Asterisk 16.1.0
I'm using hagi and SRV records for a "high availability" configuration
of AGI servers. When the first AGI server in the list is completely
down, asterisk quickly moves on to the next one. That is all good.
My concern is what will happen if asterisk can actually connect to
On 11.17.1:
The cli and the log are full of these warnings:
WARNING[12110]: chan_sip.c:4086 retrans_pkt: Timeout on 849421411 on
non-critical invite transaction.
The number is a random 9-10 digits.
What causes them? How do I stop them ?
sean
--
Hi All,
Asterisk 1.4.22.1 on CentOS 5
I've configured my dialplan to limit the maximum call length on
outgoing calls. I've done this as I get the first hour of each call
free with my bundle but I pay through the nose if the call goes over
an hour.
Family members who live overseas sometimes ask
Geoff,
I believe its actually TIMEOUT(absolute)=value. The function name is case
sensitive.
- Logan
On Dec 30, 2012 9:53 AM, Geoff Lane ge...@gjctech.co.uk wrote:
Hi All,
Asterisk 1.4.22.1 on CentOS 5
I've configured my dialplan to limit the maximum call length on
outgoing calls. I've
On Sunday, December 30, 2012, Logan Bibby wrote:
I believe its actually TIMEOUT(absolute)=value. The function name is case
sensitive.
Many thanks. I've changed my dialplan accordingly.
--
Geoff
--
_
-- Bandwidth and
No problem! Doubt check through a test extension. I don't want to be
entirely wrong. ;)
- Logan
On Dec 30, 2012 12:12 PM, Geoff Lane ge...@gjctech.co.uk wrote:
On Sunday, December 30, 2012, Logan Bibby wrote:
I believe its actually TIMEOUT(absolute)=value. The function name is
case
I have not found a solution so I am checking with the Masses here.
I have a client who has a old 5 line key system without voicemail.
Currently, I can set up a huntgroup and ring each line for 15 seconds and after
the 5th line has reached its limit, the call goes to
voicemail.
The
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] timeout with outbound calls
i have tested this solution and i have the same issue
in my case want to call a phone number 06 from my
snom phone (sip223)
the issue still the same
any
Discussion
Subject: Re: [asterisk-users] timeout with outbound calls
the CLI show this :
-- Executing [0678922645@agents:1] Set(SIP/223-6ec45a88,
CALLERID(number)
=520460587) in new stack
-- Executing [0678922645@agents:2]
MixMonitor(SIP/223-6ec45a88
Hi
i want to use timeout with asterisk 1.4 in order to hangup the outbound
calls after 25 sec
i call my mobile number 067xxx from my sip acount 223 and i want to
hangu up the call automatic after 25 sec but there is no hangup after 25
could you please help me
exten =
On Friday 08 Jul 2011, salaheddine elharit wrote:
i want to use timeout with asterisk 1.4 in order to hangup the outbound
calls after 25 sec
i call my mobile number 067xxx from my sip acount 223 and i want to
hangu up the call automatic after 25 sec but there is no hangup after 25
what can i do in order to fix this issue
regards
2011/7/8 A J Stiles asterisk_l...@earthshod.co.uk
On Friday 08 Jul 2011, salaheddine elharit wrote:
i want to use timeout with asterisk 1.4 in order to hangup the outbound
calls after 25 sec
i call my mobile number 067xxx from my
On Friday 08 Jul 2011, salaheddine elharit wrote:
what can i do in order to fix this issue
If and when an absolute timeout occurs, Asterisk jumps to the T extension.
So, in the same context as your 223 extension, you need something like
exten = T,1,NoOp(Absolute timeout triggered)
exten =
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
salaheddine elharit
Sent: Friday, July 08, 2011 6:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] timeout
elharit
Sent: Friday, July 08, 2011 6:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] timeout with outbound calls
Hi
i want to use timeout with asterisk 1.4 in order to hangup
the outbound calls after 25 sec
i call my mobile number
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
salaheddine elharit
Sent: Friday, July 08, 2011 6:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] timeout with outbound calls
Hi
i want to use timeout with asterisk 1.4
Discussion
Subject: Re: [asterisk-users] timeout with outbound calls
i have tested this solution and i have the same issue
in my case want to call a phone number 06 from my
snom phone (sip223)
the issue still the same
any help please
2011/7/8 Eric Wieling ewiel...@nyigc.com
Hi,
I set up an asterisk server that i use with iax accounts. Everything is working
fine, but, for personnal reason i need to insert a home made proxy on my server
which role is to encrypt data.
Here i encountered what i suppose to be a timeout prb : i can register with my
account but i can not
It could be, not sure, that proxy is spoofing ip adress.
http://en.wikipedia.org/wiki/IP_address_spoofing
Jose Flores Galicia
floj...@gmail.com
BriefCode Code Based Training
2010/6/2 isca...@free.fr
Hi,
I set up an asterisk server that i use with iax accounts. Everything is
working
Hi All,
suppose this:
Dial(SIP/somecarrier/somenumber/60/L(360)M(td|${EPOCH})
where 60 is the seconds to wait for the callee (the called party) to answer
L(360) is the absolute limit of the call once it has been answered, in ms
M(td|${EPOCH}) is the macro to execute when the call gets
Hi,
I was testing failover trunk with one IP trunk and 1 E1 trunk
IP trunk is the primary trunk and e1 is secondary.
I block connection to test failover for this system.
I got the msg unreachable for my IP trunk on the system as warning.
Then i tried to dial to an extension outside.
It waited
Solved..
just add qualify=yes to the trunk config.
Hi,
I was testing failover trunk with one IP trunk and 1 E1 trunk
IP trunk is the primary trunk and e1 is secondary.
I block connection to test failover for this system.
I got the msg unreachable for my IP trunk on the system as warning.
Stefan Schmidt s...@sil.at writes:
if i understand you right you have one server (peer) where thousands of
devices are connected and every device is registered to asterisk, and so
every options packet will come from asterisk to this device, right?
If you have a sip routing server like ser,
Benny Amorsen schrieb:
Stefan Schmidt s...@sil.at writes:
if i understand you right you have one server (peer) where thousands of
devices are connected and every device is registered to asterisk, and so
every options packet will come from asterisk to this device, right?
If you have a sip
Klaus Darilion klaus.mailingli...@pernau.at writes:
;timerb=32000 ; Call setup timer. If a provisional response is not
received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1
Thanks! Will try that. Just what I was looking
A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems
to not time out, or at least have a very long time out.
We have a set up where we can dial two different peers, a primary and a
backup peer. If the first one dies completely, so that no error messages
(no ICMP unreachables or
Benny Amorsen schrieb:
A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems
to not time out, or at least have a very long time out.
We have a set up where we can dial two different peers, a primary and a
backup peer. If the first one dies completely, so that no error
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Schmidt
Sent: Monday, June 08, 2009 7:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Timeout when dialing
Danny Nicholas schrieb:
There is a timeout function in the Dial command. The folks who wrote the
command obviously felt that setting a programmatic limit on this would cause
somebody a problem. If you expect a reply from your SIP peer in 30 seconds,
just do Dial(SIP/peer,30) and the line
Stefan Schmidt s...@sil.at writes:
What kind of client cant handle one packet per minute without getting a
higher load?
It isn't a client. It handles thousands of connected devices, so it'll
be handling perhaps 50 OPTIONS packets every second if I go the qualify
route.
What your are
Benny Amorsen schrieb:
Stefan Schmidt s...@sil.at writes:
What kind of client cant handle one packet per minute without getting a
higher load?
It isn't a client. It handles thousands of connected devices, so it'll
be handling perhaps 50 OPTIONS packets every second if I go the qualify
You should look at the queue() command invocation.
Thanks
l.
2009/3/12 Darrin Henshaw dhens...@ignition.bm
Hello,
We had an incident recently where a call was in queue for an extended
period of time. We use queuemetrics for reporting, and it reports that the
call was waiting for 20
Hello,
We had an incident recently where a call was in queue for an extended period of
time. We use queuemetrics for reporting, and it reports that the call was
waiting for 20 minutes. The different thing about it is that the disconnect
reason is stated as Timeout. Is there a set maximum time
Darrin Henshaw wrote:
Hello,
We had an incident recently where a call was in queue for an extended
period of time. We use queuemetrics for reporting, and it reports that
the call was waiting for 20 minutes. The different thing about it is
that the disconnect reason is stated as
I wonder which timeout will apply here: the one in master context or one from
the slave context?
[master]
exten=100,1,Dial(Local/[EMAIL PROTECTED], 20)
[slave]
exten=100,1,Dial(SIP/100, 30)
Thanks
Vadim
___
-- Bandwidth and Colocation Provided by
Hi All;
How to increase the waiting time between entering the digits for the analoge
phone device that is connected to fxs?
Is it by DigitTimeout? But how it will be apply for analoge station if the user
just pickup the handset and dialed the number?
Any help?
Regards
Bilal
On Jun 29, 2008, at 6:35 PM, bilal ghayyad wrote:
Hi All;
How to increase the waiting time between entering the digits for the
analoge phone device that is connected to fxs?
Is it by DigitTimeout? But how it will be apply for analoge station
if the user just pickup the handset and
Hello,
I have a softphone which I am using with Asterisk. Sometimes when I place a
call it works fine and sometimes the SipListener comes back with a timeout.
The timeout is a Retransmission timeout and it seems to be occurring when the
INVITE is sent. The thing is about 70% of the time it
Hi,
The dialin conference via asterisk is over, one person is still in the
conference room and accidentally does not hang up properly. Her meter at
the phone company keeps running...
I'd like to implement something to the effect of checking whether there
is only one participant in the
Way back in the day (1.2.7.1), I did this for a client.
In conf_run() in app_meetme.c, I added this code:
// if an agent abandons a caller, kick the caller after 15 seconds
// check for no agent
if ((conf-isdynamic)
(1 == conf-users)
On 02/01/07 02:15 Olle E Johansson said the following:
both channels should act the same unless there's a configuration that's
giving wrong information
to chan_sip, like you having a username= or defaultip= setting.
how does a username= entry in sip.conf affect dialling behaviour when the
30 jan 2007 kl. 06.38 skrev Yuan LIU:
When Asterisk dials an IAX destination with no registration, it
very quickly comes to the conclusion that it can't make the call
-- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, IAX2/
[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called
When Asterisk dials an IAX destination with no registration, it very quickly
comes to the conclusion that it can't make the call
-- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1,
IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]/[EMAIL PROTECTED]
[Jan 29
Hello everybody,
I want to use the TIMEOUT() function, but in the CLI the show
functions command only shows 7 custom functions:
QUEUEAGENTCOUNT
SORT
CUT
CHECKSIPDOMAIN
SIPCHANINFO
SIPPEER
SIPHEADER
In addition, sometimes I get the debug message function LANGUAGE not
registered.
How can I
Hi,
I'm using IAX2 to connect remote users to my asterisk server. Both
server and user are behind a nat. But sometimes the user registrates
correctly but sometimes doesn't.
Doing a debug i got:
Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13)
Acking anyway
Sending
Hello,
I have found that the timeout 't' takes to much time to be executed,
around 10 seconds.
Is there a place to configure this timeout ?
thanks
--
Victor Moreno
CISL SPAIN, S.L.
Parque Tecnológico de Andalucía
Edif. Bic Euronova
Avda. Juan López Peñalver, 21
29590 Campanillas (Málaga)
Fax
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+timeout
-FD
Hello,
I have found that the timeout 't' takes to much time to be executed,
around 10 seconds.
Is there a place to configure this timeout ?
thanks
___
--Bandwidth and
Hello,I am using Asterisk-java, the Manager. And I have a problem I don't know howto sort it out!:Sometimes, when I send an OriginateAction my code receives an exception withthis message:
Timeout waiting for response to OriginateI don't know what it means as Asterisk receives the action and then
Hello,
Here is part of my extensions.conf.
I set both absolute and response timeouts according to
the day context.
I wish to asterisk hangup after 60s and 10s to play or
replay the annoucement .
Asterisk doesn't jump to T extension.
How can fiox this problem ?
harry
...
[day]
exten =
In extensions.conf I have changed the queue command from:
exten = 1,1,Queue(itsupport|n|||50)
to:
exten = 1,1,Queue(itsupport|tT|||50)
Now it works as it should.
Regards
Wolfgang
Am Donnerstag, 15. September 2005 08:21 schrieb Wolfgang Lumpp:
Am Mittwoch, 14. September 2005 19:12 schrieb
Am Mittwoch, 14. September 2005 19:12 schrieb Sander:
In queues.conf
; How long do we let the phone ring before we consider this a timeout...
;
timeout = 15
This is set in queues.conf
But this is just the function how long the phones will ring you should not
set this option to long or
Hi,
I've setup a queue with 3 sip members.
I've tried with random and roundrobin and different timeout settings in
musiconhold.conf
Always after the second Nobody picked up in 15000ms I get
Exiting on time-out cycle
Stopped music on hold on CAPI/contr1/s-0
Where can I increase this timeout?
: woensdag 14 september 2005 16:25
Aan: asterisk-users@lists.digium.com
Onderwerp: [Asterisk-Users] timeout with queue
Hi,
I've setup a queue with 3 sip members.
I've tried with random and roundrobin and different timeout settings in
musiconhold.conf Always after the second Nobody picked up in 15000ms I
hello,
i've an asterisk box which is connected to an E1/PRI via a TE110P card.
incoming calls from mobile phones where the number is transfered as a
whole block work fine, but when dialing from an analog or ISDN line to
the asterisk box there is a timeout of about 3-5 seconds.
originally my
*?? Have you turned on
debugging? (pri debug span 1).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Günther Starnberger
Sent: Wednesday, June 29, 2005 10:13 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] timeout on incoming PRI call
Good day all
I have my extensions.conf configured so that it waits 8s the answers
with a message saying press 1 for... and 2 for..
How do I tell it then that if the did not press anything to should go to
the operator.
And/Or if they did not press something it will play the message again
And/Or
Hello,
On Fri, 12 Nov 2004 14:40:10 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
I have my extensions.conf configured so that it waits 8s the answers
with a message saying press 1 for... and 2 for..
How do I tell it then that if the did not press anything to should go to
the
Altus Snyman wrote:
Good day all
I have my extensions.conf configured so that it waits 8s the answers
with a message saying press 1 for... and 2 for..
How do I tell it then that if the did not press anything to should go to
the operator.
And/Or if they did not press something it will play the
Tilghman Lesher wrote:
On Monday 03 May 2004 13:56, Frank Mandarino wrote:
I have worked around this issue by storing the extension in a
variable, then restoring it using a Goto in the 'T' processing.
For example:
exten = 411,1,SetVar(ORIG_EXTEN=${EXTEN})
exten =
411,2,Dial(IAX2/[EMAIL
Hans-Henrik Andresen wrote:
Hi,
If I do this in extensions.conf
exten = 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],40,rS(10))
the line is cut of in 10 sec., thats fine, but in CDR I got dst as T,
and not 411.
How can I handle this so I still get kicked of after 10 sec., but
Hello again,
I have noticed with Queues and roundrobin policy that if even if a
timeout is set for a queue, Asterisk keeps ringing an available member
of the queue after the timeout expires. This continues a few times
before the next available agent is tried.
I am using CVS of August 17 but I
hi,
Can anybody pls tell me, how to increase the time gap between 2 digits when you
transfer a call.
ie, the operator answers the call, and presses hash key to transfer, and then enters
the extension
number, some times, it timeouts too quickly before the operator enters the whole
extension
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