Hi all,
I want to configure a SIP Channel to send Alert-Info with the INVITE. Right
now I have added an extension like so:
exten=,n(ring5),SIPAddHeader(Alert-Info: R0)
But there is no Alert-Info in the INVITE.
Any idea how I can get this working? Seems a very basic error..
Thanks.
Sandip
Hi,
Does anyone have a simple UK Extensions.conf that uses SIP that I
could have a look at? I'm trying to get a simple Asterisk system up
and running and I can't get any calls incoming or outgoing and I
don't know why.
Would it be worth posting up my extensions.conf and sip.conf?
Thanks
Hye everybody, anyone has any idea how to help me?
To resume, I just want to know how to change the IP in the URI sent by
Asterisk (first line of SIP packets)
Thanks for your time!
++
On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent lf...@leurent.eu wrote:
Hello All,
I have a little
Modify the $ru pseudovariable or use rewritehostport() out of core.
This is not the right mailing list. This belongs on the
OpenSIPS/OpenSER lists.
There is also a mailing list we operate called SER-Asterisk-Interwork
that is specifically intended to address SER* / Asterisk integration
There's nothing UK-specific about the configuration if it's SIP.
Richard Heeley wrote:
Hi,
Does anyone have a simple UK Extensions.conf that uses SIP that I
could have a look at? I'm trying to get a simple Asterisk system up
and running and I can't get any calls incoming or outgoing
Yes, I am installing on a 64 Bit OS... why, what does it make for a
difference on what or which OS it is getting compiled?!
2009/3/22 Sebastian s...@adinet.com.uy:
Are you installing on a 64bit OS?? Which Os are you using??
-Original Message-
From:
Anyone know of a good dial plan example for call in / call out?
I want to be able to call my Asterisk server, auth, and then call out
any number.
Michael
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On 23 Mar 2009, at 09:49, Tamer Higazi wrote:
Yes, I am installing on a 64 Bit OS... why, what does it make for a
difference on what or which OS it is getting compiled?!
Well you clearly know better than us what the problem is. Although, I
would have said:
On Monday 23 March 2009 11:01:40 Michael wrote:
Anyone know of a good dial plan example for call in / call out?
I want to be able to call my Asterisk server, auth, and then call out
any number.
http://www.voip-info.org/wiki-Asterisk+cmd+DISA
t.
Hello,
it is not an OpenSIPs problem I have, it's an Asterisk one,
I would like to change the URI in message generated by Asterisk.
Thanks
Le Monday 23 March 2009 10.35:09 Alex Balashov, vous avez écrit :
Modify the $ru pseudovariable or use rewritehostport() out of core.
This is not the
The Request URI generated in an INVITE originated by Asterisk is
governed entirely by the parameters passed to Dial().
For example:
Dial(SIP/1...@peer_name)
... will generate a Request URI of
1...@host.or.ip.of.sip.conf.peer.named.peer_name.
It is also possible to send requests to hosts
Good afternoon everybody.
I first would like you to excuse me for my english.
I have an issue with a SIP call ID which is not changed in the call
configuration described bellow :
I have an Asterisk Server A using only SIP protocol.
Behind A there are 2 distant clients (using softphone
Thank you, this is exactly what I needed!!
In order to Dial any number to a registered peer, I just have to enter
Dial(SIP/anynum...@sippeername)
Best Regards!
Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit :
The Request URI generated in an INVITE originated by Asterisk is
Hello, with asterisk 1.6 i am trying to make a dialplan
Which i have such entry in extensions.conf
exten = _8XXX,1,Dial(SIP/${EXTEN})
But some of my clients have both IAX and SIP accounts, to use iax clients
while outside of my Local Area, and SIP clients (or hardware phones) in
local area.
But
Thomas Stein wrote:
On Monday 23 March 2009 11:01:40 Michael wrote:
Anyone know of a good dial plan example for call in / call out?
I want to be able to call my Asterisk server, auth, and then call out
any number.
http://www.voip-info.org/wiki-Asterisk+cmd+DISA
t.
Thank
On Monday 23 March 2009 00:52:29 Rilawich Ango wrote:
On Mon, Mar 23, 2009 at 11:24 AM, Tilghman Lesher wrote:
On Sunday 22 March 2009 21:40:14 Rilawich Ango wrote:
I found that a new field lastms is used in 1.4.24. What is the
usage of that field and the datatype of it?
It's an
Maybe the Siemens DE380 IP R could help you. It's a brand new IP phone with
an integrated router.
Chris
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I have spoken to quickly,
Usually Asterisk on an incoming call sends an INVITE Reg.Contact
Number@Reg Contact IP to the Peer IP.
With the command you gave me, it is possible to send an INVITE
othernumber@Peer IP to the Peer IP.
What I would like to do is to send
I am trying this also, and authenticate and enter password, which is
accepted, but I don't get the actual DISA function to work. I get the dial
tone... and then when dialing, Congestion.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
So I'm guessing, I would disable any wrapup on the queue, and then in my
'h' extension pause the agent for a set period of time, with another
extension to unpause the agent if entered?
Or is there a better way to set the pause after the call is over?
Thanks!
--
Regards,
Robert Broyles
Hi, we are looking to roll out PBX IN A Flash at our office.
The first group will be using Soft Phones (X-Lite appears to be the best
and works in Windows, Apple Linux).
There are many types of USB Headsets to choose from and a fairly broad
price range. Is there any USB headsets people would
We've had no end of trouble with usb headsets on linux (especially cmedia
chipset), as soon as you touch the volume control the sound settings all
mess up... i'm sure there'll be an alsa seting somewhere which would solve
this but i'm not that clued up on alsa so opted for using standard 2
Edward Gray wrote:
Hi, we are looking to roll out PBX IN A Flash at our office.
The first group will be using Soft Phones (X-Lite appears to be the best
and works in Windows, Apple Linux).
tsk tsk tsk :P (I'm working for the zoiper.com :p )
There are many types of USB Headsets to choose
Thank, is there advantages to Zoiper? The interface didn't seem that
great, I haven't checked to see if it's compatible on Linux or Apple yet.
Edward Gray
Director, Vendor Management
Tucows.com Co.
eg...@tucows.com
Direct : (416) 538-5483
Work : (416) 535-0123 Ext. 1277
Fax : (416) 531-5584
I'm using
exten = s,42,SIPAddHeader(Alert-Info: info=Bellcore-r2)
and it works just fine.
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My advice?
If mandated with a USB device and softphone, I would certainly go with
Plantronics.
My question is why not pick up some real Polycom 430s or something and
realize that you really just saved yourself a great deal of time and
money in all reality.
Softphones, like inkjet printers,
I agree more than you know, I am not a fan and neither are many of the
technical folks at our office. The problem is, the business owner and my
superiors don't seem so concerned.
Suffice to say, if they come back with reports of caller quality issues,
aside from the obvious troubleshooting
On Mon, Mar 23, 2009 at 10:24:33AM -0400, Edward Gray wrote:
I agree more than you know, I am not a fan and neither are many of the
technical folks at our office. The problem is, the business owner and my
superiors don't seem so concerned.
Suffice to say, if they come back with reports of
On Mon, Mar 23, 2009 at 10:31 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Mon, Mar 23, 2009 at 10:24:33AM -0400, Edward Gray wrote:
I agree more than you know, I am not a fan and neither are many of the
technical folks at our office. The problem is, the business owner and my
superiors
I disagree with your opinion on softphones, i think they're great, saved
thousands in cabling, switch and phone costs.
I've had 50 agents running diskless/pxe linux (fedora 8), firefox,
thunderbird and twinkle and never had any problems, in the next few months i
expect to have at least 250 agents
I guess it depends on the industry and your standards. One bad call
could cost tens of thousands or much more depending on the
industry.. I have seen several hundred thousand dollar deals fall
through, first hand, because of this.
If you are just telling people to reboot their PCs and
This is always a useful site for these types of questions :)
http://downforeveryoneorjustme.com/
Sebastian wrote:
Anybody knows why is down? Or if has been moved to another page??
--
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk
On Mon, 23 Mar 2009, Tzafrir Cohen wrote:
On Mon, Mar 23, 2009 at 10:24:33AM -0400, Edward Gray wrote:
I agree more than you know, I am not a fan and neither are many of the
technical folks at our office. The problem is, the business owner and my
superiors don't seem so concerned.
Suffice
Tilghman Lesher wrote:
It will have no effect. The issue has always been that if the stream source
changed during a call, the sequence numbers could be reset, sometimes
causing audio weirdness. What has changed is that we're now able to tell
the other side to expect such a reset, thus
On Mon, Mar 23, 2009 at 03:09:54PM +, Gordon Henderson wrote:
On Mon, 23 Mar 2009, Tzafrir Cohen wrote:
On Mon, Mar 23, 2009 at 10:24:33AM -0400, Edward Gray wrote:
I agree more than you know, I am not a fan and neither are many of the
technical folks at our office. The problem is,
On Mon, Mar 23, 2009 at 11:44 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Mon, Mar 23, 2009 at 03:09:54PM +, Gordon Henderson wrote:
On Mon, 23 Mar 2009, Tzafrir Cohen wrote:
On Mon, Mar 23, 2009 at 10:24:33AM -0400, Edward Gray wrote:
I agree more than you know, I am not a fan
On Mon, 23 Mar 2009, Tzafrir Cohen wrote:
On Mon, Mar 23, 2009 at 03:09:54PM +, Gordon Henderson wrote:
On Mon, 23 Mar 2009, Tzafrir Cohen wrote:
On Mon, Mar 23, 2009 at 10:24:33AM -0400, Edward Gray wrote:
I agree more than you know, I am not a fan and neither are many of the
technical
On Mon, Mar 23, 2009 at 12:03:42PM -0400, Steve Totaro wrote:
On Mon, Mar 23, 2009 at 11:44 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Mon, Mar 23, 2009 at 03:09:54PM +, Gordon Henderson wrote:
On Mon, 23 Mar 2009, Tzafrir Cohen wrote:
A lot of the issues I've seen have
Tzafrir Cohen schrieb:
On Mon, Mar 23, 2009 at 03:09:54PM +, Gordon Henderson wrote:
On Mon, 23 Mar 2009, Tzafrir Cohen wrote:
On Mon, Mar 23, 2009 at 10:24:33AM -0400, Edward Gray wrote:
I agree more than you know, I am not a fan and neither are many of the
technical folks at our
I'm trying to get the BLF to work correctly on my Polycom phones. I
have the buddy watch working correctly, but can't get the BLF to change
based on the state...
Example:
When an extension is ringing, I get the same 'red light' that I get when
the extension is actually in use...
I
What model Polycom are you using and which BIOS level? In my *, I can tell
if the phone is ringing or in use, but I'm pretty sure the BLF records both
as inuse.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeffrey
2009/3/23 Jeffrey Phelps jphe...@mjlm.com:
I’m trying to get the BLF to work correctly on my Polycom phones. I have
the buddy watch working correctly, but can’t get the BLF to change based on
the state…
Example:
When an extension is ringing, I get the same ‘red light’ that I get when the
2009/3/23 Kevin P. Fleming kpflem...@digium.com:
Tilghman Lesher wrote:
It will have no effect. The issue has always been that if the stream source
changed during a call, the sequence numbers could be reset, sometimes
causing audio weirdness. What has changed is that we're now able to tell
According to what I've read, you have to have a fully functional MS IM
server running with phones registered to make Polycom work correctly. I
threw together a PERL script that uses core show hints and core show
channels together to give an almost live (.5 second or so delayed) line
status.
On Mon, Mar 23, 2009 at 12:39 PM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Mon, Mar 23, 2009 at 12:03:42PM -0400, Steve Totaro wrote:
On Mon, Mar 23, 2009 at 11:44 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Mon, Mar 23, 2009 at 03:09:54PM +, Gordon Henderson wrote:
On
I think we can conclude that hardphones should be used if you cannot under
any circumstances loose the call (power goes down in building, phones still
powered by PoE switches on UPS) or if you prefer/don't mind spending the
extra on hardphones... and softphones if it doesn't make a difference.
I am using Polycom IP550 with BootROM 4.1.2 and SIP 3.1.2...
It appears that the Enhanced BLF feature is what I'm looking for on
the Polycom, but it also appears that it was written to work with the MS
Live Communications server or the BroadSoft Servers...
That sucks...
Thanks,
Hi guys,
I'm looking for a affordable conference phone and a wifi phone that has a
cradle.
Polycom seems to make pretty nice conf phones but the price is a bit crazy for
us. I saw the recommendation with ATA plus an analog Polycom phone but I do
prefer a SIP phone. All because it's just too
On Mon, 23 Mar 2009, Geraint Lee wrote:
but on that subject... plantronics all the way, they seem to realise
that agents will complain if the headsets hurt (too tight, pulls hair
etc etc) and that agents don't really care about the equipment they are
using and so need to be strong.
On 20/03/2009 7:05 a.m., Klaus Darilion wrote:
Until yet I tried to avoid storing permanent data in the astDB. Did you
ever had any issues with astDB - e.g. that data gets lost, DB is corrupt
... when Asterisk crashes?
Not so far on any machines - in fact, the one I developed this on had
On Mon, 23 Mar 2009, Kelvin Chan wrote:
Hi guys,
I'm looking for a affordable conference phone and a wifi phone that has a
cradle.
Polycom seems to make pretty nice conf phones but the price is a bit
crazy for us. I saw the recommendation with ATA plus an analog Polycom
phone but I do
On 19/03/2009 6:00 p.m., Max Alex wrote:
Hi All,
I have a working asterisk 1.4.23.1 on server.
OS: Centos 5.2
Suddenly asterisk has stopped to process calls crashed.
I found that asterisk has generated coredumps.
I have restarted asterisk it started to work as expected without any
issue.
On 20/03/2009 9:27 p.m., michel freiha wrote:
Hi all,
I mentioned in asterisk.conf there is a property maxcalls...I know that
this is the max number of concurrent calls but i need to know please if this
entry is commented out, what is the default number of MAX concurrent calls
supported by
Anyone connected up to it yet?
http://www.skypeforsip.com/
It would seem to make Digiums chan_skype rather pointness, or am I missing
something?
Or is this Digiums chan_skype in a hosted box somewhere?
Gordon
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On Mon, 23 Mar 2009, Kelvin Chan wrote:
Hi guys,
I'm looking for a affordable conference phone and a wifi phone that has
a cradle.
Polycom seems to make pretty nice conf phones but the price is a bit
crazy for us. I saw the recommendation with ATA plus an analog Polycom
phone but I do
On 23/03/2009 8:22 a.m., Grygoriy Dobrovolskyy wrote:
Hello everybody, i am searching a solution for a videoconferencing, Any
solution (Free/commercial). Asterisk is a great software, but recently we
have more and more demands about videoconferencing of 3 or more peoples,
Existing solutions
Steve Totaro wrote:
On Sat, Mar 21, 2009 at 4:17 PM, Steve Kennedy steve-aster...@gbnet.net
wrote:
On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote:
Hi,
The OpenBTS Project is an effort to construct an open-source Unix
application that uses the Universal Software Radio
Just noticed you said DECT headsets... so what i wrote had nothing to do
with them, but i've used them too i think, excellent quality, tested them
with an aastra phone and worked great.
2009/3/23 Geraint Lee gera...@gmail.com
hehe, nice.
i've used those headsets hooked up to an old 4400
hehe, nice.
i've used those headsets hooked up to an old 4400 (well, via an alcatel
phone obviously)... not bad at all and i know the support department could
fix most of them - usual problems were recrimping the rj11 connections and
resoldering the bits inside the volume box thingy (assuming
Hello,
According to this page:
http://bugs.digium.com/svnstats/asterisk/trunk/2006-09.html#298 a
BACKGROUNDSTATUS variable was added in 2006 with revision 43814. As far as I
can understand it should return different values wether the file played with
a Background() command was played through or
On 24/03/2009 9:17 a.m., Harry Vangberg wrote:
Hello,
According to this page:
http://bugs.digium.com/svnstats/asterisk/trunk/2006-09.html#298 a
BACKGROUNDSTATUS variable was added in 2006 with revision 43814. As far as I
can understand it should return different values wether the file played
AFAIK, BACKGROUNDSTATUS is only meant to be a patch, not a feature. But if
your sole concern is whether the message was played, this should work:
Exten = s,1,Background(some-file)
Exten = s,n,noop(we heard the whole file)
Exten = 1,1,noop(pressed 1 before done)
Exten = i,1,Noop(pressed another
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
concrete shielding wireless phones.
So i was thinking to use one of those usb-phones, and plug it into one
of my servers there.
But what i read from the thread, i seems
The Asterisk console is pretty goodbut there was a text version of
one of the softphones once (sjphone, if I remember correctly)
PaulH
Hans Witvliet wrote:
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
On 23 Mar 2009, at 22:44, Hans Witvliet wrote:
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
concrete shielding wireless phones.
So i was thinking to use one of those usb-phones, and plug it into one
of my
For wifi phone, I tried Linksys iPhone. It works well but lacks a
cradle. My users often forget to charge it when they leave for the day
and come back to a dead wifi phone for the next morning.
I still don't get the market for this kind of phone. DECT cordless phones
can be had for
One of our local companies here in the UK are trialling a new conference
phone - the Konftel 300IP SIP however it's still as expensive as a
Polycom, but that might be the $/£ exchange - might be cheaper where you
are?
It seems like an interesting product. Compared to Polycom 7000, it's
Siemens make a range of DECT handsets under the Gigaset model range.
Yes they shit all over every wifi handset I have ever used.
Dect is way better.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001
Hi,
I have a problem with queue strategy. But only 1 of my agents ring, when
someone call.
my queue.conf:
[MyQueue]
strategy=ringall
member = Agent/201
member = Agent/202
announce-holdtime = yes
joinempty = strict
leavewhenempty=yes
my extension.conf:
exten=8708464,1,Answer
Are both of your agents logged in?
What does the CLI show?
PaulH
edw...@web.de wrote:
Hi,
I have a problem with queue strategy. But only 1 of my agents ring,
when someone call.
my queue.conf:
[MyQueue]
strategy=ringall
member = Agent/201
member = Agent/202
announce-holdtime = yes
Hi,
In last one week I have seen two servers of our organization successfully
hacked and some other under attack from some other IP addresses. We would
block one IP address on our firewall and after a few hours, they would start
getting hits from some another IP address. When I checked them on
On Mon, 23 Mar 2009 10:19:14 -0400, Steve Totaro wrote:
My advice?
If mandated with a USB device and softphone, I would certainly go with
Plantronics.
I agree. I've been using a Plantronics .Audio 615m lately and it's
great! Good long cord, and handles even wideband calls well. Also very
On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote:
Siemens make a range of DECT handsets under the Gigaset model range.
Yes they shit all over every wifi handset I have ever used.
Dect is way better.
Amen to that! Unles you have some compelling reason for VoWifi it's not
worthy of
On Mon, 23 Mar 2009, Michael Graves wrote:
On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote:
Siemens make a range of DECT handsets under the Gigaset model range.
Yes they shit all over every wifi handset I have ever used.
Dect is way better.
Amen to that! Unles you have some
Ah.. so Debian's asterisk build didn't include zaptel.
OK will give that try!
Thanks for the pointer.
Steve
On Sun, Mar 22, 2009 at 12:09:09AM -0400, Steve Gladden wrote:
I've recently installed the latest Debian Linux for powerpc onto
and old iMac (version A) the original iMac with a 233Mhz
I wonder why they only allow G.729 with this ... where's the great sound of
the skype call now ?
Martin
On Mon, Mar 23, 2009 at 2:42 PM, Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:
Anyone connected up to it yet?
http://www.skypeforsip.com/
It would
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