to debug these problems? Since it is intermittent, I am not
always able to reproduce (sometimes the calls work just fine).
Thanks,
Andrew Martin
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- Original Message -
From: Administrator TOOTAI ad...@tootai.net
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
- Original Message -
From: Administrator TOOTAI ad...@tootai.net
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
- Original Message -
From: Guenther Boelter gboel...@gmail.com
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
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
- Original Message -
From: Joshua Colp jc...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 11, 2015 1:24:53 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
- Original Message -
From: Andrew Martin amar...@xes-inc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 11, 2015 1:35:07 PM
Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
calls after
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
Thanks!
Andrew Martin
- Original Message -
From: Andrew Martin amar...@xes-inc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 8, 2015 5:12:28 PM
Subject: [asterisk-users] Retransmission Timeout results in dropped calls
after 32 seconds
seconds
Andrew Martin wrote:
snip
Joshua,
As a mitigation for this problem, could I increase the timerb option in
sip.conf
to a large value, say 1 hour (instead of the default 32 seconds)? What
other
consequences would there be from this change?
I don't know if chan_sip
- Original Message -
From: Andrew Martin amar...@xes-inc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 11, 2015 4:18:58 PM
Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
calls after
32 seconds
Andrew Martin wrote:
Since some packet loss is a possibility, I assume the protocol has
mechanisms
for dealing with it. What should be happening differently in the
communication
when packet loss occurs? Should the phone just be re-sending the OK,
instead of
printing 0
seconds
Andrew Martin wrote:
- Original Message -
snip
Most noteworthy is that the phone seems to send the OK for cseq 103, but it
seems that the asterisk server never received this OK, which is why it kept
re-transmitting the INVITE (103). Is this OK supposed to go
- Original Message -
From: Steve Davies davies...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 13, 2015 11:39:29 AM
Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
calls after 32
- Original Message -
From: John Kiniston johnkinis...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 29, 2015 11:53:13 AM
Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are
- Original Message -
From: John Kiniston johnkinis...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 28, 2015 12:12:05 PM
Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are
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 no
for additional debug information?
Thanks,
Andrew Martin
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http
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
- 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
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