Hi BB,
you could try this:
http://asterisk-outlook-dialer.voip-singapore.qarchive.org/
Never tested it deeply but apparently seems to work fine.
Giorgio Incantalupo
Bruce B wrote:
Hi Guys,
What is out there other than OutCall that works with M$ Outlook and
Asterisk 1.4/1.6 ? I prefer
Do you forward the call from SIP phone or Asterisk dialplan.
If it is from SIP Phone, above solution will not work. Infact any
solution will not work except your softphone supports call forwarding
based on some filter parameters.
--AsteriskMan
On 1/5/11, Danny Nicholas da...@debsinc.com wrote:
On 110103 2154, Barry Miller wrote:
On Mon, Jan 03, 2011 at 08:04:48PM -0800, Kirill Katsnelson wrote:
Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped?
They're now part of the wiki.
Thanks!
-kkm
--
Flavio Miranda wrote:
I really would like to understand why dont works!
The variable should be ${CALLERID(num)}, unless you have a very old
version of Asterisk.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither
Le 04/01/2011 20:50, Sebastian a écrit :
Hi,
On 01/04/2011 03:24 PM, Administrator TOOTAI wrote:
Le 04/01/2011 11:50, Gilles a écrit :
[...]
It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited
Internet plan would solve the issue.
I Would avoid OpenVPN (tested an Android)
Shaun Ruffell sruff...@digium.com wrote:
On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 01/04/2011 05:09 PM, cov...@ccs.covici.com wrote:
Hi. I
Hello!
I am trying to figure out how call transfers work in SIP. What
extension does the transferring and transferee devices go to?
Elliot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hello list,
is it possible to add the field Privacy: id to a SIP INVITE message ?
INVITE sip:32444666...@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP1 .2.3.4:5060
From: R321113 sip:3211133...@1.2.3.4;tag=2096790244
To: sip:32444666...@1.2.3.4
Call-ID:
It is indeed possible (quite common, actually) to run the wiring as you
describe. If you want to keep the data and voice traffic separate, you
can use VLANs to do so. Your switches will need to support VLANS, and
you will need to configure VLANs to separate the voice and data traffic.
As I
I'd definitely look into a phone mounted to the wall that has no actual
handset, but merely buttons and a speaker grille. It should probably
additionally be stainless steel, as I suspect it will need a good cleaning
at least daily.
The Polycom phones look great on a desk, but they are not
On 01/04/2011 09:02 PM, mgra...@mstvp.com wrote:
IMHO G.722 beats Clarity By Polycom every time.
I had an IP335 for review before they launched. The audio quality is the
same as the better models (IP450/550/650) only the user interface is
different. Very good speakerphone, too.
Review here:
I would. The whole Polycom line seems designed for desktop use, and the
speakers just don't get very loud. I have especially had this complaint
about the ring volume, even at some desktops!
In the hotels where we have installations that include busy kitchen
extensions there seems to be no
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all other SIP phones that advertise the HD voice codec like Aastra?
3- What is the main difference between the
On 01/05/2011 07:50 AM, Andy Graybeal wrote:
I'd definitely look into a phone mounted to the wall that has no actual
handset, but merely buttons and a speaker grille. It should probably
additionally be stainless steel, as I suspect it will need a good cleaning
at least daily.
The Polycom
On 01/05/2011 01:29 AM, Bruce B wrote:
Hi Everyone,
You should really spend some time learning how to use a widely available
search engine, such as Google or Bing. Many of the questions you ask
here could be quickly answered that way.
1- Are the Siren7 and Siren14 the G.722 HD voice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Wednesday, January 05, 2011 4:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Calls Transfers
Can anyone give me some pointers on the following, in our setup (ast
1.6.3) we use international carriers to terminate calls for a
callingcard system, we have an issue where there can be a very long
delay after dialing but before the far end begins to ring.
I would like to play a tone every
On 01/05/2011 05:02 AM, cov...@ccs.covici.com wrote:
Shaun Ruffell sruff...@digium.com wrote:
On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 01/04/2011
Jonas
This is how we are doing it.
exten = s,n,SipAddHeader(P-Asserted-Identity: :${siteDefaultCIDNumber})
exten = s,n,GosubIf($[${gbl_CallPrivacy}=id]?rfc-3325-CPN,1)
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten =
Hi,
On 01/05/2011 10:49 AM, Administrator TOOTAI wrote:
Le 04/01/2011 20:50, Sebastian a écrit :
Hi,
On 01/04/2011 03:24 PM, Administrator TOOTAI wrote:
Le 04/01/2011 11:50, Gilles a écrit :
[...]
It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited
Internet plan would
On Jan 4, 2011, at 12:26 PM, Tilghman Lesher wrote:
It wasn't designed to do this. While you can have the same sippeers table
for multiple servers, you really should have a separate sipregs table for
each backend server. The reason why is that some mappings depend
implicitly on the host to
Hi
We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture
I have upgraded the asterisk version in one of our test environments and
blind
If you do get a Polycom, the old 501 (discontinued) have a louder ring (or
can be configured to have a louder ring, don`t quite remember) then the
newer ones. But the others are right: it's not meant for this, at least not
in a noisy environment. What can work though is a Polycom 321, with a
On 01/05/2011 07:07 AM, Steve Underwood wrote:
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all other SIP phones that advertise the HD voice codec like
On Wednesday 05 January 2011 07:07:10 Steve Underwood wrote:
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all other SIP phones that advertise the HD
Hello list,
I'm having DTMF-troubles with a Snom phone. I want to know if it's the
Snom or Asterisk that makes the trouble.
I'm playing a prompt, then make a choice for 2 :
[Jan 5 17:06:38] VERBOSE[29172] file.c: [Jan 5 17:06:38] --
SIP/test1-0701 Playing
Hi !
I ma having trouble with my PTSN line. When I call to my asterisk I get this..
-- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn
extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'-- Hungup 'Zap/5-1'
-- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06]
On Wednesday 05 January 2011 06:50:19 Andy Graybeal wrote:
I'd definitely look into a phone mounted to the wall that has no
actual handset, but merely buttons and a speaker grille. It should
probably additionally be stainless steel, as I suspect it will need a
good cleaning at least
I have just built a new system using an HP DL360G7 with a TE420 T1 card,
and this is the first system using a generation 7 server. I'm not sure
whether that is an issue or not.
I am using Asterisk 1.2, and Zaptel 1.4.12.1 with patches for GEN5 of
the TE420 card. I have successfully used this
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone system? Might it make more sense to have a non-phone-based
intercom system, plus a phone for
Shaun Ruffell sruff...@digium.com wrote:
On 01/05/2011 05:02 AM, cov...@ccs.covici.com wrote:
Shaun Ruffell sruff...@digium.com wrote:
On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote:
Shaun
Shaun Ruffell sruff...@digium.com wrote:
On 01/05/2011 05:02 AM, cov...@ccs.covici.com wrote:
Shaun Ruffell sruff...@digium.com wrote:
On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote:
Shaun
Thanks!
Although there is no difference between SIP or any other technology,
how does Asterisk reconcile the channel variables?
For example:
1. A calls B
2. B answers
3. B transfers A to C
4. C picks up call
Now, when B makes a transfer (say by pressing the transfer button on a
sip phone), what
Shaun Ruffell sruff...@digium.com wrote:
On 01/05/2011 05:02 AM, cov...@ccs.covici.com wrote:
Shaun Ruffell sruff...@digium.com wrote:
On 1/5/11 12:46 AM, cov...@ccs.covici.com wrote:
Shaun Ruffellsruff...@digium.com wrote:
On 1/4/11 9:26 PM, cov...@ccs.covici.com wrote:
Shaun
On 01/05/2011 09:39 AM, Bryan Field-Elliot wrote:
We have one table which is serving both purposes (peers and reg). When
we want to route a call to an ATA, we first look up that ATA's regserver
in that table, and then construct a SIP URI based upon that regserver
address. In that way, we route
On Wednesday 05 January 2011 09:39:00 Bryan Field-Elliot wrote:
On Jan 4, 2011, at 12:26 PM, Tilghman Lesher wrote:
It wasn't designed to do this. While you can have the same sippeers
table for multiple servers, you really should have a separate sipregs
table for each backend server.
See the following SIP trace.
Where in the world does Asterisk get port 1025 to respond to?
This is asterisk 1.6.x.
Thanks.
-- James
--- SIP read from zzz.zzz.zzz.44:9363 ---
NOTIFY sip:pbx1.mydomain.com SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.140:9363;branch=z9hG4bK-b9a860d3^M
From: xxx-xxx-
Looks like your telco is sending you polarity reversal on sending you a
call. Which is one of the types of setups for analog lines.l
From your console output it looks like the call was handled just fine
other than the 'weird event' notification, which I'm not familiar with.
On 01/05/2011
Top link on Google for stainless steel SIP intercom:
http://www.adamtelco.com/valcom-vip-172l-st-stainless-steel-sip-intercom-
doorphone.html
Cyberdata appears to have another, too:
http://www.alloy.com.au/010935.htm
Yet another:
For some reason our Asterisk box is doing something really unusual following
applying a routine update to CentOS 5 on Monday.
We have Asterisk 1.4.2 and its been working great for years. But now when the
phone system receives an incoming SIP call, its not providing any audible dial
sound to
On 01/06/2011 01:04 AM, Tilghman Lesher wrote:
On Wednesday 05 January 2011 07:07:10 Steve Underwood wrote:
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all
On 01/06/2011 12:05 AM, Kevin P. Fleming wrote:
On 01/05/2011 07:07 AM, Steve Underwood wrote:
On 01/05/2011 03:29 PM, Bruce B wrote:
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal
across all other SIP
Some more info on this weirdness
Seems that if I play any audio out from Asterisk first the problem goes away.
its almost like the entire audio engine isn't being 'initialized' or something
on those direct calls. I found that if I execute a Swift(Please Wait) before I
call the Dial
On 01/05/2011 01:51 PM, Tom Rymes wrote:
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone system? Might it make more sense to have a
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