can you look on this from your debug
1. app_meetme.c:3030 find_conf: The requested confno is '12'?
2. == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23]
DEBUG[6872]: config.c:1306 config_text_file_load: Parsing
/etc/asterisk/meetme.conf
3. == Found
4. [May 21
Hi,
I am attempting to make about ten calls simultaneously and intermittently
get 'SIP/voipprovider is circuit-busy' followed by 'everyone is
busy/congested at this time"
I am not sure if this is related to my bandwidth to my voip provider, a
configuration issue or something else.
Has anyone seen
Hi Martin,
Yes, I do have GSM compiled for sure.
$asterisk -r -x core show codecs audio
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INTBINARYHEX TYPE NAME DESC
Hi Dhaval,
The reason confno '12' is not found in meetme.conf is because I am
using MySQL as realtime config backend.
See few lines below there is:
[May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478
mysql_reconnect: MySQL RealTime: Connection okay.
[May 21 09:33:23] DEBUG[6872]:
Hi,
To a large extend, Asterisk's /etc/asterisk/*.conf configuration files
conform to a format such as:
[section1]
key1=value1
key2=value2
[section2]
key1=value1
key2=value2
...
To increase coherence when running custom-made application in Perl, Java,
PHP, ...) and Asterisk on the same
I am attempting to make about ten calls simultaneously and intermittentlyget 'SIP/voipprovider is circuit-busy' followed by 'everyone isbusy/congested at this time"If none of the calls were going through then that would probably be an authentication issue. If some of the calls are going through
Thnx Mark ...
I think you are right, 941 doesn't support TLS at all.
Dimitris ...
M Hulber wrote:
Unfortunately, I don't have this phone and I can't find any
documentation for the 941 that refers to TLS setting. Here's what it
looks like when I set extension 4 to TLS on the 942:
On an entirely unrelated note, do you have the gsm asterisk sounds
installed? Maybe that vm-*.slin files dont exist.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris
Maciejewski
Sent: Friday, May 22,
Hi everyones,
I have a production server using asterisk 1.6.0.6
using php as an IVR and mssql server (on other machine)
My server attached a Sangoma A104 card (4xT1 card)
i have a problem with memory leak on that server
and causing a delay on IVR prompt. (Thats my assumption, memory leak
this command doesn't show the codecs present in the system do you
have g723 compiled too ?
try core show translations or something like that
Martin
On Fri, May 22, 2009 at 2:25 AM, Chris Maciejewski ch...@wima.co.uk wrote:
Hi Martin,
Yes, I do have GSM compiled for sure.
$asterisk -r
hi
i got TOS and retranssmission error on receiving SIP call
chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission
10caed68-0f1d-df82-da1e-a76c1cb3d...@172.18.100.72 for seqno 43156 (Critical
Response) -- See doc/sip-retransmit.txt.
[May 22 13:42:44] WARNING[18021]:
Hello Matt,
I do agree with you that NFS is that UNIX standard for network
filesystems and that what should essentially be used. However, I
shied away from using it, because on the surface it looks too
complicated to secure properly. It uses many ports, dynamic ports,
different background
Thanks Kinjal!
Missing sound files was the problem. There were no .gsm files in my
sounds directory. Despite console shows .slin, the actual files
required are .gsm.
Once I copied .gsm into /var/lib/asterisk/sounds everything works OK.
Regards,
Chris
2009/5/22 Kinjal Dixit
Arun Kumar schrieb:
please provide some help.
Do not repost the same question after just 1 day.
Do not cross-post.
*scnr*
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk:
Hi,
I have both codec_g726.so and format_g726.so loaded:
r...@test:~# asterisk -r -x module show | grep 726
codec_g726.so ITU G.726-32kbps G726 Transcoder 0
format_g726.so Raw G.726 (16/24/32/40kbps) data 0
But when I try to dial into Asterisk
Hi all,
I was playing with top on my Asterisk 1.4.24 server when I noticed
this strange thing:
PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND
26797 asterisk 25 0 70524 14m 6416 S 1.3 2.9 5:59.44 asterisk
...
26518 asterisk 25 0 3316 1452 1140 R 46.6 0.3
Does anybody think it would make sense for /etc/init.d/asterisk
to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk
did?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
- Ondrej Valousek webs...@s3group.cz escreveu:
Hi Vinicius.
/ 1. To enable jitter buffer on SIP channels it seems I have to
enable
// and
// force it, right?
/
Not sure about the forcing part (don't know exacly how it works),
but I always set jbforce=yes to be sure.
Ok, thanks!
Lets start from the beginning. Why are using a network share for your
voicemail in the first place?
j
On Fri, 22 May 2009, Elliot Murdock wrote:
Hello Matt,
I do agree with you that NFS is that UNIX standard for network
filesystems and that what should essentially be used. However, I
Don't be afraid about the info that I'm going to post in this mail, but
I want you to give as much info as possible. Also I want to show you
what I've tried.
What do I want
When a voicemail-message is left via the Voicemail()-application, I want
the .wav-file send to my mail-address as an
There already is a special character to tell asterisk not to parse a line...
its: ; that is why the default configuration is completely filled with
lines that start with ; its considered a comment character to asterisk and
will make it ignore the line... you'd just want to add some extra
On Friday 22 May 2009 02:25:26 Chris Maciejewski wrote:
Hi Martin,
Yes, I do have GSM compiled for sure.
$asterisk -r -x core show codecs audio
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INTBINARY
Roderick A. Anderson wrote:
Olivier wrote:
Hi,
To a large extend, Asterisk's /etc/asterisk/*.conf configuration files
conform to a format such as:
snip /
Not specific to Asterisk but there is Config::Std which, in Damian's
blurb for the module, is simple and limited. Still it could
Can anyone tell me if there is a way to vary the output levels (dB) of the
tones generated in indications.conf? I generate a few custom tones and
sometimes people tell me they are a little too loud.
Thanks
Lee
___
-- Bandwidth and Colocation
Then, my next question, is there widely available librairies to parse
Asterisk's config files-like files ?
Asterisk-Java has some support for this:
http://asterisk-java.org/development/apidocs/index.html?org/asteriskjava/config/package-summary.html
The basic things are pretty straight
Olivier wrote:
Hi,
To a large extend, Asterisk's /etc/asterisk/*.conf configuration files
conform to a format such as:
[section1]
key1=value1
key2=value2
[section2]
key1=value1
key2=value2
...
To increase coherence when running custom-made application in Perl,
Java, PHP, ...)
On Friday 22 May 2009 07:33:09 jonas kellens wrote:
My /root/.msmtprc-file has the following :
# Set default values for all following accounts.
defaults
logfile ~/.msmtp.log
There is NO entry in the logfile of msmtp (/root/.msmtp.log). No error,
no success.
Is Asterisk running as root or
I forgot to put this in my mail indeed.
[r...@asterisk ~]# ls -l /usr/sbin/asterisk
-rwxr-xr-x 1 root root 36029398 Apr 22 15:19 /usr/sbin/asterisk
[r...@asterisk ~]# ps aux | grep asterisk
root 3037 0.0 0.0 4528 556 ?SMay19
0:00 /bin/sh /usr/sbin/safe_asterisk
root
On Fri, 2009-05-22 at 13:57 +0530, DHAVAL INDRODIYA wrote:
i got TOS and retranssmission error on receiving SIP call
chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission
10caed68-0f1d-df82-da1e-a76c1cb3d...@172.18.100.72 for seqno 43156
(Critical Response) -- See
Hi Folks,
I have a few folks whom are interested in another recording session with
Alison Keenan but don't have enough work to justify her visit to the
studio.
If there's anyone whom would like her to do some work but hasn't got
around to it yet now might be the time. We need enough work to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Here is mine if it helps;
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes
[zonemessages]
David,
what is your SMTP-client then ?
Did you change the mailcommand 'mailcmd' in voicemail.conf ?? Or is it
still /usr/sbin/sendmail ??
I use version 1.4.24.
Thanks for your reply.
Greetingz,
Jonas.
On Fri, 2009-05-22 at 10:59 -0400, David wrote:
-BEGIN PGP SIGNED MESSAGE-
have you checked /var/log/maillog to see what the error might be?
2009/5/22 David da...@linuxcrazy.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Here is mine if it helps;
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
Chris Maciejewski wrote:
Found unknown media description format G726-16 for ID 102
It's right there.
And asterisk is replying with 488 Not acceptable here
Asterisk does not support G726-16, it only supports G726-32.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445
ignore me! i've just realised half this thread was deleted :)
2009/5/22 Geraint Lee gera...@gmail.com
have you checked /var/log/maillog to see what the error might be?
2009/5/22 David da...@linuxcrazy.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Here is mine if it helps;
[general]
Markus Weiler wrote:
Hi,
In VI:
In 'vi', moving the cursor over any bracket, brace, etc, and then
pressing '%' moves the cursor to the 'matching' bracket/brace character.
That can be very useful when programming, to find missing/extra brackets
and braces. It even seems to find
On 22 May 2009, at 16:34, sean darcy wrote:
Well vi is beyond my linux karma. But, you prompted me to see if nano
has the same ability. And it does: Alt-] . Who knew??
You are my hero.
S
___
-- Bandwidth and Colocation Provided by
Hi Kevin,
Thanks for your reply. I switched to G726 32Kbps but still no luck:
INVITE
[SIP headers omitted]
v=0
o=1 1291673978 653998617 IN IP4 192.168.7.55
s=-
c=IN IP4 78.105.1.131
t=0 0
m=audio 8002 RTP/AVP 104 101
a=rtpmap:104 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101
On 22 May 2009, at 16:55, Chris Maciejewski wrote:
Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer -
audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined -
0x0 (nothing)
Codec not enabled on that peer?
S
___
-- Bandwidth
Chris Maciejewski wrote:
Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer -
audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined -
0x0 (nothing)
'us' does not include g726, so you have not configured your SIP
user/peer to support G.726.
I note Got unsupported a:fmtp
Yes, I was missing allow=g726 for this peer :-(
Playback(/var/lib/asterisk/moh/fpm-sunshine)
works OK now, however I still can't get MeetMe to work.
Before I had similar problem, when MeetMe wasn't working with GSM
codec because I was missing .gsm audio files.
I suspect now it is the same
Chris Maciejewski wrote:
Yes, I was missing allow=g726 for this peer :-(
Playback(/var/lib/asterisk/moh/fpm-sunshine)
works OK now, however I still can't get MeetMe to work.
Before I had similar problem, when MeetMe wasn't working with GSM
codec because I was missing .gsm audio files.
I do have codec_g726 loaded. As I mentioned before
Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite
there is only fpm-sunshine.wav file. It is only MeetMe which is not
working:
-- SIP/OpenSER-08208098 Playing 'entering-conf-number.slin'
(language 'en')
[May 22 18:07:04]
I thought that /var/log/maillog was for sendmail ?? I'm not using
sendmail...
My /var/log/maillog is empty :
[r...@asterisk ~]# cat /var/log/maillog
[r...@asterisk ~]#
How about the system()-application ?? Why is that also not working for
me ??
On Fri, 2009-05-22 at 16:25 +0100, Geraint Lee
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with no response to our critical packet.
Calls to voicemail and internal extensions work fine.
I understand that everything points to a
Maillog is for whatever you send to it, i send clamav/spamd/qmail etc to it
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: May-22-09 1:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk
setup with outgoing calls not completing and requiring an Asterisk reset
to 'unstick' span 1.
Sorry this is a bit long but I'm completely out of my depth :-(
This system has been in use for some while and I recently upgraded
James Lamanna wrote:
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with no response to our critical packet.
Calls to voicemail and internal extensions work fine.
I understand
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
jonas kellens wrote:
| David,
|
| what is your SMTP-client then ?
|
| Did you change the mailcommand 'mailcmd' in voicemail.conf ?? Or is it
| still /usr/sbin/sendmail ??
I don't have mailcmd in voicemail.conf, I was under the impression that
is
for some reason (someone would have to look deeper) your SIP peer
sends ACK to 200 OK and Asterisk doesn't get it
so it retransmits 200 OK a couple times and then assumes there's noone there
Martin
On Fri, May 22, 2009 at 12:36 PM, James Lamanna jlama...@gmail.com wrote:
Hi,
I have a strange
I think I know what the problem is here. It's not the fault of the phone, but
of
Asterisk. The phone is sending an INVITE and then an INFO (DTMF '#',
specifically) to Asterisk. Asterisk only keeps track of the last incoming Cseq
in a dialog, so once the INFO arrives, we no longer have any
On Fri, May 22, 2009 at 12:51 PM, Russell Brown russ...@lls.lls.com wrote:
I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk
setup with outgoing calls not completing and requiring an Asterisk reset
to 'unstick' span 1.
[cut]
Can anyone suggest a course of action here?
Martin wrote:
I think I know what the problem is here. It's not the fault of the phone,
but of
Asterisk. The phone is sending an INVITE and then an INFO (DTMF '#',
specifically) to Asterisk. Asterisk only keeps track of the last incoming
Cseq
in a dialog, so once the INFO arrives, we no
Yes, this would be why I said that it is Asterisk's fault and provided
possible
workarounds.
Thank you for your helpful and constructive criticism.
LOL yes you could expect now everyone to be critical about something like this.
Asterisk has been around for quite some time now (6+ years) and
We are facing alot of problem in the DTMF. At times we are unable to do the
verification because whenever we press the numbers for verification it does
not detects and at times it detects the wrong number for instance if the
customer is having the phone no. as 1234567890 it will detect 123467890
Can this be due to G729 codec ?
If yes, how to Uninstall g729 ?
Asterisk 1.2.27 is the version.
On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote:
We are facing alot of problem in the DTMF. At times we are unable to do the
verification because whenever we press the
Hi guys,
I'm trying to write hangup causes from asterisk into the CDR record.
Using version 1.4.24.1 at the moment, but no joy so far.
Has anyone implemented this?
Neeraj Chand
Support Analyst
Fiji Islands Australia
T: +6793342526 T: +61388924326
I think you should request to get it fixed via free digium tech support
Martin
On Fri, May 22, 2009 at 12:51 PM, Russell Brown russ...@lls.lls.com wrote:
I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk
setup with outgoing calls not completing and requiring an Asterisk
Is this inbound calls to your automated attendant? Or Outbound calls to
say a bank ivr out in the pstn? What direction?
What is your interface/carrier? T1, SIP, H32? And what method are you
using for DTMF? Eg inband, out of band, what rfc, etc?
From: asterisk-users-boun...@lists.digium.com
On May 22, 2009, at 3:05 PM, Martin wrote:
Yes, this would be why I said that it is Asterisk's fault and
provided possible
workarounds.
Thank you for your helpful and constructive criticism.
LOL yes you could expect now everyone to be critical about something
like this.
Asterisk has
1)
disallow=all
allow=g729
allow=ulaw
context=default
dtmfmode=rfc2833
2) We use SIP.
3) IVR 3rd party verification.
4) VOIP
On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote:
We are facing alot of problem in the DTMF. At times we are unable to do the
verification
Then if it's a IP interface (SIP, etc) have you tried a sniffer trace
(wireshark, etc) to verify the packets are being sent correctly to
carrier?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason
Aarons (US)
Sent: Friday, May 22,
No
On Sat, May 23, 2009 at 1:01 AM, David @ULC ucoms2...@gmail.com wrote:
We are facing alot of problem in the DTMF. At times we are unable to do the
verification because whenever we press the numbers for verification it does
not detects and at times it detects the wrong number for instance
Hi Guys,
I just wanted to let you all know that you were indeed correct, it was
the SIP INFO '#'
that was causing the problem.
You'll pardon me, but I find this problem _utterly ridiculous_.
I am running asterisk v1.4.18. Are there any asterisk versions that
this is fixed on?
Thanks.
(Oh and
Hi Guys,
I would like to know if is there a way to actively stop a Background()
music?
Thanks,
Noel
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Background(file,m) will stop when user hits a DTMF digit that is an active
extension in your dialplan.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noel R. Morais
Sent: Friday, May 22, 2009 4:29 PM
To:
But I need a way to actively stop it. Without waiting for user hit a DTMF or
the background timeout.
Like StopMusicOnHold(), is there something like StopBackground()?
Thanks
On Fri, May 22, 2009 at 6:35 PM, Danny Nicholas da...@debsinc.com wrote:
Background(file,m) will stop when user hits a
You could make the background file into a MOH file in a separate class and
use the MOH commands to start and stop it. Easier than coding C and
recompiling.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noel R. Morais
Chris Maciejewski wrote:
I do have codec_g726 loaded. As I mentioned before
Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite
there is only fpm-sunshine.wav file. It is only MeetMe which is not
working:
-- SIP/OpenSER-08208098 Playing 'entering-conf-number.slin'
Noel R. Morais wrote:
But I need a way to actively stop it. Without waiting for user hit a
DTMF or the background timeout.
Like StopMusicOnHold(), is there something like StopBackground()?
No, because the dialplan does not continue running until Background
either times out or the user
Hi
I have an account with mynetphone (australia), which gives me two voip
(sip) accounts, which i used to have connected to a spa9000.
this is behind a firewall, so on the spa9000 I would listen on another
port apart from 5060. so on the firewall 5060 would go to voip1 and
5061 to voip2.
I
On Fri, 22 May 2009, Noel R. Morais wrote:
But I need a way to actively stop it. Without waiting for user hit a DTMF or
the background timeout.
What event would trigger your desire to stop the background()?
Thanks in advance,
On Fri, 22 May 2009, Kevin P. Fleming wrote:
This is not MeetMe, it's Playback. You specified a filename with '.slin'
in it to Playback, so then Asterisk attempts to find a filename called
'entering-conf-number.slin.foo' where foo is the possible formats
that Asterisk could transcode from.
We seem to be having a good bit of issues sending faxes and can't pinpoint
the issue. I'm hoping someone here may have a different outlook on the
issue that leads to a resolution.
Our call path is Sip trunk from MAX TNT - Asterisk - T1 - Adtran endpoint
converting sip trunk to copper line for
Todd S wrote:
Our call path is Sip trunk from MAX TNT - Asterisk - T1 - Adtran
endpoint converting sip trunk to copper line for house wiring.
Users at the endpoint can receive faxes without a problem. However,
sending faxes are not so friendly. 1 out of 5 faxes will send
On Fri, May 22, 2009 at 09:23:32AM +0200, Olivier wrote:
Hi,
To a large extend, Asterisk's /etc/asterisk/*.conf configuration files
conform to a format such as:
[section1]
key1=value1
key2=value2
[section2]
key1=value1
key2=value2
...
To increase coherence when running
On Fri, May 22, 2009 at 01:33:59PM +0200, Philipp Kempgen wrote:
Does anybody think it would make sense for /etc/init.d/asterisk
to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk
did?
What would you put there? When should it be run? As which user?
--
Tzafrir
I wish Polycom would hire someone with ergonomics skills. The whole
menu system is the most painful ever designed outside entry-level
phones. Polycom is an acknowledged leader in sound quality and robust
hardware but their idea of a menu sucks rocks and always has. Most of
their menus require
Yes with EFK in the latest firmwares you are able to change the on
screen button layout. I used it to bring a Do Not Disturb button to
the main screen of the SoundPoint IP330's. I may just be dense but
paired with the Administrator and Developer guides from Polycom it was
still rather
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