Hi,
I am not using FreePBX.I am using Asterisk 1.6.1.6 and TDM800P cards
for analog lines.
Warm Regards
Venugopal G
HNM-SO WiMAX CPE VoIP IOT Team
Cell : +91-99723-99437
On Thu, Mar 11, 2010 at 04:18:21PM +0800, Gopalakrishnaiyer Venugopal-Q16770
wrote:
Hi,
I am not using FreePBX.I am using Asterisk 1.6.1.6 and TDM800P cards
for analog lines.
This is an analog line. Either caller ID is sent or it isn't. Even if it
is sent, you can choose to ignore it.
On Thu, Mar 11, 2010 at 02:02:50AM +, ayodele abejide wrote:
I am using the 1.6.2.0 version
The command 'dialplan reload' comes from the module pbx_config .
Could you try running:
module load pbx_config.so
and then try:
dialplan reload
again?
--
Tzafrir Cohen
On Wed, Mar 10, 2010 at 09:27:46PM -0600, Chris Owen wrote:
On Mar 10, 2010, at 9:05 PM, Warren Selby wrote:
On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote:
This normally works fine but occasionally when someone picks up the call
other phones don't seem to realize
On 11 Mar 2010, at 08:18, Gopalakrishnaiyer Venugopal-Q16770 wrote:
I am not using FreePBX.I am using Asterisk 1.6.1.6 and TDM800P cards
for analog lines.
Right! Slowly getting the info we need now. How about some config?
S
--
JR Richardson wrote:
Zoa wrote:
On friday we finally released Attrafax under a GPL2 license.
It comes with its own set of modems and built in transparent gatewaying.
The solution should be quite stable as long as the line quality is ok.
(Some tools for measuring the line quality are
Is there a way for a client to tell a server where it is registered to
remove the registration?
/voipfc
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New to Asterisk? Join us for a live introductory
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Hi
I'm looking for a good phone SIP phone for conference room use.
My requirements are in order:
* Speaker quality
* External microphone support.
* Provisioning support / asterisk compatibility.
Does anyone have any experience on this field?
I'm
Hi,
I am restricting the caller ID from the SONUS class 5 server towards
the asterisk. But Asterisk is not considering it to be restricted and it
shows the caller ID of the SIP phone on the DAHDI lines. Kindly let me
know what all logs.config info are required for this?
I am attaching the
Hi,
Can anyone tell me at what time a realtime member with dadhi channel will
enter the state Ringing?
All other realtime DAHDI queue members on my asterisk system will always
have the state (Unknown) no matter if it is ringing or not.
But this one queue/queuemember sometimes sudden enters the
On Thu, 11 Mar 2010, Tommy Botten Jensen wrote:
Hi
I'm looking for a good phone SIP phone for conference room use.
My requirements are in order:
* Speaker quality
* External microphone support.
* Provisioning support / asterisk compatibility.
Does anyone have any experience on this
On 11 Mar 2010, at 12:43, Gordon Henderson wrote:
On Thu, 11 Mar 2010, Tommy Botten Jensen wrote:
Hi
I'm looking for a good phone SIP phone for conference room use.
My requirements are in order:
* Speaker quality
* External microphone support.
* Provisioning support / asterisk
On Thu, Mar 11, 2010 at 08:21:09PM +0800, Gopalakrishnaiyer Venugopal-Q16770
wrote:
Hi,
I am restricting the caller ID from the SONUS class 5 server towards
the asterisk.
What does (technically) a restricted caller ID mean? How should I be
able to tell a caller ID string is restricted?
Frank Church wrote:
Is there a way for a client to tell a server where it is registered to
remove the registration?
Assuming you are talking about a SIP peer (since you didn't specify),
yes, the SIP peer can cancel the registration by sending an update to
the registration and setting the
This patch adds a 'DBGetComplete' event after the 'DBGetResponse' to bring
the behavior of the DBGet Action in line with how other actions behave. I
have tested the patch against 1.4.29.1 and it worked for me.
Patch is available on the issue page.
https://issues.asterisk.org/view.php?id=16965
Is there a way for a client to tell a server where it is registered to
remove the registration?
Yes, it needs to send an UNREGISTER sip message.
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New
I am very familiar Rhino Channel Banks and what to expect from them. I
am intrigued by the Xorcom USB Channel Banks simply because I don't have
to burn a hardware port... Can anyone comment on the Xorcom Astribank
(24 FXS channels) and how well it works in an asterisk environment?
I appreciate
On Thu, 11 Mar 2010 12:56:05 +0100, Tommy Botten Jensen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Hi
I'm looking for a good phone SIP phone for conference room use.
My requirements are in order:
* Speaker quality
* External microphone support.
* Provisioning support / asterisk
I'm using Asterisk 1.4.25.1, can I do something like :
== extconfig.conf ==
hints = mysql,asterisk,hints
(got the info from https://issues.asterisk.org/view.php?id=16059 )
On Wed, 2010-03-10 at 18:26 +0100, jonas kellens wrote:
Hello list,
Can I do something like this for BLF
Voip-info.org has some BLF snippets that you can probably adapt to do what
you need.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Thursday, March 11, 2010 9:02 AM
To: Asterisk Users Mailing List -
11 mar 2010 kl. 15.17 skrev Philipp von Klitzing:
Is there a way for a client to tell a server where it is registered to
remove the registration?
Yes, it needs to send an UNREGISTER sip message.
There's actually not an UNREGISTER method in SIP.
As Kevin stated, you send a REGISTER with a
Hi
There's actually not an UNREGISTER method in SIP.
As Kevin stated, you send a REGISTER with a zero expiry to cancel a
current registration.
Yes, of course you are right there, sorry for the confusion. I was
thinking about the resulting Asterisk CLI message:
Unregistered SIP 'peername'
How can I set the prefered codec between 2 calling parties ??
My Grandstream supports G729, alaw and gsm... in this order.
The Zoiper softphone has alaw and gsm as codecs... in that order.
Although there should be a matching codec found, my Grandstream can not
call the Zoiper softphone.
CLI
Thanks.
Is there command is used for that?
I have checked the help show and there is no command like sip register
or sip unregister in the list.
Is it available on version 1.4?
On 11 March 2010 13:08, Kevin P. Fleming kpflem...@digium.com wrote:
Frank Church wrote:
Is there a way for a
The las time I had this problem it was because I had a typo in an
exclude statement in my dialplan, all the dialplan commands were gone
from the CLI except dialplan show. Go over your dialplan and maybe
start with a new extensions.conf and start pasting parts of your
dialplan until it
Hello,
I am building a bonded multi-link T1 solution and I have a few questions
about the Digium TE4xx line of cards:
0. Can hdlc0..hdlcX instances of Digium cards be bonded in the same way
that the Rhino R2T1 card knowledge base article suggests for Rhino
cards?
On 9/03/10 9:13 PM, Peter Childs wrote:
Also is there some way to get the starting end to auto pickup, (or at
least hit for this to happen (I'm using SIP if that helps))
When you make an originate request it works like this:
1. Call is made to the Channel parameter.
2. When the Channel answers
2010/3/11 Eric Wheeler aster...@ew.ewheeler.org:
4. Does anyone have a couple TE2xx or TE4xx cards that can test such a
configuration? I would like to research their capability before
purchasing a couple $1200 cards.
Hi Eric,
I have four spare TE411P but never used bonded T1 or T1 for data
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Dear Lito,
*The information in this email is given to you in
On Fri, 12 Mar 2010, Angelito Manansala wrote:
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Dear Lito,
I'm still awaiting detailed Avaya config info, but I have learned more.
Hopefully someone will have some useful input to help me out..
When we try to connect the Asterisk box directly to the Avaya (bypass the
Gatekeeper/CLAN), we get this error in the ooh323 log:
ERROR:Failed to connect to remote
On Wed, Mar 10, 2010 at 6:39 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Tue, Mar 09, 2010 at 06:20:53PM -0500, sean darcy wrote:
Receiving a fax pstn - pstn with 1.6.2.6-rc2:
-- Executing [...@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in
new stack
-- Executing
In case someone wants to see the detailed ooh323 log (which shows the failed
attempt to connect to the gatekeeper). I appreciate any help!!
21:32:06:832 Sent GRQ message
21:32:06:885 GkClient Received RAS Message
21:32:06:885 Received RAS Message = {
21:32:06:885 gatekeeperConfirm = {
I get a warning every time I run DeadAGI from the h extension:
-- Executing [...@cc:2] DeadAGI(Zap/7-1, graba_dialer.agi) in new stack
[Mar 11 20:50:10] WARNING[8598]: res_agi.c:2203 deadagi_exec: Running DeadAGI
on a live channel will cause problems, please use AGI
I use the agi
Post your Asterisk's sip.conf
On Thu, Mar 11, 2010 at 10:39 PM, jonas kellens jonas.kell...@telenet.bewrote:
How can I set the prefered codec between 2 calling parties ??
My Grandstream supports *G729, alaw and gsm*... in this order.
The Zoiper softphone has *alaw and gsm* as codecs... in
Hi,
i think that you should modify the cdr_addon_mysql module, otherwise you can
add it in the userfield.
2010/3/11 Alejandro Recarey alexreca...@gmail.com
Hi all,
I've been trying to add a custom mysql field to my CDR's, but I must
be doing something wrong.
I am using asterisk 1.4 and
Sip.conf :
[general]
;context=default
allowguest=no
allowoverlap=no
allowtransfer=yes
realm=mydomain
bindport=5060
bindaddr=X.X.X.X
maxexpiry=1800
minexpiry=60
mohinterpret=default
mohsuggest=default
language=be
useragent=mycorp
dtmfmode = rfc2833
alwaysauthreject = yes
On Thu, 2010-03-04 at 01:48 -0600, Tilghman Lesher wrote:
On Wednesday 03 March 2010 22:20:40 Pham Quy wrote:
It maybe not clear that what i'm going to do.
What i want to do is that enable user to call to a number then a
background music will be played and he/she sing to mobilephone, the
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