testExtended2.alaw
Unable to open input file: testExtended2.wav
[r...@asterisk testing]# asterisk -rx file convert testLong2.wav
testLong2.alaw
Unable to open input file: testLong2.wav
Any thoughts ?!
Jonas.
On 08/14/2010 04:30 PM, Motiejus Jakštys wrote:
On Sat, Aug 14, 2010 at 4:24 PM, Jonas
Hello list,
it seems that Asterisk is unable to convert a wav-file into an alaw-file :
[r...@asterisk testing]# asterisk -rx file convert testExtended2.wav
testExtended2.alaw
Unable to open input file: testExtended2.wav
[r...@asterisk testing]# asterisk -rx file convert testLong2.wav
Hello list,
is it normal that when adding new moh-files to the directory
/var/lib/asterisk/moh/, asterisk does not see these new files ?!
When I do a moh reload, then Asterisk is aware of the new files...
Is there a solution that does not need a moh reload ?!
Kind regards,
Jonas.
--
On 08/17/2010 08:36 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Subject:* [asterisk-users] Add play moh-files without reload
Hello list,
is it normal that when adding new moh-files
wrote:
On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
intro extended version.wav: RIFF (little-endian) data, WAVE audio,
Microsoft
PCM, 16 bit, stereo 44100 Hz
You need *MONO, 8000Hz*
$ man sox
--
Motiejus Jakštys
-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz
It is still not working :
[r...@asterisk testing]# asterisk -rx file convert test.wav test.alaw
Unable to open input file: test.wav
Jonas.
On 08/14/2010 04:30 PM, Motiejus Jakštys wrote:
On Sat, Aug 14, 2010 at 4:24 PM, Jonas
I have another file that reads :
[r...@asterisk ]# file intro\ extended\ version.wav
intro extended version.wav: RIFF (little-endian) data, WAVE audio,
Microsoft PCM, 16 bit, stereo 44100 Hz
With the same result :
[r...@asterisk ]# asterisk -rx file convert
/var/lib/asterisk/moh/test/intro\
Hello list,
I'm using asterisk 1.4.30 and realtime sip.
I notice that the field musiconhold is not working as when putting
someone on hold, the default musiconhold class is always used.
musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
;
[106002]
or directory
Questions :
1. how can I use AND class default AND class 106002 ?!
2. is it normal that Asterisk can not convert from wav to alaw/gsm ?!
Jonas.
On 08/13/2010 09:57 AM, Jonas Kellens wrote:
Hello list,
I'm using asterisk 1.4.30 and realtime sip.
I notice that the field
1. the converting is not working
[r...@asterisk testing]# file 01Long.wav
01Long.wav: RIFF (little-endian) data, WAVE audio, 20294 channels
1414676809 Hz
[r...@asterisk testing]# asterisk -rx file convert
/var/lib/asterisk/testing/01Long.wav /var/lib/asterisk/testing/01Long.alaw
Unable to
Anyone has an idea of implementation ?!
Jonas.
On 08/10/2010 09:04 AM, Jonas Kellens wrote:
Hello list,
situation :
1. incoming calls come into a queue
2. there is 1 agent logged in into the queue (not always the same agent)
3. when the caller is in the queue, he has the option to quit
Hello.
I notice that when a call that is recorded with MixMonitor is transfered
to another co-worker, the recording ends.
exten = 409,n,Macro(SDstartrecording,external,${DID})
the incoming call then goes to a queue...
[macro-startrecording]
; ARG1 = incoming DID or CALLERID(name)
; ARG2 =
Hello list,
what does the following mean ?
asterisk*CLI sip show peer test13
Mailbox : 1...@default
VM Extension : asterisk
How can this VM Extension be set to the extension of the mailbox ?!
Kind regards,
Jonas.
--
On 08/11/2010 08:39 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Subject:* [asterisk-users] VM Extension : asterisk
Hello list,
what does the following mean ?
asterisk*CLI sip show
Hello list,
situation :
1. incoming calls come into a queue
2. there is 1 agent logged in into the queue (not always the same agent)
3. when the caller is in the queue, he has the option to quit the queue
and leave a voicemail message
what I want :
when there are no more callers in the
On 08/07/2010 01:11 AM, unsero...@aol.com wrote:
Why don't you use 'real' realtime meaning to have your sip peers in your
database?
Then you would not have to do a reload after adding new peers to your db.
And you can still have sip peers additionally in sip.conf.
I have all of my sip
example listed below, but
then I always have that nasty WARNING which I find odd.
I need realtime sip registrations (so without having to do a sip reload).
Kind regards,
Jonas.
On 08/03/2010 10:13 AM, Jonas Kellens wrote:
Hello list,
scrambling different pieces of info together I've come
On 08/06/2010 06:45 PM, Carlos Chavez wrote:
You cannot use realtime static and the other realtime tables at the
same time. You will need to use realtime and then use something like
the EXEC command in sip.conf to execute a script that then pulls the
register statement from your
On 08/06/2010 06:45 PM, Carlos Chavez wrote:
Or use the realtime static table for everything.
What do you mean by everything ?! What is this everything ?!
You mean all the sip options in a database and so no sip.conf file ?!
Kind regards,
Jonas.
--
On 08/03/2010 04:21 PM, Philipp von Klitzing wrote:
Also:
There are at least two implementations of the g726 codec, i.e. g726 and
g726aal2. For this also look at the g726nonstandard setting in sip.conf.
It is quite possible that your problem is here.
I have the following setting in
Hello list,
scrambling different pieces of info together I've come with the following :
I want to have my register = statements in a MySQL-database, so I've
made the following table.
table ast_config :
id 1
cat_metric 0
var_metric 0
commented 0
filename sip.conf
category general
Hello Philipp,
thank you for your answer.
On 08/03/2010 01:21 PM, Philipp von Klitzing wrote:
Question 3 :
How can I get g726 as first preferred codec ??
Which Asterisk version are you using?
Using Asterisk 1.4.30
* check if you have disallow/allow settings in the [general]
Hello list,
Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle.
Grandstream allows for 8 different codec specifications. I have defined
them as 4 x G726 4 x alaw.
Snom allow for 7 different codec specifications. I have defined them as
3 x G726 4 x G729.
The SIP peers
Hello list,
whenever I press the #-key I hear a voice saying 'transfer'. How can I
use the #-key without this voice-message or without having it the
function of unattended transfer ?!
The T or t option is not set in my Dial()-command so I don't know where
this transfer is coming from in the
On 08/01/2010 08:13 PM, Felipe Figueiredo wrote:
uncomment the line blind transfer in features.conf, the default is
'#' and reload features: module reload res_features.so
This helped indeed. Replaced it with some awkward combination.
Thanks.
Jonas.
--
Hello list,
anyone here using Asterisk together with HTB for queing incoming and
outgoing packets ?
I've tried to subscribe myself to the Mailinglist of the Linux Advanced
Routing Traffic Control project, but I get no confirmation. This list
seems dead.
It seems my test case with HTB is
Hello list,
how come when the time is 12:31:18, the GoToIfTime-statement evaluates
to true ??
[Jul 30 12:31:18] -- Executing [...@macro-hours:42]
GotoIfTime(SIP/TELin-0067, 9:00-12:30|fri|*|*?exit) in new stack
[Jul 30 12:31:18] -- Goto (macro-hours,s,58)
The macro jumps to
My problem is that my Asterisk server is sometimes also FTP-server for
uploading of MoH-files. I don't want this FTP-traffic to interfere with
ongoing VoIP-calls. Therefore I would like to give priority to the
RTP-traffic.
I read that there is not really a way of shaping incoming traffic on
Hello list ?!
Is there anyone that can point me to the documentation please ?
I have added a new table like on
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
With the following values :
`musiconhold` (`name`, `directory`, `application`, `mode`, `digit`,
`sort`,
Thank you for your input. It seemed like a good approach, but it
confirms that Asterisk does not see the new MusicOnHold-class :
The dialplan :
exten = 60,1,NoOp()
exten = 60,n,MusicOnHold(testmoh)
The CLI :
[Jul 16 19:40:45] -- Executing [...@from-test:2]
On 07/16/2010 07:43 PM, Carlos Chavez wrote:
Here is what I use:
CREATE TABLE `musiconhold` (
`name` varchar(80) collate utf8_unicode_ci NOT NULL,
`directory` varchar(255) collate utf8_unicode_ci NOT NULL default '',
`application` varchar(255) collate utf8_unicode_ci NOT NULL
music on hold, class 'default', on
SIP/test6-0014
No moh class 'mohtest'...
Reloading, restarting Asterisk does not help...
Would it only work for asterisk 1.6 ?
Kind regards,
Jonas.
On 07/16/2010 08:16 PM, Carlos Chavez wrote:
On Fri, 2010-07-16 at 20:06 +0200, Jonas Kellens wrote
Hello,
has anybody an idea or experience with this realtime moh ?
Jonas.
On 07/14/2010 08:53 PM, Jonas Kellens wrote:
Hello list,
using asterisk 1.4.30.
When setting up the MySQL table 'musiconhold' as described in
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
One-way audio is mostly firewall problem.
Are you behind firewall ?
You can check the audio-ports that are being used in the SDP-message by
doing a /sip debug/.
Maybe you do not have enough UDP-ports open for the audio ?
Jonas.
On 07/15/2010 04:38 PM, Nasir Javaid wrote:
Hi,
I am
On 07/14/2010 08:55 AM, Gordon Henderson wrote:
On Tue, 13 Jul 2010, Paul Belanger wrote:
On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
I have no licenses and I want to avoid transcoding all together.
For terminating a call into Asterisk,
On 07/14/2010 01:39 PM, Gordon Henderson wrote:
And it's nice to have a choice of vendors to buy G729 from now too.
Doesn't help on weedy hardware though.
Gordon
I thought you could only buy licenses from Digium ? Can you install
other G729-licenses on Asterisk ?
I need the
On 07/14/2010 03:41 PM, Gordon Henderson wrote:
It's the default codec used in DECT phones. I trialled it for a while for
some backhaul applications - the users didn't notice anything different
and CPU overhead seemed very low, but I've since gone back to alaw. It
does save 32Kb/sec per call
Hello list,
using asterisk 1.4.30.
When setting up the MySQL table 'musiconhold' as described in
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf ,
what is the meaning of the fields :
`*digit*` char(1) NOT NULL default '',
`*sort*` varchar(16) NOT NULL default '',
Hello list,
when the conversation is using the G729-codec and the conversation is
recorded with the Monitor()-application in wav-format, will there be
transcoding (and thus a need for licenses ?)
Kind regards,
Jonas.
--
_
I have no licenses and I want to avoid transcoding all together.
When the phone supports G729 and the SIP provider support G729, then the
audio can just pass through...
However, in some cases the audio is recorded. Any change that we can
record in G729 format then ??
And how about
Hello list,
using Asterisk 1.4.30.
I want to set the SIP-header Remote-Party-ID to display the name of the
calling party on my phone in stead of the number.
This is the dialplan :
exten = 10,1,NoOp()
exten = 10,n,SIPAddHeader(Remote-Party-ID: eric
sip:1...@192.168.1.150;party=called )
Roger,
your answer did resolve something :
/[Jul 12 15:51:24] -- Executing [...@from-test:2]
SIPAddHeader(SIP/test6-009a, Remote-Party-ID: eric
sip:1...@192.168.1.150;party=called ) in new stack/
However this SIP-header is never send as a SIP-message to the phone from
where I'm
In my case, it shows the name eric and number 20 on the receiving
phone. As if the From-header is overwritten...
That's off course not what I'm trying to accomplish. Therefore I can use
the P-Asserted-Identity (which works well if I may add).
Jonas.
On 07/12/2010 04:05 PM,
On 07/12/2010 05:01 PM, Steve Howes wrote:
On 12 Jul 2010, at 14:09, Jonas Kellens wrote:
I want to set the SIP-header Remote-Party-ID to display the name of the
calling party on my phone in stead of the number.
Am I missing something or is this waht CALLERID(name) and sendrpid
:
On 12 Jul 2010, at 16:35, Jonas Kellens wrote:
On 07/12/2010 05:01 PM, Steve Howes wrote:
On 12 Jul 2010, at 14:09, Jonas Kellens wrote:
I want to set the SIP-header Remote-Party-ID to display the name of the calling
party on my phone in stead of the number.
Am I
Hello list,
asterisk 1.4.30
2 situations in which call-limit should work, but it does not :
[Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The
device state of this queue member, test12, is still 'Not in Use' when it
probably should not be! Please check UPGRADE.txt for
Hello,
using asterisk 1.4.30. No patch or anything else.
I just do the following in dialplan :
exten = 20,n,SIPAddHeader(Remote-Party-ID: Testing
sip:2...@192.168.1.150:5060)
When my Cisco calls my Grandstream, the name Testing appears on the
screen of my Cisco.
When my Grandstream calls
Hello list,
what is the use of realtime SIP peers when you always need to reload the
sip configuration as if you were just putting your SIP peers in sip.conf ??
My SIP peers are now defined in a mysql-DB and when I add a mailbox in
the field 'mailbox', the change is not active untill a do a
Hello,
this is my configuration :
;- REALTIME SUPPORT
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
; source code.
;
On 07/06/2010 12:00 PM, Ishfaq Malik wrote:
On 06/07/10 10:34, Jonas Kellens wrote:
Hello list,
what is the use of realtime SIP peers when you always need to reload
the sip configuration as if you were just putting your SIP peers in
sip.conf ??
My SIP peers are now defined in a mysql-DB
Hello Gareth,
echo also appears when making calls with a SIP phone. These are outgoing
calls.
Another site now also gives feedback on echo, telling they sometimes
also have echo on outgoing calls and if they recall right then sometimes
also on incoming calls (coming from a queue).
This
Hello list,
this is the dialplan :
snip
exten = s,n,Dial(SIP/test1SIP/test2,,t)
snip
exten = 10,1,Dial(SIP/test1)
exten = 20,1,Dial(SIP/test2)
So there is an incoming call that rings SIPaccounts test1 and test2.
Account test1 answers and wants to transfer the call to test2.
Transfer is : #20
Danny,
thank you for you feedback.
I have the following setting in sip.conf :
limitonpeer = yes
and for every sip peer definition I have :
asterisk*CLI sip show peer test1
* Name : test1
Realtime peer: Yes, cached
Secret : Set
MD5Secret: Not set
Context :
Hello list,
this is the setup :
analogue phone -- Grandstream GXW4008 -- Linksys WAG160N --
Asterisk-server (public)
and
Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public)
When calling with an analogue phone + Grandstream GXW and also when
calling with the Zoiper softphone, we
Hello,
I also thought about echo because the Zoiper softphone is used with a
headset. But that didn't explain why the echo also appeared on the
analogue phone + gateway.
I have the same Grandstream GXW 4008 gateway with 5 analoge phones
attached in another environment and there, there are
Hello,
I stated in my first post that both ends hear an echo when one speaks to
the other...
The only place where echo cancellation is being applied is in the
Asterisk server. I have the following in sip.conf :
;-- JITTER BUFFER CONFIGURATION
2010 10:28, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
Hello,
I also thought about echo because the Zoiper softphone is used
with a headset. But that didn't explain why the echo also appeared
on the analogue phone + gateway.
It will present
On 06/30/2010 12:20 PM, Gareth Blades wrote:
So you get echo when calling from the softphone to the analogue phone?
From softphone to analogue phone is echo.
What if they call a regular telephone number?
Calling to a cellphone number or a fixed number on another Telco-network
: echo
Internet Telephony Service Provider = SIP provider. The company that
connects the Asterisk-server via a SIP trunk with the other networks
like GSM, analogue carriers...
Jonas.
By ITSP do you mean a SIP provider?
--
_
Gareth,
multiple users/SIP-accounts use this asterisk server from many
locations. Like I said: in another location with a similar setup, there
are no echo-complaints on received or made calls.
If you say that it has nothing to do with the Cisco-router, I don't
really know what to go looking
Hello list,
I notice on the wiki that it is possible to execute a macro or a gosub
within the queue-command in asterisk 1.6.x
1. Does this mean the macro/gosub is executed everytime a queued call is
answered by a queue member ?
2. I'm using asterisk 1.4.30. Is there a backport or other way
Will turning off the jitter buffer affect the quality of the other calls ??
jbenable = no
I must say I'm not really into these jitter-settings in asterisk. I made
jbenable=yes as it can do no harm...
Jonas.
On 06/30/2010 04:24 PM, Gareth Blades wrote:
Try the SIP phone. If it is better
Taking my first steps into AGI then :
[r...@asterisk agi-bin]# cat sample.agi
#!/usr/bin/php -q
?php
$MYSQLSERVER2=localhost;
$MYSQLUSER2=user;
$MYSQLPASSWD2=passwd;
set_time_limit(30);
require('phpagi/phpagi.php');
$agi = new AGI();
$db=mysql_connect($MYSQLSERVER2, $MYSQLUSER2,
Danny,
1. I only know php, I'm no programmer
3. the query works in normal PHP.
Can I debug to know what's going wrong ?
Jonas.
On 06/30/2010 05:42 PM, Danny Nicholas wrote:
1. (personal preference) I wouldn't use PHP
2. that out of the way, I comment out the AGI stuff and run my
Thank you for your help. It works now. So these were my first steps into
AGI...
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Hello list,
why is it that GoToIfTime thinks a date of **|*|29-*|jun *is not valid ??
[Jun 29 14:06:34] -- Executing [...@macro-vac:10]
*GotoIfTime*(SIP/testcorp-0036, **|*|29-*|jun*?onvac) in new stack
[Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day
'*',
Does the 1.4.26.2-patch also work with asterisk 1.4.30 ??
I have reported a codec-issue, but there is no solution. Will this patch
also answer my question ??
https://issues.asterisk.org/view.php?id=17020
Jonas.
On 06/29/2010 09:42 PM, Mindaugas Kezys wrote:
Try this:
Hello.
I'm using asterisk 1.4.30.
I've found this patch for app_queue.c :
https://issues.asterisk.org/view.php?id=11700
Can I easily implement this by issuing : */wget
'https://issues.asterisk.org/file_download.php?file_id=17192type=bug'
-O - | patch -p0/* ??
Does this mean I have a
Does this mean I have a patched asterisk ? (I ask this because some
applications require a non-patched asterisk version)
Yes.
What is then the unpatched version of Asterisk 1.4.30 ??
Jonas.
--
_
-- Bandwidth
Hello,
my Asterisk CLI is flooded with the following message :
[Jun 25 21:24:57] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit
of 0x409cd8f8 (len 519) to 109.236.137.138:1025 returned -1: Operation
not permitted
[Jun 25 21:25:01] WARNING[15174]: chan_sip.c:1851 __sip_xmit: sip_xmit
of
Hello list,
using asterisk 1.4.30
I have the strangest problem that some SIP accounts can register to my
Asterisk and others not. I see no connection between all those that can
register or all those that can't.
It's not a firewall problem as all register to port 5060 and the range
5060 --
Giorgio,
there is just no registration coming in. SIP debug shows nothing on the
SIP peers that do not register. TCPdump shows nothing on incoming
registrations. Firewall is down.
I'm using SIP realtime and I'm starting to think it is a problem with
the MySQL-DB of the MySQL-driver or
As I said earlier, the firewall was down. I'm using LFDCSF.
If it had anything to do with the firewall, how can the problem be
resolved by recompiling asterisk and asterisk-addons ?
Yesterday everything went well, this night something happened I guess,
and this morning on some locations some
The only solution I see to have a PICKUPMARK-variable created on an
incoming channel, and have the same PICKUPMARK on another created
channel (the one that does the pickup) is to work with a database like
MySQL.
I see no other way to separate multiple incoming channels (with their
own
Hello.
Can I check if a variable contains some string ??
suppose Set(var=foobar)
can I check if ${var} contains foo ??
Like the php-function strstr.
Jonas.
--
_
-- Bandwidth and Colocation Provided by
This also gives result 1 but is not correct :
exten = 1234,n,Set(test2=${REGEX([fot] ${footest})})
exten = 1234,n,verbose(test2 returned ${test2})
[Jun 17 16:59:01] -- Executing [1...@from:7]
Verbose(SIP/13-0096, test2 returned 1) in new stack
[Jun 17 16:59:01] test2 returned 1
Indeed, this gives a correct result.
Thank you all.
Jonas.
Being a PERL Weenie I don't delve into regular expressions any more
than I have to. That caveat given, these changes produced the
desired (and correct?) testing results
exten = 1234,n,Set(test=${REGEX(foo ${footest})})
exten =
If I have one incoming call that rings IPphone-1 and another incoming
call that rings IPphone-2, then the Set(_PICKUPMARK=whatever) is
overwritten.
When I do a pickup I'm always picking up the conversation that rang
IPphone-2.
Is there a way to pick up the first conversation to IPphone-1 ??
exten = start,n,Set(_PICKUPMARK=${CALLERID(num)})
In your example this means that the calling number (for example
3291234567 in Belgium or 49215790657 in Germany) will be set as PICKUPMARK.
If I want to pick up Phone-1 which rings on an incoming call, how can I
know on which incoming number
exten = **XX
-- This is a local extension, a certain phone which is monitored with
BLF-lights. So if I press the button I want the phone call that made
this phone ring, not another phone.
exten = 1234567,n,Set(_PICKUPMARK=${EXTEN:5})
-- If I set the PICKUPMARK-variable the same as the DID
Hello.
This is what I have :
suppose ${SIPaccounts}=SIP/testcorp5SIP/testcorp6
exten = group,1,Set(_PICKUPMARK=${SIPaccounts})
exten = group,n,Dial(${SIPaccounts})
This is what happens when I try to pickup an extension :
[Jun 16 20:39:33] -- Executing [...@sub-routing:13]
Hello list,
using Asterisk 1.4.30.
[Jun 16 21:35:12] -- Executing [...@sub-routing:12]
Dial(SIP/user110-005a, SIP/user2|999) in new stack
[Jun 16 21:35:12] -- Called user2
[Jun 16 21:35:12] -- SIP/user2-005c is ringing
[Jun 16 21:36:12] WARNING[1991]: chan_sip.c:13073
Yes, so I've noticed that I can name _PICKUPMARK anything I want... OK
so the name does not mather and has nothing to do with the different
SIPaccount that it holds...
Another problem is that when another call come in, the _PICKUPMARK
variable is overwritten and I can no longer pick up the
I have read some info on PICKUPMARK and that I need to set this when a
call comes in.
But what happens when there are multiple calls coming in ??
How will Pickup(1...@pickupmark) know which channel to pick up ??
In stead of PICKUPMARK (which is a global variable) I would rather like
to use a
What happens in my extensions.conf is that an incoming call makes a
group of SIPaccounts ring :
[sub-routing]
snip
exten = s,n(group),NoOp()
exten = s,n,Macro(GetGroupDetails,${ganaarID})
exten = s,n,GoToIf($[${sequencenr}==1]?callit)
exten = s,n,Answer()
exten =
Philipp,
thank you for your willingness to help me.
In a previous mail I gave a part of my dialplan: an incoming call rings
a group of extensions/SIPaccounts :
[sub-routing]
snip
exten = s,n(group),NoOp()
exten = s,n,Macro(GetGroupDetails,${ganaarID})
exten =
Philipp,
I have read this wiki, but I still don't know how to implement it in my
example...
The wiki says :
exten = 1234,1,Set(__PICKUPMARK=1234)
So how do I do this with my dialplan :
exten = s,n,Dial(${SIPaccounts},${timeout})
or translated :
exten = s,n,Dial(SIP/testcorp1SIP/testcorp2)
Rob,
it's not a macro but a sub. In my previous post I posted more info, I am
not going to post the whole output every time.
I read on the wiki that you set the PICKUPMARK equal to the extension
for that channel, but in my case I'm not using extensions but multiple
SIPaccounts in my dial
Hello list,
I noticed today that the last logfiles dates 3 days ago !
The logfiles are rotated every night. The logfiles of 2 days ago, 1 day
ago and today are empty !
vps*CLI module show like logger
Module Description
Use Count
0
I don't think it's a disk space issue :
bash-3.2# df -h
FilesystemSize Used Avail Use% Mounted on
/dev/sda1 25G 5.0G 19G 21% /
tmpfs 256M 0 256M 0% /dev/shm
bash-3.2# df -h /var/log/
FilesystemSize Used Avail Use% Mounted on
Hello list,
I try to pick up a ringing extension but nothing works.
To be clear, I'm trying to pick up extension 10.
[Jun 14 17:37:34] -- Executing [*...@from-testcorp:4]
Pickup(SIP/testcorp3-0041, 1...@123456) in new stack
[Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159
Using asterisk 1.4.30
sip.conf is realtime sip_buddies in mysql database. What settings here
affect the Pickup() ?? If you think about pickup/call-groups, I have
none defined.
extensions.conf :
exten = _**XX,1,NoOp()
exten = _**XX,n,Macro(GetKlantIDfromCALLnum,${CALLERID(num)})
exten =
I'm getting a lot of these on the CLI :
[Jun 10 13:41:36] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of
0x1b381bb0 (len 554) to 192.168.12.120:2056 returned -1: Operation not
permitted
[Jun 10 13:41:37] WARNING[4286]: chan_sip.c:1817 __sip_xmit: sip_xmit of
0x1b381bb0 (len 554) to
I have commented out case 5, case 2 and case 3, leaving case 1, 4,6,7,8,9.
But when I press 1 on the menu, I hear: I'm sorry, I did not
understand your response
if (play_auto) {
cmd = '1';
} else {
cmd = vm_intro(chan, vmu, vms);
}
Hello,
I noticed that changes to realtime sip peers are not applied until a
'reload'. A 'sip reload' does not make any changes to realtime sip peers.
When changing for instance the mailbox-parameter in the realtime
sip_buddies table, the change is not applied with a 'sip reload'.
For every
Using 1.4.30
Jonas.
On 06/08/2010 05:53 PM, Zeeshan Zakaria wrote:
On my asterisk 1.4.22-4 sip reload works just fine, and occasionally I
have to use it when sip secrets are updated. What version are you using?
Zeeshan A Zakaria
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Off course...
Jonas.
On 06/08/2010 07:44 PM, Zeeshan Zakaria wrote:
Do you have rtcachefriends=yes in your sip.conf?
Zeeshan A Zakaria
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New to
I made some changes to app_voicemail.c and recompiled asterisk. Now my
caller is only presented with the menu-choice I want.
However, the caller can still give another dtmf-input and be taken to
that specific menu.
How can I disable dtmf-input 2,3,4 if I only want the menu behind option
1
Hello list,
using asterisk 1.4.30 and trying GROUP() and GROUP_COUNT() for the first
time... Having some troubles.
This the dialplan (using a sub) :
exten = s,n,Set(_custID=${custID})
exten = s,n,GROUP(${custID})
exten = s,n,NoOp(grouppcount = GROUP_COUNT(${custID}))
exten = s,n,GoToIf($[
I made your adjustments, but still the same result .
dialplan :
exten = s,n,Set(GROUP()=${custID})
exten = s,n,NoOp(This channel is member of group: ${GROUP()})
exten = s,n,GROUP()
exten = s,n,NoOp(groepcount = GROUP_COUNT(${custID}))
The CLI shows :
[Jun 5 16:50:04] -- Executing
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