Questions:
[1] Can I use oslec for echo cancellation? I'll have beefy hardware.
Is echo cancellation necessary?
Yes you can use oslec provided that either your distribution has a zaptel
package with the oslec patch (or you build the zaptel drivers + oslec
yourself)
Well without echo cancelation
Steve Repo wrote:
Hello,
Please forgive me for i'm not an asterisk user yet. I've done as much
research as I can .. and have the following questions.
I'm setting up a new office and a home office and i'm shopping for
hardware.
Office: 2 analog lines
Hardware: TDM412B (2 FXO, 1FXO)
Alan Lord wrote:
If you only have one analogue line why not just get a simple x100p card?
When you use OSLEC with them they work great here in the UK. I bought my
card from a USA based eBay seller. Total cost for card and shipping was
about £17.00
Respectfully, I don't agree. I've
Hi All;
I have an IP Trunk between two asterisk box (A and B),
when side A originate calls via the digium card from
the fxo port, and need to talk with side B at Polycom
320, then there is a disturbance will be heared on
Polcyom 320.
Note that used codec for the trunk between the side A
and B is
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
the X100P card, on the long period they almost always have some
drawbacks... Faxing have been troubling for me. Don't know if it was for
the line or else, but with a Digium card I had no problem at all.
No
Hi,
Would just like to ask about cdr, i have an asterisk and i would like to bill
only outbound calls not extension to extension, when i'm looking at the CDR, i
can't figure out which fields i need to filter all outbound calls only.
e.g if i dial 00. or 9XX (for local pstn calls) those
Hello ppl,
Anyway in Asterisk to update a DB/ do some action on
events like ringing.
The issue is I need to be able to hangup/cancel a
call, if it's ringing(decided by the admin). This is
independant of the timeout that we can specify in the
Dial command.
If I could somehow update a DB with
On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote:
Tzafrir Cohen a écrit :
On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:
Here it is, but since the AsteriskNow release has stripped the binary
i fear it won't be of much use:
Is there any -debug
Tzafrir Cohen a écrit :
On Tue, May 06, 2008 at 09:42:17PM +0200, Benoit Plessis wrote:
Tzafrir Cohen a écrit :
On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote:
Here it is, but since the AsteriskNow release has stripped the binary
i fear it won't be of
Hi!
I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls)
and 15 SIP extensions.
The receptionist has a SNOM-360.
How many SIP accounts would you configure on that phone?
Only one would be enough?
Yes.
One SIP account, has a limit on concurrent calls?
No, not that I am
Hi All,
How can i enable time condition on meetme? below i would like to deny
callers if the time is not yet the scheduled time of the conference, but
it seems like its still goes to 600,2, hope anyone can help.
[meet-me-test]
exten = 600,1,GotoIfTime(10:00-11:00|*|19|Apr?meet-me-test,600,3)
Concise summary: When I set the TOS to Minimize-Delay the DSCP field
in the packet changes from Expedited Forwarding to Unknown
Here are the details:
Scenario 1: IpTables is not used to set the TOS
This is what the packet looks like using wireshark:
Internet Protocol, Src: 59.93.192.xx
Vinícius Fontes wrote:
Sorry, my fault. I did a
$ grep -R -i TE410P *
before asking, but in the README it was listed as TE410, so no match.
The TE410P and TDM410 (no P) are very different; I don't think you
actually searched for the TE410P :-)
Yes, wctdm24xxp is the correct driver for
On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote:
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
the X100P card, on the long period they almost always have some
drawbacks... Faxing have been troubling for me. Don't know if it was for
the line or
This happens because the TOS and DSCP are the same field. TOS is the first
implementation of QoS on the IP header, DSCP is it's evolution and uses the
same field on the IP header, you can use only one of the two at the same
time.
Best Regards,
On Wed, May 7, 2008 at 8:59 AM, Vikas [EMAIL
I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I
can see the channel bank with lsusb, but when I tried to use
zaptel_hardware, or when I try the /etc/init.d/script, they don't see my
Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's
dependecies, fxload
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX
in a flash, etc. etc.
If you have trouble finding it let me know and I can send you it.
I can;t really vouch for its quality, but I do use it and it does work... but
i;m not sure how well it handles multiple results.
7 maj 2008 kl. 04.34 skrev Brian J. Murrell:
Does anyone have a better ENUM lookup handler than the built-in
ENUMLOOKUP() function? The built-in function does not properly handle
multiple return values such as:
8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP
Why are you trying to change the ToS from 46 (0x2e) Expedited for the RTP/RTCP
packets to 16 (0x10)?
I mean... these values really only need to be meaningful to yourself, your
switches, your routers etc however
ToS 46 (0x2e) is the standard value for RTP / RTCP as it is basically the
Hi everyone,
I've got the same problem described in
http://bugs.digium.com/view.php?id=10647 (unfortunately, the bug is closed
and I could not find the way to reopen it).
Wiki says, When options t, T, h, H, w, W or L (with multiple
arguments) are applied, Asterisk will remain in the media
Is there a method from the dialplan that I
can turn on a voicemail indicator on a polycom phone. Like a blinking
light or something.
Then I would also need to turn it off.
Is there a way to do that?
Jerry
___
-- Bandwidth and Colocation Provided by
On Wed, May 7, 2008 at 8:17 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote:
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
the X100P card, on the long period they almost always have some
On Wed, May 7, 2008 at 9:00 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Wed, May 7, 2008 at 8:17 AM, Michael Graves [EMAIL PROTECTED] wrote:
On Wed, 07 May 2008 09:58:04 +0100, Alan Lord wrote:
Marco wrote:
Respectfully, I don't agree. I've purchased an original clone :-P of
So a quick update on this since I haven't had any feedback... I've
just grabbed the latest trunk from the digium subversion repo; I've
completely cleaned out the asterisk server and rebuilt from scratch
with CentOS 5, all pre-reqs have been yum installed and the whole box
has been yum update'd.
Jerry,
I'd imagine that you can achieve this through SIP Event Notify, via
AGI using
sipsak (www.sipsak.org)
I'm doing a similar thing with Cisco phones, and it works great.
Here's an example of what I pass to the phones.
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
From: sip:asterisk;tag=2427962554
On Wednesday 07 May 2008 08:00:17 Steve Totaro wrote:
If your budget is tight and you want a decent card (not an X100P) with
room to upgrade, then check out
http://www.openvox.com.cn/products.php?genre_id=25 or
http://store.getvoicecards.com/index.php?cPath=66 they are the
reference design
Hi all,
i'm trying from several days to setup a SLA on my machine with some
THOMSON 2030.
My goal is to bind every F key to an extension (NOT a trunk).
So, F1 = 201, F2 = 202, F3 = 203, and so on...
I'm googled thousand of pages and many more confusing concepts are in my mind.
My server uses
Al Baker schrieb:
Are you saying the * server does NOT TRY to re-establish the BD connection ?
The MySQL Realtime driver _does_ reconnect.
(Search for mysql_reconnect() in res_config_mysql.c)
If NOT, what happens to you CDR records ?
Same thing with cdr_addon_mysql.c - it tries to reconnect.
On Wed, 2008-05-07 at 14:26 +0200, Johansson Olle E wrote:
Quoting RFC 3824:
Only one SIP URI, ideally, appears in an ENUM record set for a
telephone number. While it may initially seem attractive to
provide multiple SIP URIs that reach the same user within ENUM,
if
Steve Totaro schrieb:
I would not run MySQL on the local box. I would simple use Asterisk's
csv CDRs and then use some script to import the CSVs into a database
residing on another server using some sort of script. Depending on
your needs, you could probably run that during low call volume.
ronald ramos schrieb:
Would just like to ask about cdr, i have an asterisk and i would like to bill
only outbound calls not extension to extension, when i'm looking at the CDR,
i can't figure out which fields i need to filter all outbound calls only.
e.g if i dial 00. or 9XX (for
On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 08:00:17 Steve Totaro wrote:
If your budget is tight and you want a decent card (not an X100P) with
room to upgrade, then check out
http://www.openvox.com.cn/products.php?genre_id=25 or
On Wed, May 07, 2008 at 09:20:59AM -0300, Guilherme Loch Waltrick Góes wrote:
I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I
can see the channel bank with lsusb, but when I tried to use
zaptel_hardware, or when I try the /etc/init.d/script, they don't see my
On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Note that purchasing Digium boards helps pay for full time Asterisk
development, and purchasing clone boards does not pay for even a part-time
Asterisk developer.
--
Tilghman
BTW, I am all for having payed
Hi all,
I want to get the first three Via-Header of an INVITE request to commit
them into an AGI script:
In the documentation is stated, that there are two possibilities to call
this function, the first one using only one parameter for the
SIP_HEADER-function is working:
exten =
On Wed, 2008-05-07 at 08:21 -0400, Matt Watson wrote:
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX
in a flash, etc. etc.
Yeah, I had gotten that impression somewhere too.
If you have trouble finding it let me know and I can send you it.
If you would be so
On Wed, May 7, 2008 at 10:22 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Wed, May 7, 2008 at 9:40 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Note that purchasing Digium boards helps pay for full time Asterisk
development, and purchasing clone boards does not pay for even a part-time
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB/ do some action on
events like ringing.
The issue is I need to be able to hangup/cancel a
call, if it's ringing(decided by the admin). This is
independant of the timeout that we can specify in the
Dial command.
If I could somehow
Benoit Plessis wrote:
Well i tried a debian/lenny with an mISDN patched for 2.6.24
Are those patches available somewhere? Pointers?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
FaberK schrieb:
I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls)
and 15 SIP extensions.
The receptionist has a SNOM-360.
How many SIP accounts would you configure on that phone?
1
One SIP account, has a limit on concurrent calls?
Of course there is _some_ kind of limit
On Wednesday 07 May 2008 09:40:21 Steve Totaro wrote:
Interesting results in Google for TDM400P TigerJet reference design.
http://www.google.com/search?hl=ensafe=offclient=firefox-arls=org.mozill
a:en-US:officialhs=h9Ppwst=1sa=Xoi=spellresnum=1ct=resultcd=1q=Tiger
Philipp Kempgen a écrit :
Benoit Plessis wrote:
Well i tried a debian/lenny with an mISDN patched for 2.6.24
Are those patches available somewhere? Pointers?
Regards,
Philipp Kempgen
It's a patch i got from the gentoo portage site, should be made of some
mISDN commit
in
I'm using Zaptel 1.4.10 compiled from source, it's an Astribank with
6FXS+2FXO here's the output of some commands:
[EMAIL PROTECTED]:~# invoke-rc.d asterisk stop
* Stopping Asterisk PBX: asterisk
...done.
[EMAIL PROTECTED]:~# invoke-rc.d zaptel restart
Unloading zaptel hardware drivers:.
Basic process:
1) Build the A*k server so that it has tftp installed (or another box
that does)
2) Build up the SIPdefault.conf and get the firmware files in place (see
Cisco docs on this, plus theres loads on the wikis).
3) Test with a single phone, change its tftp server to the asterisk.
Check
Benoit Plessis wrote:
Philipp Kempgen a écrit :
Benoit Plessis wrote:
Well i tried a debian/lenny with an mISDN patched for 2.6.24
Are those patches available somewhere? Pointers?
It's a patch i got from the gentoo portage site, should be made of some
mISDN commit
in the git tree. but I
Steve Johnson wrote:
Hi everyone,
We have a particular user on our Asterisk 1.4.x system who always
listens to his voicemail messages via email.
- Is there some way to send the voicemail ONLY to email and not retain
them on the phone?
- Alternatively, can the voicemail system only keep,
Steve Johnson schrieb:
- Is there some way to send the voicemail ONLY to email and not retain
them on the phone?
delete=yes in voicemail.conf I believe.
Grüße,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all
see voicemail.conf.sample all the options you need are documented there.
maxmsg delete
On Wed, May 7, 2008 at 12:49 PM, Steve Johnson [EMAIL PROTECTED] wrote:
Hi everyone,
We have a particular user on our Asterisk 1.4.x system who always
listens to his voicemail messages via email.
-
On my SIP carrier, I register to a proxy sipconnect.dal0.cbeyond.net
which ends up being 192.168.22.212 (They supply a T1 bundle)
#sip show peers
Name/username HostDyn Nat ACL Port Status
snip
Generic-8174691929/817469 192.168.22.212 N 5060 OK (41 ms)
[EMAIL PROTECTED] wrote:
On my SIP carrier, I register to a proxy sipconnect.dal0.cbeyond.net
which ends up being 192.168.22.212 (They supply a T1 bundle)
#sip show peers
Name/username HostDyn Nat ACL Port Status
snip
Generic-8174691929/817469 192.168.22.212
The leading 0 is not part of Caller*ID. Remove it.
Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten =
On Wed, May 07, 2008 at 01:03:53PM -0300, Guilherme Loch Waltrick Góes wrote:
I'm using Zaptel 1.4.10 compiled from source, it's an Astribank with
6FXS+2FXO here's the output of some commands:
[EMAIL PROTECTED]:~# invoke-rc.d asterisk stop
* Stopping Asterisk PBX: asterisk
...done.
We also have a script available (on www.generationd.com) which allows a user
to reply to an emailed voicemail, which then deletes the associated VM file
on the asterisk box.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas van
dem Helge
Sent:
Hi Marcelo,
Sorry, just realized I responded to you directly rather than the list. So
for the record, here's the list response.
_
Hi Marcelo,
What format are the recordings in? Have you tried converting them to the
same format?
Thanks
_
From: Marcelo Freitas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: 07 May 2008 20:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Out-Going Calleriid
The leading 0 is not part of Caller*ID. Remove it.
Tim Guy schrieb:
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Maybe CALLERID(num) works?
Grüße,
Philipp Kempgen
--
amooma GmbH -
Slightly off-topic:
On Wed, May 07, 2008 at 10:29:47AM -0400, Brian J. Murrell wrote:
I guess a code audit will tell. :-) Although I got an impression that
it was written in PHP. I'm not much of a fan of PHP. Don't really see
the point for something so simple. Bash, Perl (without the
On Wed, 2008-05-07 at 22:54 +0300, Tzafrir Cohen wrote:
Slightly off-topic:
Yeah.
On Wed, May 07, 2008 at 10:29:47AM -0400, Brian J. Murrell wrote:
I guess a code audit will tell. :-) Although I got an impression that
it was written in PHP. I'm not much of a fan of PHP. Don't really
Tim Guy wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: 07 May 2008 20:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Out-Going Calleriid
The leading 0 is not part of
At 10:04 AM -0400 2008/5/7, Brian J. Murrell wrote:
On Wed, 2008-05-07 at 14:26 +0200, Johansson Olle E wrote:
Quoting RFC 3824:
Only one SIP URI, ideally, appears in an ENUM record set for a
telephone number. While it may initially seem attractive to
provide multiple SIP
On Wed, May 07, 2008 at 07:46:59PM +0100, Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten =
Hello,
I have given up hope of finding a solution for my problem and I think this
is my last resort.
At my company I have a Trixbox box and I used freePBX to configure the pbx.
They have a queue with 15 static members.. they are not online all the
time.. still when ever a call comes in the
On Wed, 2008-05-07 at 13:40 -0700, John Todd wrote:
1) The ENUMLOOKUP function is currently being fixed for TRUNK.
Ahhh. Sweet. I wonder how difficult a backport will be.
Take a look at http://bugs.digium.com/view.php?id=8089 for the
current status. Testing would be appreciated.
Will
Hello users,
I had developed several patches that allows to monitor current status
of queues/channels in realtime db. For example specifying realtime
family channels will make asterisk to keep current channel list in
realtime database engine. The same would be for queue callers and
queue members
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB/ do some action on
events like ringing.
The issue is I need to be able to hangup/cancel a
call, if it's ringing(decided by the admin). This is
The Asterisk development team has released Asterisk version 1.4.20-rc2.
This release is a release candidate for the upcoming official release of 1.4.20.
It includes a fix for a SIP channel driver regression introduced in 1.4.20-rc1,
among a number of other changes. For a full list of changes
Atis Lezdins schrieb:
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB/ do some action on
events like ringing.
The issue is I need to be able to hangup/cancel a
call, if it's ringing(decided by the
On Wed, 2008-05-07 at 11:44 +1000, Paul Hales wrote:
Tomorrow night is the monthly Asterisk night...in melbourne
(australia)...
The usual stuff - get together, eat, show off tech toys.
At the Pint on Punt, from 7pm.
later,
PaulH
Love to come, but as my bike got a
Hello all.
I'm encountering an issue whereby a Mediatrix 2102 is able to register,
authenticate, and place a call into an asterisk box. However, the
problem happens when the asterisk box tries to proxy the call to another
Mediatrix 2102, or back to the other port on the same Mediatrix 2102.
No
Hi,
In asterisk is there a way of saving the voice codec used in the call in the
CDR.
I am having mostly SIP calls, with few H323 calls.
I have been trying for the past 2 weeks to figure it out on my own, but with no
luck.
There are no channel variables that can give the current codec used in
On Thu, May 8, 2008 at 12:34 AM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Atis Lezdins schrieb:
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB/ do some action on
events like ringing.
Atis Lezdins schrieb:
I had developed several patches that allows to monitor current status
of queues/channels in realtime db.
[...]
+1 as long as the user can choose whether they want realtime status
data in the database.
*** Supporting this feature
If You find that those features would be
The subject should read Asterisk 1.4.20-rc2 Now Available
On Wed, May 7, 2008 at 11:24 AM, The Asterisk Development Team
[EMAIL PROTECTED] wrote:
The Asterisk development team has released Asterisk version 1.4.20-rc2.
This release is a release candidate for the upcoming official release of
Gentlemen,
Dean Collins alerted me to this thread which I had skipped over.
(Thanks, Dean.) I thought I'd offer my viewpoint on the matter; please
take it for what it is - just another opinion, although I hope it is an
informed one. From my personal experience with buying software,
On Wednesday 07 May 2008 16:11:05 Atis Lezdins wrote:
However I encountered a resistance from Asterisk developers, as they
don't wish to accept my patches - because they don't wish to support
another interface. As I read in Bug Guidelines, if enough users
request this, it should make into
Thanks Adrian and all the other guys that gave me tips on this
Femi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian Marsh
Sent: 07 May 2008 17:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco to
Antoine Megalla schrieb:
In asterisk is there a way of saving the voice codec used in the call in the
CDR.
I am having mostly SIP calls, with few H323 calls.
I have been trying for the past 2 weeks to figure it out on my own, but with
no luck.
There are no channel variables that can
On Thu, May 8, 2008 at 1:07 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 16:11:05 Atis Lezdins wrote:
However I encountered a resistance from Asterisk developers, as they
don't wish to accept my patches - because they don't wish to support
another interface. As I
Similar system and situation here
I finally taught mine to log off and on without logging in phantom
extensions or external numbers (THAT was almost a disaster) But prior
to that Do Not Disturb helped except for not allowing internal,
non-queue calls to be answered without going to VM.
I'm still
Brian J. Murrell wrote:
Right. Which to me at least, tightly couples it. IOW, the security
fix, while yes, it fixes the security problem, is quite useless without
this other fix as it makes iax2 unstable.
I agree with you.
I am in the process of working on the Asterisk 1.2.20 release, which
Hello everyone,
We are building a new * server based on a Supermicro motherboard with a 2.8
Xeon processor and a TE220B card. We're using the PBX In a Flash distribution.
What we've found is that with a 4 user MeetMe conference, the CPU usage is
consistently around 16%. This in comparison
Andrea Spadaccini wrote:
I've set up DUNDi between two asterisk boxes, and sometimes happens that calls
from machine A can't reach peers in machine B, but calls from B to A work
correctly.
The strange thing is that the CLI command 'dundi show peers' shows correctly
the registered peer in
Sanjay Rajdev wrote:
I just want to Run Asterisk with the basic required modules, What can I do to
achieve so?
My only requirement is to run SIP clients and the Dictate Module.
2 options:
1) Before compiling and installing Asterisk, run make menuselect to select
only the modules that you
JR Richardson wrote:
I have 1.4.9.1 setup, with the compiler flags enabled for T38, and
have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes
between devices but can't seem to invoke T38 pt UDPTL. It's enabled
in sip.conf [general] and well as the [peer].
I get an error at the
There is a bug in 1.4.19.1 with IAX. That's your issue.
On Wed, May 7, 2008 at 12:38 PM, Kevin Ragsdale [EMAIL PROTECTED] wrote:
Hello everyone,
We are building a new * server based on a Supermicro motherboard with a 2.8
Xeon processor and a TE220B card. We're using the PBX In a Flash
It was a big pain for me, I change all to Linksys spa2102
I had 20 Mediatrik as a paper weight
Sorry
Polo
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Daniel Lynes
Enviado el: Miércoles, 07 de Mayo de 2008 04:37 p.m.
Para:
On Wednesday 07 May 2008 17:27:33 Atis Lezdins wrote:
So all together - I'm saying there could be really simple interface
for all this - no troubles with locking of lists or keeping persistent
connections. Why would user application need to take care of all this,
if DB engine can do that.
On Thu, May 08, 2008 at 12:19:52AM +0300, Atis Lezdins wrote:
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB/ do some action on
events like ringing.
The issue is I need to be able to
Tilghman Lesher a écrit :
On Wednesday 07 May 2008 17:27:33 Atis Lezdins wrote:
So all together - I'm saying there could be really simple interface
for all this - no troubles with locking of lists or keeping persistent
connections. Why would user application need to take care of all this,
Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten =
On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
Tilghman Lesher a écrit :
Your question leads to this question: why don't you create a proxy
application that listens on AMI and populates a database outside of
Asterisk,
then do all your queries to that database?
On Thu, May 8, 2008 at 3:49 AM, Ex Vito [EMAIL PROTECTED] wrote:
On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
Tilghman Lesher a écrit :
Your question leads to this question: why don't you create a proxy
application that listens on AMI and populates a
Hi list,
I'm planning a private DUNDi network for a cross-country
distributed PBX. Initially it will be composed of about 10
systems, growing to about 20.
Current requirements point to a topology of two interconnected
DUNDi hubs, each peering with half the PBXs... This would
lead
On Wed, May 7, 2008 at 11:38 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 09:40:21 Steve Totaro wrote:
Interesting results in Google for TDM400P TigerJet reference design.
http://www.google.com/search?hl=ensafe=offclient=firefox-arls=org.mozill
Ex Vito wrote:
Now, how to move on to acheive some kind of fault tolerance ?
According to the docs we've studied, DUNDi does not like loops
(which we assume one can limit with low enough TTLs).
Which documentation are you referring to? You may have misunderstood
something,
or there
I don;t have any answers for you...
But I would love to hear about the results after you get this working and what
road blocks you hit and how you overcame them.
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ex Vito [EMAIL PROTECTED]
Sanjay Rajdev wrote:
I have Asterisk 1.4.15 installed on a Fedora Core 8 machine. Asterisk is
snip
In the dialplan we have used MixMonitor() to record the calls.
Can anyone help me on getting to the root of the problem or fixing it?
We have fixed a _lot_ of issues in that area of the
Are you using IAX2 as your transport between the 2 servers or SIP?
If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either
machine? If so, you may be encountering the IAX2 bug that some have been
discussing on the list recently you can read it here:
Besides the Background() app mentioned, you might like the WaitExten()
app
Thanks guys for your response.
I have had much success with Read() as below so that whenever I press a
key before the sound file finishes playing, it will read the digit and
move to the next line.
exten =
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