I have two servers, each connected to the PTSN via PRI. When I call from site
A (951-999-) to site B (555-1212) and the phone at site B is on the phone,
I hear the normal ring tone for about 20 seconds, then the message all
circuits are busy now. please try your call again latter followed
09, 2014 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] PRI congestion instead of busy
I have two servers, each connected to the PTSN via PRI. When I call from site
A (951-999-) to site B (555-1212) and the phone at site B is on the phone,
I
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 09, 2014 10:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI congestion instead of busy
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 09, 2014 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI congestion instead of busy
I tried changing the dialplan to use Hangup(17
Hi List,
I have a DID number which is routed to my production server. Problem is
that when I dial that DID number from my production number then it's gives
DIALSTATUS to CONGESTION if I don't pick the calls. As per the asterisk it
should reply NO ANSWER.
*extensions.conf *:-
exten =
-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, December 21, 2011 6:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Sam Govind
Subject: [asterisk-users] Why **CONGESTION** not *NOANSWER** ?
Hi List,
I have a DID number which is routed to my
, December 21, 2011 6:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Sam Govind
Subject: [asterisk-users] Why **CONGESTION** not *NOANSWER** ?
Hi List,
I have a DID number which is routed to my production server. Problem is
that when I dial that DID number from my
Hi,
I had a breif telco outage with one of my sip providers.
Is there a way to add failed calls to the cdr aswell as the connected ones?
I was also thinking about having an automated process that monitored congested
calls vs Succesful ones on a carrier and weight the dial plan using
Kenny Watson wrote:
Hi,
I had a breif telco outage with one of my sip providers.
Is there a way to add failed calls to the cdr aswell as the connected ones?
I was also thinking about having an automated process that monitored
congested calls vs Succesful ones on a carrier and
Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 30 June, 2010 11:44:58 AM
Subject: Re: [asterisk-users] Adding Congestion to CDR logs
Kenny Watson wrote:
Hi,
I had a breif telco outage with one of my sip providers.
Is there a way to add failed calls to the cdr aswell
list-aster...@skycomuk.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 30 June, 2010 11:44:58 AM
Subject: Re: [asterisk-users] Adding Congestion to CDR logs
Kenny Watson wrote:
Hi,
I had a breif telco outage with one of my
Blades list-aster...@skycomuk.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 30 June, 2010 12:14:29 PM
Subject: Re: [asterisk-users] Adding Congestion to CDR logs
Using standard AGI will add a fair bit of load and most
Hi,
I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and
try to make a call I get the following error message:
-- Executing [6781...@default:1] Dial(IAX2/iaxy-7477,
DAHDI/g1/96781948)
On Sun, Apr 25, 2010 at 08:37:22PM +0300, Chris Datfung wrote:
Hi,
I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk and
try to make a call I get the following error message:
--
On Sun, Apr 25, 2010 at 8:43 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sun, Apr 25, 2010 at 08:37:22PM +0300, Chris Datfung wrote:
Hi,
I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk
and
try to make a call I get the following error message:
Hi,
I'm using an half T1 line on a asterisk (obviously :)) 1.6.2.4 system,
up to recently everything
was fine but we are starting to experience the call limitation of the
line (15).
So as to warn user of the problem i attached a vocal notification to the
CONGESTION status after a Dial(),
but it
To: Asterisk Users List
Subject: Re: [asterisk-users] Understanding Congestion to incoming caller
2009/11/17 Michelle Dupuis supp...@ocg.ca
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without network load on my trunk. How would I do this?
The voipinfo wiki shows playing a congestion tone to the caller, but that
seems stupid since
2009/11/17 Michelle Dupuis supp...@ocg.ca
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without network load on my trunk.
On which tech, does this trunk rely ?
Is it a SIP trunk ?
How
Just to answer your side issue:
On Sun, Sep 27, 2009 at 04:05:30PM -0500, Andy Howell wrote:
The only Warning or Error I see is when asterisk first starts a new call.
logger.c: -- Starting simple switch on 'DAHDI/1-1'
[Sep 27 15:55:50] WARNING[4199] chan_dahdi.c: Unable to enable echo
Hi!
I had the same problem...
dahdi_transcode command solved the problem!
CheerS!
Ismael.
On Mon, Sep 28, 2009 at 1:25 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Just to answer your side issue:
On Sun, Sep 27, 2009 at 04:05:30PM -0500, Andy Howell wrote:
The only Warning or Error
...@xorcom.com
Subject: Re: [asterisk-users] DAHDI congestion problem
To: asterisk-users@lists.digium.com
Date: Monday, September 28, 2009, 2:25 AM
Just to answer your side issue:
On Sun, Sep 27, 2009 at 04:05:30PM -0500, Andy Howell
wrote:
The only Warning or Error I see is when asterisk
On Mon, Sep 28, 2009 at 04:56:01AM -0700, Landy Landy wrote:
I have a similar problem with DAHDI. If my server gets rebooted, I can't make
any calls until the a call come in from outside. From there I can answer the
call and DAHDI works fine afterwards.
In your case: is the problem reset by
In your case: is the problem reset by restarting asterisk?
'dahdi
resstart'?
The problem does not reset by restarting asterisk.
I've noticed that I can call other sip phones but, when trying to call out, I
get the same (Busy/Congested/Not-Available) congested messege.
On Mon, Sep 28, 2009 at 02:26:29PM +0200, Tzafrir Cohen wrote:
On Mon, Sep 28, 2009 at 04:56:01AM -0700, Landy Landy wrote:
I have a similar problem with DAHDI. If my server gets rebooted, I can't
make any calls until the a call come in from outside. From there I can
answer the call and
: [asterisk-users] DAHDI congestion problem
On Mon, Sep 28, 2009 at 02:26:29PM +0200, Tzafrir Cohen wrote:
On Mon, Sep 28, 2009 at 04:56:01AM -0700, Landy Landy wrote:
I have a similar problem with DAHDI. If my server gets rebooted, I can't
make any calls until the a call come in from outside
On 09/28/2009 01:06 PM, Danny Nicholas wrote:
Funny. The first thing I always do after a reboot is call in from my
cell to make sure things work. But last night I rebooted and immediately
tried dialing out (with a TDM842B) and got:
WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to
On Mon, Sep 28, 2009 at 04:09:43PM -0500, Shaun Ruffell wrote:
On 09/28/2009 01:06 PM, Danny Nicholas wrote:
Funny. The first thing I always do after a reboot is call in from my
cell to make sure things work. But last night I rebooted and immediately
tried dialing out (with a TDM842B) and
I am unable to dial out over a Wildcard TDM400P. This was working previously,
so must have
messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX
2.5.2.2.
When I dial, I see:
-- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1,
DAHDI/g0/9239220,300,) in
new stack
Andy Howell wrote:
I am unable to dial out over a Wildcard TDM400P. This was working previously,
so must have
messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX
2.5.2.2.
When I dial, I see:
-- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1,
Hello, all!
I've noticed a peculiar situation and I am hoping someone can shed
some light on it for me. We have an Asterisk (1.4.18 ) box talking to
the world via Zaptel on a PRI from a telco (USA). I have an extension
that returns busy signal (fast-busy or regular busy) (using US tones).
On Wed, 16 Apr 2008 08:40:42 -0500, Mark Gimelfarb [EMAIL PROTECTED]
wrote:
why do cell phones and Gizmo both detect busy tones and terminate the
call? Is that a standard behavior?
It *is* standard procedure for a cellphone to terminate a call immediately
it discovers that the called number
What country are you in?? Yes, it is common for cell phones to
disconnect the call if they receive CONGESTION, but not BUSY.
Horwich IT Services (Godwin Stewart) wrote:
It *is* standard procedure for a cellphone to terminate a call immediately
it discovers that the called number is busy. It
I'm in the US, so I was originally using the US tones.
Looks like I'm getting a disconnect with both CONGESTION and BUSY. In
fact, I wasn't actually using Congestion() and Busy(), I just did
Playtones() for both of those. There is no reason to send PRI messages
to cell phones, is there? The
Hi, I use:
Trixbox-2.2.4
FreePBX-2.3.1.0
Asterisk-1.2.17
BRIstuffed-0.3.0-PRE-1y-e
Zaptel-1.2.19
..with two ISDN cards, often but occasionally the dial out failed but is
possible to receive external call.
My zapata.conf conf is:
[trunkgroups]
[channels]
language=it
context=from-pstn
Hy all!
Your Asterisk server is return this log :
*CLI -- Executing Dial(Khomp/B0C0, IAX2/*.*.*.*/9834|30|r) in new stack
-- Called *.*.*.*/9834
Mar 14 15:35:40 NOTICE[4212]: chan_iax2.c:2836 auto_congest: Auto-congesting
call due to slow response
-- IAX2/*.*.*.*:4569-1 is
Were getting messages auto_congest:
Auto-congesting . I cant find much information about
what this means, and Ive had a look at the source code but that didnt
help me much. Could anyone point me to a description of auto congestion?
Thanks
David
check incominglimit and outgoinglimit in sip.conf http://www.voip-info.org
On 1/20/06, Kristian Larsson [EMAIL PROTECTED] wrote:
Hey!
I'm having a small problem. I'm using Realtime to
store SIP account information. Dialing works just
fine, but when dialing a person already on the
phone I
Hey!
I'm having a small problem. I'm using Realtime to
store SIP account information. Dialing works just
fine, but when dialing a person already on the
phone I don't get a busy tone.
Eg, Phone 100 calls 200 and they chat with each other
phone 150 calls 100, and gets a regular ringing tone
what I
Tomislav Parcina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
When somebody calls me on fxo4 port * sents that call to SIP 214 phone.
The problem is that when call ends and SIP user hangs up, the line stays
up. Now I don't use Congestion any more. Can sombody tell me do I
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
When somebody calls me on fxo4 port * sents that call to SIP 214 phone.
The problem is that when call ends and SIP user hangs up, the line stays
up. Now I don't use Congestion any more. Can sombody tell me do I
realy need that
Customer is having problems with his internet connection, I
have in my context:
[jimballboutiques]
exten = 1235690251,1,SetGroup(customer)
exten = 1235690251,2,CheckGroup(3)
exten = 1235690251,3,Dial(SIP/jimball,20,r)
exten =
1235690251,4,VoiceMail([EMAIL PROTECTED])
exten =
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