On Sun, Apr 20, 2008 at 07:37:32PM -0700, Chris Brentano wrote:
When I ran ./configure, which completed successfully, I noticed that it
complained about the PKG_CONFIG_PATH and not being able to find libcurl:
(lines omitted)
...
checking for curl-config... /usr/bin/curl-config
Package
As recommened I got the new firmware for my echo cancellers and it
solved hte problem with the agressive echo cancelling causing half
duplex audio. I have to say, so far these cards are far superior to the
previous models. The sound quality is hugely improved (enough to really
notice which is
Generally I'd agree. But it could at least more adequately notify the
user, even if they are compiling on a different system than where it
will be running on. It just seems that in most cases people will be
compiling on the system they will be installing on. This is what they
teach at the
Hi all, i have a basic question on asterisk.The below is my scenerao.
I have my sales offices around the globe.Theyare all connected with Speed
Internet connection.I don't mind installing 1 asterisk box in each site.i don't
mind using IP phone.i just wanted to call them for free at the cost of
Hello ppl,
Is there any other way to detect states like Ringing on SIP channels on
Asterisk?
Nvlinedetect is one way, but it seems to have disappeared from the face of the
earth!
Any pointers or does anyone have the code for NV* features?
Thanks in advance
- Ben.
Hello ppl,
I am using the Astman API Originate command to initiate a call to a user. On
connect of the user, I dial another user to bridge the call between the two.
I am using the Async option with the Originate command, as I don't want to use
Astman proxy yet. Is there any way to invoke a
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe Brevda wrote:
will this do?
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/tb-trixboxgraph-0-0.1.0-2.html
btw, it has (almost) nothing to do with trixbox
I added the following to the voicemail config files:
imapserver=imap.gmail.com
imapport=993
and imapuser=user|imapsecret=pass to the mailbox details. However that just
causes asterisk to hang as soon as i try to use that extensions...
___
-- Bandwidth
On 21/04/2008, Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after
Sure - read up on IAX for a few good points.
PaulH
rupak shrestha [EMAIL PROTECTED] wrote:
Hi all, i have a basic question on asterisk.The below is my scenerao.
I have my sales offices around the globe.Theyare all connected with
Speed Internet connection.I don't mind installing 1
On 18/04/2008, Rosa De Santis [EMAIL PROTECTED] wrote:
Hi all.
Please, how can I configure an Asterisk PBX using an outbound proxy (that
resolve NAT Traversal)
I'm trying using the outboundproxy and outboundproxyport values in sip.conf
but the PBX don't get registered on the outbound
On 20/04/2008, robert boardman [EMAIL PROTECTED] wrote:
Hi All
I'm having problems with outboud ISDN calls,
They setup OK , and ring the other end OK, but when the call is answered
I get a disconnect cuase 17 with an error message in the console of
[Apr 15 08:06:13] DEBUG[4361]
On Mon, Apr 21, 2008 at 5:11 AM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 04:19:16AM -0400, Matthew Gibson wrote:
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen
[EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe Brevda wrote:
will this
On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson
[EMAIL PROTECTED] wrote:
Ex Vito wrote:
Matthew,
...is there any specific test you'd like us to perform on this revision ?
(considering that currently we have no PSTN line to attach to... we
can cross-connect the
Hi,sorry for off topic post, struggling to find any information on UMA in
the UK. I have a Blackberry 8320 phone with wi-fi and UMA
capability, its actually an unlocked Orange branded phone.
T-Mobile don't support UMA in the UK, is it possible to do anything else
with the UMA feature of this
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after dialing the user because
most of the users are behind PBXes
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after dialing the user because most of
the users are behind PBXes
On 21/04/2008, Benjamin Jacob [EMAIL PROTECTED] wrote:
Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after
On Mon, Apr 21, 2008 at 04:19:16AM -0400, Matthew Gibson wrote:
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe Brevda wrote:
will this do?
Dear All
I am using gtalk features with my own XMPP server OpenFire
I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls
from clients registered on my XMPP server to SIP devices by calling the xmpp
accounts registered as clients on asterisk.
So far so good. So if I want
Dear All
I am using gtalk features with my own XMPP server OpenFire
I have setup gtalk.conf and jabber.conf on asterisk and now I can make calls
from clients registered on my XMPP server to SIP devices by calling the xmpp
accounts registered as clients on asterisk.
I have sent a previous email
Hello,
Is here an easy way to get RTCP Stats in channel variables after the call
ends?
Or source should be edited to accomplish this?
I would like to know this before developing this feature.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
On Mon, Apr 21, 2008 at 11:02:13AM +0100, Mike Dent wrote:
sorry for off topic post, struggling to find any information on UMA in
the UK. I have a Blackberry 8320 phone with wi-fi and UMA
capability, its actually an unlocked Orange branded phone.
T-Mobile don't support UMA in the
Hi!
I need to implement click-to-talk web application.(not click-to-call or
callback)
I try to use njiax, and iaxclient but I can´t made it work.
Has anybody other solution??
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
I've been working on a visualization tool that would be totally open
source and I'd share it now if I wasn't embarassed by how quickly and
shoddily I assembled it. Here's an example of the sample extensions.conf
file that was part of 1.4.19:
http://www.mbs3.org/extensions-conf-sample.jpg
On 21 Apr 2008, at 14:31, equis software wrote:
Hi!
I need to implement click-to-talk web application.(not click-to-call
or callback)
I try to use njiax, and iaxclient but I can´t made it work.
Has anybody other solution??
Yep. We can help on a commercial basis. Contact me off-list if
The E1 use ALAW, if you want to avoid trans-coding use ALAW in your
phones as well. In any call you have 2 call legs, callee and caller,
try to isolate the problem and determine if the audio is really coming
that bad from the E1, you can use ztmonitor to hook into the E1 and
listen to the audio.
Hello guys,
I thought it would be neat if we had a SIP client for Asterisk working in Adobe
Flash, but as far as I know, Flash only supports TCP. I know that Asterisk (at
least v1.6) can handle SIP communication over TCP, but I was wondering is there
a possibility to route audio stream over
Thanks, I´m interested in non comercial solutions.
On Mon, Apr 21, 2008 at 11:00 AM, Tim Panton [EMAIL PROTECTED] wrote:
On 21 Apr 2008, at 14:31, equis software wrote:
Hi!
I need to implement click-to-talk web application.(not click-to-call
or callback)
I try to use njiax, and
Ex Vito wrote:
On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson
[EMAIL PROTECTED] wrote:
Ex Vito wrote:
Matthew,
...is there any specific test you'd like us to perform on this revision
?
(considering that currently we have no PSTN line to attach to... we
can
I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record
all incoming calls. One of the box that have Asterisk 1.4.18 is properly
merging calls and the other box that has Asterisk 1.4.15 is recording the calls
but not merging them, I have made sure that SOX is installed on
On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote:
One of the box that have Asterisk 1.4.18 is properly merging calls and
the other box that has Asterisk 1.4.15 is recording the calls but not
merging them, I have made sure that SOX is installed on the box.
It might be worth giving the
On Mon, 2008-04-21 at 10:31 -0300, equis software wrote:
I need to implement click-to-talk web application.(not click-to-call
or callback)
I try to use njiax, and iaxclient but I can´t made it work.
Has anybody other solution??
You can try with jiax:
http://www.hem.za.org/jiaxclient/
Newer version of sox don't seem to have soxmix anymore, but you can
use sox -m and I think asterisk should be changed to use that instead.
on Monday 04/21/2008 Jared Smith([EMAIL PROTECTED]) wrote
On Mon, 2008-04-21 at 21:11 +0530, Sanjay Rajdev wrote:
One of the box that have Asterisk
As long as you renamed it to indications.conf when you copied it then it
should be working after you do a reload of Asterisk. If it's not
working then the problem was not indications.conf
aby azid wrote:
Hi Eric,
i copy the indications.conf.sample from the asterisk source and paste it in
John,
Is their something that I can change on my side to get this working ?
Jared,
I thought MixMonitor() was for Queue, Can you let me know how to use it?
Thanking you for replying.
Regards,
Sanjay Rajdev
- Original Message -
From: John covici [EMAIL PROTECTED]
To: [EMAIL
http://www.voip-info.org/wiki/view/MixMonitor
On Mon, Apr 21, 2008 at 7:43 PM, Sanjay Rajdev
[EMAIL PROTECTED] wrote:
John,
Is their something that I can change on my side to get this working?
Jared,
I thought MixMonitor() was for Queue, Can you let me know how to use it?
Thanking you
I would look at setting up OpenVPN on each of the Asterisk boxen and
running SIP between them. I have read that IAX2 is much better now,
but I have had many major voice quality issues with it.
With OpenVPN, all Asterisk boxen appear to each other as being on the
same subnet. This gives you ease
Hi Ali,
I have sent a previous email with a problem that I solved by using component
mode. In this mode the asterisk server acts as a subdomain. So I can call
[EMAIL PROTECTED], [EMAIL PROTECTED]
That's a nice way of using Asterisk's component capability. Which
XMPP/Jingle client are you
Hello everybody.
Is there a way how to setup asterisk to notify caller's phone?
Example:
I have some numbers and names in asterisk database ( cidname, cidnum),
and I want to display the name of person on my phone ( which has no
addressbook, but can display chars ) which I am calling to be sure
I'm looking for some configuration help. I'm currently running Asterisk 1.4
on Centos 5. I have a server that has two network cards, the first card is a
public ip that does sip trunking to our sip provider. The second network
card is an internal ip that is a seperate voice vlan. The problem that
What is good for recording all the incoming and outgoing calls, Monitor() or
MixMonitor().
Regards,
Sanjay Rajdev
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
AnDY wrote:
Hello everybody.
Is there a way how to setup asterisk to notify caller's phone?
Example:
I have some numbers and names in asterisk database ( cidname, cidnum),
If I understand you correctly, you'll be interested in this bug:
http://bugs.digium.com/view.php?id=8824
Doug
--
I changed the codeof the monitor app to use sox -m instead of soxmix
which I no longer have. Mixmonitor would work as well, but the
one-touch recording was set to the other, so I am using that.
on Monday 04/21/2008 Sanjay Rajdev([EMAIL PROTECTED]) wrote
John,
Is their something that I can
This will be my first major asterisk experiment and I'm trying to
choose a PoE switch for 15-24 phones. I was going to spend $400 on
this:
http://www.newegg.com/product/product.asp?item=N82E16833124053
but then I see this on ebay:
MixMonitor.
And please stop posting the same question to the list over and over.
Sanjay Rajdev wrote:
What is good for recording all the incoming and outgoing calls,
Monitor() or MixMonitor().
Regards,
Sanjay Rajdev
I am probably not too qualified to answer this question but I would go with the
linksys.
Fred
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hilary Miller
Sent: Monday, April 21, 2008 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
external ip for an internal server ? sounds too dangerouse to me.
i would suggest you put the server back to local lan use a router to hold
your external ip do port forwarding to internal servers. it will solve
your dilema keep your server safe.
___
Woah,
How weird. I JUST bought this off of ebay 2 minutes ago. The exact one.
This will be my first time playing with PoE. I have all cisco phones
here but I'll let you know how it goes.
This will be my first major asterisk experiment and I'm trying to
choose a PoE switch for 15-24 phones. I
On Fri, 18 Apr 2008 11:30:46 -0400
Steve Totaro [EMAIL PROTECTED] wrote:
n Fri, 18 Apr 2008 18:40:17 +0300
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, 19 Apr 2008 15:07:47 +0100
Alan Lord [EMAIL PROTECTED] wrote:
thanks to everybody; I'm happy to know no responses were
hostile :-) but I will
Hi folks,
I have some asterisk 1.2 box with self-made billing, and I need to
disable call transfer on all calls and directions.
I turned it off in features.conf and there is no 'tT' option in all my
Dial() commands, but users still able to transfer call using transfer
function in ip of
Hilary Miller wrote:
This will be my first major asterisk experiment and I'm trying to
choose a PoE switch for 15-24 phones. I was going to spend $400 on
this:
http://www.newegg.com/product/product.asp?item=N82E16833124053
but then I see this on ebay:
canreinvite=no should just work, if it doesn't then maybe you want
to post parts of your SIP conf.
Thanks,
Steve Totaro
On 4/21/08, linuxian iandsd [EMAIL PROTECTED] wrote:
external ip for an internal server ? sounds too dangerouse to me.
i would suggest you put the server back to local lan
On Mon, Apr 21, 2008 at 4:38 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Ex Vito wrote:
...when can one expect to have your new code available in a zaptel
release ?
In the next one or maybe later because the branch you're working on
has lots of different things to
We've been very happy with the SRW224Ps we've deployed.
(noisy as hell... good for either the datacentre / computer room or
for installation in a noise-cancelling cabinet... but then again, are
there any PoE switches that aren't ?)
--
exvito
http://sipsak.org/ has enabled people to display many different things
on their phones. I have yet to do this but have seen it mentioned
more than once.
Thanks,
Steve Totaro
On 4/21/08, Doug Lytle [EMAIL PROTECTED] wrote:
AnDY wrote:
Hello everybody.
Is there a way how to setup asterisk
On Mon, Apr 21, 2008 at 5:54 PM, Sean Dennis [EMAIL PROTECTED] wrote:
The Cisco 3524 switch doesn't support 802.3af which is what your Linksys
phones are going to want.
Thank you for sharing Sean! When I saw them I felt a disturbance in
the force, and now I know why!
--
Just Hil
The answer to your question depends on the QOS you desire. If you are
concerned less with call quality, and more with price, then the previous
solutions are certainly adequate. If call quality is important to you, I
would recommend going with a VOIP provider that has a private network, that
way
If you use voicemail.conf to configure the voicemail users, instead of using
users.conf, you can just specify the voicemail box without an e-mail address
and it will not send the e-mail notification of the voicemail to the user. I
am not sure if this was your exact question, or if you wanted to
We've been using D-Link DES-3028P and DES-3052P switches. They can supply full
power to EACH port unlike the Linksys switches we've tried. They're also rock
solid from our experience.
-Jon
- Original Message -
From: Hilary Miller [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
As the previous person said, Asterisk will only accept one phone for each
SIP account.
In order to do what you are trying to do, you will want to create 2 entries
in sip.conf such as [1000] and [1001]
After that, you will need to set it up in extensions.conf you will create an
extension that
Jonathan C. Bailey wrote:
We've been using D-Link DES-3028P and DES-3052P switches. They can
supply full power to EACH port unlike the Linksys switches we've
tried. They're also rock solid from our experience.
I echo that recommendation. The Linksys switches are probably the
loudest that I've
On Monday 21 April 2008 20:44, Kyle Gibbons wrote:
On Thu, Apr 17, 2008 at 1:14 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
Is there a way to specify per user attachment options for voicemail,
from within users.conf?
I know I can enable or disable it globally in voicemail.conf, but I have
Assuming I only operate in one municipality (I do), and assuming I made
some sort of connection to the emergency services center in this area,
via SIP or a T1 or whatever, does asterisk have a way for me to send the
E911 address data?
___
--
Apologies for not explaining the set up .
Using AstMan API, I Originate a call to user A. User A is a conference bridge
which needs pin authentication. So post 200 OK, I need to send DTMFs for that
pin.
After sending the pin, I Dial (using the Originate context) user B. Now user B
is behind
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