It's a Digium single-port job. No other timing sources aviailable (the * box IS
the pbx).
qrss wrote:
What kind of card are they using? Is there only 1 telco circuit?
If so, then I'm thinking their card should have detected the loss of
service and switched to it's internal clock. Do they
Does anyone have a MAX/APX with working ingress PRI calling name?
I recently acquired a MAX TNT on the cheap and it's integrating fine
except for one thing. In the 11.0.0 release notes, it is stated that
ISDN calling name will, if present and permitted by presentation
flags, be added to the
On Thu, 23 Jun 2005, Massimo De Nadal wrote:
Have you planned to integrate some echo cancel feature ?
Echo cancelling (if the card supports it) is already implemented.
As far as I know the Eicon Diva Server cards are the only cards supporting
echo cancel via onboard DSPs.
Armin
Armin
I have been trying to open an account with voipreach.net for over
a week now and I have not gotten any response from them as yet.
None of their phone numbers are working.
They didn't respond to my emails either... Tixter is right, forget
about them if they don't even care to reply to take your
Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:
brands
+--+--+--+--+--++--+--+
| name | language | inc | publishednum | did | markup |
Ypek,
I would like to know how can I manage to implement a table which translates
an extension number into a phone number. Let see an example:
There are many ways of doing this. You could map the extensions to
phones in extensions.conf, via the internal database or via an
external database, or
Hi,
I'm pulling my hair down and getting bold :-) . I have Asterisk between
Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
Asterisk)
I'm trying to do just plain transfer of call from pbx to ISDN through
Asterisk...
It seems like PBX hangsup, when call is
Hi,
It
seems that my flash panel only works when I specify my ip address and not the
host name.
I've
tried quite a few things (change host file, dns resolve, proxying.) but couldnt
get it to work.
Anyone
knows how to solve this?
Thanks,
Ohad
Sorry 4 a.m. I'm kind of tired and I slipped a password. :S
Already changed it. Sorry!
Juan Luis Moyano wrote:
Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:
brands
Hi all,
Thanks anyway for helping me to install h323 and it
work i think. my problem now ..i dunno the
configuration from cisco and oh323.conf coz i have
tried several time ans still get error message from
asterisk voip-h323...failed so falling back to
'exten' s.
did anyone here have the
I am running [EMAIL PROTECTED] and have a digium
tdm04b (4 fxo)
The problem we have is we have 3 incoming pstn lines that step down from the telco,
then a spare line and a fax line. The office is now looking to add a second
0800 (free dial in NZ) to terminate to the spare line and the
try to clean and retry co configure strp by step dont
forget to mark your step. I already try and installed
my problem was the same with you so ...i tri to use
another version. I tri 0.7.1 at the first time then
0.7.0 also have a same error but then work after i use
0.6.5. my machine runing RH9
How do you get the called number on incoming SIP calls? I've never
had multiple DID's via SIP from one provider before and somehow I
never realized that with IAX it just works, and SIP is different.
If I don't set an extension in the register command the incoming
invite has sip:[EMAIL PROTECTED]
Armin Schindler ha scritto:
Have you planned to integrate some echo cancel feature ?
Echo cancelling (if the card supports it) is already implemented.
I think he was talking about the software echo suppressor
As far as I know the Eicon Diva Server cards are the only cards supporting
Date: Thu, 23 Jun 2005 08:50:50 +0200
From: Robert Rozman [EMAIL PROTECTED]
Subject: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *
- Euroisdn Italy
I'm pulling my hair down and getting bold :-) . I have Asterisk
between
Panasonic KXTD816 and Euroisdn in Italy (beronet
On Thu, 23 Jun 2005, Robert Rozman wrote:
I'm pulling my hair down and getting bold :-) . I have Asterisk between
Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff
Asterisk)
I'm trying to do just plain transfer of call from pbx to ISDN through
Asterisk...
It
Have you planned to integrate some echo cancel feature ?
Besides the Eicon-CAPI feature there is an echosquelch in the driver.
Elmar
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Daniel,
we have the same problem when our PRI line drops and Zapras has to
reconnect. You will also notice that the pppd process does not die
when Zapras does and the ppp connection cannot re-establish itself.
What we normally do is restart asterisk and then kill the pppd process
with the
I have a second-hand 7960 which I am attempting to upgrade to use a SIP
image.
The phone currently has a firmware release which doesn't seem to be listed
in Cisco docs - P003AM30. On reboot, it finds the tftp server and requests
the firmware image listed in OX79XX.txt correctly, displaying
Yes, I know. I was meaning the software thing.
Diva server cancels echo via dsp only with new revisions boards (older
boards are not able to run newer drivers with echo cancellation).
Fritz cards don't cancel echo anyway.
And echo squelch is only a trick that doesn't really solve the problem.
Hey does anyone know how to configure the 4 port fxo to work with
Asterisk? I have the updated firmware. All ports register, however
incoming calls are never handled properly by the fxo. I even set
hotline.
Does anyone have any info, or perhaps a web site?
Sergio Chersovani wrote:
As far as I know the Eicon Diva Server cards are the only cards
supporting
echo cancel via onboard DSPs.
AVM active cards do not support it?
No.
Avm active cards are basically multi fritz boards running the same
firmware onboard instead of charging pc cpu.
On Thu, 23 Jun 2005, Massimo De Nadal wrote:
Yes, I know. I was meaning the software thing.
Diva server cancels echo via dsp only with new revisions boards (older boards
are not able to run newer drivers with echo cancellation).
Which boards don't support that? If DSPs on board, echo-cancel
I have an asterisk box connected to two isdn lines via an AVM c2 card,
the ISDN boxes have the 0227006XXX and 0227007XXX numbers, and are
configured both p2p, with the first one as file-leader.
(I don't know if file-leader is the correct term, it's a literal
translation from the italian term
Doing further tests, I discovered that I can successfully do MeetMe
on both server B and server C, AS LONG AS all parties are SIP
extensions registered on the same server (e.g. server B or server C).
However, when I try to bring a call from server A into a MeetMe in
server B or server C,
Armin Schindler ha scritto:
Which boards don't support that? If DSPs on board, echo-cancel should be
available.
I have in my hands right now a DIVA Server BRI-2M-PCI (not the 2.0
version) which own its dsp but doesn't echo cancel, due to old capi
drivers which don't support this feature.
Hello all
i am using asterisk 1.07 with mpg123-0.59r but still i get very choppy sounds, any suggestions?
extensions.conf
---
exten = 444,1,WaitMusicOnHold(120)
modules.conf
[modules]
autoload=yes
load
On Thu, 23 Jun 2005, Massimo De Nadal wrote:
Armin Schindler ha scritto:
Which boards don't support that? If DSPs on board, echo-cancel should be
available.
I have in my hands right now a DIVA Server BRI-2M-PCI (not the 2.0 version)
which own its dsp but doesn't echo cancel, due to
Yes, that should be possible. But I don't think a channel driver (and each
channel driver) should do that on its own. Software echo cancelling
belongs in a common part of Asterisk.
I strongly agree. But asterisk doesn't seem to work this way. Zap channel
has
it's own
I'm trying to figure out how much call load I can put on a Dual Xeon
2.4 Ghz Asterisk server acting strictly as an IAX2 call director, as
show in the diagram below.
The idea is that I have N number of gateway asterisk servers
connected to the PSTN using T1 Digium boards. Then, I have M
Hello,
I am searching for a working solution for Least Cost Routing usable in
France with asterisk. Does Anyone have any tip?
Regards,
Daniel ANDRE
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On Thu, 23 Jun 2005, Klaus-Peter Junghanns wrote:
Yes, that should be possible. But I don't think a channel driver (and
each
channel driver) should do that on its own. Software echo cancelling
belongs in a common part of Asterisk.
I strongly agree. But asterisk doesn't
If I connect to a provider using iax, and that provider connects to
his provider using only sip, the provider I am connecting to isn't
going to be able to bridge the call and drop out of the media stream
correct?
Correct.
If I'm understanding how bridging works, you lose the ability to
On 23 Jun 2005, at 10:48, Waldo Rubinstein wrote:
I'm trying to figure out how much call load I can put on a Dual
Xeon 2.4 Ghz Asterisk server acting strictly as an IAX2 call
director, as show in the diagram below.
The idea is that I have N number of gateway asterisk servers
connected
do you have VAD enabled?
On Thu, 23 Jun 2005 12:23:15 +0300, Mahmoud Badran wrote
Hello all
i am using asterisk 1.07 with mpg123-0.59r but still i get very
choppy sounds, any suggestions?
extensions.conf
---
exten = 444,1,WaitMusicOnHold(120)
Shouldn't [Asterisk] be smart enough to go to Zap/4 as the only available
port in the group [with a live trunk]?
Adam Goryachev wrote:
No, asterisk doesn't do dialtone detection.
But this isn't an issue of dialtone detection but one of detecting
battery (a much easier task).
I have a second-hand 7960 which I am attempting to upgrade to
use a SIP
image.
The phone currently has a firmware release which doesn't seem
to be listed
in Cisco docs - P003AM30. On reboot, it finds the tftp server
Here's how I performed the upgrade:
Downgrade from the stock P003AM30
Matt,
I've done it several times for customers in India using E1s with
EuroISDN and:
loadzone = nl
defaultzone = nl
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Matthew Gibson wrote:
Hi,
Is anyone successfully using Asterisk in India hooked up to the
I am looking into using a Cisco T1 device that uses MGCP. Asterisk is
talking to it fine, but I am having a hard time figuring out how to
handle channel grouping like Zap does. With Zap, I can take channels
1-23 and make a group g1 out of it and then simply dial Zap/g1. Does
MGCP have this
Dear All,
I was searching voip-info for Failover and load balancing for
Asterisk, my goal here is to have a system where the SIP traffic is being
divided on five central servers with Asterisk on, and if an asterisk server
fails another asterisk server will assume it's place , from my readings
I
Am Donnerstag, den 23.06.2005, 12:41 +0200 schrieb Armin Schindler:
I strongly disagree. :-) You dont want to do echo cancelation in
userspace. Especially not on a non-realtime operating system.
To make echo cancelation work it has to be as close to the line
interface as possible. Also
Peter Svensson wrote:
On Tue, 21 Jun 2005, Leandro Morgado wrote:
Steve Underwood wrote:
Robert Rozman wrote:
I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of quadbri
I have been having a problem for a while where an internal user will be
calling out via SIP through the Asterisk box which has a TDM400 and when
they pick up they have an inbound caller on the line. The lines then
become bridged and they stay that way until you do a soft hangup on
one of the
I got current stable release in CVS repository, and I think that Ok.
See below:
/var/log/messages
Jun 22 17:04:35 darthvaden kernel: PCI: Found IRQ 9 for device 02:09.0
Jun 22 17:04:35 darthvaden kernel: PCI: Sharing IRQ 9 with 00:1f.5
Jun 22 17:04:35 darthvaden kernel:
Hello,
I am interested in some management-performance issues:
1st Scenario:
A management tool (for example a webbased one) has the following process:
- write in database
- read with script (for example perl) data from db and write conf files
- reload asterisk
I was reading around in the
Leandro Morgado wrote:
Steve Underwood wrote:
Robert Rozman wrote:
Hi,
I'm getting unreliable dtmf recognition (it works fine for 4-5
digits, errors (duplicates) on more), when transferred inband from
gsm gateway to NT port of quadbri under bristuffed Asterisk.
Since Asterisk
I think I have a similar problem...When I dial out using TDM400P,It sometimes
stucks on and backs me as the line is busy although it is not...When I reboot
the Asterisk box , it becomes ok..
-Original Message-
From: Colin E. McDonald [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 23,
Hi,
the following from extension.conf does not work correctly:
exten = 301, 1, Dial(SIP/455SIP/456, 15)
That is the console output:
-- Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10) in
new stack
-- Called 455
-- Called 456
-- SIP/455-46a8 is ringing
== Spawn extension
On Wed, 2005-06-22 at 18:49, Mike M wrote:
Congratulations. What were you using prior your pull from CVS? Maybe
something old that didn't recognize the the TDM400P and its daughters?
My prior went changing zaptel.conf like you said to:
fxoks=1,2
fxsks=3,4
I got from CVS - azaptel
Hello,
Does anyone now, if there are any legal requirements for setups of
Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
interested, if a system does need to hang on a UPS?
Thanks,
Roland
Roland Welker
Moray Office Supplies
Edgar Road, Elgin, IV30 6YQ
T: +44/(0)1343/549869
As a newbie to Asterisk, I'm in love.
There is no information discussing the way to use the FTP program vsftpd
which is need for phone configurations. So far I've been able to add a user
with useradd and add that user to the VSFTPD.USER_LIST and now I can FTP to
my [EMAIL PROTECTED] server but
Something is not quite right - your extensions.conf is specifying
Dial(SIP/455SIP/456, 15)
but the console is showing
Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10)
note the extra SIP/456 (as in SIP/456SIP/456) and the 10 instead of the
15 in the extensions.conf.
Are you sure
Hi J.
On Wed, 2005-06-22 at 19:22, Jerry wrote:
Hi Alessandro,
I think he means the daughter card color, not the LED on the card slot.
What color are the actual daughter cards?
You are correct! The actual daughter cards are green(FXS) and red(FXO).
Greetings!
Roland Welker wrote:
Hello,
Does anyone now, if there are any legal requirements for setups of
Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
interested, if a system does need to hang on a UPS?
Unless things have changed, you need either:
- 7 hours of UPS support
Two approaches come to mind -- 1) Using DBPut/DBGet to associate a fixed
amount of phone-numbers with a given extension and dial, all from
extensions.conf, or 2) Using a small mySQL table and a short agi script
to accomplish the same thing. The former solution has the advantage
that it's rather
On 6/14/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Hi
Has this been resolved?
Not as such but I noticed I get the error only when I run asterisk in
the foreground with the arguments -vvvc.
However I get no segfault error when asterisk restarts when running in
the background.
So in short,
[EMAIL PROTECTED] wrote:
We have 1 queue called helpdesk and are setting up a second one called isp.
The helpdesk queue is for internal support calls and isp for our ISP customer
calls. Both of these queues will be directed to the same agents (helpdesk
phone extensions).
We want to have
I need an AGI to monitor the quality of two connections and return a
yes/no based on packet loss, connectivy, provider being there, so I can
rollover the dial plan and dial the next available method. We have two
internet connections, two providers, and PSTN for backup.
My main concern is to
On Thu, 2005-06-23 at 21:43 +0800, Steve Underwood wrote:
Roland Welker wrote:
Hello,
Does anyone now, if there are any legal requirements for setups of
Digital (i.e. Linux/Asterisk) based PBXs in the UK? I am especially
interested, if a system does need to hang on a UPS?
Unless
Hi Chris.
You have been facing the same problem of mine. I was encouraged to use the
CVS HEAD version that includes an application called SIP_HEADER. With
SIP_HEADER we can handle SIP Headers fields.
If you get success on it, please let me know. I will do the same.
Regards,
Deniss Galvãao.
Hi,
does anyone know what Asterisk variable must be set to manipulate the
line under From:-line with a polycom 500 ip phone?
Thanks + regards,
Kib
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Hello,
I have got a new project about an asterisk instalation in a Morocco travel Company.
I thought to use a quadBRI card to connect it to the Morocco PSTN, but I do not know exactly witch kind of port signalling should I use in Morocco.
Another thing is that I do not know exactly how the ISDN
yes, you are right - the extension.conf wasn't the same as debug output
but it is solved anyway. There was just a missing registration for the
extension 456
Thanks
Asterisk wrote:
Something is not quite right - your extensions.conf is specifying
Dial(SIP/455SIP/456, 15)
but the console
Perhaps my deffinition of multi-threaded is skewed/wrong...
I've got asterisk HEAD running on a 4 proc machine.
I'm using top as my guide (and yes I know top sucks, but what else do I
use?).
I just watched asterisk hit 63% cpu usage for about 5 seconds. There
were 5/5 G729 licenses in use
Hi;
Does anybody knows what is the recommended
settings for Turkey
to configure dialtone of ata186 sip version 3.1.1
Thanks in advance
Betul
HASSAN GROUP
HIGHTEX 2005
Ist. Uluslararasi Teknik Tekstiller ve Nonwowen Fuari bünyesinde sizleri agirlamaktan gurur duyar.
13-16 Temmuz
Hello Steve,
Wich will be the version I will need to install to solve this problem??
Is this version already finished
Best Regards,
Jônatas Amorim
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My company is facing a similar situation. The agents/queue system in
Asterisk 1.0.x is badly designed to meet such needs. Temporary I am
working around the problem by giving each employee that answers a call
one AgentID. I then set them up as callback agents. They are then
members of both
That's what I'm confused about:
* two 4 port FXS cards
* one 24 port FXS channel bank
both, neither, and if both -- why do you need the dual digium cards?
shouldn't your channel bank just take MGCP or SIP or something?
What am I missing?
[EMAIL PROTECTED] said:
Shawn guessed correctly; Most
Here's something I haven't been able to discover as of yet - I need to
set up a direct link from my Asterisk box to an external line...
basically I need to be able to pick up an internal extension and have
it call a local phone number.
This is call forwarding, I know - the question that I have is
Rosario,
Unfortunately this problem doesn't just affect you, I'm also affected
and have been since 1.0.5. If you set the debugging high enough and use
mysql, you'll see the insert statements being generated by asterisk, but
they never make it to the DB.
I'm glad to know I'm not the only
This works for me:
to display the following on the polycom phone:
From: Support-Group
x
--- the caller id number
you can use the following code in extension.conf:
exten = 301, 1, Dial(SIP/456SIP/455SIP/457, 30)
exten = 301,
On Thu, 23 Jun 2005, Klaus-Peter Junghanns wrote:
Am Donnerstag, den 23.06.2005, 12:41 +0200 schrieb Armin Schindler:
I strongly disagree. :-) You dont want to do echo cancelation in
userspace. Especially not on a non-realtime operating system.
To make echo cancelation work it has to
I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried
looking on VOIP-info.org's ChanSpy page
(http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and
also referred to the link regarding bug 3836
(http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I
Leave what you have in place. Install Asterisk and investigate the
various line interface options. Enlist early adopters on your campus to
participate in the trial. Connect Asterisk to your existing system with
PRI. Gradually ramp up the Asterisk system and ramp down the existing
system.
Hi all,
How are things going ?
Is there a way for me to individually identify
each BRI channel on the Sirrix quad BRI board.
The reason I ask is because our client uses the
"Asterisk Flash Operator Panel" to monitor its external lines and transfer calls
from the lines to the various SIP
I just want to be sure that it's possible to do this, and that im not wasting
my time.
No time wasting. This is a great fit for you. Stay away from some of
the Dell servers until you know that they work well. (interupt and
ACPI issues)
--
Thanks for the advice on staying away from Dell
Hi List,
I've managed to install this great sript and it's working fine.
I am using this in the US, just want to know if this is possible and if so,
how:
1- Remove the '!' before the name when the calling number is a Cell phone
2- Remove the '1' before the number. I'd like the number to appear
[EMAIL PROTECTED] 06/22/05 12:55PM
I do understand that I would need to replace all of
my existing telephones with VoIP-capable
phones, and that I'll need to re-wire most of the
campus telephone infrastructure (it's still
all cat-3) -- these arent problems.
why do you think that
THANK YOU NANCY DREW!!! Could be a bit more vague about this eh?
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jun 22, 2005, at 6:30 PM, trixter http://www.0xdecafbad.com wrote:
http://www.frsirt.com/english/advisories/2005/0851
Just use CVS-HEAD.. stable is a pile of crap. let the flames being
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jun 23, 2005, at 10:09 AM, Tim Karl wrote:
I am trying to find the app ChanSpy for Asterisk v1.0.7. I have
tried
Take a look at the unicall.conf file, in the line with the
protocolvariant=br,XX,YY be sure you're using the right amount of digits...
XX should be the length of the ANI you're receiving and YY should be the
length of the DNIS...
if it doesn't work please try a debug of the unicall (in
Hi Emanuele,
you are right. I installed the CVS HEAD now (tried out asterisk -V) but sipredirect is unknown.
Do you have any hint for me, where I can have a look in which version it will be included?
Kind regards,
AxelEmanuele Pucciarelli [EMAIL PROTECTED] schrieb:
Axel Schemberg wrote: I use
If yes, then I have to disagree here. Something like 'playing' with
audio-data is nothing the kernel should be concerned with.
This belongs in user-space and if you need realtime, then you should use
a
realtime OS or use RT-scheduling. Just putting such a code into
Why wait? And why use agi? and why in the hell use parking?
Call comes in without callerid:
; call gets answered
exten = s/,1,Answer
exten = s/,2,Set(SCREENFILE=/tmp/screen-${CALLERIDNUM})
; ask the callers name and records it
exten = s/,3,Playback(screen-record)
exten =
On Thu, 2005-06-23 at 00:46 +0200, satchid wrote:
Dear List members,
I have an asterisk box whereon 45 GXP-2000 telephones from Grandstream are
connected at my work. This works fine.
Now I want to take 5 GXP-2000s to different homes on internet and want them
to be part of the same internal
What's the best way to get 1.0.8? I've downloaded the latest from CVS but when
I compile it it says 1.0.6!! Is that right?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: 23 June 2005 16:45
To: Asterisk Users Mailing
Haha, fun.
Why use the bufferoverflow if you already have the permissions to
execute any linux command using the manager interface :p
Brian West wrote:
THANK YOU NANCY DREW!!! Could be a bit more vague about this eh?
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan:
You're trying way too hard on this one.
exten = 555,1,Dial(Zap/g1/18005551212)
its no different than anything else in the PBX just set it up.. no
need to forward it.
replace 555 with the internal extension you wished to
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan:
On Thu, 2005-06-23 at 08:54 +0200, [EMAIL PROTECTED] wrote:
Hi,
It seems that my flash panel only works when I specify my ip address
and not the host name.
I've tried quite a few things (change host file, dns resolve,
proxying….) but couldn’t get it to work.
Anyone knows how to
That is not a typo. One is the loader, the other is the firmware.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Geoff
Manning
Sent: Thursday, June 23, 2005 6:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
I was reading up on DUNDI and although it sounds like it would solve
some of my problems, I don't know if it will do everything I want to
do. For example, among the things I wanted to do is something like:
the call director will have access to a database that will tell it
which server a
Hi Zhu!
Please check if you assign ratecard(s) to the tariffgroup!
Go on list Tariffgroup, click edit on the one u re using,
then check if one of the ratecard, at least, has been added.
Schusss,
Areskaille la canaille
On 6/21/05, zhu [EMAIL PROTECTED] wrote:
hello, guys:
my areskicc can
I've seen the embedded posts.
Is anyone running Asterisk on the MINI ITX?
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
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I had the same issue with a Sipura 2002 and the Firefly
softphone. The Sipura had another code applied to *77. I
was able to change default setting for *77 via the web interface of the
sipura to *777 and after that I was able to get into the digital
receptionist. As for the Firefly softphone I
Then you sip.conf is not defined properyly, or your extensions.conf does
not have any end, the only way the call should really go back is if
asterisk is telling it to, see what is called in what order in your
dialplan, remeber the dialplan is not called in the order it is written
in the file
On Thu, Jun 23, 2005 at 12:20:34AM -0600, Kevin Blackham wrote:
Does anyone have a MAX/APX with working ingress PRI calling name?
I recently acquired a MAX TNT on the cheap and it's integrating fine
except for one thing. In the 11.0.0 release notes, it is stated that
ISDN calling name will,
I have a second-hand 7960 which I am attempting to upgrade to
use a SIP
image.
The phone currently has a firmware release which doesn't seem
to be listed
in Cisco docs - P003AM30. On reboot, it finds the tftp server
Here's how I performed the upgrade:
Downgrade from the
On Thu, Jun 23, 2005 at 11:39:21AM -0500, jltaylor wrote:
I've seen the embedded posts.
Is anyone running Asterisk on the MINI ITX?
Yes, no problems, I have an X100P in the PCI slot, but its only
a single POTS line. I used the MII board, but only because thats
what I had avaliable.
Iain
Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:
brands
+--+--+--+--+--++--+--+
| name | language | inc | publishednum | did | markup |
jltaylor wrote:
I've seen the embedded posts.
Is anyone running Asterisk on the MINI ITX?
Not directly related, but I got OpenBSD to boot on a CF card , on my
Soekris this weekend.
Soekris is also selling units with the sangoma card as a daughterboard,
might be a cheaper/quiter
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