks,
Andrew Martin
--
_
-- Bandwidth and Colocation Provided b
22 matches
Mail list logo
