Make sure this is in your xml config:
ip.of.ntp.server
Unicast
On Thu, Jul 2, 2009 at 10:27 AM, mahboob zaman wrote:
>
>
> -- Forwarded message --
> From: mahboob zaman
> Date: Tue, Jun 30, 2009 at
I think this is what you want: http://bugs.digium.com/view.php?id=8824
On Fri, Oct 3, 2008 at 4:21 AM, Olivier <[EMAIL PROTECTED]> wrote:
> Hi,
>
> When dialing a number, I use :
> exten => _123X, 1, Dial (SIP/${EXTEN})
>
> Then, I get TRYING and RINGING SIP messages which both include this kind
Does anyone have any suggestions on what to use to monitor a vendor doing
remote support?
On the windows side things are typically done via screen sharing (
gotoassist.com, bomgar or similar) so at least you can see what the other
end is doing.
In working with linux (especially hardware vendors f
Hello,
I have a PRI coming into a Digium TE122B with hardware echo cancel,
but we are still experiencing echo on the first 10 seconds of a call.
Is there anything that can be done about this?
I have tried contacting digium support, but have not heard back from
them (placed a support incident about
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Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm not
getting any CRC, framing or other errors on the T1.
Using asterisk 1.4.19 and zaptel 1.4.10
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Does anyone have a solution for remotely getting the newer cisco phones
(7941, 7961, 7970, etc ) to reread their configs (or even rebooting). I am
running SIP firmware connected to asterisk.
Check-sync doesn't seem to work anymore, I can't login to the phones as root
because I am given a "challen
RIs but in some locations a PRI is not
> > affordable and these provide the same DID functionality for a small
> > fraction of the price.
> >
> >
> >
> > Darren Wiebe
> >
> > [EMAIL PROTECTED]
> >
> >
> >
> >
> >
> > Wed Feb 1
We are interested in getting something working also, let me know how it
goes. We are currently using LCS 2005 for IM, the only thing we want to add
is the ability to update the "On the Phone" status in communicator. I have
a test system on 1.6, but so far have been unable to update the presence
i
Looks like it is part of the 1.6 Beta.
>From the Change Log:
2008-01-18 22:04 + [r99080-99085] Russell Bryant <[EMAIL PROTECTED]>
* CREDITS, include/asterisk/http.h, main/tcptls.c (added),
main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
main/Makefile, main/http.
Does anyone have any suggestions for connecting analog DID trunks? I have
some small locations that will have 2 analog DID trunks each, the only
solution that I can see will work will be using a channel bank and T1 card,
but it will be close to $1500 to terminate these DID trunks. Was hoping
some
I will try to answer it this way:
G.711 is toll quality voice, if everything is functioning properly should be
almost identical to a regular phone call.
You will need to do trouble shooting to (in the words drilled into me by an
old boss): isolate, identify and quantify the issue. I would start
Anybody have this file or find any documentation on it? Thanks.
On 5/10/07, Joe Pukepail <[EMAIL PROTECTED]> wrote:
I found on the web that there is way to customize the softkeys for the
7941/7961 phones. In the SEP.xml there is a section called
"softKeyFile" where you ca
I found on the web that there is way to customize the softkeys for the
7941/7961 phones. In the SEP.xml there is a section called
"softKeyFile" where you can specify an xml file for the softkeys. I
couldn't find any examples of this softkey file or the format to this file.
Does anyone have a c
There was a patch to get this working, looks like it has been abandoned, though. Should give you a starting point to get it working, or perhaps a bounty would get someone interested in getting it usable and committed.
http://bugs.digium.com/view.php?id=4845
On 10/4/06, Joel Hill <[EMAIL PROTE
I believe asterisk for the most part is single threaded, you will get some advantages by having other system processes use the extra Processor/Core, but I don't think asterisk will use alot of the other CPU.
On 9/25/06, Tomislav Parčina <[EMAIL PROTECTED]> wrote:
> Asterisk is very happy on dual co
I seen something in the bug tracker and svn about SMDI. Not sure if it was included it 1.4 though. Would be interested if anyone knows if this will work with nortel system (option 11 in particular).
On 9/22/06, Bruce Reeves <[EMAIL PROTECTED]> wrote:
There are a couple more that I have run acros
Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive.
On 7/27/06, Brian Vincent (C) <[EMAIL PROTECTED]> wrote:
Two questions:
We need to run Ethernet ou
I don't know about what our LEC is calling a "digital trunk" but verizon tried to offer me something like this for a location that they couldn't offer a PRI, basically it was a just a voice T1 (24 channels), didn't have features like Caller ID, setting outbound caller ID, ANI, etc. YMMV.
On 7/25
Why don't you detail what you are trying to accomplish on the list, perhaps someone will do it for free. If it is a legitimate bug you could add an incident to the bug tracker.
On 7/24/06, Bart Fisher <[EMAIL PROTECTED]> wrote:
I need someone to "patch" what I believe to be a simple change toch
Or you could use web-meetme, it has this feature.
On 7/6/06, Alexander Lopez <[EMAIL PROTECTED]> wrote:
I would like to walk you through it but I have much on my plate right now that requires my attention.
I will point you in the right direction.
Look at the menu options in MeetMe, the ex
This is known issue, we fixed it by putting an answer() in the dial plan before it gets forwarded, the fix transcode_via_sln=no (detailed in the bug tracker) didn't work for me. YMMV.
http://bugs.digium.com/view.php?id=4101
On 6/28/06, Kevin Savoy <[EMAIL PROTECTED]> wrote:
Sorry if this
I am in the same situation, I have heard the hw echo can is much better, easier to configure, etc. But it seems like an overkill to use a quad span card when we will only be using 1. Anyone know if digium or sangoma will release a dual span card with hw echo can?
On 6/14/06, Cory Andrews <[EMAIL
On 4/27/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
Joe Pukepail wrote:> I have a question, we have some locations were I'm just planning on> putting in a PRI, management also wants analog lines incase the PRI is
> down and someone calls 911. Is there a way to use asterisk to
I have a question, we have some locations were I'm just planning on putting in a PRI, management also wants analog lines incase the PRI is down and someone calls 911. Is there a way to use asterisk to seize a phone line from the fax machine?
I don't want to have to have an analog line that o
Best way is to have a PRI interface to your PBX, I don't have any experience with NEC, but with our nortel system this is what we did. You program your PBX to send extension 123 out the PRI, asterisk sees the call and routes it accordingly.
On 3/15/06, John Padovano <[EMAIL PROTECTED]> wrote:
Fo
As far as I know, you will need to do this yourself with some creative scripting. There was some talk on the list awhile ago to move the recording to voicemail, but I dont' know if anyone has made a patch to do it yet.
On 3/7/06, Tomislav Parčina <[EMAIL PROTECTED]> wrote:
How can I send recor
I had the same problem, I just set the voicemail button on the phone to dial the voicemail extension, but you will still have the problem (at least on the 360) if the user uses the "Soft buttons" below the display to access the voicemail.
On 3/3/06, Nabeel Jafferali <[EMAIL PROTECTED]> wrote:
> I
I like the specs on this, the only thing that it seems to be missing is POE. Anyone know if POE is going to be supported on the 300? Looks nice and I could see it for low use areas, but would suck for wall mounting if it can't do POE.
On 2/22/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
Clint -
Ok, I got through that error, after recompiling with app_cbmysql asterisk doesn't want to start up. I renamed the app_cbmysql.so file and it came up ok.. Anyone have any advise?
[app_cbmysql.so]Feb 16 13:08:17 WARNING[21558]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_cbmysql.
I'm getting the error on the bottom of pages, I'm running this in tandem with 1.4, so not sure if this is an issue, but 1.4 still works (using the same user, password and database as version 2).
Warning: mysql_pconnect(): Access denied for user: '[EMAIL PROTECTED]' (Using password: YES) in /var/w
What you are doing is changing the priority of packets that you are sending to the internet, you'll have to throttle the bandwidth for incoming packets (or better yet, have sprint do it on their router). What you are doing will help if you are getting bad calls when someine is uploading something
user number not screened (0) '3251' ]
Your user number being sent is just the caller ID of the SIP channel.
Regards,
- Brad
-Original Message-From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Joe Pukepail
Sent: Tuesday, February 07, 2006 3:26 PMTo: Asteri
't
> see a valid hangupcause, it might be best to get your carrier on the> line and have them monitor the circuit while you dial 911. They might> be able to tell you what the problem is.>>
>> -MC>>>> >> *From:* [EMAIL PROTECTE
Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk?
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Is there a way to network the asterisk voicemail system between offices? We would like the ability to forward a voicemail to another user at a branch office (each office would have their own asterisk server connected via iax), I guess I would prefer not to use one central server for voicemail for
I see the announcement for the snom 300 on the website, any idea of the street price for that phone?
On 2/1/06, Christian Stredicke <[EMAIL PROTECTED]> wrote:
Hey we have made a new version of our soft phone which fixes animportant bug in the SRTP SSRC part... It is compatible with our latest
versi
Do they have an 800 number? If so perhaps their 800 number provider is doing it via DTMF. Search around on the internet, I believe the standard format for the DTMF is *CALLERID*CALLEDNUMBER* (or perhaps reversed).
On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I have been discussing
I'm having problems with the Message waiting indicator on my Snom 360 that I'm using for testing. I got the button and message waiting indicator working, the problem is : when I hit the voicemail button (or use the menu on the display to access voicemail) it seems to clear the message waiting ind
Perhaps I'm an idiot, but I looked through the readme and changelog but can't figure out what asterisk-netsec is all about? Anybody figure it out?
On 1/18/06, Mr. James W. Laferriere <[EMAIL PROTECTED]> wrote:
Hello Announce & All ,On Wed, 18 Jan 2006, Asterisk Development Team wrote:> Gre
Anyone know if a Sip phone with bluetooth for a wireless headset exists? If so does anyone have any recommendations? Or maybe a Wifi/Sip headset?
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My snom phone shows how many messages, it is a snom 360, there is the MWI light and on the lcd display it will say "5 New and 5 old messages".
On 1/4/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
If the voicemail is stored locally on the server that the phone isregistering to, the phone should autom
I agree, I liked the old ringtone 2 also (just a beep), I use it at my desk, If I'm there I can pick it up and it wasn't obnoxious enough to disturb others. Please email it to me if you get it in the format needed.
On 1/2/06, Remco Barende <[EMAIL PROTECTED]> wrote:
Hi Usman,Thanks for the expla
I am using T1 Signaling and seeing the same problems (I think), so I don't think its just E1.
On 12/29/05, Javier Ergas <[EMAIL PROTECTED]> wrote:
I have tried both inband and outofband too unsuccessfully. I think the priindication parameter says how Asterisk reports Busy and Congestion to the P
I heard on the radio about 1-800-FREE411 and tried it out, I was very suprised to hear allisons' voice for the digits. Not sure if they are using asterisk for the backend on this or not.
Try it out its Free!
http://www.snopes.com/inboxer/nothing/free411.asp
(not afflicated with it in any way)
I have tried both inband and outofband, doesn't seem to make a difference. I added the congension and playtones(congestion) to the dial plan after the dial, but the users just get a busy instead of "Do-De-Dah The number of have reached is not in service ". PRI Debug below.
-- Executing
I am having a similar problem, I thought it was because the PRI card is in another server that I connect to via IAX from my server, but we are seeing the same problem, ie getting a hangup instead of unavailable when calling a number that is not in service. I'm using T1 and Asterisk
1.21
On 12/28/
Is there a way to have control go back to the dialplan after a call gets to voicemail?
I'm looking to implement findme and campon, but I want the options to be "hidden", so if someone calling got a voicemail they could key in "*1" (or whatever) and it would go back to the dialplan so I can implem
You can't use a ethernet crossover cable, make sure you are using a T1 crossover cable. (you will definately need to use a T1 crossover cable).
I'm running a Nortel Option 11 and Asterisk connected in this manner.
On 12/8/05, Steve Totaro <[EMAIL PROTECTED]> wrote:
He said that he is using a
Look into the findme feature, this will require the person receiving the call to push a button "hit 1 to accept this call" before a call gets transfered to a cell phone (or home phone for that matter), if nobody hits "1" it continues in the dialplan, this will prevent calls from being transfered to
I tried to use DISA 1.2 with regular asterisk (not [EMAIL PROTECTED]), and had problems with it (losing the last digit or occasionally other digits), YMMV.
On 12/4/05, Richard Smith <[EMAIL PROTECTED]> wrote:
Hi all,
I was wondering whether the DISA function on the latest asterisk 1.2 stable re
I have been wondering about echo canceling, it seems to be one of the major problems people have with asterisk. I've gotten it to acceptable levels (using mark2 aggressive), but everything I've read indicates that the echo canceling software isn't very effective.
My question would be, what do
Look into the findme feature, there is a patch on the bug tracker to add this feature. I believe that someone shows how to do it in the dial plan. I plan on implementing this, but haven't gotten around to it yet.
On 11/30/05, Benjamin Lenard <[EMAIL PROTECTED]> wrote:
Hi,I'm trying to have an
I haven't heard of this product before so I did some searches on the Internet, this card is $5,400 for a single span T1 card? ouch!
http://www.eiconworks.com/DivaServerV-PRI_T1%20.asp
On 11/25/05, David Waugh <[EMAIL PROTECTED]> wrote:
Hi John,I'm going to have to disagree with some previous posts
For an example here is what I setup to call out, we have a job that runs on our mainframe, when the job completes it ftps flag1.txt to our asterisk server, the .bash program is run from the crontab at a certain time and notifiy staff if the job is not complete at that time. It will keep calling (u
Here is what we use for our helpdesk, on saturday morning we have other people fill in on the helpdesk so we ring other extensions between 8am-3pm on Saturday, otherwise it rights 392 and 6001 when people call the helpdesk (x355 or x4357)
exten => 4357,1,GotoIfTime(8:00-15:00|sat|*|?default,43
I was wondering what the concusses is for building a server for asterisk, we are looking at installing it in about 7 locations (all within an hour of each other). I prefer Dell servers, but have seen there is some incompatibility with digium hardware. (
http://www.digium.com/index.php?menu=compat
Since it doesn't look like any of the FXS cards supported by asterisk
support analog DID trunks, would it work if I used a T100P connected
to an adtran channel bank (atlas 550?) with an FXS card installed?
Anyone ever try this configuration?
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I'm working on putting together some Ideas about using Asterisk in our
environment, one of the things I want to consider is DID trunks
(analog), what hardware do I need to terminate these trunks? I'm
looking at the voicetronix openswitch6 or openswitch12.
On the openswitch, I'd like to use some o
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