Friday, June 17, 2016, 11:56:34 PM, Mike wrote:
> I've got a device that seems to become unreachable for about 2 minutes, every
> hour. From what I can tell, it isn't due to network or server issues. Any
> ideas?
The default registration time in spa112 is 1 hour. If registering is
slow in your
Hi,
back in the old analog telephony days there was digital PBX-es and
digital system phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blinks) or busy (constant
light), and the user
Hi,
back in the old analog telephony days there was digital PBX-es and
digital system phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blinks) or busy (constant
light), and the user
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card with CAPI in my first
installation (1 BRI), and a hfc 4 port bri card with misdn on other.
The first
Tuesday, August 27, 2013, 8:41:18 PM, Patrick wrote:
On 08/27/2013 08:04 PM, Gergo Csibra wrote:
Hi,
is anybody out there who can set the outgoing caller id on ISDN (CAPI
or misdn) channels? I've tryed everything what I found in forums, os
voip-info.com but no luck. I use a fritz card
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
Tuesday, August 20, 2013, 6:08:19 PM, Jonas wrote:
On 08/20/2013 06:03 PM, Gergo Csibra wrote:
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan
Sunday, December 30, 2012, 5:13:30 PM, Patrick wrote:
On 12/30/2012 04:26 PM, Ron Wheeler wrote:
I participate in a lot of lists and top posting is now the norm since
people want to see quickly if the message is worth reading.
Isn't it a bit of a stretch to extrapolate your experience with
Monday, September 26, 2011, 11:33:50 PM, Kristijan wrote:
Gergo, why do you want to use mISDN? Use Dahdi. Or do you want to
use a very exotic isdn card which is only supported by mISDN? tell
us more.
Well, I want to use one channel driver for all my installations. Now I
have to reinstall a
Hi,
are there anybody, who using the chan_misdn included with Asterisk
v1.8? If yes what mISDN version used v1 or v2? Yes, I can read on many
pages for mISDN2 I need to use chan_lcr, but this informations are 2-3
years old, and I can't imagine asterisk v1.8 chan_misdn works only
with linux kernel
Monday, September 26, 2011, 7:20:10 PM, Kevin wrote:
On 09/26/2011 11:35 AM, Gergo Csibra wrote:
Hi,
are there anybody, who using the chan_misdn included with Asterisk
v1.8? If yes what mISDN version used v1 or v2? Yes, I can read on many
pages for mISDN2 I need to use chan_lcr
Thursday, August 4, 2011, 9:35:50 PM, Dan wrote:
Is there any technical reason for this 63 group limit?
Currently the number of callgroups/pickupgroups is limited to 63, because the
variable type that holds the group bits is a long long.
Ah, so it's a major change to get it to hold a larger
Friday, July 8, 2011, 3:13:01 PM, deeps wrote:
What is the maximum size limit of Master.csv file and what happens when it
reaches limit?
That is a text file. Only limited by filesystem.
--
Best regards,
Gergomailto:csi...@gmail.com
--
Sunday, May 29, 2011, 10:57:00 AM, virendra wrote:
I have stupid question but I want to know it. Why we use the PRI insted of
BRI ? Just for the sake of number of lines or any thing else ?
Yes, because of much more channels. But if you need only 2 or 4
channels BRI is cheaper. From 10-12
Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote:
How can I minimize this time lapse? Can Asterisk notify all SIP
clients in its sip.conf that they need to acknowledge being on-line
or not (thus forcing re-registration in my scenario)?
If you have two identical servers online, it is better
Tuesday, February 1, 2011, 11:22:30 PM, Juan wrote:
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650,
Thursday, September 9, 2010, 4:32:29 PM, Gopalakrishnan wrote:
I am sending FAX from one extension to another extension. I am not able to
send.
Preferred Codec:G711u
You forget to mentoin where do you live? In some countries the G711a
codec and in onther countries the G711u codec useable.
Wednesday, August 4, 2010, 6:02:45 PM, Danny wrote:
R5 would use 3 out of 4.
You can have R5 across 10 drives too. Yes, the writes will be slow,
but it possible.
--
Best regards,
Gergomailto:csi...@gmail.com
--
Friday, June 11, 2010, 12:27:08 AM, Tzafrir wrote:
On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote:
Okay. There's some problems with mISDN v2: I'm unable to compile
zaphfc, because there's no source for it. mISDN v2 works with hfcpci
too?
Certainly there is.
It's also part
Tuesday, July 6, 2010, 8:11:46 PM, bruce wrote:
Can someone please explain what Y-cords are available out there and how they
can be used with Aastra or other VoIP phones? Maybe with or WITHOUT
headsets?
Isn't a Y-cord traded for soft Barge in these days?
I think Y-cords only for PSTN. Or
Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:
i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on
CentOS 5.5. The only thing, i want to do is a call-redirection from an
isdn-call to my mobile via sip-account.
Unless you are using mISDN v2: Do yourself a favour and
Hello Asterisk,
This is only a test, because I can't start new thread in this list...
--
Best regards,
Gergo mailto:csi...@gmail.com
--
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Wednesday, March 24, 2010, 9:42:52 PM, Dave wrote:
On 03/24/2010 03:56 PM, Miguel Molina wrote:
Gergo Csibra escribió:
This is only a test, because I can't start new thread in this list...
If you can send an email, you can start a new thread on this list.
What's the point of all this?
He
Wednesday, March 24, 2010, 10:35:20 PM, Karl wrote:
Gergo is using Gmail,
He is also using an IMAP client to author his messages (as I do).
No. I don't use any IMAP thing :)
I use The Bat! but only to download messages with POP3, and I send
messages through my ISP's SMTP server.
--
Best
Wednesday, February 24, 2010, 3:56:50 PM, David wrote:
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote:
Hi Guys
We are using asterisk 1.4 on all of our platforms for a while now.
Some of our partners recommended to use asterisk 1.6 in order to improve
overall
Wednesday, February 17, 2010, 10:40:18 PM, Steve wrote:
Congratulation. Doesn't it feel great to help yourself rather than
bothering the mailing list with questions that have nothing to do with
Asterisk? And it only took you 17 minutes!
Much better than cool dude :)
--
Best regards,
Gergo
Tuesday, February 16, 2010, 12:55:12 PM, cool wrote:
call comes it should be received by extenion 2000, n if person wants to
talk to Sales, receptionist should put the caller on hold than connect
to Sales i.e exten 2001, while on hold the caller should hear music on
hold,now sale exten can
Friday, February 12, 2010, 9:57:42 PM, cool wrote:
how to allow some extenstions to call outside and some extensions
cant call outside. i am attaching sipand extensions.conf
thx
Put the extensions into different contexts, and create outside call
extensions only in the allowed context.
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote:
Forget about virtualization!
...
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass and makes stuff bork beyond imagination.
Well. Why do you use computer? There're slide-rule. You can
Thursday, January 21, 2010, 12:53:09 AM, Jeff wrote:
On Thu, 21 Jan 2010, Gergo Csibra wrote:
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote:
Forget about virtualization!
...
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass
Sunday, January 10, 2010, 11:24:22 AM, hadi wrote:
You are not willing to help me anymore ?
Why do you think this?
--
Best regards,
Gergomailto:csi...@gmail.com
--
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Friday, January 1, 2010, 7:12:54 PM, Alex wrote:
On 01/01/2010 01:06 PM, Warren Selby wrote:
Also, shouldn't the .php script be located in /var/lib/asterisk/agi-bin?
Fact.
And on a live channel must use AGI instead of DeadAGI.
And man should not topposting on a maillist...
--
Best
Wednesday, December 30, 2009, 6:48:37 PM, Robert wrote:
Tilghman Lesher wrote:
On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote:
Just curious if anyone has successfully patched cdr_addon_mysql to use
accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'?
Thursday, December 24, 2009, 5:41:46 PM, Danny wrote:
Just my opinion; unless you are recording long or many long calls, you
should record to your local drive, then copy the files to the USB drive.
Asterisk is a very good tool - you don't need to mess it up by introducing
an easy point of
Wednesday, December 23, 2009, 11:17:39 AM, ABBAS wrote:
when compiling asterisk with Postgresql we need to specify directory where
the postgresql is installed.
I need to know once asterisk is ready to use(ie compiled and installed ).
Do it still refer the postgresql files that are not part of
Hi,
something very strange here. I've downloaded asterisk 1.6.1.11 and
dahdi-linux-complete-2.2.0.2+2.2.0. My linux kernel is 2.6.30.9. After
compiling and installing dahdi it found my hfc based card, but at
reboot it says driver should be 'zaphfc' but is actually 'hfcpci'.
Ok. This hfcpci comes
Saturday, April 4, 2009, 3:13:12 PM, Puskás wrote:
Got it working with Asterisk 1.2 installed on the same PC as Asterisk 1.4 [
in
different directorys and username of course :) ] . Using isdn4linux kernel
module and Dial(Modem/ttyI0/1234567:${EXTEN}) command.
Használj MISDN-t, és ne
Wednesday, March 4, 2009, 5:43:13 PM, Joseph wrote:
On 03/04/09 15:56, Gergo Csibra wrote:
Wednesday, March 4, 2009, 3:22:59 PM, Joseph wrote:
FAX Passthru Codec: G711u
for me FAX works better with G711a
Can you folks compare my setting below with your settings and let me know
Wednesday, March 4, 2009, 3:22:59 PM, Joseph wrote:
FAX Passthru Codec: G711u
for me FAX works better with G711a
--
Best regards,
Gergomailto:csi...@gmail.com
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Friday, December 5, 2008, 2:49:59 PM, Andrew wrote:
Address added to spam filter. Please do NOT e-mail me again.
A: Because it messes up the order in which people normally read text.
Q: Why is top-posting such a bad thing?
A: Top-posting.
Q: What is the most annoying thing in e-mail?
--
Best
Sunday, October 26, 2008, 12:31:16 PM, Hans wrote:
On Sat, 2008-10-25 at 11:54 -0600, Joseph L. Casale wrote:
I need to increase reliability at an office as SIP/Internet provider outages
are causing some issues.
What would be the least expensive analogue card that people are using
Tuesday, September 23, 2008, 11:57:00 AM, Thanos wrote:
Hello
I have just set up Asterisk Asterisk 1.4.21.2 on a CentOS 5.2 machine.
I am using the OpenVox B200P ISDN card.
My problem is that even though chan_misdn module seems to be loaded
correctly with
Asterisk (I can see it using
Hello asterisk-users,
My SIP phones are in pickupgroup, and if some of them ringing from
other phone can pick up with *8 as usual. But I want to know if this
happen. I've tried the a extension, but seems not working.
Any other idea?
--
Best regards,
Gergo
Hi,
The Dial command has the g option, voip-info.org says:
If the g option is specified, and the called party hangs up before
the calling party, then Dial continues execution at priority n+1.
and this works well. But I need to continue the execution if the
caller hangs up first too.
What do I
Wednesday, February 13, 2008, 2:56:42 PM, Michiel wrote:
On 13:14, Wed 13 Feb 08, Gergo Csibra wrote:
Well?
Is it impossible to detect BUSY on SIP channels?
not in stock 1.2
Bristuff has a function for it, and russell created a
function for it in current trunk that is also available
Saturday, February 9, 2008, 10:29:08 AM, Csibra wrote:
My problem is in subject. As I read in documentations and
voip-info.org I can't user ChanIsAvalil because it not detects BUSY
information on SIP channel. I've tried to use SIPPEER function, but it
gives OK (9 ms) back on BUSY SIP channel.
Friday, December 14, 2007, 5:47:38 AM, Paul wrote:
Umm - you could just buy a SPA-3000/3102/3666/etc.
What is SPA-3666?
--
Best regards,
Gergomailto:[EMAIL PROTECTED]
___
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Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote:
Hi all,
I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404
or 2 TDM808 to get 16 FXO? What is the difference (in performance and
control) in using 4 x TDM404 and 2 x TDM808 if possible?
ango
Well, using more than one
Wednesday, June 13, 2007, 9:44:08 AM, Nick wrote:
I'd love to get it working... if you could share a sample config or other
advice, I'd appreciate it.
http://www.voip-info.org/wiki/view/Linksys-Cisco+SPA400
http://voxilla.com/voxilla-tips/voxilla-tips/the-linksys-spa400-and-asterisk-886.html
Friday, May 4, 2007, 10:42:13 AM, Phil wrote:
On Thu, May 03, 2007 at 08:17:25PM +0100, Kyle Gordon wrote:
With the gamut of FXO cards out there, I'm looking for a recommendation for
home use. I have a nicely working Asterisk 1.4 system that just requires an
FXO card to connect my NTL PSTN
Friday, May 4, 2007, 3:06:02 PM, Dave wrote:
On Fri, 2007-05-04 at 08:35 -0400, Joe acquisto wrote:
Gergo Csibra [EMAIL PROTECTED] Wrote: 5/4/2007 8:12 AM:
Yes it is an ATA with an FXS and an FXO port, and you can use as many
as you want instead of one TDM400/TDM800/TDM2400.
It has two
Monday, April 23, 2007, 12:44:08 PM, Diego wrote:
you need to use apt-get install asterisk.
If you MUST HAVE 1.217 or your cats die, there are repositories available.
For
example, read this: http://www.buildserver.net/
If you still MUST build asterisk yourself, I wish you good luck.
Monday, April 2, 2007, 7:30:57 PM, Giedrius wrote:
Has anybody debian and misdn working fine? Maybe you can advices , what
kernel and misdn versions to use...
I use kernel 2.6.20.1 and misdn 1.1.0 with fritz card, and working
fine. The kernel, asterisk (1.2.15) and misdn also compiled from
Friday, March 30, 2007, 5:02:08 AM, Matt wrote:
On 3/29/07, Alan Chandler [EMAIL PROTECTED] wrote:
I have a linksys SPA 3102 with a DECT phone connected into its Telephone
port.
It has been working, but something I've done (and I don't know what)
means that now everytime asterisk tries to
Thursday, March 29, 2007, 7:18:43 PM, LKS wrote:
Hi folks!
Does anybody know how to receive send faxes throw mISDN? It's almost
impossible!
Describe your problem, but read this before:
http://www.catb.org/~esr/faqs/smart-questions.html
It works for me, in 3 places, the analogue fax machines
On 3/20/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote:
I want to install Asterisk on a Debian machine.
I need to download the sources or just with apt-get install is enought???
It depends on what version do you want to use. In sarge is only the
version 1.0.7. In etch is 1.2.13, but the 1.2
On 3/9/07, Wai Wu [EMAIL PROTECTED] wrote:
Two things.
1) This is a bug(feature) of standard analog switchs which only clear the talk
path when both sides of the call are terminated.
Well, not exactly. The call will not terminated until the caller (not
both) hangs up. I don't knew the
Hi!
Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to
show the Caller number on the phone.
There's a Caller ID Method: option on Regional settings, but I
tested all options, and my CLIP phone never shows the Caller number...
:(
Any idea?
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