On 04-11-14 11:50, Ishfaq Malik wrote:
On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
Hello,
I have 5 Asterisk servers all using mysql realtime to store queue
log information.
There is 1 out of 5 servers which stores
Hello,
I read on the wiki :
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
*${HASH(SIP_CAUSE,channel-name)}*. Additionally make sure you're using
the destination channel, not the source channel.
But when I use this in my dialplan, this 'variable' is empty.
Dialplan
Hello,
is it possible to reload just a context in stead of the whole dialplan ?
I see this on the tracker :
https://issues.asterisk.org/jira/browse/ASTERISK-19934
But is it possible in some Asterisk version ?
Kind regards,
Jonas.
--
Hello,
I have added the following to the peer definition :
ignorecryptolifetime=yes
But still Asterisk tells me :
[Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244
sdp_crypto_process: Crypto life time unsupported: crypto:1
AES_CM_128_HMAC_SHA1_80
On 09-10-14 14:11, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
Kia ora,
I have added the following to the peer definition :
ignorecryptolifetime=yes
This is not an option within the official tree so unless you've added
a patch this won't actually do anything.
But still Asterisk
On 09-10-14 14:28, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
any idea where and what to change in the source code then ?
I am able to change the source code, but to do minimal damage I would
like to know where to change what exactly.
Yes. In channels/sip/sdp_crypto.c where the line
On 07-10-14 12:32, Jonas Kellens wrote:
Hello,
I am trying to setup a Grandstream GXP2160 IP-phone with secure
calling (SRTP).
Secure signaling SSIP for registration is working great !
I follow this guide :
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
But when I try
Hello,
I am trying to setup a Grandstream GXP2160 IP-phone with secure calling
(SRTP).
Secure signaling SSIP for registration is working great !
I follow this guide :
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
But when I try to make a call with SRTP, I get stuck.
Hello,
I have a situation where a call comes in to my Asterisk server B. This
call comes from another Asterisk server A. I want to tell to this server
A why my server B hangs up.
So just before hanging up, I add a custom SIP-header :
exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten =
On 02-09-14 11:34, Steven Howes wrote:
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
So just before hanging up, I add a custom SIP-header :
exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten = s,n,Hangup()
SIPAddHeader only works
On 02-09-14 14:22, Eric Wieling wrote:
Try Hangup(123) where 123 is whatever hangup cause you want to send
back to the caller. The calliing Asterisk server will get the valuse
back in HANGUPCAUSE variable.
Hello,
I have tried sending Hangup(321) on Asterisk server B to Asterisk A but
Hello,
If the phone of my colleague rings, I can see this with BLF-lamps on my
Snom IP-phone. I would also like to see *_who_* is calling. I would like
to see the external number on my screen so I can choose whether to
pickup the call with BLF.
Therefore I have in sip.conf : notifycid = yes
Hello,
I notice that Asterisk always sends Subscription-State: active, even
when the SIP-peer is offline (IP-phone cut from power) :
/[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer:
Peer 'testacc77000' is now UNREACHABLE! Last qualify: 49//
//[Jul 31 11:56:58] Really
On 31-07-14 12:13, Joshua Colp wrote:
Jonas Kellens wrote:
I notice that Asterisk always sends Subscription-State: active, even
when the SIP-peer is offline (IP-phone cut from power) :
/[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer:
Peer 'testacc77000' is now
On 31-07-14 14:28, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
I read on Yealink support that Yealink IP-phones expect
Subscription-State:terminated for there Presence/BLF-functionality.
So how can I get Subscription-State:terminated on Asterisk ?
That would be a bit strange
On 31-07-14 15:06, Joshua Colp wrote:
Jonas Kellens wrote:
On 31-07-14 14:28, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
I read on Yealink support that Yealink IP-phones expect
Subscription-State:terminated for there Presence/BLF-functionality.
So how can I get Subscription
On 31-07-14 16:14, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
I was reading this post :
http://forum.yealink.com/forum/showthread.php?tid=894
http://forum.yealink.com/forum/showthread.php?tid=894pid=4794#pid4794
Has the explanation.
Since they are using dialog-info+xml there's
2, 2014 at 4:39 AM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
Hello,
I am trying to create a dynamic call parking lot using
https://wiki.asterisk.org/wiki/display/AST/Application_Park
But this manual is not enough to fix my problem : Asterisk
Hello,
I am trying to create a dynamic call parking lot using
https://wiki.asterisk.org/wiki/display/AST/Application_Park
But this manual is not enough to fix my problem : Asterisk keeps trying
to park the call in the default parking lot :
[Jul 2 11:32:14] -- Executing
Hello,
how can I create the following scenario :
I have a Call Queue and I want to play an announcement, but only once
after about 10 seconds.
The current option |periodic| |-| |announce| |-| |frequency| keeps on
playing the announcement indefinitely. (it should have an option 'once'
like
Hello,
using asterisk 1.8.12.2 and realtime architecture with mysql.
I get the following message on CLI when changing the value in the strategy
/[Mar 26 11:02:24] WARNING[10648]: app_queue.c:2030 queue_set_param:
Changing to the linear strategy currently requires asterisk to be
restarted.//
Hello,
I execute the following php script when a call ends and the h-context is
executed :
/exten = h,n,System(/usr/bin/php
/var/log/asterisk/loggingAST/loggingAST.php
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)/
The script loggingAST.php writes some information in a MySQL database on
On 13-02-14 17:33, Steven Wheeler wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, February 13, 2014 7:12 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re
On 12-02-14 16:58, Steven Wheeler wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Wednesday, February 12, 2014 3:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject
Hello,
I'm using MySQL realtime Call Queues (table /queues/ and table
/queue_members/).
I would like to ring the members of the call queue in a certain order.
Therefore I use ring strategy /lineair /and I put the members into the
table /queue_members/ in the order in which they have to be
Hello,
is there a mailinglist where I can post questions regarding Digium
IP-phones ?
I have the following question :
I'm trying to provision a Digium D70 IP-phone from a https provisioning
server.
The Digium D70 contacts the provisioning server correctly but seems to
log in with the
Hello,
I see the strange behaviour that outgoing calls end after 15 minutes.
I didn't knew there is some kind of call duration limit that can be set ?
Is there ?
Using Asterisk 1.8.12.2
Kind regards,
Jonas.
--
_
--
On 08-01-14 16:47, Markus wrote:
Am 08.01.2014 16:07, schrieb Jonas Kellens:
Hello,
I see the strange behaviour that outgoing calls end after 15 minutes.
I didn't knew there is some kind of call duration limit that can be
set ?
Is there ?
Look at session-timers in sip.conf. I had to set
Hello,
I am getting the following error when compiling dahdi :
make[2]: Entering directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64'
Building modules, stage 2.
MODPOST 0 modules
make[2]: Leaving directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64'
make -C
Hello,
I have the following construction :
Provider -- SipAgent (asterisk) -- Asterisk Server_A -- IP-phone
(Snom 370)
If a call comes in from the Provider to my SipAgent, then my SipAgent
send the call to the correct Asterisk Server_A (dialplan logic based on
number). The Asterisk
Hello,
What does it mean when rtp set debug ip shows RTP packets that have
been send, but there is no audio ?
There was no audio on my call in both directions, but rtp set debug
shows that there were RTP packets send.
There is no firewall active on my Asterisk server :
[root@sip
On 28-11-13 11:45, Jonas Kellens wrote:
Hello,
What does it mean when rtp set debug ip shows RTP packets that have
been send, but there is no audio ?
There was no audio on my call in both directions, but rtp set debug
shows that there were RTP packets send.
There is no firewall active
Hello,
Using asterisk 1.8.24 on CentOS 6.4
I notice that the asterisk process is using between 105 en 110 % CPU :
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk
2682 mysql 20 0 627m 29m 6204 S
On 27-11-13 12:26, Jonas Kellens wrote:
Hello,
Using asterisk 1.8.24 on CentOS 6.4
I notice that the asterisk process is using between 105 en 110 % CPU :
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk
voicemail...
How can I find out if there is trancoding ??
Kind regards,
Jonas.
On 27-11-13 13:27, Andrew Colin wrote:
Are you transcoding?
What is your server spec?
Regards
Andrew Colin-mobile
Vsave(PTY)Ltd
Original message
From: Jonas Kellens
Date:27/11/2013 13:48 (GMT+02
Hello,
I have installed asterisk 1.8.24 (from source) but I can not start up
Asterisk :
[root@sip32 admin]# /usr/sbin/asterisk -r
Illegal instruction
[root@sip32 admin]# /sbin/service asterisk status
asterisk dead but subsys locked
[root@sip32 admin]# /sbin/service asterisk restart
Stopping
Hello,
I have installed asterisk 1.8.24 (from source) but I can not start up
Asterisk :
[root@sip32 admin]# /usr/sbin/asterisk -r
Illegal instruction
[root@sip32 admin]# /sbin/service asterisk status
asterisk dead but subsys locked
[root@sip32 admin]# /sbin/service asterisk restart
Stopping
Jonas.
On 20-11-13 14:11, Ron Wheeler wrote:
Is it possible that in your build you mixed 32 bit and 64 bit libraries?
Ron
On 20/11/2013 8:06 AM, Jonas Kellens wrote:
Hello,
I have installed asterisk 1.8.24 (from source) but I can not start up
Asterisk :
[root@sip32 admin]# /usr/sbin
, Jonas Kellens wrote:
Hello,
how can I mix libraries ?
I have installed prerequisites from yum and asterisk from source
(make make install).
My kernel :
[root@sip32 asterisk-1.8.24.0]# uname -a
Linux sip32.domain.tld 2.6.32-358.23.2.el6.x86_64 #1 SMP Wed Oct 16
18:37:12 UTC 2013 x86_64
be caused by that.
You might want to check the build logs to be sure that you do not have
a 32 bit library installed.
32 bit libraries will work on 64 bit Linux but not when mixed with 64
bit applications.
Ron
On 20/11/2013 8:15 AM, Jonas Kellens wrote:
Hello,
how can I mix libraries ?
I
On 20-11-13 14:43, A J Stiles wrote:
On Wednesday 20 November 2013, Jonas Kellens wrote:
Hello,
I have installed asterisk 1.8.24 (from source) but I can not start up
Asterisk :
[root@sip32 admin]# /usr/sbin/asterisk -r
Illegal instruction
Are you using a VIA C6/C7 processor (often found
Mohammad wrote:
Hello,
you can check the asterisk binary with.
file /usr/sbin/asterisk
and linked library
ldd /usr/sbin/asterisk
On Wed, Nov 20, 2013 at 2:51 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 20-11-13 14:43, A J Stiles wrote
:
Hello,
you can check the asterisk binary with.
file /usr/sbin/asterisk
and linked library
ldd /usr/sbin/asterisk
On Wed, Nov 20, 2013 at 2:51 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 20-11-13 14:43, A J Stiles wrote:
On Wednesday 20
Hello,
when calling a group of SIP peers like this :
Dial( SIP/inno0SIP/inno4SIP/inno6,30)
is it possible to have a SIP header added for just 1 of these SIP peers,
like only for SIP/inno0 but not for SIP/inno4 and SIP/inno6 ??
I know the function SipAddHeader(), but when I use this in the
On 11/13/2013 11:48 AM, Johan Wilfer wrote:
2013-11-12 17:42, Jonas Kellens skrev:
X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%)
0. 000136 00 ( 0.00%) 0.0031
X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0.
007301 00
Hello,
can I use include-statements in the calendar.conf configuration file ?
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Hello,
what could be causing the issue of poor sound quality ? Some calls,
certainly not all of them, sound like if the caller is standing next to
a very busy road with lots of cars passing.
To be clear : the person calling is not standing next to a highway.
But there seems to be a noise on
On 11/12/2013 04:29 PM, jg wrote:
Did you have a look at the codecs that are involved?
There are about 40 à 45 simultaneous calls (using G711a).
Jonas.
--
_
-- Bandwidth and Colocation Provided by
say more about network problems, but first let's see what
channelstats says.
jg
Am 12.11.2013 16:34, schrieb Jonas Kellens:
On 11/12/2013 04:29 PM, jg wrote:
Did you have a look at the codecs that are involved?
There are about 40 à 45 simultaneous calls (using G711a).
Jonas
problems.
I could say more about network problems, but first let's see what
channelstats says.
jg
Am 12.11.2013 16:34, schrieb Jonas Kellens:
On 11/12/2013 04:29 PM, jg wrote:
Did you have a look at the codecs that are involved?
There are about 40 à 45 simultaneous calls (using G711a
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
If for instance an incoming call makes 10 IP-phones ring, does this mean
that Asterisk preserves 10 x 2 RTP ports for audio ?
I guess Asterisk sends in the SIP INVITE an SDP body with an RTP
On 10/29/2013 05:14 PM, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
short question : does Asterisk reserve RTP ports for every IP-phone that
is being called ?
It uses 2 ports per channel under normal circumstances, 1 for RTP and
1 for RTCP.
If for instance an incoming call makes 10 IP
', =, 'context2', '');
INSERT INTO `my_extensions_table` VALUES (NULL, 'includecontext',
'include', =, 'context3', '');
This would then replace the following in extensions.conf :
[includecontext]
include = context1
include = context2
include = context3
Possible or not ?
Thanks,
Jonas Kellens
Hello,
I want a caller who is waiting in the queue to be able to exit this
queue (and the waiting) by pressing a digit.
I read in the wiki :
/Context//
//; A context may be specified, in which if the user types a SINGLE
digit extension while they are in the queue, they will be taken out of
Hello,
I have defined that I want to receive audio (RTP) on port 11500 till
11954 (rtp.conf).
The same range I have defined in my firewall.
I now see that an IP-address gets blocked by my firewall because there
are packets coming onto port 11955.
How come the client sends audio on port
On 09/13/2013 11:41 AM, Andrew Colin wrote:
Normally you should open ports 1-2 udp
On 9/13/2013 11:37 AM, Jonas Kellens wrote:
I now see that an IP-address gets blocked by my firewall because
there are packets coming onto port 11955.
Why do I need such a big range ? That's like
, Jonas Kellens wrote:
On 09/13/2013 11:41 AM, Andrew Colin wrote:
Normally you should open ports 1-2 udp
On 9/13/2013 11:37 AM, Jonas Kellens wrote:
I now see that an IP-address gets blocked by my firewall because
there are packets coming onto port 11955.
Why do I need such a big
Could be... is there no way to be sure ? Is there no way to calculate this ?
Thanks,
Jonas.
On 09/13/2013 12:11 PM, Johann Steinwendtner wrote:
Maybe you should open 11955 on you fw as well. This could be the rtcp
port.
Regards
Hans
On 2013-09-13 11:49, Jonas Kellens wrote:
Hello
Hello,
how can I cut off the last character of the EXTEN-variable with
variating length ?
So I have :
112233#
123#
123456789#
I want to cut off the last character.
${EXTEN:-1} gives me #, but that is the character I want to cut off.
Kind regards,
Jonas.
--
On 08/20/2013 10:47 AM, jg wrote:
How about ${EXTEN:-1:1}?
The Definitive Guide has a special paragraph with the title *More
Advanced Digit Manipulation.*
jg
Same result : #
Jonas.
--
_
-- Bandwidth and Colocation
On 08/20/2013 10:40 AM, Gareth Blades wrote:
On 20/08/13 09:29, Jonas Kellens wrote:
Hello,
how can I cut off the last character of the EXTEN-variable with
variating length ?
So I have :
112233#
123#
123456789#
I want to cut off the last character.
${EXTEN:-1} gives me
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
On 08/20/2013 06:03 PM, Gergo Csibra wrote:
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT
Hello,
is it possible to use the #include - syntax to include several
configuration files situated in one directory ?
Something like :
extensions.conf :
#include extra/*
#include addons/*
Is this possible ?
Using asterisk 1.8
Thanks.
Jonas.
--
Hello,
I notice that it takes 4 to 6 seconds between someone pressing a cipher
and Asterisk continuing inside the dialplan. How come ???
Taken from verbose logfile :
(attempt 1)
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin '1' received on
SIP/SipAgenT01-1eb0
[Jun 11 15:29:25]
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why doesn't Asterisk continue immediately inside the dialplan after having
received
On 06/11/2013 04:39 PM, Richard Mudgett wrote:
On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between
On 06/11/2013 04:46 PM, Eric Wieling wrote:
The only way to resolve this is to redesign your dialplan so you do not have
ambiguous matching, This is not an Asterisk issue, this is an issue with the
way you designed your dialplan and would apply to any IVR on any system.
I understand that I
Hello,
when picking up an incoming call from one ip phone on another ip phone,
the call terminates after about 5 to 10 seconds.
When reading out the hangup cause variable in the h-extention of the
dialplan, the hangup cause seems to be 111.
In the dialplan output, you can see that
On 04/02/2013 05:42 PM, Matthew Jordan wrote:
On 04/02/2013 06:37 AM, Jonas Kellens wrote:
On 04/02/2013 12:50 PM, A J Stiles wrote:
(Message re-ordered for readability. The beginning is *not* the right place
for your response -- answers come *after* questions, or *between* points
Hello,
when compiling dahdi (CentOS 2.6.18-348.3.1.el5) I get the following error :
In file included from
/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xpd.h:26,
from
/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/card_bri.c:29:
Hello,
can you use Progress() in the dialplan for outgoing calls ? For example
just before the Dial()-command ?
Is there a risk involved when using the Progress()-command ?
Kind regards,
Jonas.
--
_
-- Bandwidth and
Hello,
any idea why the Asterisk CLI gets flooded by these messages ? How can
the SIP peer /vita3/ cause this flood ?
[Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] --
SIP/vita3-10af requested media update control 26, passing it to
SIP/708708-10b3
[Apr 2 11:45:48]
The SIP peer vita3 is a realtime sip peer, installed in a hardware
IP-phone (Siemens Gigaset N510 pro).
Jonas.
On 04/02/2013 12:35 PM, A J Stiles wrote:
On Tuesday 02 April 2013, Jonas Kellens wrote:
Hello,
any idea why the Asterisk CLI gets flooded by these messages ? How can
the SIP peer
On 04/02/2013 12:50 PM, A J Stiles wrote:
(Message re-ordered for readability. The beginning is *not* the right place
for your response -- answers come *after* questions, or *between* points.)
On Tuesday 02 April 2013, Jonas Kellens wrote:
On 04/02/2013 12:35 PM, A J Stiles wrote:
On Tuesday
Hello,
what is the equivalent parameter of X in the ConfBridge()-command ?
How can you exit ConfBridge by pressing a digit ?
Concerning MeetMe() :
Verbosity is 25 and I still don't see anything on the console or in
the logs when pressing '0' (zero).
Kind regards,
Jonas.
On 02/20/2013
Hello,
using Asterisk 1.8.12.2
I am having trouble with exiting the conference room by entering a
single digit.
option X of the Meetme()-application should do this.
I have following in extensions.conf :
/exten = _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)//
//exten =
: Jonas Kellens jonas.kell...@telenet.be
But nothing happens when pressing 0 (zero).
Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log
enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it.
You can also push those to the console
:32 PM, Rusty Newton wrote:
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
But nothing happens when pressing 0 (zero).
Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log
enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type
Hello,
using Asterisk 1.8.12.2
case :
I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer
to my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other
side.
Receptionist transfers
Hello,
thanks you for your answer.
The IP-phones in this case are Yealink T32G.
What setting is needed in this IP-phone ?
Jonas.
On 02/04/2013 02:29 PM, Steven Howes wrote:
On 4 Feb 2013, at 12:53, Jonas Kellens wrote:
I call with my cellphone to our public telephone number
Our
Hello,
and is there any setting in Asterisk to turn this functionality on/off ?
Maybe mine is not enabled.
Jonas
On 02/04/2013 03:30 PM, Steven Howes wrote:
On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
The IP-phones in this case are Yealink T32G.
What setting is needed in this IP-phone
Hello,
at certain time inside my dialplan I would like to have an external php
script executed. Asterisk should not wait for the end of the execution
to continue with the rest of the dialplan. It should just start the
execution of the php script (which inserts an entry into a remote
vote for system() on two accounts. #1 AGI requires more
overhead and protocol #2 you are not expecting a result to return to
the dialplan.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Wednesday, January 23
and
hangup. The failure of Jonas.php due to database or any other problem
would not affect the execution of the dialplan.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Wednesday, January 23, 2013 8:32 AM
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Wednesday, January 23, 2013 8:54 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Execute a script outside Asterisk
Hello
Hello,
what exactly is the function of the parameter 'sayduration' in the
voicemail box configuration ?
Whether I put this to 'yes' or to 'no', nothing changes. I do not get
the announcement of duration at the beginning of the voicemail message.
Kind regards,
Jonas.
--
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 5:36 AM
*To:* Asterisk Users
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 9:00 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer
until you try it on your box.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 9:15 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set
are using so that I can verify whether or
not my changes could have affected you.
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, December 8, 2012 5:55:39 AM
Subject
it might work...
How come app_queue is suddenly so unstable ?
Which version has a stable app_queue ?
I thought unstable versions are released with rc- added ?
Kind regards,
Jonas.
On 09-12-12 19:19, Jonathan Rose wrote:
Jonas Kellens wrote:
Hello,
using Asterisk 1.8.12.2
I think
On 09-12-12 19:49, Joshua Colp wrote:
Jonas Kellens wrote:
it might work...
Without labbing things up with your exact scenario Jonathan can't
confirm it. I did a quick search of the issue tracker for anything
open similar to the issue you specified and nothing came up. The
functionality
On 09-12-12 20:10, Joshua Colp wrote:
Jonas Kellens wrote:
On 09-12-12 19:49, Joshua Colp wrote:
As well - if the log you provided has not been altered then you are
attempting to add an interface member3 to the queue. While this will
succeed it is ultimately not a valid interface and would
Hello,
I add a member to a queue with AddQueueMember, but the Queue still
indicates joinempty :
Add member to queue :
/-- Executing [queueadd@sub-GetParams:2]
AddQueueMember(SIP/sip17-5c1e, myqueue11,member3) in new stack
-- Executing [queueadd@sub-GetParams:3] NoOp(SIP/sip17-5c1e,
Hello,
at the moment I am logging queues into a MySQL DB, but this can quickly
become a lot of information.
Is there a way to exclude certain queues from being logged into the
queue log ?
Thanks,
Jonas.
--
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-- Bandwidth
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, November 27, 2012 10:21 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Queue logging
Hello,
at the moment I am logging queues
...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, November 27, 2012 12:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Queue logging
Hello,
I am not using triggering (what is this ?).
Just using extconfig.conf
Asterisk 1.8.12.2
Kind
Hello,
what exactly does hangupcause 111 mean ?
I read on the wiki : 111 protocol error 500 Server internal error
Is the the SIP response that was received form the other end ? Or is
this an internal server (Asterisk) error ?
Kind regards,
Jonas.
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