Hi Neoh,
All you have to do is configure your VoIp provider as another SIP
extension on your Asterisk server and then use extensions.conf to set
dialout rules, so when you do dial a number your asterisk server
forwards it to the VoIp provider.
Examples of extensions.conf can be found at
Dear all
I want to restrict outgoing call from specified extention so
is there any configuration for this setup ?? please send me example file
-
Boardwalk for $500? In 2007? Ha!
Play Monopoly Here and Now (it's updated for today's
Dear all
I have fax machine which is connected with audiocode FXS port
and audiocode connected with my asterisk server now what configuration i have
to configured on asterisk ?? can any one suggest me what would be best for this
kind of setup ??
Hi,
I've just built my first asterisk server. Current information:
OS Version:
Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10
06:50:22 EDT 2007 i686 i686 i386 GNU/Linux
Asterisk Build:
Asterisk 1.4.11
Asterisk GUI-version Revision: 1479 $
Server Date TimeZone:
On Wed, Sep 05, 2007 at 01:36:02AM -0300, Ricardo Gemignani wrote:
Hi all,
My name is Ricardo and unfortunately I'm just crawling in this
telecomm/asterisk world. So, after reading all day long i still don't
understand a few things. :D
I'm trying to develop a call recorder for a
Adrian Marsh wrote:
When you access the A*k console, is this via a tty connection
(ssh/telnet), or actually on the physical console of the server?
I don't think it's A*k that's directly logging to the console - the
config doesn't show that... I'm guessing, that you're accessing A*k via
the
Hi Devraj,
May i have your extension.conf working sample??
Thanks you very much.
Date: Wed, 5 Sep 2007 16:14:36 +1000
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to make call from asterisk?
Hi Neoh,
All you have to do is configure
Ricardo Gemignani wrote:
Hi all,
My name is Ricardo and unfortunately I'm just crawling in this
telecomm/asterisk world. So, after reading all day long i still don't
understand a few things. :D
I'm trying to develop a call recorder for a costumer. He has a
small call center ( 10
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board, to terminate POTS lines and use all SIP Phones?
Due to circumstances, I end up with a 1u server that has no aux power
Joe Acquisto wrote:
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board, to terminate POTS lines and use all SIP Phones?
Yes, you only need to connect a power supply
On 9/4/07, Devraj Mukherjee [EMAIL PROTECTED] wrote:
Hi Atis,
Is your code open source, or are you willing to share your PHP code
snippets with me? And thanks for the information on Asterisk's
stability. Do you think there is an issue in the implementation or
just network/traffic issues?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi to all
I'm using Asterisk 1.4 and I'd like to use LDAP instead of sip.conf...
Now you suggest to use asterisk realtime (res_config_ldap) or astirectory??
Can I use one of them with version 1.4?
thx
On 8/30/07, Gavin Henry [EMAIL PROTECTED] wrote:
No probs.
On 29/08/2007, Abhishek M S
Mike Hammett wrote:
Pong
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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I've been trying to send messages to the list for the past 24 hours, but they
just aren't going through.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Wednesday,
Quoting Doug Lytle ([EMAIL PROTECTED]):
Pong
The list seems to act weird. I mailed to the list earlier, the message
was accepted, but does not appear on the archives nor did i get a bounce
or my own listmail back.
Though i do see other people posting :/
--
| Only those who will risk going
Carlos Chavez wrote:
I have a customer that has two buildings that are connected with a
fiber link. We have a single Asterisk server to cover both buildings.
Now the customer went and bought an overhead paging system for the
remote building and they want to integrate it with Asterisk.
Agreed. This conversation is working just fine, but the important messages
I'm trying to get to go through aren't.
I've never had consistent success from posting to asterisk-users.
Asterisk-biz seems to work all of the time.
-
Mike Hammett
Intelligent Computing Solutions
Sander Smeenk wrote:
Quoting Doug Lytle ([EMAIL PROTECTED]):
Pong
The list seems to act weird. I mailed to the list earlier, the message
was accepted, but does not appear on the archives nor did i get a bounce
or my own listmail back.
Though i do see other people posting :/
Same
Hi
i generate a call from the dialplan in this mode:
exten = 1002,1,Answer()
exten = 1002,2,Dial(SIP/[EMAIL PROTECTED])
the call is generated, but after some seconds it is interrupted, here
the asterisk log:
*CLI -- Executing Answer(SIP/host1-0819d0d0, ) in new stack
-- Executing
I have a customer that has two buildings that are connected with a
fiber link. We have a single Asterisk server to cover both buildings.
Now the customer went and bought an overhead paging system for the
remote building and they want to integrate it with Asterisk. Is there a
device
ACK
Mike Hammett wrote:
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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The phones you are using might support it already (and not even need
the system)
the grandstreams I have do, but I can't speak for any others.
Quoting JR Richardson [EMAIL PROTECTED]:
I have a customer that has two buildings that are connected with a
fiber link. We have a single
Hello,
my configuration is the following:
a TDM400P board with an fxs and fxo daughter boards on it.
I thus connect a fax to my FXS port, after having verified that this port
was correctly functioning. For this, I had tried before with a simple phone,
and with some basic voicemail exten scripts.
Dave Fullerton wrote:
Same thing happened to me a while back. I sent a new message asking a
question ..twice.. and neither made it through. However replies to other
peoples messages went through just fine.
This may not be the problem, but I've seen this on my NEW post a few times
and it was
Hi:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made
modprobe wctdm the fxs modules is lightened but there is no dial tone came from
it .
Can i get some help please.
Best Regards;
Wassim
_
Windows Live
*nods* I verified more than once and even copied + pasted to make sure.
Obviously my ping message went through, but my others have not.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Bill Andersen [EMAIL PROTECTED]
To: Asterisk
On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote:
Hi:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i
made modprobe wctdm the fxs modules is lightened but there is no dial tone
came from it . Can i get some help please.
do you have the power cable
On Wed, 2007-09-05 at 09:11 -0500, Mike Hammett wrote:
*nods* I verified more than once and even copied + pasted to make sure.
Obviously my ping message went through, but my others have not.
I'm working with Digium's IT department to try to track down the
problem.
--
Jared Smith
Community
On Wed, 2007-09-05 at 14:09 +, wassim darwish wrote:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when
i made modprobe wctdm the fxs modules is lightened but there is no
dial tone came from it .
Once you've loaded the wctdm kernel module, you should get battery on
the
and I appreciate it much.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 05, 2007
You are using safe_asterisk, it will restart automatically Asterisk
after it crashes.
Original Message
Subject: [asterisk-users] Asterisk Died message
From: Nitesh Divecha [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Joe Acquisto wrote:
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board, to terminate POTS lines and use all SIP Phones?
Due to circumstances, I end up with a 1u
Dear All,
I'm integrating avaya commuication manager difinity ver 1.0 with asterisk
using B2B E1. following are the details of my H/W, zaptel configs and
software installed.
Digium TE110p
asterisk 1.2.19
cent OS 4.4
zaptel 1.2.18
libpri 1.2.4
etc/zaptel.conf
span=1,0,0,cas,hdb3
bchan=1-15,17-31
It will not after some types of crashes.
/b
On Sep 5, 2007, at 9:43 AM, Perssy Llamosas wrote:
You are using safe_asterisk, it will restart automatically Asterisk
after it crashes.
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Hi guys,
is it possible to set caller presentation with mISDN? I tryie with
SetCallerPres() and CallingPres without success...
exten = s,1,ChanIsAvail(mISDN/1)
exten = s,2,CallingPres(32)
exten = s,3,Set(CALLERID(num)=e.164_number)
exten = s,4,Dial(${CUT(AVAILCHAN||1)}/${ARG2})
Anyone can help
Hi,
Can anyone tell me why the below dialplan doesn't filter off dialed
numbers for 01793520158, and jump to local,priority1
If I change it to :
exten = 01793520158,1,Goto(local,${EXTEN:-3},1)
then it works fine (but that's too specific)...
exten =
On Wed, Sep 05, 2007 at 09:43:20AM -0500, Perssy Llamosas wrote:
You are using safe_asterisk, it will restart automatically Asterisk
after it crashes.
Or will contantly die, clog the logs and make debugging the problem more
difficult than it is.
Or you might have two safe_asterisk processes
I had to turn Sipbroker off at one point, as I found that some Conf.
Calls on a 3rd party system didn't like the DTMF being passed (users
unable to enter conferences). I traced all the failures to calls
passing out via SIPbroker, disabled it so the calls went via PSTN and
all was well..
Now I'm
On Wed, Sep 05, 2007 at 08:26:25PM +0530, Vidura Senadeera wrote:
Dear All,
I'm integrating avaya commuication manager difinity ver 1.0 with asterisk
using B2B E1. following are the details of my H/W, zaptel configs and
software installed.
Digium TE110p
asterisk 1.2.19
cent OS 4.4
On Wed, Sep 05, 2007 at 10:27:48AM -0400, Jared Smith wrote:
On Wed, 2007-09-05 at 14:09 +, wassim darwish wrote:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when
i made modprobe wctdm the fxs modules is lightened but there is no
dial tone came from it .
Once
On 9/5/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Or you might have two safe_asterisk processes trying to restart asterisk.
A symptom of this (when Asterisk is not actively crashing) is constant
remote UNIX connection messages on the console every few seconds (assuming
you have nothing that
Sin you have sangoma card , it will act as timer . You need to install
meetme ( app_conference is not very stable last time i read ) .
On 01/09/07, fateme fatah [EMAIL PROTECTED] wrote:
Hi:
I want to have conference call service and I use A102d sangoma's card.Do I
should install ztdummy or
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Hi,
Can anyone tell me why the below dialplan doesn't filter off dialed
numbers for 01793520158, and jump to local,priority1
If I change it to :
exten = 01793520158,1,Goto(local,${EXTEN:-3},1)
then it works fine (but that's too
Asterisk-Java 0.3.1, a free Java library for Asterisk PBX integration,
has been released.
The Asterisk-Java package consists of a set of Java classes that allow
you to easily build Java applications that interact with an Asterisk
PBX Server. Asterisk-Java supports both interfaces that Asterisk
Hi,
I have a client setup where a T1 is terminated into a Cisco IAD2430 Series
device which then interfaces with a Digium Wildcard TE110P card in a server
running Asterisk 1.2.23. I am having a problem with the DTMF tones being
passed to the Asterisk server. Wrong tones are being passed to
Hi to all,
I've installed Asterisk 1.4 and all function very well.
Now I need to use LDAP or RADIUS instead of sip.conf since all the trusted
users have an account on LDAP/RADIUS.
Any suggestions...try astirectory (but is for asterisk 1.2.x, I've 1.4.9) or
Asterisk realtime LDAP (it is only for
Many thanks for that!! I didn't know that the order worked quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9] worked fine).
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis
Sent: 05 September 2007
Dear Sirs,
we ...
1) buy minutes from other providers
2) sell minutes to out clients
some calls terminate to our equipment, others - to h323 proxies.
we want calls to be routed according to costs (a route is chosen from many
by lowest cost).
at the end of it, we'd like to bill our clients and
On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote:
Dear All,
I'm integrating avaya commuication manager difinity ver 1.0 with
asterisk using B2B E1. following are the details of my H/W,
zaptel configs and software installed.
Digium TE110p
asterisk 1.2.19
cent OS 4.4
zaptel
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Many thanks for that!! I didn't know that the order worked quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9] worked fine).
You can also impose a finer level of control over the order extensions are
RADIUS does two things
1) authentication
2) accounting
(well, actually, 3 things, but I see no difference of authorising and
authentication)
accounting is easy for asterisk-1.4, there're CDR (call detail record) which
stores call in radius out of box.
as for authentication/authorising against
Thomas Kenyon wrote:
Joe Acquisto wrote:
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board, to terminate POTS lines and use all SIP Phones?
Yes, you only need to
Vidura Senadeera wrote:
Dear All,
I'm integrating avaya commuication manager difinity ver 1.0 with asterisk
using B2B E1. following are the details of my H/W, zaptel configs and
software installed.
Digium TE110p
asterisk 1.2.19
cent OS 4.4
zaptel 1.2.18
libpri 1.2.4
etc/zaptel.conf
Just to be clear, I thought that dialtone provision didn't require the
power cable, just generating ring voltages? Can anyone say?
Moj
Anthony Messina wrote:
On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote:
Hi:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my
Hi all,
Configuration: Analog phone connected to TDM400p.
I'd like the phone to give a half-ring (chirp) periodically when there
is a message waiting. Can this be done? How is it configured?
The visible Message waiting indicator and the stutter dial tone are
working fine, but are not
or a man in the middle...
http://www.tuxtone.com/index.php/VoIP:T1_man_in_the_middle
On 9/5/07, Steve Totaro [EMAIL PROTECTED] wrote:
Ricardo Gemignani wrote:
Hi all,
My name is Ricardo and unfortunately I'm just crawling in this
telecomm/asterisk world. So, after reading all
Hi,
There isn't an astirectory driver for Asterisk version 1.4. So I guess
you'll have to use the asterisk realtime (res_config_ldap) driver.
cheers
Abhishek
On 9/5/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi to all
I'm using Asterisk 1.4 and I'd like to use LDAP instead of sip.conf...
In article [EMAIL PROTECTED],
Joseph Begumisa [EMAIL PROTECTED] wrote:
I have a client setup where a T1 is terminated into a Cisco IAD2430 Series
device which then interfaces with a Digium Wildcard TE110P card in a server
running Asterisk 1.2.23. I am having a problem with the DTMF tones being
Hi all,
Please mention your real life experience with Asterisk about how many
concurrent calls a single server has handled for you.
Please don't tell me it depends on the Hardware or ., I want your
experiences.
Someone might used it as a calling card with a2billing on a single box with
60
Quoting Jared Smith ([EMAIL PROTECTED]):
*nods* I verified more than once and even copied + pasted to make sure.
Obviously my ping message went through, but my others have not.
I'm working with Digium's IT department to try to track down the
problem.
As it may help you follow the message
The answer, I believe, is yes... but I'm not sure how We had
this working on some SPA-2002s from Sipura... but then after an
asterisk upgrade it stopped working. I'm not sure if it's a setting
in the ATA or asterisk, and we just never needed to pursue it. So the
answer is.. yes it can be
Denied.
On 9/4/07, Moshe at Talk'n'Save [EMAIL PROTECTED] wrote:
please unsubscribe
Moshe Wahrhaftig
IT Manager
Talk'n'Save
Israel: 02-655-0313
Cell: 052-2771738
USA: 516-204-
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
For my wife I recently set up a cron schedule that, every ten minutes,
greps the output of show voicemail users for a new message waiting.
Upon finding one, it dumps a call file into asterisk's outgoing
directory that rings the house phone and, when one is picked up, it
connects the user to
The SIPuras support it, Asterisk analog does not, as far as I know.
Matt wrote:
The answer, I believe, is yes... but I'm not sure how We had
this working on some SPA-2002s from Sipura... but then after an
asterisk upgrade it stopped working. I'm not sure if it's a setting
in the ATA or
Hi, thanks for the reply. This capability is provided by Sipura ATAs
(apparently they do it each time they process SIP REGISTER messages with MWI).
The periodic ring works when the same analog phone is connected the Sipura ATA.
But not when it is connected to the TDM400p.
So to reiterate,
Do Linksys PAP2Ts support it and if so, where is the setting?
On 9/5/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
The SIPuras support it, Asterisk analog does not, as far as I know.
Matt wrote:
The answer, I believe, is yes... but I'm not sure how We had
this working on some
On 9/5/07, Ricardo Gemignani [EMAIL PROTECTED] wrote:
Hi all,
My name is Ricardo and unfortunately I'm just crawling in this
telecomm/asterisk world. So, after reading all day long i still don't
understand a few things. :D
I'm trying to develop a call recorder for a costumer. He has a
On 9/5/07, Kate Kretz [EMAIL PROTECTED] wrote:
Dear Sirs,
we ...
1) buy minutes from other providers
2) sell minutes to out clients
some calls terminate to our equipment, others - to h323 proxies.
we want calls to be routed according to costs (a route is chosen from many
by lowest
On 9/5/07, Pirlouwi [EMAIL PROTECTED] wrote:
Hello,
my configuration is the following:
a TDM400P board with an fxs and fxo daughter boards on it.
I thus connect a fax to my FXS port, after having verified that this port
was correctly functioning. For this, I had tried before with a simple
On 9/5/07, Matt [EMAIL PROTECTED] wrote:
Do Linksys PAP2Ts support it and if so, where is the setting?
I don't know about PAP2T but SPA2102 does. Basically anything that is
similar to the Sipira-SPA firmware, I don't know how familar you are
with them but if your webinterface looks like this:
Bingo! That was it. Well, it's got it to 98% there. I can play with it
now and tweek it.
Todd
- Original Message -
From: Kai-Uwe Jensen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 05, 2007
On Wed, 2007-09-05 at 12:57 -0400, James FitzGibbon wrote:
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Many thanks for that!! I didn't know that the order worked
quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9]
On Wed, 2007-09-05 at 22:44 +0600, Kate Kretz wrote:
Dear Sirs,
we ...
1) buy minutes from other providers
2) sell minutes to out clients
some calls terminate to our equipment, others - to h323 proxies.
we want calls to be routed according to costs (a route is chosen from
many by
That's a clever idea, and it sounds like a viable solution. But (and not
knocking your inventiveness in any way), its a bit of a hack to get around what
seems like a clear limitation.
I'll keep looking for a more elegant solution over the next couple of days, and
give this a go if nothing
Date: Wed, 5 Sep 2007 09:21:19 -0800
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re:
[asterisk-users] No Dial tone came from fxs modules Just to be clear, I
thought that dialtone provision didn't require the power cable, just
On 9/5/2007 at 1:06 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Thomas Kenyon wrote:
Joe Acquisto wrote:
I need to ask, to refresh, is the aux power connector on the TDM400P card
*only* to power the ringer on any
analog phones/devices on the system?
Can I still use this board, to
Yeah, it's a hack for half-rings, but a little less so for putting
someone right into voicemailmain without delay.
Moj
Justin Ridge wrote:
That's a clever idea, and it sounds like a viable solution. But (and not
knocking your inventiveness in any way), its a bit of a hack to get around
Agreed. I appreciate your suggesting it!
- Original Message
From: Mojo with Horan Company, LLC [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 5, 2007 5:55:27 PM
Subject: Re: [asterisk-users] Can
Hi Carlos/All,
Thanks for your reply. I can remove dchan=16 from zaptel.conf
But according to the documentation of Digium and sangoma they mentioning to
use dchan=16.
Are there any specific reason you have experiance regarding this and I am
confusing that what this is included to the
The Polycom hardphones do it by defaultBUT a colleague of mine
worked in a large office and she said that monday morning people would
be driven mad by almost every phone on the floor making that beeble-bup
noise...over and over and over
PaulH
On Wed, 2007-09-05 at 10:32 -0700, Justin
On 9/4/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Do you know where to find clear developers' guides (with some
examples)
for developing apps that run *on* Cisco 79xx phones (especially the
7970)? Examples that can run against Asterisk (not CallManager) with SIP
firmware (not
Thanks to all who responded. My hunt for cheap, err, inexpensive, Polycom's
continues.
How cheap?
PaulH
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Let me quote oej:
Make sure that you remove unnecessary text when you reply
I don't need messages to tell me *5* times about Astricon,
who provides the bandwidth and how to unsubscribe.
I'm sure this has been posted a dozen times but please
http://learn.to/quote
Thanks,
Philipp Kempgen
--
On Sep 5, 2007, at 7:42 PM, Philipp Kempgen wrote:
I don't need messages to tell me *5* times about Astricon,
who provides the bandwidth and how to unsubscribe.
You sure about that unsubscribe part? People do seem to miss it :P
/b
___
Sign up
You need to log your agents in - or set your queue members to be SIP
accounts. (which is probably the best solution)
PaulH
On Wed, 2007-09-05 at 16:53 +1000, Joshua Small wrote:
Hi,
I’ve just built my first asterisk server. Current information:
OS Version:
Linux
Brian West wrote:
On Sep 5, 2007, at 7:42 PM, Philipp Kempgen wrote:
I don't need messages to tell me *5* times about Astricon,
who provides the bandwidth and how to unsubscribe.
You sure about that unsubscribe part? People do seem to miss it :P
Good point. :)
Regards,
Philipp
Hi,
I've been doing some testing on moving from 1.2 to 1.4 and one issue I've
encountered is re-transmits whenever an INVITE is cancelled. I have a stateless
SIP proxy in fron of my asterisk servers (all it does is direct requests to one
asteisk server or another) and the re-transmits do not
Have you tried asterisk -rvvv?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nik600
Sent: Wednesday, September 05, 2007 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Spawn extension (default, 1002, 2)
On Thu, Sep 06, 2007 at 05:48:57AM +0530, Vidura Senadeera wrote:
Hi Carlos/All,
Thanks for your reply. I can remove dchan=16 from zaptel.conf
But according to the documentation of Digium and sangoma they mentioning to
use dchan=16.
Please leave dchan=16 , and replace 'cas' with 'ccs' in
Hello ppl,
Any way to specify alphabetical exten patterns in the dialplans on Asterisk?
All my users would have alpha/numerical ids. I don't want to add a line
for every user in my dialplans.
I searched around, but couldn't get anything useful. Any way to get
around this?
Thanks in advance
-
I have the same setup asterisk-1.4.11 with TE120P two port E1 card with is
connected with avaya system but signaling is Qsig becase i want unified dialplan
my configuration
/etc/zaptel.conf
### Digium TE120P Card Configuration #
# E1 port 1
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
Hi there
I europe alaw is usual. I have a SIP Phone which perferes ulaw.
When my * box has to transcode alaw to ulaw the sound get's one way choppy.
(alaw = ulaw is choppy, ulaw = alaw is fine).
I managed to fix the issue by forcing my SIP phone to use alaw only, but is
this a know issue with
Off-topic to the original thread. I just wonder what you meant in your
configuration:
On Wed, Sep 05, 2007 at 09:58:19PM -0700, satish patel wrote:
I have the same setup asterisk-1.4.11 with TE120P two port E1 card
with is connected with avaya system but signaling is Qsig becase i
want
yes, i've tried asterisk -r
i've also tried sip debug, but i can't reach any error... only that
the cmmunication is finished.
On 9/6/07, Shonga_Kerz [EMAIL PROTECTED] wrote:
Have you tried asterisk -rvvv?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Dear all
I have FAX machine connected with audiocode SIP device i am
trying to send fax and when negosiation going on and i start send fax button
then my after half page it got stuck in fax machine so is there any codec
problem i am useing ulaw/alaw is it fine or not anybody
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