Anyone got any clues or patches on how to make these work with the
latest svn trunk. The only versions of app_txfax.c don't compile
Julian.
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We have used both IP501, IP601 (Polycom), Snom 320 and Snom 360. All of
them are good phones with very good quality of voice and full of features.
However, SNOM phones have a feature (missing from Polycom) that most of
our customers really require: with SNOM phones you have leds for
presence
Hi
What is the best solution for H323 in asterisk
-- h323 in source,
-- oh323 or
-- ooh323c?
which is most robust and reliable? Which supports gatekeeper functionality?
Best wishes
Andrutto
--
Najnowsze fakty!!!
On 26 Aug 2006, at 07:57, Martin Joseph wrote:
Now, the fact you can't easily get these phone in the US, that's a
conspiracy ;~)
... and if you take them to the US you realize you should have gotten
a quad-band phone because your E60 can't deal with the common US
frequency of 850 MHz,
Hi,
i used to work ooh323 with my asterisk. it gives better performance
than other oh323 or H323 comes with asterisk...
i got H323 channel and oh323 with a lot of error.( like codec
selection )but ooh323 works fine with me
thanks
atik
On 26 Aug 2006 12:13:52 +0200, andrutto
Our MOH died, so I finally had to kill my * process and restart it.
Interestingly, stop now didn't work. I had to kill the process. It
used to work, but it had been up so long that it must have gotten
corrupted somehow. Here is the show uptime before I killed it:
Asterisk-A*CLI show uptime
On Sat, Aug 26, 2006 at 01:35:36AM +0800, Sam Tam wrote:
WE can provide you with budget GSM Gateway if you are interested?
which is commercial nope? wrong list again? could have been private
Email?
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715
Hello all,
I am trying to test if the length of a dialed number is greater than 7. When
i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial)
Thanks for your reply Steven.
I appears to me that that the extens in Intercom Group are patterns
requiring an initial underscore
but the extens in 2) One to Many Paging and 3) One to Many Intercom
are named extensions and should not have an initial underscore as
(mistakenly) shown.
That
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John Millican wrote:
Hello all,
I am trying to test if the length of a dialed number is greater than 7. When
i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the
If Asterisk was used to set up and tear down calls, and
using canreinvite allowing the RTP to pass from end-point to end-point, how
many calls could Asterisk handle at once?
I ask because I have been utilizing OpenSER but find myself constantly
needing Asterisk to do this or that, and
Hi list!
Im using Tycho software to see my voicemail, y
can see de detail from the message but i cant hear de message.
Somebody use that software any time ? have you the
same problem ?
Thanks
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Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer
On 8/26/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Who says * isn't stable enough for prime time? At least it is on 1.0.3.
What kind of abuse does that box take?
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Use the m option at the end of the dial string. Google told me so.On 8/26/06, Crazy Boy [EMAIL PROTECTED]
wrote: Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring
Is it possible to get sip to listen on two ports (say 5060 and 5061)?
Maybe its not necessary, but I'm trying to get a PAP2 to work with 2
lines configured behind a Linksys router with NAT.
I've noticed the default config in the PAP2 is to use 5060 for line 1
and 5061 for line 2.
I'm guessing
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeetingOn 8/26/06, atik khan
[EMAIL PROTECTED] wrote:Hi,i used to work ooh323 with my asterisk. it gives better performance
than otheroh323 or H323 comes with asterisk...i got H323 channel and
Please disregard this message.
Evidently changing the port required a power cycle on the PAP2.
On 8/26/06, Mr. Jones [EMAIL PROTECTED] wrote:
Is it possible to get sip to listen on two ports (say 5060 and 5061)?
Maybe its not necessary, but I'm trying to get a PAP2 to work with 2
lines
There aren't a lot of phones. There are 50-60 SIP phones and SIP
connections to two Cisco PRI gateways. About 10,000 calls / month and
about 15,000 mins of LD/month. I know when I started with *, I head how
it had to be restarted every week and ours just ran and ran.
Justin Tunney wrote:
At 07:24 AM 8/26/2006, you wrote:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial)
in new stack
I'm guessing that you need to remove the
Here is a detailed install guide for FreePBX but helps
even if your not using FreePBX.
http://powerontech.com/freepbx-on-debian.htm
From: Christopher Aloi
[mailto:[EMAIL PROTECTED] Sent: Friday, August 25, 2006 7:09
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re:
I gotta put in a plug for my favorite phone the Aastra 9133i which also has
BLF for each programmable button. Best all around reasonably priced
business grade phone IMHO.
-Original Message-
From: Mario [mailto:[EMAIL PROTECTED]
Sent: Saturday, August 26, 2006 2:09 AM
To: Asterisk Users
To anyone having problems installing SoundTouch or libsoundtouch4c,
I've improved the build system for libsoundtouch4c and updated the
install instructions. Please let me know if you continue to have
problems.
http://www.lobstertech.com/code/voicechanger/
- Justin
I've been struggling with this issue for over a year.
I wish there were some kind of automatic gain control built in to set
the rx/tx gain on the fly based on the volume of the two channels.
Probably not realistic though.
Is there other hardware other than digium's that better deals with this
That worked great!.
I was using Ring_WaveForm and I guess it's case sensitive and the
correct form should be Ring_Waveform.
Thanks,
Daniel
On Aug 25, 2006, at 11:48 PM, Shanon Swafford wrote:
This works for me on my SPA-3000 ver 3.1.10(GWd).
Ring_WaveformTrapezoid/Ring_Waveform
Then
I dug into my config (I do use paging over the phone but remember playing with
it) and came up with this reference.
-
exten = 5488,1,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
[ext-pager]
include = ext-paging-custom
exten =
Along the same lines as this question... Are there any Voip phones that
have dual gigabit ethernet ports?
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Sergio R. D'Ippolito wrote:
I’m using Tycho software to see my voicemail, y can see de detail from
the message but i cant hear de message.
Please send me or post here:
- Client Version / os platform
- Server Operating System
- HTTP Server + php version
- which version of the scripts you
On 2006-08-25 10:35:36 -0700, Sam Tam [EMAIL PROTECTED] said:
Hello
WE can provide you with budget GSM Gateway if you are interested?
Sam
Hey Scumbag,
How many timed do you need to be told that this isn't the place to sell
your wares?
Please Stop it!
On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5
over Grandstream HT488 ATA.
snip
Personally I found the FXO port on the HT-488 to unworkable except as a
backup for power outages.
I
If one would visit with knowledgeable transmission engineers that work
full time in the telephone industry, one would find telephony standards
that govern exact transmission levels at each point throughout a
country's telephone network (including the long distance facilities, pbx
trunk loss,
Martin Joseph wrote:
On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said:
Hi list!
I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk
1.2.5 over Grandstream HT488 ATA.
snip
Personally I found the FXO port on the HT-488 to unworkable except as a
backup for
Does ANYONE have any clues?
This is annoyng me no end :-(
Thanks
- Original Message -
From:
Paul A Brown
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, August 25, 2006 5:22
PM
Subject: [asterisk-users] 7970 'LoadID
incorrect'
Does ANYONE have any clues?
Only played with 7940 and 7960, but I will try to help since nobody
comes forward
loadInformationSIP70.8-0-3S/loadInformation
Shouldn't that be something like P0S3-08-2-00 ?
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- Original Message -
From: Time Bandit [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, August 26, 2006 10:37 PM
Subject: Re: [asterisk-users] 7970 'LoadID incorrect' problem
Does ANYONE have any clues?
Asterisk-Java 0.3-m1, a Java library for Asterisk PBX integration,
has been released.
The Asterisk-Java package consists of a set of Java classes that allow
you to easily build Java applications that interact with an Asterisk
PBX Server. Asterisk-Java supports both interfaces that Asterisk
I use a PAP2 device to register both lines to my Asterisk server ... Both
the server and the PAP2 are inside my firewall and are on the same IP subnet
... In Asterisk's SIP.CONF I have the second line setup expecting SIP to
use 5061 ... One rings a wireless phone while the other rings a CO line
Hi,Is there a way to determine the MeetMeAdmin User number?I am using the MeetMeAdmin function from within the dialplan.I
would like one of my admins to be able to drop out of the conference
and be able to kick the last user that joined the conference.
I believe that I can do this
I am fighitng with this problem since last
week.
We use sipura 2100 ATA configured with rtp lenght
about 20ms.
Asterisk is connected to our upstrim using pri
(Sangoma aft104d)
During the call the pstn side hear a lot of ticks,
I changed all kind of jitter buffer into the ata and patched
On Sat, 2006-08-26 at 12:53 -0700, Martin Joseph wrote:
Along the same lines as this question... Are there any Voip phones that
have dual gigabit ethernet ports?
Was wondering about that myself.
I've wired every room with gb (for speeding up nfs) and i hate to loose
that speed, because an
On Fri, 2006-08-25 at 14:50 -0400, [EMAIL PROTECTED] wrote:
I'm faced with the need to create forensic test data for an Exchange 2007
server with unified messaging. Microsoft has a list of tested PBX and IP
gateway products that are known to work (below) but I'd prefer to use
Asterisk if
On 2006-08-24 06:32:27 -0700, Jon Schøpzinsky [EMAIL PROTECTED] said:
I also have this phone, and have stumbled in to the same problem.
I just think that it isn't in nokia's interest to change this, as it
forces consumers to have some sort of local hardware, that (possibly)
only the
You have me there. The onlu thing I can think of is to offer some one money
to help build it for you. I had an issue with the p option in the dial
command and paid some one on the developers list to patch it for me and then
had him add it to SVN. Asterisk as a whole I think is good. It needs
You first step is going to be figuring out its an xfered call which
you can do by checking the ${BLINDTRANSFER} variable.
Then you just add the m option in the dial command.
http://www.voip-info.org/wiki/view/BLINDTRANSFER
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+dial
On
Sorry to badger everyone on the list but I never heard from even a
single person on this so felt maybe I'll repeat it, just in case, it
got unnoticed.
Any ideas if it's possible to either record greetings/names in a
different format than GSM OR be able to convert these voicemail
subscriber
Hi All,Could anyone give me idea, How i can set Call Max Time, so in pariticular time the call should disconnect automatically.I will be appriciate for your helps.Abdul
Get your email and more, right on the new Yahoo.com
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Set(TIMEOUT(absolute)=seconds)
Change seconds to the number of seconds you want to allow a
call to last
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
AbdulSent: Sunday, August 27, 2006 1:21 AMTo:
Asterisk-Users@lists.digium.comSubject: [asterisk-users] Call
Hi,
hint is used to monitor the status channels by using extensions in
the dialplan.
When an IP phone holds a call, there aren't any extensions sent to Asterisk.
Does anyone know how I could monitor Hold ?
For example, when an IP phone holds an existing call, the button on the
phone that
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