That should do it, it tells Asterisk to override the contact field which
includes the private IP, and use the public IP and port it received the
packet from instead.
Try a 'sip debug peer ' and see what it is coming in as.
From: [EMAIL PROTECTED]
[mailto:[EMAIL
elcome notices to mobile users : whenever a
mobile user comes in, a SIP MESSAGE is sent by a software application which
has previously subscribed to be notified of any registration event related
to this mobile user.
It appears Asterisk supports SIP NOTIFY-SUBSCRIBE methods.
But I couldn't fi
hello,
I use trixbox.I had define a feature code testfeature:
[applicationmap]
#include features_applicationmap_additional.conf
testfeature => *3,callee,Macro,vote
[featuremap]
blindxfer => ## ; Blind Transfer
disconnect => **; Disconnect Call
automon => *1
We bought a few IAX ata's a while ago (virbiage) and they worked quite
wellone of those with a standard cordless phone would be an idea...
PaulH
On Wed, 2007-08-22 at 15:27 +0100, Ade Vickers wrote:
> Does such a beastie exist?
>
> I've tried a couple of UT Starcom WiFi SIP phones (the F10
Sorry for not indenting, I am stuck using OWA for the moment.
If your customer seriously wants to pursue that option, please let me know what
they have. Model numbers, used/new, how many, and any other details. I can
probably get them a much better price than Ebay or something.
Thank,
Ste
For this week's conference, the two founders of Voicepulse, Ravi
Sakaria and Ketan Patel, will be joining us. For those of you who
are not aware, Voicepulse is an asterisk friendly VOIP provider that
has won awards for service and innovation.
We will also have Trixbox news, updates, as well as di
I use a centralized database (with replication) for several servers, and it
works very well. I keep all the mysql traffic on a separate network from
the SIP traffic. It makes it easy to add capacity. If you are doing all the
mySQL on one box anyway, I don?t see any adavantage to using separate
da
Is there any way to get the channel of the first agent called in a
queue? Say I have a queue with 5 agents setup in roundrobin. I want
the voicemail to go to the first person that was called. Say a call
comes in and rings 1,2,3, then I want it to go to vm for 1. Say the
next call rings 4,5,
On Wed, Aug 22, 2007 at 01:19:14PM -0500, Asterisk Development Team wrote:
> The Asterisk.org development team has announced the release of Zaptel
> versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in
> the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel
> rel
Matthew Rubenstein wrote:
> Imagine if the world's largest online marketplace operated the world's
> largest alternative (and one of the largest in general) telco and an
> unregulated global online banking monopoly. And the telco suddenly went
> down, unexplained, for hours or days.
>
>
Dear all, I have an Asterisk server with SIP and IAX softphones clients, and I
need to encrypt the voip calls among them:
*For SIP clients I use Twinkle which implements the ZRTP/SRTP encryption
mechanism client-2-client; I read it's the better security mechanism nowadays
created by Phill Zimme
Zane C.B. wrote:
> 1: Software RAID on Linux is way less than impressive. Plus last a I
> checked Linux can't handle mirroring a entire disk. Last I looked at
> it around a year ago you were limited to only mirroring partitions,
> which is a joke from a administrative standpoint.
How is this any d
Gordon Henderson wrote:
> You do (sometimes) need the hardware RAID controller to be supported by
> Linux and this is a weak area. Some controllers just look like a standard
> drive, so they are transparent to the system, but then you need to use
> either the BIOS utilities to set it up in the f
Jon Pounder wrote:
> Quoting Steve Prior <[EMAIL PROTECTED]>:
>
>
> personally my favourite still is phone in intercom mode listening at
> all times, if you have something to say, say it.
>
> otherwise pickup and dial for control or to talk or whatever.
>
> nothing preventing you from ignorin
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in
the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel
releases, as well as a handful of other issues. See the respective
Changelogs for mor
Jason Parker a écrit :
> Administrator TOOTAI wrote:
>
>> Jason Parker a écrit :
>>
>>> Administrator TOOTAI wrote:
>>>
>>>
Hi all,
I receive this error while compiling chan_mobile:
gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
chan_mobile.c:
Seems to me that as long as all the contacts / reachability info / URIs
are distinct for each user, there is not a problem with using one big
database, and that it certainly presents less of a maintenance headache.
It also provides easier migration path to future options you may want to
explore
I am in the process of setting up several * servers using realtime and
connecting to mysql. I am trying to figure out if I should just use one
database and one set of tables for all of the user data. Or if I should
have separate databases for each * box. The boxes are independent of
each oth
Jason Parker wrote:
> > Administrator TOOTAI wrote:
>> >> Hi all,
>> >>
>> >> I receive this error while compiling chan_mobile:
>> >>
>> >> gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
>> >> chan_mobile.c: In function `mbl_load_config':
>> >> chan_mobile.c:1745: erreur: trop d'arguments p
Administrator TOOTAI wrote:
> Jason Parker a écrit :
>> Administrator TOOTAI wrote:
>>
>>> Hi all,
>>>
>>> I receive this error while compiling chan_mobile:
>>>
>>> gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
>>> chan_mobile.c: In function `mbl_load_config':
>>> chan_mobile.c:1745: er
Quoting Steve Prior <[EMAIL PROTECTED]>:
personally my favourite still is phone in intercom mode listening at
all times, if you have something to say, say it.
otherwise pickup and dial for control or to talk or whatever.
nothing preventing you from ignoring one of the options if you don't
l
I've gotten burned by software raid so I'll probably be sticking with
hardware in the future. If your drive dies for some reason it could affect
the SATA bus and cause the system to crash. That's what happened to me. It
wouldn't come back up on the second Raid 1 drive until I removed the bad
one
Jason Parker a écrit :
> Administrator TOOTAI wrote:
>
>> Hi all,
>>
>> I receive this error while compiling chan_mobile:
>>
>> gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
>> chan_mobile.c: In function `mbl_load_config':
>> chan_mobile.c:1745: erreur: trop d'arguments pour la fonction
Steve Edwards wrote:
> Personally, I hate voice recognition systems. Voice prompts are
> great, but don't take away my keypad.
I never proposed to take away your keypad, I just wanted to add the
voice option as well. What I do want to get rid of is the step
below where you press ** to get into
Thanks Diego :)
It works now. For the rest of the people, the commands needed were
/var/lib/asterisk/bin/retrieve_conf and then asterisk -rx reload
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To
Administrator TOOTAI wrote:
> Hi all,
>
> I receive this error while compiling chan_mobile:
>
> gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
> chan_mobile.c: In function `mbl_load_config':
> chan_mobile.c:1745: erreur: trop d'arguments pour la fonction «
> ast_config_load »
> make[1]:
Hi all,
I receive this error while compiling chan_mobile:
gcc -g -c -fPIC -fPIC -o chan_mobile.o chan_mobile.c
chan_mobile.c: In function `mbl_load_config':
chan_mobile.c:1745: erreur: trop d'arguments pour la fonction «
ast_config_load »
make[1]: *** [chan_mobile.o] Erreur 1
make[1]: Leaving
Hi Dovid, I had the same problem with the same configuration 3 lines
worked fine but the fourth couldn't detect disconnect supervision (it was
a Foriegn Exchange Line).
There are two things you should try before anything else.
1. Reverse the Tip and Ring connections on the CO Lines and test again.
Brandon Kruse wrote:
> /me goes to work.
>
> There are none that I know of. There are only a couple of
> IAX(2) hard phones, and none of them, that I know of, are
> manufactured in the US anyways, and have problems.
> (Of course, what is manufactured in the US these days)
>
> That would be
Answered my own question - use buddy watch on the Polycom and create a
hint priority extension for the agent channel...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Wednesday, August 22, 2007 8:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi List,
I have a client who has a TDM400P with 4 FXO. He has a problem them when some
one calls, then hangs up it takes a good 10-15 seconds or more of the card to
realize that the line was hung up on. The phones keep reigning After a bit it
hangs up on the line. Also there has been some hangi
On Tue, 21 Aug 2007, Steve Prior wrote:
> Steve Edwards wrote:
>
>> "To control the tv in this room, press 1. To control a tv in another room,
>> press 2. To control the outside lights, press 3. To control the
>> sprinklers, press 4, ..."
>
> A while back I was thinking along the lines of using a
Hello All,
Stable release of A2Billing has solved most of my problems and so far
everything is OK...
Right now the only problem I am facing with my SIP clients are: -
- Three-way Calling
Three-way calling works fine, but when SIP client hangs up the
call, the other two channels are s
Has anyone designed a method to allow callback agents (Asterisk 1.2) to
log in on a Polycom SoundPoint IP phone and have the phone visually
indicate the agents logged in status?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
ast
/me goes to work.
There are none that I know of. There are only a couple of IAX(2) hard phones,
and
none of them, that I know of, are manufactured in the US anyways, and have
problems.
(Of course, what is manufactured in the US these days)
That would be a great device, would love to see it
Does such a beastie exist?
I've tried a couple of UT Starcom WiFi SIP phones (the F1000g and F3000
respectively), and found them both to be seriously lacking - regular crashes
(especially the F3000), poor battery life, and poor reception in particular.
However, whilst SIP phones are great, I'd re
On Wed, 2007-08-22 at 08:50 -0500,
[EMAIL PROTECTED] wrote:
> Date: Tue, 21 Aug 2007 21:01:50 -0400
> From: "David Cook" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] 99 bottles of beer
> To:
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="us-ascii"
>
> On 8/21/07, St
Have you tried installing this board in another PC to test your FXOs ?
What motherboard are you using ?
Are your FXOs boards original digium or they are chineses versions ?
Luis A P Barbosa
2007/8/15, [EMAIL PROTECTED] <[EMAIL PROTECTED]>:
>
> Hello,
>
> I have a TDM400P with 4 FXO ports, curren
On Wed, 22 Aug 2007, Steven wrote:
> For RAID1, I am not sure.
>
> But for RAID 5, You should always use hardware RAID.
>
> If you use software RAID and your CPU spikes for too long, you can
> corrupt your disks. I have seen this several times.
Please report this to the linux-raid mailling list,
For RAID1, I am not sure.
But for RAID 5, You should always use hardware RAID.
If you use software RAID and your CPU spikes for too long, you can corrupt your
disks. I have seen this several times.
--
--
Steven
http://www.glimasoutheast.org
"Vidura Senadeera" <[EMAIL PROTECTED]> wrote
I've been working on an X10 component already. It works, but I wish the
CMA15 would work correctly in Linux (I know it's suppose to, but for
whatever reason the one I have just doesnt.) It's just a little AGI
script that I have working with Cepstral that throws http PUTs to the
Windows box that
Henry L.Coleman wrote:
> I think what Alex was trying to say was that Polycom IP Phones are an
> example of immature product development. While they look very nice and
> have a nice display the product doesn't compete very well compared to
> other manufacturers.
> The two most obvious flaws are tha
On 8/22/07, fateme fatah <[EMAIL PROTECTED]> wrote:
> Hi:
> Which one is better and easier for configure asterisk,directly or by GUI ?
> I'd appreciate any idea.
> Regards.
It's up to you to decide what's easier for you and your needs. For
beginners GUI is ok, but if you need some fancy functio
I have both of those command lines for my natted sip device.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl
Dunkin
Sent: Wednesday, 22 August 2007 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom and NAT
In your
Dear all
I have setup of asterisk 1.2.14 and this is working fine.
first i want to explain you my setup of asterisk on network i have connect my
asterisk with mediant 2000 gateway and PRI terminated on mediant.
[fax_machin]--[audio_code_fxs]-[Asterisk]---[median
Thanks for replying, Raj.
Do you think such feature should, ideally, be implemented in Asterisk should
it be implemented in a dedicated software (presence ?) ?
It seems to me it should, though I'm not aware of many devices using this
feature, beside SIP hardphones.
Would it be difficult to extend
FYI about cisco firmware:
http://www.cisco.com/warp/public/707/cisco-sr-20070821-sip.shtml
A.
On Wed, 2007-08-22 at 12:26 +0100, Adrian Marsh wrote:
> Hi All,
>
> A question for those with Cisco 7940/60 SIP phones. I used to load
> POS3-06-03-00 Firmware to the cisco phones. A month or so ag
Well done! It's top-news on AstPligg right now.
http://oinko.net/astpligg/story.php?title=DUNDi_So_Easy_A_Caveman_Could_Do_It
Thanks
l.
On Wed, 22 Aug 2007 03:51:51 +0200, JR Richardson
<[EMAIL PROTECTED]> wrote:
> Here you go folks:
>
> ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf
>
> If someone
On 8/21/07, Matthew Fredrickson <[EMAIL PROTECTED]> wrote:
> Matt Florell wrote:
> > Hello,
> >
> > A client has asked for Two B channel Transfer capability (known as
> > TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG
> > Path Replacement) in a new Asterisk system and so I resea
Hi All,
A question for those with Cisco 7940/60 SIP phones. I used to load
POS3-06-03-00 Firmware to the cisco phones. A month or so ago, I ran
some tests and found that latest 3.8.6 firmware worked well, and solved
an issue or two on the phones.
I've a number of users who work outside of the L
Olivier,
This feature is not supported in Asterisk. I can tell this looking at the code.
If you want to test this yourself, send Asterisk a SUBSCRIBE message
with Event: reg header in it. You can either use an off-the-shelf UA
that supports RFC 3680 to do this or you can use SIPp (an open-source
Hi:
If any body use meetmemanager or conman or web-meetme please say how about is
it.I'd appreciated any idea.
Regards.
-
Be a better Heartthrob. Get better relationship answers from someone who knows.
Yahoo! Answers - Check it out. _
Hi:
Which one is better and easier for configure asterisk,directly or by GUI ?
I'd appreciate any idea.
Regards.
-
Building a website is a piece of cake.
Yahoo! Small Business gives you all the tools to get online.__
Steve Totaro wrote:
> I guess I am just lucky to have 24 hour manned data centers with staff
> that walk around looking for flashing LEDs.
>
> I am sure there is some error thrown in /var/log/messages about a
> failure that could be used to trigger a notification quite trivially.
>
Both smar
You are updating the MySQL config, which is not propagated to the Asterisk
config files. Only after you regenerate the configuratios, you can reload
asterisk.
Dirty hack: "need_reload" flag must be set to true.
Real solution: retrieve_conf + "asterisk reload"
On Wednesday 22 August 2007 10:22,
Hello guys,
I'm using Trixbox and I have a PHP application that updates a value in
the MySQL asterisk database as an interface to have a dynamic
customizable IVR.
After execute the UPDATE SQL query, the php application is supossed to
reload asterisk or restart amportal in order to get the change
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