I have TDM card with FXS on port 1, and the other ports are FXO's
After isntalling the driver I execute the following commands:
- modprobe wctdm
- ztcfg -v
ztcfg relpy:
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Michiel van Baak wrote:
I thought the wcfxs module is used for fxo cards? Anyhow, if I load the
wcfxo module, then I get errors with ztcfg (below).
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
Normally, if I load the wcfxs module and the zaptel module, then this is
what I
On 5/16/06, Edu [EMAIL PROTECTED] wrote:
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb
RAM. It was working 24/7 without any for a month, but for not related causes I
Just for your info, I have experienced the same issue (just once) on a
Dell PE 2850 also. Same
Hi All,
We experienced this issue some time ago on our 2850.
Question - Are you using queues ?
Callum
stoffell wrote:
On 5/16/06, Edu [EMAIL PROTECTED] wrote:
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850
with 4Gb
RAM. It was working 24/7 without any for a month, but
Or more specifically, are you using the AgentCallBackLogin function ?
Callum McGillivray wrote:
Hi All,
We experienced this issue some time ago on our 2850.
Question - Are you using queues ?
Callum
stoffell wrote:
On 5/16/06, Edu [EMAIL PROTECTED] wrote:
We have an Asterisk Bussines
Michiel van Baak wrote:
If you load the wcfxs module and everything works (cept for
the asterisk answering the phoneline) all is correct.
wcfxs is for connecting an analog phone, not a PSTN
connection. I think you have the wrong module on you
wildcard to interface with the PSTN net.
Sorry.
Rodney G. McDuff wrote:
Hi All
Before I go out and buy a DELL PowerEdge 2850 has anyone had
problems (or any other useful experience) getting a TE411P to work with
it. I also have a legacy TE110P. Has anyone had problems with this combo.
I can not talk about Digium, but 2850 with Sangoma
On Tue, 16 May 2006, Hughes, Sam wrote:
The context can be set when the agent(s) log in.
AgentCallbackLogin([AgentNo|][Options|[EMAIL PROTECTED])
Many thanks, I knew I must be missing something simple.
___
--Bandwidth and Colocation provided by
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Now I'm playing around with WaitExten , but I would prefer collecting digits
in the channel.
Why in the channel driver? The channel driver just provides the possibility
for Asterisk to 'speak' CAPI (or any other API). The logic
We have a 2850 running with a Sangoma A102 , 50 agents using
Agentcallbacklogin and around 4000 calls per day. No problems at all.
With a te410 / 405 (we've got both, can't remember which one was in the
dell) we had lockups almost every day.
Julian.
Klaus Darilion wrote:
Rodney G. McDuff
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi!
I have problems with the ToN configurations in chan_capi-cm. I understand how
incoming calls are rewritten using national and international prefix. But for
outgoing calls - what is the ToN?
I never really needed ton in TE
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Does someone have any hints?
I would need a 'capi debug' log to say more, but chan-capi receives
the command for dial these digits twice.
Just a wild guess, but do you have softdtmf detection activated in
addition?
Yes -
First, you can remove the quotes aorund your variable
reference. I've seen examples with it, but you don't need
it.
I'm not sure: if variable is empty, you got an error.
In addition, double quotes around text that may contain spaces
will force the surrounded text to be evaluated as a
Hi all,
This is my first post! I'm newbie, yesterday I installed [EMAIL PROTECTED], and Igot lot of Kernel panics, after trying to reinstall it, this messages dissapeared. My problem now is that the default password for maint user in AMP is not working... I got this error message when I try to
2006/5/16, Rich Adamson [EMAIL PROTECTED]:
So I was telling myself : what if I could buy the most inclusive fax-modem, connect it to a PC, and run a bunch of test scripts to gather useful information on both production and preparation systems ?.
Total waste of money as the problem isn't the fax
Hi all,
i am playing around with several optipoint4x0 and run
into trouble trying to get hint functionality to work.
I notice that there is no status notifications. But
afaik this should be implemented via the
SUBSCRIBE/NOTIFY mechanism.
I can see INVITE, TRYING, RINGING, ACK, BYE but no
On 17/05/2006, at 7:36 PM, richard Coco wrote:
[local]
exten = 2001,1,Dial(SIP/2001,10,tr)
exten = 2002,1,Dial(SIP/2002,10,tr)
[notify]
exten = 2001,hint,SIP/2001
exten = 2002,hint,SIP/2002
Try this:
[local]
exten = 2001,1,Dial(SIP/2001,10,tr)
exten = 2001,hint,SIP/2001
exten =
Hi everybody,
I don't know how to do this,
I redirect a call and then dial someone, but I dont want the ringing
tone to be listened while the dialing part is waiting for the dialed
part to pick up the phone.
exten = 555,2,Dial(${STRING4},30)
I have tried when the option 'm' , but I don´t want
I am trying to register my Asterisk server to a SIP server which
doesn't accept an Expires: field smaller than 1800 seconds and
indicates it correctly with a Min-Expires: in an error response when
Asterisk tries to use its default of 120 seconds.
Is Asterisk supposed to honor this field and retry
On Wednesday 17 May 2006 08:48, Florian Overkamp wrote:
Michiel van Baak wrote:
If you load the wcfxs module and everything works (cept for
the asterisk answering the phoneline) all is correct.
wcfxs is for connecting an analog phone, not a PSTN
connection. I think you have the wrong
On 05/17/06 04:00 Noah Miller said the following:
only one registration. You can register from multiple devices, but
only the one that has most recently registered will receive calls.
Put another way, when the second device registers it will unregister
the first device.
exactly as you've put
Hi,
first of all, sorry for this long thread... I have
changed my extensions.conf like you suggested and
delete the line with subscribecontext=notify. But
unfortunately i still don't see subscribe request in
the sip debug trace.
SIP Debugging enabled
kingcoco*CLI
-- SIP read from
Im trying to compile Zaptel driver with the
ztdummy. I have no hardware cards from digium.
I tried following steps:
http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation
http://www.voip-info.org/wiki/index.php?page=Asterisk+timer+ztdummy
Im running : Linux version
Hi,
I'm actually using a slightly old version of AAH with Asterisk 1.2.1,
because at first install it was perfect for my moderate knowledge of
Asterisk. It is working well but I gradually introduced many changes to
dialplan during normal use and now I'm feeling like in a straitjacket!
Moreover I'd
Hi,
Asterisk version : 1.2.7.1 stable version
We try agent.conf setting of
createlink=yes
We always can not see this link value to be filled in MySQL's
table filed : userfield
But we can see the record file has been created correctly.
In debug mode, no userfiled shown in SQL command,
On Wed, 17 May 2006, Klaus Darilion wrote:
Armin Schindler wrote:
On Tue, 16 May 2006, Klaus Darilion wrote:
Hi!
I have problems with the ToN configurations in chan_capi-cm. I
understand how
incoming calls are rewritten using national and international prefix.
But for
Hi,
I suspect that either the permissions are wrong for /main or there are
no files in it and directory listings are denied.
It sounds like an incomplete install to me.
try and ssh onto it and do an ls -lh in /main and see if there are any
files in there
Alasdair
Laura Barquín wrote:
Hi
On 17/05/2006, at 8:27 PM, richard Coco wrote:
unfortunately i still don't see subscribe request in
the sip debug trace.
Have you configured your phone to subscribe to the extension? :)
cYa,
Avi
___
--Bandwidth and Colocation provided by
On Wed, May 17, 2006 at 08:29:14AM +0200, Terry Wade wrote:
Why not just run the genzaptelconf file, or
is that specific to AAH?
genzaptelconf is not specific to AAH. Not originally from there,
actually. A recent version of it could be found in latest debian
packages and Xorcom Rapid
On 12:30, Wed 17 May 06, Mimmus wrote:
Hi,
I'm actually using a slightly old version of AAH with Asterisk 1.2.1,
because at first install it was perfect for my moderate knowledge of
Asterisk. It is working well but I gradually introduced many changes to
dialplan during normal use and now I'm
Hello!
Unfortunately we are seeing lately (2-3 times during a day) that
asterisk seems to hang up somehow - no new calls can be made and sip
show peers and other commands show no obvious problem. We then
recompiled 1.2.7.1 with all the DEBUG_ turned on in the makefile and
now we see the
Sometimes (once per 4000 lines or so - depending on speed of network) manager
improperly returns events. For example, one QueueMember will get overwritten by
(or as part of) another, like this (see third line):
Event: QueueMember
Queue: 09
LocatiEvent: QueueMember
Queue: 09
Location:
The correct module to load for a TDM Cards interface is wctdm
see http://www.digium.com/en/docs/misc/quick_install_zaptel_asterisk.pdf
Regards,
Moutaz
-- Original Message ---
From: Pieter Claassen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Ok,
I got rtptimeout setup to 60 in sip.conf and it go better ; came back to
normal
as soon as I put it.
If anybody knows if rtpkeepalive and rtptimeout can work in conjunction,
please share your toughts !
Thanks,
Fred
- Original Message -
From: Frederic Jean
To: Asterisk Users
Hi again,
what do you mean exactely with Have you configured
your phone to subscribe to the extension? :).
I have several optipoint410 and eyebeam. On one of the
Optipoint(exten 2001) i have configured a selected
dialing bottum with the extensions of the
eyebeam(exten 2002). Do i need more
Callum McGillivray schrieb:
We experienced this issue some time ago on our 2850.
Question - Are you using queues ?
In my case we did not use queues and we had the problems with a custom
made machine with a Intel Torrey Pines Mainboard.
Chris
___
I've been having problems with my A20002D lately - callers from the PSTN
don't hear me when I answer, but I hear them. Disabling echo
cancellation in zapata.conf brings the audio (and echo) back. This used
to work fine, until two days ago.
Well, just to complete my own thread, this seems to
So I was telling myself : what if I could buy the most inclusive
fax-modem, connect it to a PC, and run a bunch of test scripts to
gather
useful information on both production and preparation systems ?.
Total waste of money as the problem isn't the fax modem as noted
Hello all.
This is my very first message to the list. I have a TDM400P card, It
has 2 FXO channels which are connected to extensions of my PBX
(Ericsson BP250), so I can dial from any SIP softphone directly to
physical (analog and digital) extensions on my company.
My PBX is configured
Hello Julian!
As we are using the same HW (2850 and Sangoma cards) but have some problems
with AMI (* manager interface) I wonder which OS (version of Linux, kernel
version) are you using?
Thanks!
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
Hi,
Following last thread onunifying blind and
attendedtransfers (http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/146002/focus=146683)
I think it would be great if a user could either
:
1. transform a transfer into into a
blind-and-forget transfer one pressing # key
What problems are you having ?
We are using CentOS 4.2 Linux 2.6.9-22.0.1.ELsmp
Asterisk SVN-trunk-r7353M
I know, time to update, but I am not in the office currently and do
*not* want to do it remotely ;)
Julian
Asterisk wrote:
Hello Julian!
As we are using the same HW (2850 and Sangoma
Title: Running commands from dialplans
Hi,
Can anybody tell me which version of
res_perl I have to install on Asterisk 1.2.4.
I tried to compile res_perl version 3.5 on
Asterisk 1.2.4 and I got the following error.
gcc -Wall
-DRES_PERL_BASE=\/usr/local/res_perl\
-DMULTIPLICITY -
Going to AMP, Setup - General - Extension of fax machine for receiving
faxes = disabled *should* disable fax detection by causing it to use a
different branch of the AMP macro's...
I did set it to disabled, but it still called NVFaxDetect() with a
parameter of zero.
We have problems with 'loosing' parts of messages sent from Asterisk Manager to
our CTI Server (we also tested it with another test program, but the problem
persisted). Sometimes (once per 4000 lines or so - depending on speed of
network) manager improperly returns events. For example, one
Giorgio Incantalupo wrote:
I cannot make my TDM2400P (with Echo Canceller module) detect faxes. I
tried with a TDM400P and it worked at 80% (20% of faxes were lost). My
test conf.files are:
This is a known problem with the hardware echo canceler. For the time
being, load the wctdm24xxp module
This is an old version of trunk though - be careful :)
Julian
Asterisk wrote:
We have problems with 'loosing' parts of messages sent from Asterisk Manager to
our CTI Server (we also tested it with another test program, but the problem
persisted). Sometimes (once per 4000 lines or so -
Steve Davies wrote:
In the cases previously mentioned, the user is doing an attended
transfer using the handset features, and not Asterisk. I do not know
whether SIP even allows the Caller ID to be changed at the point when
two separate calls are bridged to one...
It does, but Asterisk does
Mimmus wrote:
Hi,
I'm actually using a slightly old version of AAH with Asterisk 1.2.1,
because at first install it was perfect for my moderate knowledge of
Asterisk. It is working well but I gradually introduced many changes to
dialplan during normal use and now I'm feeling like in a
Hi,
Are there any good free win32 apps for reading queue_logs?
Regards,
Jan
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Cory
The 480i-CT does not state DECT to my knowlege as the EU DECT standard
uses reseved frequency space in the US. I have heard rumblings about
a US DECT standard, would this be the DECT you are refering to and if
so could you provide a link to information on compatablity.
Andrew
On
Hi!
I have tried when the option 'm' , but I don?t want the default music on
hold to be listened neither. I want nothing (silence) to be heard instead of
ring, ring.
Any idea how to do this???
The answer is in your question: Create a MOH music file with... silence
in it. Then make a new MOH
I have a cisco VPN from router to router over a Data T-1.
The ping times are consistently 32ms with random ping responses of 295ms -408ms
about every 30 secs to a minute, I have jitter buffer enabled. The connection
goes like this Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B
I apologize about doubling these up, I forgot the subject!
I have a cisco VPN from router to router over a Data T-1.
The ping times are consistently 32ms with random ping responses of 295ms -408ms
about every 30 secs to a minute, I have jitter buffer enabled. The connection
goes like
Wes Santee wrote:
The first problem is obviously that the curly braces used in regex patterns
to
denote repeating patterns means something different to Asterisk. I would
expect
back-slashing to fix this. So...
This was just recently fixed in SVN branch 1.2, and the fix will be part
of
I have a cisco VPN from router to router over a Data T-1. The ping times
are consistently 32ms with random ping responses of 295ms -408ms about
every 30 secs to a minute, I have jitter buffer enabled. The connection
goes like this… Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B
and
Samuel Tardieu wrote:
Is Asterisk supposed to honor this field and retry with the proposed
minimum Expires: field? It looks like it doesn't, and I had to change
the default_expirey globally.
Yes, it should. Please open a bug report on bugs.digium.com with a 'sip
debug' trace of this
I currently have a single server with a few SIP and
IAX upstreams for origination and termination with IAX clients. I am
adding a second server that will have a much higher capacity and will be
handling a larger call volume. However, this second server is not going to
be geographically near
Hi!
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with 4Gb
RAM. It was working 24/7 without any for a month, but for not related causes I
rebooted it a week ago. Yesterday the machine suddenly stop working, with a
kernel panic. We was watching logs, and found in
Well, looks like we had a similar issue. Replaced the Sangoma and it worked.
We have asked for a failure analysis from Sangoma on the defective card.
Dr. Michael J. Chudobiak wrote:
I've been having problems with my A20002D lately - callers from the
PSTN don't hear me when I answer, but I hear
Ok, I going to do it, thanks
El Martes, 16 de Mayo de 2006 17:06, Joshua Colp escribió:
Edu wrote:
Hi!
We have an Asterisk Bussines Edition ABE-A.1-6 on a PowerEdge 2850 with
4Gb RAM. It was working 24/7 without any for a month, but for not related
causes I rebooted it a week ago.
For your configuration to be like this
RRDNS and Realtime. I believe someone made a patch for realtime to work
correctly with RRDNS you would
have to check the wiki or mantis to find
it.
_.._
Brian Fertig - dCAP, MSCE, CCNA,
Andrew - I am not sure of the exact spectrum frequency which constitutes
DECT in the US, but several major manufacturers are developing and selling
DECT devices for the US market.
Plantronics recently release a DECT 6.0 wireless headset, the CS55, which
you can see here
I know for a fact that the Aastra 480i-CT is not available in the UK/Europe at the moment. There is no program in place to get in over into Europe however I think it could happen in the next 4+ months.Does anyone know if the UNIDEN UIP1868 is available in the UK? If so how do I get my hands on one
Not sure about overseas distribution for
the Aastra 480i-CT, I am not aware of any VARs or distys across the
pond. The UIP1868 has actually had production discontinued at Uniden. They
are continuing to manufacture a Vonage Locked version of the
unit, but have stopped producing the unlocked
I think around Q3/Q4 of this year, you'll see some very interesting new
products which incorporate DECT for wireless. For consumer products with
limited mobility, it seems to make a bit more sense than WIFI.
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
According to all of my sources, the UIP1868 has been
discontinued. Kind of a shame, it was a neat product. -Mike
Michael Crown Managing Partner
www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED]
From: Colin MacMillan
[mailto:[EMAIL PROTECTED] Sent: Wednesday,
Hi group,
i'm brand new and i would like to ask about soekris hardware. I read
along the web but i have some doubts that i think can be solved here.
My question are the following:
1) does the Digium TDM400P fit in the soekris box with a 4801 SBC or a
bigger box is needed? Any suggestions about
Hi!
I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not
accept the 200 OK responses. E.g in the following example, Asterisk
retransmits the CANCEL although the 200 OK is received.
There is no log message, why this packet is not accepted/processed. Is
there a ways to
I'm wondering if anyone has a solution to this before I begin looking
at making some changes to the SIP channel. Basically when calling
SIPAddHeader() twice from the Dialplan or an AGI script with the same
header name it adds duplicate headers instead of overwriting the
existing one.
Hi,
I have a problem with fax in asterisk version 1.2.7. The application rxfax
doesn't work triggering these errors :
channel.c: Dropping incompatible voice frame on ... of format slin since our
native format has changed to ulaw
chan_sip.c: sip_write: Asked to transmit frame type 64, while
Hi,
is it possible to let asterisk issue a SIP redirect?
A SIP invite command by a SIP client should be answered
by 30X Temporarly moved to SIP/
Is this possible with asterisk, maybe from within the dialplan?
(reinvite is not what I'm looking for, because it does not
completely release
Klaus Darilion wrote:
I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not
accept the 200 OK responses. E.g in the following example, Asterisk
retransmits the CANCEL although the 200 OK is received.
SVN trunk is not Asterisk 1.2.
There is no way to help you with this
Roger Schreiter wrote:
is it possible to let asterisk issue a SIP redirect?
A SIP invite command by a SIP client should be answered
by 30X Temporarly moved to SIP/
Have you read any documentation on the applications available in
Asterisk, or on the voip-info wiki? The Transfer()
, Inc. and Telecenter Inc.
Use of this information by anyone other than the recipient or
sender will be considered in breach of agreement.
-- next part --
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060517/d9af5983
What do you mean exactly?
Do you want to flash Asterisk on this Wireless Router?
If so, I would like to hear more such as what kind of end-user
applications would you like to run on it?
Cheers
Fifou
Frank Tarczynski a écrit :
I'm looking for a recent asterisk package for the Linksys WRT54G.
Google and voip-info.org will have answers to all of your questions.On 5/17/06, Jonathan Gonzalez
[EMAIL PROTECTED] wrote:Hi group,i'm brand new and i would like to ask about soekris hardware. I read
along the web but i have some doubts that i think can be solved here.My question are the
more kindly :
http://www.astlinux.org/
Olivier
Christopher Snell a crit:
Google and voip-info.org
will have answers to all of your questions.
On 5/17/06, Jonathan Gonzalez
[EMAIL PROTECTED] wrote:
Hi
group,
i'm brand new and i would like to ask about soekris hardware. I read
Hi all,
I am running an Asterisk server behind a NAT.
I want to forward the calls from PSTN to a SIP phone (no nat and also an
asterisk).
I set the externip and localnet in sip.conf already. I opened the ports
in my firewall. (I changed SIP port from 5060 to 5065 and limited the
rtp port to
Check out http://www.openwrt.org
I run their prebuilt asterisk-1.0.7 package on a Linksys WRT54GL and it
works like a charm, just don't expect too much performance out of it :)
-Evan
Phi Fou wrote:
What do you mean exactly?
Do you want to flash Asterisk on this Wireless Router?
If so, I
On 5/17/06, Mimmus [EMAIL PROTECTED] wrote:
I was thinking to this plan:
- install another server with Red Hat 4 U3
- install PHP, MySQL and other usefuls stuffs
- download latest version of Asterisk and third parts applications I use
- compile all
- copy /etc/asterisk from old server to new,
It also looks like they've got a 1.2 package for openwrt now:
https://dev.openwrt.org/browser/trunk/openwrt/package/asterisk/Makefile
-Evan
Evan Borgström wrote:
Check out http://www.openwrt.org
I run their prebuilt asterisk-1.0.7 package on a Linksys WRT54GL and it
works like a
I wouldn't knock the third party friendly interfaces to Asterisk too hard. They will evolve and improve over time. The adoption of Asterisk as a mainstream PBX is dependent upon a user friendly interface.
On 5/17/06, Strom Carlson [EMAIL PROTECTED] wrote:
On 5/17/06, Mimmus [EMAIL PROTECTED]
What I did with AMP was take the best parts of it and copy/paste to a clean
extensions.conf, then add my modifications onto it. Worked for me.
-Original Message-
From: Strom Carlson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 17, 2006 10:21 AM
To: Asterisk Users Mailing List -
Does anyone know if/how I can get Asterisk to listen for SIP/RTP/IAX/whatever
on multiple network interfaces?
Thanks,
Doug
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Kevin P. Fleming wrote:
Wes Santee wrote:
This was just recently fixed in SVN branch 1.2, and the fix will be part
of the 1.2.8 release that will appear later this week.
In the meantime, you can work around the problem by storing the regex
string itself in a variable, then using a
Hi All:
Assume that FreePBX can configure a remote MySQL server, is any way
that I could use FreePBX to configure second Asterisk by using the same
dialplan deployed for configuring the first Asterisk server? If answer
is no, could anybody help me some ideas to resolve this?
Thanks,
Tielin
Hello,
anyone is using the Asus P5GD1-Pro with Asterisk ? For a customer I need
to use this mainboard for 2 PCI Boards (1 BNS8 Beronet 8xRDSI and 1
TDM2400 ).
Any reference for bad or good mainboard for asterisk ? This is a Intel
915P Chipset.
Greetings to all.
Juan Carlos Valero.
just have to say WOW
I got a new voip wifi handset.
Not yet on the market, but the constructor promised me international
versions if I can have a 50+ order.
Well, the constructor is a very well known handset provider in the
Pstn/Isdn world and Skype world :( ...
It's a german one...
Kevin P. Fleming wrote:
Klaus Darilion wrote:
I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not
accept the 200 OK responses. E.g in the following example, Asterisk
retransmits the CANCEL although the 200 OK is received.
SVN trunk is not Asterisk 1.2.
Of course -
Hello!
Does anyone know how I can use two diferent databases with ODBC with the
same erealtime family?
Something like this:
res_odbc.conf:
;;; odbc setup file
[ast_cnf1]
dsn = ORACLE
username = asterisk
password = asterisk
pre-connect = yes
[ast_cnf2]
dsn = MySQL
username = asterisk
password =
Hello Everyone,
I just picked up 10 new 7960G. Im trying to load 8.2 on them
but getting a weird error during the tftp transaction when reading OS79XX.TXT
Here is what im getting.
Ben Blakely, CISSP
Security Architect
Blink Communications
div. of Oakville
Hydro
905-825-6369
I have IAX on a server with dual nic in 2 subnets, I can connect to the IAX channel on both subnets, I do not have a bind setting in my IAX.conf file. I don't know if that will work for SIP/RTP, but I would think so.
On 5/17/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Does anyone know if/how I
Hello All,
I just picked up 10 new 7960Gs and having a hardtime
upgrading the firmware on them. We already have 30 or 40 of these phones in
production. Typically when I get a new phone, I just plug it into the voice
vlan and it auto grades to the firmware in OS79XX.txt. Now whats
Just curious...
Does anyone know if any companies using Asterisk on embedded hardware (out at
the customer premisis), such as the Soekris Net4801, to provide VOIP service?
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
Just turned up a system with an A8N/Athlon 64 on Monday, 40 extensions, 6
IAX locations w/ TE-110P and TDM400. Works perfect, no echo, ZTTEST runs
steady at 99.9873%. Had to disable everything, onboard LAN, RAID, USB,
serial, etc. Replaced the onboard LAN with an Intel 82557 based card,
because
On 5/17/06, David K Parker [EMAIL PROTECTED] wrote:
I wouldn't knock the third party friendly interfaces to Asterisk too hard.
They will evolve and improve over time. The adoption of Asterisk as a
mainstream PBX is dependent upon a user friendly interface.
Well, as soon as a GUI shows up that
Douglas Garstang wrote:
Does anyone know if/how I can get Asterisk to listen for SIP/RTP/IAX/whatever
on multiple network interfaces?
Yes, I have a system setup to listen to both a 10.10.10.x network and
192.168.102.0 network. I don't specify a bind address. The phone on
site are on
Hello:
I have tested the Zyxel wifi phone with locustworld AP's. The roaming between the AP's is seemless but works.
MohamoudOn 5/17/06, olivier.taylor [EMAIL PROTECTED] wrote:
just have to say WOW
I got a new voip wifi handset.
Not yet on the market, but the constructor promised me
I have this exact same issue with the SPA3000, I'm assuming it must be a
SPA3000 bug?
Dave Hawkes
Alchaemist wrote:
Hi,
These days I run into something quite odd.
I have an [EMAIL PROTECTED] that was modified to meet our requirements.
We have a completely funtional
1 - 100 of 174 matches
Mail list logo
