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 09:33:23
hello, i made a experimental patch for libpri to have NT/PTMP mode,
answers please on asterisk-dev at:
http://lists.digium.com/pipermail/asterisk-dev/2009-May/038455.html
Kristijan
2009/5/14 Kristijan Vrban
> good news, i just made my isdn device ring! ok, after it ring, any
> timout then hang
Thanks all. I figure out to exit the queue by setting context in queue.conf.
On Thu, May 21, 2009 at 11:20 PM, Kevin P. Fleming wrote:
> Mark Michelson wrote:
>
>> Not to undermine Kevin's requests to read what is documented, I can say that
>> what you want actually will not be presented by runn
it should work just fine; do you have the GSM codec compiled/loaded
core show modules like codec_gsm ... ?
OR that particular version has a BUG...
Martin
On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski wrote:
> Hi,
>
> I am not sure if I am doing something wrong, but I can't get MeetMe
Y,
Because the scheduler usually uses the dahdi timer to run ... and if
the timer has stopped
then the frames/events will not go out and finally you get the scheduler full
Martin
On Thu, May 21, 2009 at 9:14 AM, Hose wrote:
> What you say...Martin (asteriskl...@callthem.info):
>
>> check if you
Hi Nikhil,
Several of these "out of sync" issues have been resolves in many recent
versions
of Asterisk. I'm not sure if many of the out of sync issues were reported
against 1.2 when it was receiving bug updates, so you may need to move to
Asterisk 1.4 in order to get these updates.
Additiona
(monitor legs are out of sync)
On Thu, 21 May 2009, Nikhil Nair wrote:
> I'm running Asterisk 1.2.13...
A more modern version wouldn't hurt.
> I've been using monitor() to record calls, with fairly satisfactory
> results - at least until the last few months.
If you don't need the legs separat
Hi Gang,
I've got 1.4.25-rc1 up and running pretty good now. The only
difficulties I have left to conquer are:
1. FFA won't receive a fax from a DELL A990 (failed on 6 out of 7
attempts with wrong protocol or timeout).
2. DAHDI dial makes a clicking/static sound on line du
On 05/21/2009 09:11 AM, Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Robin Rodriguez wrote:
>
>> still rather frustrating getting the EFK working. If needed I could
>> post that portion of sip.cfg to get you started.
>
> Please do! Just having the example could be he
Here's a little story on all the cheap guys trying to get the best rate on
any route out there ( lcr and others).
Anyone have 0.01 to Mexico billed 1/1 ?
"
When customers call us to ask if we sell Cuba termination for 50c/min, I
sometimes joke and tell them "sure, I'll
On 21 May 2009, at 22:02, Nikhil Nair wrote:
> I'm pretty stumped here; I can only imagine that, for some reason,
> not all
> silence is being recorded in the sound files.
Silence suppression might be enabled somewhere? Asterisk doesn't like
that generally, so might screw recordings too..
Ste
Hi guys,
I'm running Asterisk 1.2.13 on a Debian Linux system (that was just the
version that was packaged for it). I've been using monitor() to record
calls, with fairly satisfactory results - at least until the last few
months.
I've been recording VoIP calls, and using monitor() with no arg
Gavin Henry wrote:
> Is there any document on the reasons for the 1.6.0 and 1.6.1 branches?
> I remember reading something but can't find it again.
>
> Was it stability versus new features?
>
> I'm currently playing with 1.6.1
The difference is in regards to new features.
Instead of waiting 1-2
I have one asterisk server where most of the calls that go through AMD get
stuck in it, even if the analysis time of 3 seconds has already ended.
It doesn't move to the next priority (which is checking the AMD_STATUS).
Executing 'show channels' shows that the calls are stuck in the AMD app.
I ha
The Asterisk Development Team is pleased to announce the release of
Asterisk-Addons 1.6.0.2. Asterisk-Addons 1.6.0.2 is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves a potential crash issue in the ooh323 channel driver, and
resolves
The Asterisk Development Team is pleased to announce the release of Asterisk
1.4.25. Asterisk 1.4.25 is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves several crash issues, DTMF related issues, and CDR related
issues.
For a summary
Hi Lenz,
Here is my objective. Planning to implement queue using Asterisk 1.2.18.
So created a queue named testqueue in queues.conf and then created
agents for this. Now, our actual requirement is to collect the callerid
from the inbound call and search in the DB (customer list) and display
the in
Couldn't he also just do a "sip set debug" to view the responses coming
back?
Jeff LaCoursiere wrote:
> On Wed, 20 May 2009, John Regal wrote:
>
>
>> Thanks for the reply and apologize for the double post. My original post
>> landed in another thread and thought it may have been missed...
>>
>
It is already a macro, not sure about passing an array of numbers.
Alex Samad wrote:
> On Wed, May 20, 2009 at 03:16:34PM -0400, M Hulber wrote:
>
>> Alex Samad wrote:
>>
>>> On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
>>>
>>>
>
> [snip]
>
>
>>>
>>>
>>
Hi,
We are looking for the best outbound rate to US48 termination, in any
quality lines (for call centers resale).
If you offer volume discounts, please quote for:
- up to 1 million min/month
- over 1 million min/month
We currently use in total up to 3 million min/month and are planning on
growing
> Hi, in MFC-R2 signaling there is a value Calling party category signal
> (e.g., normal subscriber, high-priority subscriber, operator, coin-operated
> telephone)
>
> How can I get that information in my Asterisk??
That depends on which MFC-R2 solution are you using for Asterisk. The
2 most known
Danny Nicholas wrote:
> To clarify:
> Inbound - Answer
> Outbound - Answer (again)
>Dial.
Hmmm... that seems like it would be from the department of redundancy
department but I gave it a try, both before and after the Monitor()
command with the same result... it fails.
Thanks!
Barry
Hi, in MFC-R2 signaling there is a value Calling party category signal
(e.g., normal subscriber, high-priority subscriber, operator, coin-operated
telephone)
How can I get that information in my Asterisk??
Is there any similar value in SIP?
Thanks
___
-
openSuse 11
Asterisk 1.4.23.1
Asterisk GUI 2.0 Latest SVN version
I set up some page groups using the Asterisk GUI and found that when I
hang up the paging phone it causes Asterisk to restart. So far no one
has been on the phone at this time so I am unsure if it hangs them up
but it definatly
Mark Michelson wrote:
> Not to undermine Kevin's requests to read what is documented, I can say that
> what you want actually will not be presented by running "core show
> application
> Queue" in the CLI.
As file would say... 'osnap'
In my haste to respond this morning while eating breakfast
I'd be interested in this as well... I;m coming up to an upgrade cycle and
trying to decide if I should upgrade to the latest 1.4 or 1.6.1
When others that have commented on this say they have had problems with PSTN
connections, are you referring to T1 or POTS? I;m in a T1 scenerio, so if
problem
Not that I;m exactly a big fan of NFS but... why would you choose to
implement a filesystem that was designed to emulate Windows shares for your
UNIX-type environment? You have to kind of expect odd problems like this
when you choose to use things for other than their intended purpose. Samba
I wo
On Thu, May 21, 2009 at 09:32:02AM -0500, Tim Nelson wrote:
> yum -y install kernel-devel kernel-headers
kernel-devel is the one you'll need . Sadly you'll get one of a newer
version . If booting to a newer kernel is not an issue, I suggest you
install the newer kernel (which is recommended anyw
Hi David,
>
> That's very similar to a setup I made. And I was troubleshooting
> similar problems. Let me ask you a question:
>
> Are you quite confident that the inbound faxes that fail are going to
> succeed on an ordinary fax machine?
At least I'm sure of a couple of calling numbers that I kno
Lyle Giese wrote:
> Manoj Panicker - FOES wrote:
>>
>> Hi
>> Which is the best interface card to connect* PSTN* line with
>> Asterisk. Can somebody please help. My intention is to route the
>> incoming PSTN calls to internal IP Phones through Asterisk and Vice
>> versa. The Asterisk is
Kevin P. Fleming wrote:
> Rilawich Ango wrote:
>
>> I want to allow user to press 0 to the voicemail if the user don't
>> want to wait in the queue. Below is what I set but it doesn't work.
>> Anyone can help? ango
>
> None of that is necessary, but reading the documentation is. app_queue
> a
On opensuse 11.0, I had to install my kernel source using "zipper" (that's y
not I - email self corrected).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Farooq Hussain
Sent: Thursday, May 21, 2009 9:27 AM
To: asterisk-use
"Farooq Hussain" wrote:
>
Hello Everyone,
I am receiving following error message will making Zaptel on Cent OS 5.2.
make[1]: Entering directory `/usr/src/zaptel-1.4.12.1'
> echo "You do not appear to have the sources for the 2.6.18-92.el5 kernel
> installed."
> You do not appear to h
Hello Everyone,
I am receiving following error message will making Zaptel on Cent OS 5.2.
make[1]: Entering directory `/usr/src/zaptel-1.4.12.1'
echo "You do not appear to have the sources for the 2.6.18-92.el5 kernel
installed."
You do not appear to have the sources for the 2.6.18-92.el5 kernel
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Robin Rodriguez wrote:
> still rather frustrating getting the EFK working. If needed I could
> post that portion of sip.cfg to get you started.
Please do! Just having the example could be helpful for those of us
preparing to tackle this kind of pr
To clarify:
Inbound - Answer
Outbound - Answer (again)
Dial.
If I missed that, please disregard.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline
Sent: Thursday, May 21, 2009 9:05 AM
To
What you say...Martin (asteriskl...@callthem.info):
> check if your dahdi card still takes interrupts at this point
> dahdi_test should return some numbers close to 99%
>
> Martin
Thanks, I'll try running d_t next time it happens. Are you suggesting
that either 1) the card is no longer generati
Danny Nicholas wrote:
> You should try Answer before Dial on the Monitored call. Bridging can be
> very unhappy.
Hi Danny.
Already done earlier in the dial plan, when the call first comes in but
before it gets routed to the part that I showed. Thanks for looking
though!
Barry
__
sasirekha jaganathan wrote:
> Did anyone tried static build of asterisk 1.6 version?
> Installation fails when tried with static build.
>
>
> warning: Using 'initgroups' in statically linked applications requires
> at runtime the shared libraries from the glibc version used for linking
> aster
FWIW, asterisk processes its' voicemail in FIFO (First in First out) fashion
using msg.* to store the messages, so it sends msg, then msg0001,
etc. You could write a shell or perl or C script to do a "bubble sort" on
all voicemails for a user.
Here is a listing of two voicemails
ll /
My issues are all DAHDI/POTS related. Unfortunately, our present
communication depends on the POTS lines, so Im back to 1.4.25-rc1 as stated
earlier.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday
2009/5/18 Danny Nicholas
> I'd love to see this as well. After a few days of trying 1.6.1 (from
> 1.4.21) I dropped back to 1.4.25-rc1 and that is going pretty well.
Which issues did you get ?
I'm about to deply a 1.6.1 system it does seem to work ok in a pure SIP
environment.
>
>
> -Orig
Rilawich Ango wrote:
> I want to allow user to press 0 to the voicemail if the user don't
> want to wait in the queue. Below is what I set but it doesn't work.
> Anyone can help? ango
None of that is necessary, but reading the documentation is. app_queue
already supports the caller using a DT
Neeraj Chand escribió:
> 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.
>
>
Maybe something like this could do the job:
exten => h,1,Set(CDR(userfield)=${HANGUPCAUSE})
You can use the accountcode f
Sangoma as well.
Also ATA's such as what used to be called the Sipura 3000
Cisco 3810's with SIP IOS will give you up to 6 analog ports, and on the
really low end, if you can still find an X100 card, at least for US PSTN
lines.
John Novack
--[ UxBoD ]-- wrote:
> - "Paul Hales" wrote:
>
>
What are you getting if you do a dahdi_cfg -vv?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Bareiro
Sent: Thursday, May 21, 2009 4:38 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-user
You should try Answer before Dial on the Monitored call. Bridging can be
very unhappy.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline
Sent: Wednesday, May 20, 2009 8:52 PM
To: asterisk-users@lis
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!
>>/ 2. If I enable and force jitter buffer, Asterisk
Manoj Panicker - FOES wrote:
>
> Hi
> Which is the best interface card to connect* PSTN* line with
> Asterisk. Can somebody please help. My intention is to route the
> incoming PSTN calls to internal IP Phones through Asterisk and Vice
> versa. The Asterisk is in LAN and is reachable from a
Hi,
Did anyone tried static build of asterisk 1.6 version?
Installation fails when tried with static build.
warning: Using 'initgroups' in statically linked applications requires at
runtime the shared libraries from the glibc version used for linking
asterisk.o: In function `cli_prompt':
warni
We did an opensource Java Applet that plays GSM files _very_ simply if
that helps.
I'd accidentally removed it from our website, but it is back now -
improved
with a javascript interface supporting load, play and pause actions.
http://www.westhawk.co.uk/software/playGSM/PlayGSM.html
The de
Danny Nicholas wrote:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
> Sent: Wednesday, May 20, 2009 4:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [ast
- "Ondrej Valousek" escreveu:
> Hi List,
>
> I have a question regarding jitterbuffer in Asterisk 1.4.24. I see
> that
> jitterbuffer is only effective on the receiving channels.
> My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch
>
> office.
> Questions:
> 1. To enabl
On May 21, 2009, at 5:59 AM, Karl Fife wrote:
> While I have not needed to do this for myself, I believe you can
> create this
> functionality quite easily using Polycom's 'Enhanced Feature
> Keys' (EFK's).
> IIRC, EFK's are available in the newest firmware revision 3.1.x and
> newer.
> -Ka
hi,
i'm searching solution for playing media(moh,prompts,voicemail,recordings
- wav format) from adobe flash player (web browser)
flash cannot play wav directly (imho)
i must convert files to any other format on-the-fly
- i cannot use mp3 because of royalties
- next option is swf (with ffmpeg
On Wed, May 20, 2009 at 6:58 AM, Santiago Gimeno
wrote:
> We have been working with the ReceiveFax application for some weeks now in
> order to receive faxes in T.38 and it works fairly well, but there are some
> faxes that for some reason we are not able to receive correctly.
>
> The asterisk ver
While I have not needed to do this for myself, I believe you can create this
functionality quite easily using Polycom's 'Enhanced Feature Keys' (EFK's).
IIRC, EFK's are available in the newest firmware revision 3.1.x and newer.
-Karl
- Original Message -
From: "Matt Darnell"
To: "As
On Thu, May 21, 2009 at 06:38:27AM -0300, Daniel Bareiro wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> El miércoles 20 de mayo del 2009 a las 21:19:18 -0300,
> Daniel Bareiro escribió:
>
> > I load the modules wctdm and dahdi. But when I execute in Asterisk
> > CLI "dahdi s
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
El miércoles 20 de mayo del 2009 a las 21:19:18 -0300,
Daniel Bareiro escribió:
> I load the modules wctdm and dahdi. But when I execute in Asterisk
> CLI "dahdi show channels", I get the following error message:
>
>
> No such comm
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?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-us
What exactly are tyou trying to achieve?
l.
2009/5/20 Kurian Thayil
> Hi All,
>
> I am trying to implement ACD using Asterisk 1.2.18 and I've chosen
> AgentCallbackLogin for login purpose. One AGI is written which will actually
> get executed when agent dials '1001' (say) from his SIP phone and
Hello!
Thanks...I set up a Samba mount, which works ok, except that Asterisk
confuses a wave file as a wav49 file. I think it may have something do with
the way Samba supports case sensitivity. Since Windows is not very
aggressive when it comes to being case sensitive, I am thinking that Samba
i
Hi List,
I have a question regarding jitterbuffer in Asterisk 1.4.24. I see that
jitterbuffer is only effective on the receiving channels.
My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch
office.
Questions:
1. To enable jitter buffer on SIP channels it seems I have to enabl
Hi,
I am not sure if I am doing something wrong, but I can't get MeetMe to
work with GSM codec (Asterisk 1.6.1 SVN r190371).
My config files below:
sip.conf:
[general]
context=common
canreinvite=no
bindport=5060
bindaddr=78.105.1.127
disallow=all
allow=alaw
allow=gsm
rtptimeout=600
rtp
On Thu, May 21, 2009 at 10:04 AM, Matt Darnell wrote:
> 1. Set the phone to automatically record all calls to the USB stick,
> now you have to press three keys.
Not possible AFAIK.
> 2. Put Record on the main screen when a call is active. This would
> eliminate having to press the 'more' softkey.
Hi All,
please provide some help.
I'm just posting this questions to both forums as its related to both. In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones > Asterisk SIP Trunk > Call Manager -> CME
Has anyone been able to do the following:
1. Set the phone to automatically record all calls to the USB stick,
now you have to press three keys.
2. Put Record on the main screen when a call is active. This would
eliminate having to press the 'more' softkey.
Thanks,
Matt
- "Paul Hales" wrote:
> Digium PSTN cards seem to work.
>
>
>
> PaulH
OpenVox works well.
Best Regards,
--
SplatNIX IT Services :: Innovation through collaboration
___
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