Basically I have some background noise like keyboard stoke or clicking
sound in random basis, I need to measure that, when I check my IPLC its
fine, and with my Telco service provider its fine...
So am trying to conclude with some solution... trying to identify the root
cause.
Any advice would be
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.7.0.1
DAHDI-Tools-v2.7.0.1
dahdi-linux-complete-2.7.0.1+2.7.0.1
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi
I just checked the Snom phone on my desk and it turned out that the phone has an internal timing
value of 120s for invitations (ringing_time). This value cannot be set using the web interface
for my phone. I also checked a Cisco phone, but I did not find a similar setting. It doesn't
matter, but
sip set debug on and see trace of upload on pastebin.
On Wed, Aug 21, 2013 at 8:25 PM, jg wrote:
> At first I also thought this might be a phone setting. But then I found
> the same 60s to be true for a variety of SIP phones (Snom, Cisco, ...),
> despite the 300s timeout value in the Dial cmd.
At first I also thought this might be a phone setting. But then I found the same 60s to be true
for a variety of SIP phones (Snom, Cisco, ...), despite the 300s timeout value in the Dial cmd.
So it is likely to be Asterisk. The Asterisk Admin Guide says that the default value is 136
years, so th
Does anyone know what knob I need to turn to adjust how long the phone will
ring before giving up?
Mike.
On Wed, Aug 21, 2013 at 8:18 AM, Eric Wieling wrote:
> Asterisk is not timing out. The phone is rejecting the call after 60
> seconds. This is a phone configuration issue.
>
> -Orig
Asterisk is not timing out. The phone is rejecting the call after 60 seconds.
This is a phone configuration issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Wednesday, August 21, 201
On 20/08/13 17:48, Gergo Csibra wrote:
can I echo this variable ?
Like : exten =>> s,n,NoOp(${LAST_INSERT_ID()})
No, this is a mysql query, so:
exten => s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET C1="${ARG1}",
C2="${ARG2}", timestamp="${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)}
Hi,
I think I encountered a bug in the qualify timers for IAX on asterisk
1.8 but I'd like to check if I'm not messing up in my config somewhere
before reporting a bug.
In my IAX peer configuration I have this:
[remote-host]
type=friend
host=172.16.6.45
username=remote-host
secret=test
notran
You do not need to calculate the jitter values yourself. For a quick check you can use the CLI
cmd "sip show channelstats". For external monitoring you could capture the RTCP AMI events.
jg
--
_
-- Bandwidth and Colocation Prov
Hi all,
I've got a user with a couple of Cisco SPA303's. When I dial their phones
with a dial string like:
dial(sip/phone-a,300,rwkxttT)
The phone rings, as expected.
However after exactly 60 seconds, I get:
[Aug 21 02:09:56] -- Got SIP response 480 "Temporarily not available"
back from a
You got to set event off while connecting to AMI to get rid of AMI responses on
each event. There are ways you can suppress the events
http://www.voip-info.org/wiki/view/Asterisk+manager+API
Ask your provider to send 180 instead of 183.
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