Greetings all,
I have a problem with a PBX that I manage. The system has 2 AVM Fritz boards
connected to two BRI ISDN services using chan_capi in addition to several SIP
trunks going out to Internet based providers for call termination via the
Internet.
They experience problems when the
lee,Thanks for the feedback.in most diagrams explaining t38, it shows, the sending fax machine connecting to a pots before connecting to a gateway,then the internet. but if i've read and understood correctly, the sending end can use an ATA with t38 support instead of a pots. in that case,
Hi,
use AOC. See here:
http://www.snom.com/wiki/index.php/FAQs#Q:_How_to_show_billing_information_on_the_phone_display.3F
Regards,
Sven
On Friday 22 September 2006 17:31, Ale wrote:
Hi all,
Can anyone help me... i need to display the cost of a call during a
conversation on a sip or iax
linksys spa3102 or 2100 are known to work.
Grandstream also should do it with recent firmware.
Don't be fooled by what is written on the box, lot of ata's out there
claim t.38. (while the firmware doesnt contain anything related to t.38)
Zoa
Christopher Corn wrote:
lee,
Thanks for the
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I have a few dual core that I have installed Asterisk on without any issues.
Hi Bill!
Sure you don't have any issues, but do you take any advantage of dual core
processor? Why would I pay for something if I can't profit from it?
--
Asterisk is very happy on dual core. It greatly reduces load. We just
put a Pentium-D in poduction last week and it is working verry well.
We have a Core 2 Duo on order that we should be putting in production
next week.
MATT---
Hi Matt!
Thank you for this information. Can you please
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
My home Asterisk server is running dual proc dual core zeon 3ghz, seems
happy, no crashes that I didn't bring about myself. ;)
mpg123 does occasionally hang a pid at 100% now and then, but it does that
on single proc/single core
On Sat, Sep 23, 2006 at 09:22:50PM +0200, Morten Isaksen wrote:
Hi!
I was trying to upgrade my Asterisk 1.2.1 at home (not the [EMAIL PROTECTED]
dist)
to 1.4b2 but ran into problems with zaptel. The OS is Fedora Core 3.
When I start zaptel it fails with this error:
[EMAIL PROTECTED]
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm sure other people are using 7960 phones so maybe someone could have
a quick look at what time sip show peers reports? When I do a 'sip show
peers' all my cisco 7960 phones report times 150ms. Every single one.
I've scoured the
Hi,
I've recently got a Snom 300 phone.
When I set it up and a VM was left the light flashed as excpected.
I can use the soft button with the Tick sign on it to go straight to
voicemail, all fine so far.
However once the message is picked up and listened to, the light still flashes?
I'm using
On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm sure other people are using 7960 phones so maybe someone could have
a quick look at what time sip show peers reports? When I do a 'sip show
peers' all my cisco 7960 phones report
Hi to all.
I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and
I've some problem with outgoing calls: there is a big delay for
bidirectional audio flow.
By traces, I've observed that several 200 OK SIP messages are sent by
my SIP Provider until ACK is riceved.
Maybe the 200 OK
I can confirm the same.
It doesnt mean the audio will be delayed, the phone is just slow with
replying to the sip messages.
Zoa
Michiel van Baak wrote:
On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm sure other people
Michiel van Baak wrote:
On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm sure other people are using 7960 phones so maybe someone could have
a quick look at what time sip show peers reports? When I do a 'sip show
peers' all my
Hi,
On Centos IRQBalance should already be available.
You should be able to run 'setup' from a console/terminal, go to System
Services enable irqbalance. It will then be enabled on boot.
To start it without re-booting, use
service irqbalance start
If it's already marked as enabled in the
I'm not sure irqbalance is a good idea. (although i'm not familiar with
it, its sounds like it balances it all the time, not just spreads it and
leaves it).
Maybe its best to do it manually ?.
have a look at something i wrote ages ago:
On 04:11, Mon 25 Sep 06, Brian Capouch wrote:
That is a classic (and, AFAIK innocuous) behavior of the original Cisco
ATA-186 ATAs as well.
Nobody was ever able to explain why they are that way, but it seems to
normal behavior.
It really is something in the SIP image of the 7960.
I have
hi,I try The asterisk1.4 beta2, it said Callback mode (AgentCallbackLogin) is now deprecated, and let us use dialplan logic.I checked the docs/queues-with-callback-members.txt
file, this example is complex, i can't understand it.I want use dialplan logic for AgentCallbackLogin only. Anybody have
Hi,
When I use Voicemail function, there is a default system greeting
before voicemail recording. Is it possible to change that greeting?
How?
unplug
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In sip.conf for one friend (Cisco 7970 phone) I have define this
dtmfmode=inband
And in xml.conf of that phone I have
preferredCodecnone/preferredCodec
dtmfAvtPayload101/dtmfAvtPayload
dtmfDbLevel3/dtmfDbLevel
dtmfOutofBandnone/dtmfOutofBand
But DTMF doesn't work for that phone.
Phone
On Fri, 22 Sep 2006 17:51:10 -0400
Juan Pablo Abuyeres [EMAIL PROTECTED] wrote:
Wow... I am looking for exactly the same feature. Did you find out how
to do it???
Not yet, but maybe our chances rise now as we are already two...
regards
christian
On Thu, 2006-09-21 at 18:41 +0200,
Hi Friends,I got a strange doubt and problem. Is there any problem for SIP protocol, if we configure Intercom (SIP), VoIP (SIP) and PSTN in a single server (that may be Asterisk or Trixbox).My First Experience:Initially I configured Asterisk in a system and created SIP extensions and VoIP with
On 9/25/06, Mike Dent [EMAIL PROTECTED] wrote:
Hi,
I've recently got a Snom 300 phone.
When I set it up and a VM was left the light flashed as excpected.
I can use the soft button with the Tick sign on it to go straight to
voicemail, all fine so far.
However once the message is picked up and
Dear Users
I setup asterisk in my home and i want to just using asterisk for outgoing call
from internet to my PSTN line
but when i connect fxo port to phone line asterisk pickup the line first , i
want to asterisk wait to somebody pickup the line if line not picked up after
proper time the
First hi
I am trying to make a call back system
I am using callme.php(click to call) to write the file at
/var/spool/asterisk/outgoing
And at the incoming context ,I match the
incoming did as follow
exten = 009613504768,4,system(elinks -dump
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm sure other people are using 7960 phones so maybe someone could have
a quick look at what time sip show peers reports? When I do a 'sip show
peers' all my cisco 7960 phones report times 150ms. Every single one.
Tomislav Parčina wrote:
In sip.conf for one friend (Cisco 7970 phone) I have define this
dtmfmode=inband
And in xml.conf of that phone I have
preferredCodecnone/preferredCodec
dtmfAvtPayload101/dtmfAvtPayload
dtmfDbLevel3/dtmfDbLevel
dtmfOutofBandnone/dtmfOutofBand
But DTMF doesn't work for
unplug wrote:
Hi,
When I use Voicemail function, there is a default system greeting
before voicemail recording. Is it possible to change that greeting?
How?
Call into voicemail as though you were going to listen to your messages,
and press 0 for Mailbox Options. Then press 3 to record your
Paging: You can also use the server's audio card for paging, if it is
close to the main amp. Beware of Bogen or Valcom as they mainly make
FXO paging interfaces for trunk lines. Viking makes the paging toys
you would want to look at.
ABE: ABE is the supported version of Asterisk, same code but
Lee Howard wrote:
On Sun, Sep 24, 2006 at 01:58:21PM -0700, Christopher Corn wrote:
A couple of faxing methods im confused about.
The pass through method, sending fax data over G711 codec
versus
Relay method, t30 to t38 conversion
Can someone explain to me why the pass through
Hi,
I am experiencing some similar difficulties with voicemail. I have an IP phone on extension 101 and I do not know how to dial in to access the voicemail options. When I dial 101 I have tried pressiong * and 0 but I do not get to the mail box menu. Can someone please help?
--
Hi guys,
I'm tryng to estabilish a trunk with an Alcatel 4200 Pabx. In asterisk i'm using a Digium TE110P as E1.
The span is ok with green led, but when pabx make calls to asterisk, i received this error:
asterisk*CLI
!! Unexpected Channel selection 3
-- Accepting call from '3069' to
You can put fastagi-mapping.properties in the root dir of the classes of
your project.
[]'s,
Edmilson Santana
Unitech Tecnologia de Informação (http://www.unitech.com.br/)
[EMAIL PROTECTED] wrote:
We try to work with asterisk-java and FastAGI (for our diploma).
We did everything like on
On Mon, 25 Sep 2006 [EMAIL PROTECTED] wrote:
Hi,
I am experiencing some similar difficulties with voicemail. I have an IP phone
on extension 101 and I do not know how to dial in to access the voicemail
options. When I dial 101 I have tried pressiong * and 0 but I do not get to the
mail box
Hi!
I have problems when bridging from SIP to PRI. As soon as the setup
message is sent, Asterisk replies with 183 to the sender.
Although there is nor PROGRESS message received, the 183 is sent as the
SIP channel received a voice frame and thus activates early media.
I wonder why Asterisk
I believe asterisk for the most part is single threaded, you will get some advantages by having other system processes use the extra Processor/Core, but I don't think asterisk will use alot of the other CPU.
On 9/25/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
Asterisk is very happy on dual
Hi, folks. I've got an X100P Wildcard here. I get an odd error when
running ZTCFG on it.
===
pbx1:~# asterisk -V
Asterisk SVN-branch-1.2-r43509
pbx1:~# lsmod
Module Size Used by
wcfxo 13184 0
zaptel
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
In asterisk sip.conf, use dtmfmode=rfc2833 for that extension, and in
the SIPDefault.cnf boot file for the cisco, include:
dtmf_inband: 1
dtmf_outofband: avt
dtmf_db_level: 3
(you'll need to translate the above 7960 parameters
For what we do with Asterisk(lots of meetme and Zap - IAX2) It does
spread the load across both cores. In our initial comparisons for
equal call traffic, the P4-D had half or the average loadavg for a 6
hour time period of the P4 of the same speed.
MATT---
On 9/25/06, Tomislav Parčina [EMAIL
On Mon, Sep 25, 2006 at 08:14:27PM +0800, Steve Underwood wrote:
Now introduce VoIP telephony... where a small amount of audio
corruption (jitter) is anticipated on the UDP channel... and mix it
with faxing and hopefully you can see how it just doesn't work well.
VoIP is packetized audio
On Sun, Sep 24, 2006 at 11:00:41PM -0500, Jason Parker wrote:
- Jay R. Ashworth [EMAIL PROTECTED] wrote:
I will assume that you are a native speaker; I'm not equipped to
evaluate whether ... well, anyway. Anyone know where those prompts
actually *came* from? :-)
The Spanish
Hi Steve,
On 9/14/06, Steve Davies [EMAIL PROTECTED] wrote:
On 9/14/06, Steve Underwood [EMAIL PROTECTED] wrote:
Steve Davies wrote:
[snip]
This looks pretty good I have to say - The ECM seems as if it may be a
little intolerant... On a fax machine where I got 100% success in the
past
On Mon, Sep 25, 2006 at 09:04:41AM +0200, Tomislav Par?ina wrote:
Sure you don't have any issues, but do you take any advantage of dual
core processor? Why would I pay for something if I can't profit from
it?
Well, it would seem to me that with a little attention to processor
affinity, you
On Mon, Sep 25, 2006 at 04:11:33AM -0400, Brian Capouch wrote:
I have the same here. All between 150 and 250 ms.
The phones do work perfectly, only the time in sip show
peers is higher then any other phone/device.
That is a classic (and, AFAIK innocuous) behavior of the original Cisco
On 9/25/06, Klaus Darilion [EMAIL PROTECTED] wrote:
Hi!
I have problems when bridging from SIP to PRI. As soon as the setup
message is sent, Asterisk replies with 183 to the sender.
Although there is nor PROGRESS message received, the 183 is sent as the
SIP channel received a voice frame and
Looks good, great price:
http://www.aastratelecom.com/ipphones/pro_243.asp
Anybody using these? How's the cordless? Does it play nice with * ?
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I have a asterisk box with some queues for a call center and need help on
two points:
1. I have a scenario where if a queue has no agents logged in, an inbound
call should immediately failover to the failover destination for that queue.
However, this does not seem to be working in that, even if
I would like to know if any of you have a cell phone like a pci card to install in one slot to my asterisk server?, I want to make a connection from my asterisk to the cellular network.Does anybody has a solution like this?Regards,Yrving
Do You Yahoo!?
La mejor conexión a Internet y 2GB
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm sure other people are using 7960 phones so maybe someone could have
a quick look at what time sip show peers reports? When I do a 'sip show
peers' all my cisco 7960 phones report times 150ms. Every single
SEÑORES DIGIUMLA PRESENTE ES PARA CONFIRMAR QUE REQUERIMIENTOS DE HARDWARE Y/O SOFTWARE SON NECESARIOS PARA CONECTAR UNA CENTRAL TELEFONICA PANASONIC TDA620 QUE TIENE INSTALADA UNA TARJETA SE5E18 (E1) CON UNA TARJETA TE110P.
ATENTAMENTEDIEGO FERNANDO GÜIZA ARCE
It's excellent home phone. I wouldn't use it in a business environment. No
hold, no one-touch voicemail. However, it works great!
/R
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Monday, September 25, 2006 10:25 AM
To:
I have this phone on my desk. It works very very well!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Monday, September 25, 2006 10:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Spam? [asterisk-users]
i have all files in the same directory: c:\agi
(asterisk-java-0.2.jar, fastagi-mapping.properties, HelloAgiScript.class and
HelloAgiScript.java). My slasspath is also c:\agi
Did you mean this?
But i get still the following errors:
if i start it with eclipse:
...
INFO: Received connection.
Hi Steve!
The problem is following
PSTN PSTN
| |
| |
E1 E1
| |
PBX1--E1--Asterisk1---SIP---Asterisk2--E1--PBX2
2 offices.
On 9/25/06, Michelle Dupuis [EMAIL PROTECTED] wrote:
I have a asterisk box with some queues for a call center and need help on
two points:
1. I have a scenario where if a queue has no agents logged in, an inbound
call should immediately failover to the failover destination for that queue.
- Original Message -
From: Steve Glaus [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 22, 2006 4:25 PM
Subject: [asterisk-users] Very high ping times from 7960 phones
I've asked this here before
On Mon, Sep 25, 2006 at 08:25:10AM -0600, Colin Anderson wrote:
Looks good, great price:
http://www.aastratelecom.com/ipphones/pro_243.asp
Anybody using these? How's the cordless? Does it play nice with * ?
Well, anyone who thinks that a 4-p,4-c modular jack *has* an RJ
designation makes
I don't have experience using the 480i CT, only using the 9112i, so you
should take what I say with a grain of salt.
I have been nothing but impressed with this phone. In terms of being
friendly with *, they dedicate a section of their manual to asterisk
configuration, which makes things go
On 9/25/06, yrving rivas [EMAIL PROTECTED] wrote:
I would like to know if any of you have a cell phone like a pci card to
install in one slot to my asterisk server?, I want to make a connection from
my asterisk to the cellular network.
Does anybody has a solution like this?
Regards,
Yrving
It's excellent home phone. I wouldn't use it in a business environment.
No
hold, no one-touch voicemail. However, it works great!
aw crap, that's a biggie but I think I can work around it, teach the user to
dial *98 for voicemail, *700 for park and hash to transfer, currently the
users dial
Colin Anderson wrote:
Looks good, great price:
http://www.aastratelecom.com/ipphones/pro_243.asp
Anybody using these? How's the cordless? Does it play nice with * ?
___
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asterisk-users
On 9/25/06, Colin Anderson [EMAIL PROTECTED] wrote:
It's excellent home phone. I wouldn't use it in a business environment.
No
hold, no one-touch voicemail. However, it works great!
aw crap, that's a biggie but I think I can work around it, teach the user to
dial *98 for voicemail, *700 for
Thanks for the feedback. More questions:
1. How's the range on the wireless?
2. Is there a soft key that can be programmed on the wireless handset?
3. Can I make a soft key basically do anything, any keystroke?
4. How's the call log detail?
-Original Message-
From: Mike Clark
yrving rivas wrote:
I would like to know if any of you have a cell phone like a pci card
to install in one slot to my asterisk server?, I want to make a
connection from my asterisk to the cellular network.
Does anybody has a solution like this?
Regards,
Yrving
On the 7960 with a SIP image, Press the Settings button and go to
option 4 SIP Configuration. Scroll down to line 24 NAT Enabled and
set it to yes. Then set 25 NAT Address to the external IP address.
This will need to be manually changed every time the phone's router
pulls a new DHCP lease. In
I'd like to know if anyone has sucessfully managed to run multiple instances of
Asterisk on the same system.
- Did you run each instance as a separate user?
- Did you have any install or config problems?
- It looks like the G729 codec registration utility doesn't work when files
aren't
On Mon, 25 Sep 2006 08:25:10 -0600, Colin Anderson wrote
Looks good, great price:
http://www.aastratelecom.com/ipphones/pro_243.asp
Anybody using these? How's the cordless? Does it play nice with * ?
Very good phone. The range of the cordless unit is not the greatest but
enough to be
SEÑORES DIGIUMLA PRESENTE ES PARA CONFIRMAR QUE
REQUERIMIENTOS DE HARDWARE Y/O SOFTWARE SON NECESARIOS PARA CONECTAR
UNA CENTRAL TELEFONICA PANASONIC TDA620 QUE TIENE INSTALADA UNA TARJETA
(E1) SE5E18, NO SE SI SEA ESA LA REFERENCIA, CON UNA TARJETA DIGUIM TE110P.
VI EN ALGUNOS FOROS QUE TIENE QUE
Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local
lookup to see if a number is available locally, in order to find out if the
number is available on one of the other 9 servers, this peer has to query all 9
remaining peers.
Is that true?
Is there a way to have
Hi Folks,
Has anyone seen these errors repeatedly in the CLI?
Incoming call: Got SIP response 415 Unacceptable Content-Type back
from 192.168.1.209
We're using GXP-2000s.
TIA,
Brian
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At 07:25 AM 9/25/2006, you wrote:
Anybody using these? How's the cordless? Does it play nice with * ?
I have 3 of them here, we're very happy with them. The cordless is
fine, about the range of my old Panasonic cordless. Sound quality is
good and the speaker phone seems good. Plays fine with
I just downloaded asterisk 1.4beta2, and did a:
./configure --prefix=/home/pbx/1.4
[11:[EMAIL PROTECTED](pbx1):asterisk-1.4.0-beta2]# ls /home/pbx/1.4
bin include lib sbin share
What happened to etc? If I do a 'make samples', the default conf files get
thrown in /etc/asterisk.
Doug.
On 09:35, Mon 25 Sep 06, yrving rivas wrote:
I would like to know if any of you have a cell phone like a pci card to
install in one slot to my asterisk server?, I want to make a connection from
my asterisk to the cellular network.
Does anybody has a solution like this?
1/2/4 simslot pci
Hi Doug,On 9/25/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Lets say you have a cluster of, say, 10 Asterisk servers. After doing a local lookup to see if a number is available locally, in order to find out if the number is available on one of the other 9 servers, this peer has to query all 9
one that also offers support for it. thanks.___
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http://lists.digium.com/mailman/listinfo/asterisk-users
At 07:48 AM 9/25/2006, you wrote:
It's excellent home phone. I wouldn't use it in a business environment. No
hold, no one-touch voicemail. However, it works great!
No Hold? Mine has a hold button and programming one touch voice mail
would be no problem at all.
Ira
At 08:31 AM 9/25/2006, you wrote:
aw crap, that's a biggie but I think I can work around it, teach the user to
dial *98 for voicemail, *700 for park and hash to transfer, currently the
It has a dedicated hold button and you can easily program dedicated
Park and voice mail buttons. I've not
Asterisk does not support this, as it already has features for
multi-client configuration within a single Asterisk installation/process.
Douglas Garstang wrote:
I'd like to know if anyone has sucessfully managed to run multiple instances of
Asterisk on the same system.
- Did you run each
Bidirectional SIP trace usually helps in these situations.On 9/25/06, Mr. Jones [EMAIL PROTECTED] wrote:
Hi Folks,Has anyone seen these errors repeatedly in the CLI?Incoming call: Got SIP response 415 Unacceptable Content-Type back
from 192.168.1.209We're using
Thanks for the input
Yes I have nat=yes and qualify=yes I know in the SIPMacAddress.cnf file
I have
# NAT/Firewall Traversal
nat_enable: 1
nat_received_processing: 1
nat_address: phone's public IP Address
Do I still need to set it again in SIP Configuration ?
Thanks all
Barry
Hughes, Sam
At 09:23 AM 9/25/2006, you wrote:
2. Is there a soft key that can be programmed on the wireless handset?
Not really, there's a function key menu and you can set that up any
way you want, but what you can assign to the functions is very
limited. The cordless is very handy, but the
All my Cisco phones show less than 75ms except for one (mine of
course). I do have a switch in my cube that I use for extra ports and
that's the only real difference.
Do you have anything plugged into the extra network port on the phone?
Yes, I have workstations plugged into the extra ports
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Monday, September 25, 2006 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk
Asterisk does not support this, as
My asterisk server has 2 NICs . One with a public IP and one with an
internal LAN IP. All the phones configure to the LAN IP so there's
basically nothing between them. A 3com switch and that's it.
basically nothing is wrong. I have a 3com switch in front of the one phone
that reports the
Doug,
Why do you want to do this to begin with? I think the best solution is
to use the realtime stuff, and build your own management tools, which
would allow you to do this (you could drastically cut the complexity
with the right tools). Even if you could run them together, how
would you put
I'm seeing channel.c: Nobody there, continuing... in the asterisk
full.log. This error is repeated 20+ times per second when it occurs. I
thought this problem was specific to one PBX that performs call
recording on all the call queues, but after disabling all call
recording, the error
-Original Message-
From: Brian Rogan [mailto:[EMAIL PROTECTED]
Sent: Monday, September 25, 2006 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Running Multiple Instances of Asterisk
Doug,
Why do you want to do this to begin
On Mon, Sep 25, 2006 at 12:28:30PM -0600, Douglas Garstang wrote:
How easy do you think the management of the configuration files is
going to be if your trying to host several dozen companies on the one
Asterisk instance? Sure, you can split things into contexts, but just
try and imagine how
On Mon, Sep 25, 2006 at 12:52:43PM -0600, Douglas Garstang wrote:
-Original Message-
From: Brian Rogan [mailto:[EMAIL PROTECTED]
Sent: Monday, September 25, 2006 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Running Multiple
I know asterisk 1.4 has t.38 pass through, but I don't think trixbox does. i run trixbox. looks like for now i will have to setup my fax machine to connect directly to my t38 provider. anyone know when trixbox may have this update?___
--Bandwidth and
Doug,
I actually see this as a pretty logical way to solve the problem.
Please keep us posted if you have any luck sorting out running multiple
instances, or mail me off-list if no one else is interested.
Thanks,
On 9/25/06 1:52 PM, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original
Mailing List wrote:
My asterisk server has 2 NICs . One with a public IP and one with an
internal LAN IP. All the phones configure to the LAN IP so there's
basically nothing between them. A 3com switch and that's it.
basically nothing is wrong. I have a 3com switch in front of the one
Best of luck getting multiple instances of Asterisk to play nice when
accessing Zap channels.
James Texter wrote:
Doug,
I actually see this as a pretty logical way to solve the problem.
Please keep us posted if you have any luck sorting out running multiple
instances, or mail me off-list
On 9/25/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] zaptel-1.4.0-beta1]# ztcfg -vvv Notice: Configuration file is /etc/zaptel.conf
line 235: Unable to read Zaptel version information. Zaptel Version: $êþP¦0 Echo Canceller: Configuration == Channel map:
On 14:31, Mon 25 Sep 06, Mailing List wrote:
My asterisk server has 2 NICs . One with a public IP and one with an
internal LAN IP. All the phones configure to the LAN IP so there's
basically nothing between them. A 3com switch and that's it.
basically nothing is wrong. I have a 3com
you didn't listen. SIP only. Anyone can understand that multiple
instances on the same machine can't touch the same hardware.
I can see how this would be very easy - dedicate an IP to an instance,
and it'll play nice.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Greetings List,
I'm putting together a plan for a new Asterisk system and I'm trying to
decided on an interface card to use. I was originally planning on using
a Sangoma A200 but now I'm considering a Digium TDM2400P. The server is
large enough to accommodate the full sized TDM and I'll be
We aren't accessing ZAP channels. No Digium hardware is installed!
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Monday, September 25, 2006 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Running
But if I segment my zap channels, that shouldn't be an issue, correct? I.e.
Instance 1 = Port 1, Instance 2 = Port 2, etc. Of course, you are also
assuming there is Zap channels, as I believe he is using a gateway, which
takes that out of the equation.
On 9/25/06 2:23 PM, Eric ManxPower
Hi,
we're writting interface module for our speech recognition system. We would
like to export stream of audio samples to external app, but to preserve dtmf
recognition and dialplan progress.
I wonder if recording application would be a good start for that (recording
application obviously
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