Thanks for all answers.
One further question: If I run Asterisk as root, and set its group in
asterisk.conf to apache, and make no changes to file/folder permissions,
will I be able to run asterisk -rx 'clicmd' from a php-script (running as
user apache with group apache)?
BR,
Torbjörn
2011/10/19 Paul Belanger pabelan...@digium.com
snip Later out I found to properly use snmp you actually need to run
asterisk as root.
Hi,
Can you elaborate a bit ?
Which SNMP feature requires to run asterisk as root ?
Regards
--
Hello Terry,
thank you for your answer.
2011/10/19 Terry Wilson twil...@digium.com
If I had to guess, I'd say that you don't have canreinvite/directmedia=no
in sip.conf and there is possibly a NAT between the phones and Asterisk.
When they have the same codec and directmedia is enabled, the
Hi,
I have been testing MixMonitor and Monitor to record some calls in Asterisk and
I have noticed that MixMonitor works fine whereas in the Monitor files of the 2
separate channels, we can find little cuts of the audio. We are using U law
codec and wav files for the recording.
Anyone have
Hi,
I've done some similar thing in one of my testing, using MixMonitor and
monitor at the same time. Everything worked perfectly well no issues even on
Vmware. Can you check if the CPU utilization is normal. Also which version
of asterisk you are using?
--
Regards,
Sammy
On Thu, Oct 20, 2011
Hi Everyone,
I am in search of a reliable open source tool for the real time monitoring of
ASR ACD, so any help or suggestions will be highly appreciated.
Regards,
Faisal Rehman--
_
-- Bandwidth and Colocation Provided by
Something must have changed since asterisk 1.2 ... ;)
I used to be able to run a simple sipp test using
./sipp -sn uac -d 2 -s sipp-client 127.0.0.1 -l 1
however, with asterisk 10 I am getting a call failure with the words
1319103341.109277: Aborting call on unexpected message for Call-Id
Hi Julian,
1319103341.109277: Aborting call on unexpected message for Call-Id
'20-18295@127.0.0.1': while expecting '100' (index 1), received
'SIP/2.0 407 Proxy Authentication Required
What I am trying to figure out is how / why / what is different that
now asterisk requires proxy
Some of you may be interested in a project I've been keeping myself busy
with whilst convalescing - or rather, two closely related projects.
callPopPy is an incoming call popup notifier written in Python. It's
probably most useful to Linux users, but was written to be portable and
has been tested
Hi all,
After having done some successful tests with 3G in combination with the
G729a codec, I plan to use this as a failover path for when the main
internet connection goes down.
However, on this usual connection, G711a is used. I could have the
script that monitors the main line also sed
Hi,
I am planning to setup a video call with Asterisk 1.4.17 and eyebeam
softphone, can any one suggest some link or configuration to setup the
things.
Thanks.
--
_
-- Bandwidth and Colocation Provided by
Hi,
CPU usage does not change when a call is served by Asterisk. I performed
several there was no influence.
Version is Asterisk 1.6.2
Regards,
Date: Thu, 20 Oct 2011 13:30:20 +0500
From: Sammy Govind govoi...@gmail.com
Subject: Re: [asterisk-users] Monitor does not work well (little cuts
Hello Everyone,
The documentation suggests using unixodbc for asterisk realtime. Is
there any way
we can just use native database clients such as libmysqlclient from
MySQL? The native
clients tend to be more up-to-date.
Thanks in Advance,
Nick.
I've used the MySQL addon for years
Hello,
if this is the complete sip trace the UAS (client) have to reply with an
ACK after the 401 response from asterisk. thats why asterisk thinks the
request is still alive.
regards
stefan
Am 19.10.11 22:22, schrieb markus_wei...@mailworks.org:
Hallo,
any idea what's wrong with that
Dear all,
Do you know if there is a way to know the 2 RTP ports that Asterisk is using
for audio flow in a call in the dialplan?
I would like to launch a Linux shell command tcpdump to capture audio flow in
those 2 RTP ports before call starts and stop capturing at the end of the call.
You could temporarily change rtp.conf to use just 4 ports (say 10001-10004)
and monitor 10001 and 10002. On a production system you would have to do
something with a tool like netstat to try and predict which ports in the
range would be used.
From: asterisk-users-boun...@lists.digium.com
Hi Isabel,
Could you not just filter out after the fact using something like Wireshark?
Regards
On Thu, Oct 20, 2011 at 3:28 PM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Dear all,
** **
Do you know if there is a way to know the 2 RTP ports that Asterisk is
using for audio flow in a
Hello JR,
Thank you so much for your response. I would greatly appreciate
having the chance to look over those papers. Dated or not, past
approaches is what drives our efforts forward ;).
Nick from Toronto.
On Thu, Oct 20, 2011 at 8:59 AM, JR Richardson jmr.richard...@gmail.com wrote:
Hello
Have you tried changing/upgrading asterisk version.?
On Thu, Oct 20, 2011 at 5:34 PM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi,
CPU usage does not change when a call is served by Asterisk. I performed
several there was no influence.
Version is Asterisk 1.6.2
Regards,
Date: Thu, 20 Oct
Hello list,
what does this mean ?
[Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data
for 'account9' (Out of memory or socket error)
[Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data
for 'account10' (Out of memory or socket error)
[Oct 20 15:23:08]
I have canreinvite and directmedia to 'no' - and there is no NAT
between the phones and asterisk...
Hmm. In that case, I'm not sure. You could take a look at the output of rtp
set debug on when the call is going on to see what is going on with the audio.
--
On 11-10-20 10:28 AM, Jonas Kellens wrote:
Hello list,
what does this mean ?
[Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data
for 'account9' (Out of memory or socket error)
[Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data
for 'account10' (Out of
On 10/20/2011 05:07 PM, Paul Belanger wrote:
On 11-10-20 10:28 AM, Jonas Kellens wrote:
Hello list,
what does this mean ?
[Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to build sip pvt data
for 'account9' (Out of memory or socket error)
[Oct 20 15:23:08] ERROR[1496] chan_sip.c: Unable to
hi list
i have an issue related to queues.conf
i have configured the code like below
extensions.conf
[default]
exten = 800,1,AgentLogin()
exten = 666,1,Answer()
exten = 666,2,Queue(hotline)
#include aheeva_dialplan.conf
===
queues.conf
[hotline]
member = Agent/1000
On Thu, Oct 20, 2011 at 11:01 AM, salaheddine elharit
salah.elharit...@gmail.com wrote:
[Oct 20 15:34:00] WARNING[19179]: pbx.c:1832 pbx_extension_helper: No
application 'Queue' for extension (agents, 666, 2)
This line here indicates to me that you don't have app_queue.so loaded.
Try, from
On Thu, Oct 20, 2011 at 5:53 AM, Bart Coninckx bart.conin...@telenet.bewrote:
Hi all,
After having done some successful tests with 3G in combination with the
G729a codec, I plan to use this as a failover path for when the main
internet connection goes down.
However, on this usual
Very Good day to everybody, I had a problem with an implementation that
recently made in my Company, dealing in a Grandstream gateway
which will enable echo cancellation but until you hear noise
point appears in the middle of words creating noise interference,
when you disable the echo in this
You cannot echo cancel SIP. Removing echo must be done before PSTN is
converted into SIP. i.e. your PSTN/SIP gateway.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Diego Alejandro
Sanchez Quiroga
Sent:
Hi!
I am interisted to dial out from the console with chan_alsa. Can
somebody of you help me according this problem?!
I added user the asterisk to pulse and pulse-access, and it didn't
change anything. alsa applications are routed by default to pulse.
cat /etc/asound.conf
pcm.!default {
type
Hi List,
Another dumb conversion question (I hope). I installed 10.0
and copied my 1.4 configuration files over. With a few tweaks everything
works great except for 1 feature that I specifically went to 10.0 for. When
I do an attended transfer, I still get the receptionists caller
Greetings,
If you are planning on attending Astricon, please take the time to
attend the GPG key signing event. More information can be found on the
wiki page[1].
[1]
https://wiki.asterisk.org/wiki/display/~pabelanger/Astricon+2011+Key+signing+event
--
Paul Belanger
Digium, Inc. |
thanks you Eric, thanks, I understand your contribution, and you're
right because
in my internal network I have no problem handling a grandstreamgateway gw
4108, someone might suggest to take intoaccountparameters for this device, I
tried to contact the manufacturer and is very difficult .
When
On 10/20/2011 05:16 PM, Paul Belanger wrote:
Greetings,
If you are planning on attending Astricon, please take the time to
attend the GPG key signing event. More information can be found on
the wiki page[1].
[1]
[trunkgroups]
[channels]
[my-phones](!)
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
echocancel = yes
Patrick:
would you mind sharing how you're using the menuselect.makeopts exactly? I
first had this problem with 1.8.7.0 and again with 1.8.8.0-rc1. I've since
updated to 1.8.8.0-rc2 but for some reason I am still unable to get
menuselect to use the .makeopts that I put in my ~home dir. Maybe
On 10/21/2011 04:33 AM, Luke Hamburg wrote:
Patrick:
would you mind sharing how you're using the menuselect.makeopts exactly? I
first had this problem with 1.8.7.0 and again with 1.8.8.0-rc1. I've since
updated to 1.8.8.0-rc2 but for some reason I am still unable to get
menuselect to use the
On Thu, Oct 20, 2011 at 04:41:17PM -0500, Danny Nicholas wrote:
... I installed 10.0
and copied my 1.4 configuration files over. With a few tweaks everything
works great except for 1 feature that I specifically went to 10.0 for. When
I do an attended transfer, I still get the receptionists
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