9:29, John Covici <cov...@ccs.covici.com> wrote:
> I always set it to no, but set the registration time to 60 seconds,
> and that has always worked for me.
>
> On Wed, 23 Aug 2017 17:23:38 -0400,
> Gary Reuter wrote:
>>
>> Hello,
>> We've had dozens of Polycom
Hello,
We've had dozens of Polycom 3.x firmware phones deployed and working
great for years.
Now I've finally been charged with the long-overdue task of figuring
out why newer Polycom devices with 4.x firmware register fine but do
not respond to SIP OPTIONS request and therefore always become
?
Thanks,
Gary Kuznitz
WPM$LEX5.PM$
Description: Mail message body
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Thanks again for your time!
Kind Regards,
Gary Shergill
- Original Message -
From: Amit Patkar a...@avhan.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, May 21, 2014 4:55:57 PM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external
asterisk
any logs required, I have some logs from when it works
and doesn't.
Thank you for your help.
Kind Regards,
Gary Shergill
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itself, it is just
talking to an Asterisk server (and that asterisk server is the one which talks
to the webrtc client).
Thank you.
Kind Regards,
Gary Shergill
- Original Message -
From: Amit Patkar a...@avhan.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, May 21, 2014 04:41
).
Unsure what would be causing this, because it does work sometimes and doesn't
at others, with no obvious reason either way.
Thanks again.
Kind Regards,
Gary Shergill
- Original Message -
From: Gary Shergill gsherg...@gltd.net
To: Asterisk Users Mailing List - Non-Commercial
) disconnection of calls
6) de-initialization
And perhaps showing how two channels are connected to create a conversation?
Thanks in advance,
Gary
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Is it possible to issue the POKE to a end point from the CLI? Our
asterisk servers is not seeing some end points drop off and I would like
to create a script to manually check end points.
Thanks!
Gary
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I received the same spam myself.
Regards,
Gary Carr
List users,
Did anyone else recently receive spam from DIDForSale with the subject
DIDForSale 2012 achievements? I suspect that they are using this
list to harvest email addresses and think they should be called out on
this poor business
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Gary Carr
Sent: Wednesday, October 03, 2012 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call extension play
I am trying to setup a context to take a inbound call, hold the call,
connect to an external number, play a sound file to the external number,
then connect the inbound caller to the external number.
My thought was to accept the call and place them in a parking lot. Then call
the external
I am trying to track down a white noise problem we are having in our conference
rooms. If there are 3 or 4 users in the conference the quality is good. After
we get more users in the conference we develop a white noise that gets louder
as more users come online. I have tried both meetme and
Currently running version 1.8.16.0 and trying to manage confbridge rooms and
users. When I try to use the confbridge cli command I get a command not found
error.
CLI confbridge
No such command 'confbridge' (type 'core show help confbridge' for other
possible commands)
I've tried googling
? Why is it firing? I saw a similar
post from 2007 where the person had the same issue.
http://forums.digium.com/viewtopic.php?p=60477sid=caa115851aab005f6e56a218a81618b9
Any help anyone can provide would be greatly appreciated.
Gary
I have a couple of questions about asterisk 1.6:
Can you change codecs mid-call upon re-invite?
Can you handle the SDP offer-answer in the 200-ACK instead of the usual
INVITE-200?
Thanks in advance for any insight.
Gary
...@evaristesys.comwrote:
On 05/03/2011 12:43 PM, Gary Graves wrote:
Can you change codecs mid-call upon re-invite?
Do you mean: can Asterisk be configured to _initiate_ such a change at
some point, mid-call? Or do you mean: Will Asterisk properly react to such
a re-INVITE and change codecs if asked to do so
, the called number
answers, but the server only detects the answer about 3 seconds later ..
The outgoing line is TDM on port 2 of the TDM800P.
The same hardware was running Asterisk 1.4 recently and we didn't
have this problem.
Where should I look?
Thanks
Gary B
Well... Looks like he's trying to use a streaming MOH solution like an online
radio station or something, so the files are irrelevant. Too bad the original
post didn't specify that. I still think there is a different source selected
for the call queue than for the rest of the system.
Sorry
On Sat, Jan 15, 2011 at 7:20 AM, James Miller paramedi...@gmail.com wrote:
I have it all configured and it should work, and it did briefly several
weeks ago, however now, it doesn't work at all and only plays the default
hold music.
the middle, and still can not get MOH to work.
Did you
On Sun, Jan 16, 2011 at 11:41 AM, Warren Selby wcse...@selbytech.com wrote:
MOH plays the default class unless specified by a channel variable to play a
different one. In queues.conf you can specify the MOH class on a queue by
queue basis, but that's the hold music for someone waiting to be
I have it all configured and it should work, and it did briefly several
weeks ago, however now, it doesn't work at all and only plays the default
hold music.
If it is playing the default music, then the MOH function is working. What
do you get from moh show files in Asterisk?
--
-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gary Allen
*Sent:* Saturday, January 15, 2011 11:33 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Music on Hold not working?
I have it all
RTP always uses a random even numbered port, then RTCP will use the next
port, which will always be odd numbered. Symmetric RTP only needs two
ports, while asymmetric RTP uses four.
http://www.armware.dk/RFC/rfc/rfc4961.html
On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote:
pbx_extension_helper: No application 'SetVar' for
extension (voicemenu-custom-4, 106, 1)
I'm running Asterisk/1.4.22.
Does anyone have any idea what I need to do to either make SetVar work or
replace it
with something else?
Thanks you,
Gary
)
-- Executing [s@macro-stdexten:3] Dial(DAHDI/7-1, SIP/106|20|) in new
stack
What did I do wrong in adding the first two lines?
Thank you,
Gary
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What type of phones? Easy to do with Polycom and several others from Asterisk
CLI.
Sent from my BlackBerry® smartphone
-Original Message-
From: Nikhil d.nik...@cem-solutions.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 28 Dec 2010 08:42:22
To:
=0.0.0.0/0.0.0.0
permit=192.168.1.201/255.255.255.255
All the other local phones here
snip
One WanIP address
Thank you,
Gary Kuznitz
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Thank you for the reply.
On 10 Dec 2010 at 8:53, Jeroen (Jeroen Eeuwes jeroeneeu...@gmail.com)
commented about Re: [asterisk-users] Configuring Softphone:
Hi Gary,
I not using anything to create my dialplan. I'm trying to add a softphone
to a dialplan
that was created a couple years
On 10 Dec 2010 at 9:31, Jeroen (Jeroen Eeuwes jeroeneeu...@gmail.com)
commented about Re: [asterisk-users] Configuring Softphone:
Hi Gary,
That is a great suggestion. Yes I did try that. I might be having router
issues with a
SonicWall. I'm working with a port sniffer now to try
this Asterisk box let me know. I need to
find a
way to block these also.
Gary
On 9 Dec 2010 at 7:57, Joe (Joe Greco asterisk-users@lists.digium.com)
commented
about [asterisk-users] Asterisk SIP attac:
Hello,
We had been seeing SIP-guessing attacks on our Asterisk server here.
While
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Configuring Softphone:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary
On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz docf...@theoffice.la) commented
about
[asterisk-users] (Fwd) Re: Configuring Softphone:
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Configuring Softphone
Thanks for the reply.
On 9 Dec 2010 at 20:56, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] (Fwd) Re: Configuring Softp:
On Thu, 9 Dec 2010, Gary Kuznitz wrote:
I'm getting closer. Express Talk is now making the call.
I'm getting an error
I see in sip debug it says Audio is at port 10342
Express Talk suggests Audio at UDP 8000-8020
I've tried changing Express Talk to 1 and forwarded 1-10400.
Is there a possibility Express Talk won't work in the 1 range?
Is it possible to limit Asterisk to 8000-8020?
Thank you,
Gary
On 9 Dec 2010 at 22:32, Gary (Gary Kuznitz docf...@theoffice.la) commented
about [asterisk-users] Audio ports:
I see in sip debug it says Audio is at port 10342
Express Talk suggests Audio at UDP 8000-8020
I've tried changing Express Talk to 1 and forwarded 1-10400
on the cmd line saying:
NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to
extension '91AreaCodePhone#' rejected because extension not found.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2,HangUp()
exten = 120,1,Dial(SIP/Gary
Hi,
I'm trying to get a softphone configured. In Sip.conf [general] I found an
example
that said I need:
nat=yes
localnet=192.168.xxx.xxx
Is localnet supposed to be a LAN IP or a WAN IP?
Thank you,
Gary
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I have no idea the correct way to configure this software phone.
It's called Express Talk
The Asterisk box is at IP = WanLocation
Software phone is at IP = WanSoftware
They are not on the same LAN.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten = 1001,2
Shouldn't Asterisk be listening on UDP port 5060?
I'm working with an Asterisk installation running in Ubuntu. Asterisk is
running but
non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I
supposed to see something listening?
Thank you,
Gary
On 2 Dec 2010 at 15:11, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Asterisk ports:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz
Sent: Thursday
Thank you for the reply.
On 2 Dec 2010 at 16:23, Jeff (Jeff LaCoursiere j...@sunfone.com) commented
about Re: [asterisk-users] Asterisk ports:
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
Shouldn't Asterisk be listening on UDP port 5060?
I'm working with an Asterisk installation running
Thanks for the reply.
On 2 Dec 2010 at 14:11, Steve (Steve Edwards asterisk-users@lists.digium.com)
commented about Re: [asterisk-users] Asterisk ports:
On Behalf Of Gary Kuznitz
Shouldn't Asterisk be listening on UDP port 5060?
Yes. Unless configured otherwise, that's the SIP port
Port: 5060
UDP Bindaddress:0.0.0.0
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
In sip.conf bindport = 5060
'Sip show settings' doesn't work in 1.4.22
I don't have access to a '1.4' instance right now, but 'sip show settings'
works in 1.2 and 1.6 so I'm guessing
Thank you very much for the reply.
On 2 Dec 2010 at 17:06, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] Asterisk ports:
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
You get extra points today. I think you found where the problem is. It
found /etc
I did wrong?
Thank you,
Gary
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Thank you for the reply.
Comments below...
On 30 Nov 2010 at 20:21, Warren (Warren Selby wcse...@selbytech.com)
commented about Re: [asterisk-users] Trying to configure a SIP so:
On Tue, Nov 30, 2010 at 8:04 PM, Gary Kuznitz docf...@theoffice.la wrote:
I have been told that my logic
On 23 Nov 2010 at 16:54, Joseph (Joseph syscon...@gmail.com) commented about
Re: [asterisk-users] Someone has hacked into our :
On 11/23/10 14:18, Gary Kuznitz wrote:
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk-
us
Thank you for the reply.
On 23 Nov 2010 at 18:51, John (John Novack jnov...@stromberg-carlson.org)
commented about Re: [asterisk-users] Someone has hacked into our :
Gary Kuznitz wrote:
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk-
us...@lists.digium.com) commented about Re: [asterisk-users] Someone has
hacked
into our :
On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote:
I have the log now. I'd like to know
Someone has hacked into our system and is making calls overseas.
How can I:
1. Find out the where the calls are originating from?
2. Block all calls that are not authorized?
Our system is in the USA.
Only calls from inside our LAN are allowed.
Thank you,
Gary Kuznitz
how
to get in
to our box.
Thanks you,
Gary Kuznitz
On 22 Nov 2010 at 13:11, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Someone has hacked into our :
From: Gary Kuznitz [mailto:docf...@theoffice.la]
Sent: Monday, November 22, 2010 12:20 PM
To: Danny
I have remote access to the server so I checked the canreinvite .. they
are all set to no. I can't try the call from here, I will get back to you.
Gary Baribault
On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote:
Do you agree something is blocking the audio in one direction? Can you
do a 'rtp
I have checked, the users have ulaw, then alaw, the phones are set to
711u then 711a which is the same thing (I think).
Gary Baribault
On 06/02/2010 08:32 AM, taimur hasan wrote:
Also check the codecs as if you are using g729 or g723, there is a
chance that they are not available in codecs
I don't know if this makes any difference, I created a lot of this
configuration with the Asterisk-GUI (SVN-branch-2.0-r4980) and when I
edit the users.conf file, there are two entries 'type = peer' for each
extension and they are highlighted in red!
Gary Baribault
On 06/02/2010 08:32 AM, taimur
work fine, sound is correct.
Voice mail works fine as well, the IVR works great.
Any ideas?
Gary Baribault
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Incomming calls are on TDM lines connected to the Digium card. Calls
between extentions are on the LAN for SIP registered users/ip phones.
Gary Baribault
On 06/01/2010 03:32 PM, Zeeshan Zakaria wrote:
Incoming and outgoing calls are on SIP or on ZAP?
Zeeshan A Zakaria
--
Sent from my
This is done while the calls are active? I just issued the command and
got nothing, but there where no active calls.
Gary Baribault
On 06/01/2010 03:45 PM, Danny Nicholas wrote:
My assumption is that inbound/outbound calls are DAHDI and that internal
calls are SIP. Can OP post core show
As I stated, the incoming calls are on TDM DS0s connected to the Digium
card, and the extensions are on the same local network as the Asterisk
server. There is currently no NAT anywhere.
Gary Baribault
On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
Output of 'iptables -L -n' would also
I've tried a number of solutions, but I've been unable to get Asterisk
working with streaming MOH without running into the buffer issue.
I've tried using various combinations madplay, mpg123, mpg321. I've
also tried streamplayer by itself, and in combination with play-fifo (
Please forgive this off-topic post... I've been on this list since
2005 (over 45k messages in my archive) and this is obviously really
not something I normally do.
If you have a minute and are feeling generous, please visit
http://bailout.chipin.com/ and consider helping me out.
Sorry if I've
Hi Folks, sorry for the delay ... I found that the documentation was
rather iffy .. I finally found the defines.php in the lib subdirectory
and figured out how to give the MySQL port with the host and it all
works fine now.
Gary Baribault
Courriel: g...@baribault.net
GPG Key: 0xFA812835
GPG
for open source, the server is not
commercial, and I have very little budget.
Thanks
Gary B
--
Gary Baribault
Courriel: g...@baribault.net
GPG Key: 0xFA812835
GPG Fingerprint: 8597 4D3D 3C3D 4247 077C 9FF9 E412 CAC4 FA81 2835
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using 1.6.1 SVN, r216266.
Gary H
--
Gary Hawkins MBCS gary.hawk...@garyhawkins.me.uk
OpenPGP Key ID: 0x9A1037BB
Web: http://www.garyhawkins.me.uk
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Tilghman Lesher wrote:
On Friday 04 September 2009 12:08:26 Gary Hawkins wrote:
I've just had to enable the requirecalltoken=no option in iax.conf for
one of my IAX2 trunks, and I don't think it works properly in the
realtime version.
[snip]
Please try the attached patch.
I've just tried
this work with the GUI .. any help or
suggestions would be appreciated!
Thanks
Gary B
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this work with the GUI .. any help
or suggestions would be appreciated!
Thanks
Gary B
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to take effect. When I reboot the server, the trunk is gone!
WTF???
--
Gary Baribault
Courriel: g...@baribault.net
GPG Key: 0xFA812835
GPG Fingerprint: 8597 4D3D 3C3D 4247 077C 9FF9 E412 CAC4 FA81 2835
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Thanks ahead,
Best Regards,
Gary
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Gary G.
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,
it does not see or register with my home Asterisk server after I change
it's proxy to point to my home IP address.
Any Ideas? - Is this a router issue (sure seems like it)? - Much thanks in
advance.
Gary G.
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this will fix the problem you were seeing, too.
I've just tested with this revision and all seems to be well again.
Thanks for finding and fixing the bug!
Gary H
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problems to this?
Thanks
Gary H
--
Gary Hawkins [EMAIL PROTECTED]
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I am running asterisk 1.2.27 and it dead today. The following is the backtrace
of core file. Can anybody help me to identify what is the possible cause of
crash?
It seems the mysql connection causing problem in Thread 2. But I can not tell
what exactly happened.
This asterisk is using as ACD
gary wrote:
I am running asterisk 1.2.27 and it dead today. The following is the
backtrace of core file. Can anybody help me to identify what is the
possible cause of crash? It seems the mysql connection causing problem in
Thread 2. But I can not tell what exactly happened. This asterisk is
using
I am running asterisk 1.2.27 and it dead today. The following is the backtrace
of core file. Can anybody help me to identify what is the possible cause of
crash?
It seems the mysql connection causing problem in Thread 2. But I can not tell
what exactly happened.
This asterisk is using as ACD
interested in your results.
Also, if anybody wants to take this off-forum and discuss/help me out, I'll
be greatly thankful. - I have a Broadvoice account and we can even establish
a phonecon.
Thanks VERY MUCH in advance.
Gary Guthary
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- Original Message -
From: Lee, John (Sydney) [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 19, 2008 11:48 PM
Subject: [asterisk-users] Newbie IVR: How to read() before playback()
isfinished?
I am working on a menu to accept input from a caller like as
Asterisk 1.2.26.2
On an ACD call, I can press 0 to do attendant transfer. After talking to the
transfered party, I want to cancel the transfer and get back to the original
party. If I press *, it will disconnect me and complete the transfer. How can I
set it up so I can press * and get the call
I will be out of the office on Wednesday, January 9, 2008. If this is an
emergency, please call Customer Service at (877) 791-7700. Thank you.
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To
I will be out of the office until Wednesday, January 2, 2008. If this is an
emergency, please call Customer Service at (877) 791-7700. Thank you have a
great holiday season!
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I used ChanSpy( ) recorded some test conversations. It has .raw extension.
What kind of audio file is this? How can I play it?
Gary
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hoping this is something that is
configurable in the XML or on the phone UI.
Thanks
Gary
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asterisk-users
David,
Yes, I'm aware of that, but unfortunately it does two calls on each
line appearance (button), so the first two calls go on line 1, and the
third will appear on line 2. I'd like to limit it to 1 call per line.
Any ideas?
Gary
On 9/25/07, David Cook [EMAIL PROTECTED] wrote:
Gary, if you
Sorry to drag up an old thread, but the backport of ringinuse is a
godsend for those of use stuck using asterisk 1.2 (trixbox 2.2). Many
thanks, Gavin
GTG
On 1/21/07, Gavin Hamill [EMAIL PROTECTED] wrote:
Nothing much to be said.. I backported ringinuse, autofill and the QueueLog
application
I am using a2billing as calling card platform with asterisk 1.2.17.
After running for several days, if I issue 'sip show channels' command, I got a
lot of dead sip channels although 'show channels' command only show 5
channels. What cause these dead channels? How can I clean out these dead
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 22, 2007 10:51 PM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 37, Issue 88
Send asterisk-users mailing list submissions to
(2)
exten = 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|HL(12:61000:3))
exten = 8111001001,n,Hangup()
It worked with previous version 1.2.18 on this box. I am new to 1.4. Do I miss
something? Or is it just a bug?
Gary Chen
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* *
Park Call
Dynamic Feature Default Current
--- --- ---
(none)
Call parking
Parking extension : 700
Parking context : parkedcalls
Parked call extensions: 701-720
What else do I need to do to make the features work?
Gary Chen
I recorded some sound files using Asterisk record() app as ulaw file. I need to
edit these sound files. What kind of audio editor can I use to edit these files?
Gary Chen___
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asterisk
to give it a shot.
If anybody wants to take me by the hand and lead me to a solution, I'll be
truly gratefull! - If you want to take it off-line (off-list), please e-mail
me: gary at guthary dot com
Oh Yeah! - Whatever I learn from this adventure will be fully documented an
made freely available
?
Thanks in advance.
Gary Guthary
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Sorry to clutter up the mailiing list, but I've been unable to post to this
list for the past 2 WEEKS!
My ISP's blocking SMPT from other than his own servers.
I think I've worked around it. - But if I see this message in the digest
then I know I'm okay.
Again. - Sorry for any inconvenience.
Gary
- Original Message -
From: Darrick Hartman (lists) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, June 24, 2007 11:25 AM
Subject: Re: [asterisk-users] inband DTMF for g729
Gang Chen wrote:
On 6/22/07, Gary
This is the required dial plan:
0+61|XXX.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall
Sent: Friday, June 22, 2007 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] international numbers...
Using
type of incoming call this
is.
Just thinking. - Is this do-able?
Thanks in advance
Gary Guthary
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Everyone is going to have their sacred cow on this one so suspect you might
have opened a can of worms ...
I can tell you that I have very good results using a number of different
Intel based SuperMicro servers ... these seem to be very mundane and
extremely well behaved ... I have used both
Does anybody know why Asterisk does not support inband DTMF for G.729?
Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it
for our Asterisk IVR system.
Any suggestion to solve this problem?
Gary___
--Bandwidth and
-users] inband DTMF for g729
Sounds like you need a new SIP carrier. G.729 has a way of
destroying inband DTMF tones.
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
On Jun 22, 2007, at 1:20 PM, Gary Chen wrote:
Does anybody know why Asterisk does not support inband DTMF for G
Hi!
Hope someone can help me. I'm trying to pass SIP traffic from one asterisk
to another through a third server. Here is the desired scenario:
ServerA -- SIP -- ServerB -- SIP -- ServerC
When a call is placed on a ServerA local, I can see that ServerB receives
the call and dials ServerC. But
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