I have a process that runs on a server and does a simple 'asterisk -rx
sup show peers' /tmp/peers
and then looks for any (Unspecified) items and reports them as having
lost connection.
My server is running 1.4.43 and the two boxes I am monitoring are also
running 1.4.43.
Once in a great while
If I understand correct you need to increase qualify value.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, May 22, 2012 5:02 PM
To: Asterisk Users Mailing List
On 05/21/2012 11:43 PM, Don Dawson wrote:
Does asterisk support gr-303?
Seems to be undocumented if so.
Yes, chan_dahdi has support for connecting GR-303 channel banks to
Asterisk via T1 spans. It's pretty rare that someone tries to use it,
though, and as you say, there is little
yeah, put qualify=2000 to ensure that you shall get the latency perfectly.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email:
On 05/19/2012 03:52 PM, Tarek Sawah wrote:
Thank you, Any idea how?
Need to be able to control the codecs in use through soem bandwidth
tests. so i need to be able to set the SIP_CODEC and still be able to do
Video.
any suggestions?
Unfortunately you can't do what you want using SIP_CODEC; if
Thanks Kevin and Yaroslav,
Sorry was out of town.
Sorry I forgot to mention that Iam using an VOIP GSM gateway to connect to PSTN.
Kevin,
I am have decided to use Sangoma CPA. Do you know of any other options
that are easier to integrate with?
Yaroslav,
Yes I had set the header ASYNC to yes.
Hi list,
we face a strange behavior on an Asterisk 1.6.2.24: from time to time,
around 5 times a day (time when it happend totaly unpredictable),
asterisk doesn't see incoming INVITE SIP packets, those being catched by
a tshark capture :-(
So incoming calls are CANCELED by other party as
hello
have a problem
have voip-gateway with one ip and 2 ports (1fxo and 1fxs), which
registred on asterisk separately (2 sip accounts, working with 2
different contexts)
sip.conf was written for asterisk 1.4... (where working right) and
replased to 1.8.10 while hard/soft-upgrade of server.
Le 22/05/2012 19:48, Administrator TOOTAI a écrit :
Hi list,
we face a strange behavior on an Asterisk 1.6.2.24: from time to time,
around 5 times a day (time when it happend totaly unpredictable),
asterisk doesn't see incoming INVITE SIP packets, those being catched
by a tshark capture :-(
Le 22/05/2012 22:12, Administrator TOOTAI a écrit :
Le 22/05/2012 19:48, Administrator TOOTAI a écrit :
Hi list,
we face a strange behavior on an Asterisk 1.6.2.24: from time to
time, around 5 times a day (time when it happend totaly
unpredictable), asterisk doesn't see incoming INVITE SIP
Hello,
I was checking how to DELETE old voicemail from Asterisk, for my extension
300, i have 20 MB
[root@pbx INBOX]# pwd
/var/spool/asterisk/voicemail/default/300/INBOX
[root@pbx INBOX]# du -s -h
20M
There are 4 files for each voicemail:
msg.gsm
msg.txt
msg.wav
msg.WAV
I've
On 05/22/2012 04:54 PM, Danny Dias wrote:
There are 4 files for each voicemail:
msg.gsm
msg.txt
msg.wav
msg.WAV
That is perfectly normal. The .txt file is metadata that contains things like
caller ID and duration. Asterisk will also save voicemails into every format
you
Thanks Jason,
But how to delete them? there are a lot of old voicemails, but i don't want
to break the app_voicemail.
2012/5/22 Jason Parker jpar...@digium.com
On 05/22/2012 04:54 PM, Danny Dias wrote:
There are 4 files for each voicemail:
msg.gsm
msg.txt
msg.wav
You can delete old voicemails. Why not your install webvmail?
This is web based GUI for voicemails. You can select and delete from font
end without breaking anything.
http://www.voip-info.org/wiki/view/Asterisk+gui+vmail.cgi
--
regards,
abdul basit
On Wed, May 23, 2012 at 3:03 AM, Danny Dias
I cannot find it
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 2012-05-21 10:25
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
On 23/05/2012 10:46 AM, Ruddy Gbaguidi wrote:
I cannot find it
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny
Nicholas
*Sent:* 2012-05-21 10:25
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:*
This is a hard one to explain. My home PSTN line is connected via an
Openvox A400P card to my Asterisk 1.6.2.23 box which then routes
incoming calls to my 2 SCCP extensions.
The calls are routed just fine, but when a call is answered at one of
the extensions or externally (by a home telephone)
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