Hi,
I am using FXO 8 Ports AudioCodes. Asterisk is able to accept incoming
outgoing calls. However, I noticed that the caller ID of the caller coming from
the FXO displays its endpoints assigned number and not the actual caller's ID
coming from PSTN.
Hope someone is using the same scenario
asterisk-users@lists.digium.com
hi,
I install asterisk1.4.0 , when I use the meetme application. The console show
that
WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for
extension .
I found that there havn't app_meetme.so in the directory of moudles.
On 02/01/07 02:15 Olle E Johansson said the following:
both channels should act the same unless there's a configuration that's
giving wrong information
to chan_sip, like you having a username= or defaultip= setting.
how does a username= entry in sip.conf affect dialling behaviour when the
Richard Lyman wrote:
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because
Hello,
I got a little interogation about these 3 points.
I want to write something like this sample in my extension.conf. I have
tested and it works but I don't know if it is a good way to make a menu.
I don't want to put number as it is boring to maintain.
Does anyone know if there is some
Hi All,
I've installed Enhanced PickupChan on asterisk 1.2.14 using howto from
http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp .
from extensions.conf:
exten = 0,1,Dial(SIP/eosoiris|20|tTrR)
exten = 200,1,Dial(SIP/dzalewski|20|tTrR)
exten = _7.,1,Pickup2(${EXTEN:1})
When I try to
Is uniqueid globally unique? I have three Asterisk installations and I need to
store data from all of them in same database, in same table. Will this uniqueid
field be unique?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP:
This depends on your application. As you say you are able to do everything you
require in dialplan at that is great. I have used AGI fairly extensively
becuase the stuff I want to do can't be done in dialplan alone. For instance i
have written a auto attendants that can be dynamically
You have to compile and install Zaptel first, for asterisk to build meetme.
--
--
Steven
http://www.glimasoutheast.org
Àî¾ý [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
asterisk-users@lists.digium.com
hi,
I install asterisk1.4.0 , when I use the meetme application. The
Have you tried
phpagi
http://phpagi.sourceforge.net/
Jon Farmer
Telford, Shropshire, UK
- Original Message
From: Michelle Dupuis [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, 26 January, 2007 5:52:27
Mitch Thompson wrote:
I'm looking for some help from any Asterisk heavy who might be doing
something similar to what I'm trying to do...
Background:
I work for a research lab, testing telephony products and tools.
Historically, we used Ameritec Crescendos and Fortissimos to act as load
Hi Guys..
I want to see what the R2mfc community has been up to. Anybody moved to 1.4?
what have you done regarding unicall? Any updates or are you stuck with
1.2.X too?
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Irvin
Sent: Wednesday, January 31, 2007 10:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Toll-free dialing via PRI problem
The point is that the SIP carrier side gets the abort *before the SIP
carrier can complete the connection*. That doesn't take 45s. It takes
something like a few seconds. What is causing my (Asterisk) side to
abort right after completing registration?
On Thu, 2007-02-01 at 02:28 -0500,
On 2/1/07, Chung-lai Chan [EMAIL PROTECTED] wrote:
Hello all,
Can I send fax from PAP2 through a zap channel to PSTN? I have tried but
it is not successful.
Thank you for your help!
Lai
Try to remove echo cancellation (any type of cancellation) and VAD.
I got good answer receiving fax as
Steven,hello!
Thank you so much, but I have installed Zaptel before Asterisk.
You have to compile and install Zaptel first, for asterisk to build meetme.
--
--
Steven
http://www.glimasoutheast.org
李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi Jon,
Did you find any solution for your problem?
-- Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schopzinsky
Sent: Friday, January 26, 2007 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users]
Mitch Thompson wrote:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at
the beginning of [from-trunk-custom] to the full dialed digits in
${EXTEN}, before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has already
called this
hi,
i think, by default, 'uniqueID' is created by the asterisk.
if this is correct, you would (eventually) have non-uniqueID's
i saw somewhere in the wiki that someone suggested a change (in the code ?) so
that 'uniqueID' would be generated by the database. unique-id being the
primary key and
Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in
menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It
then compiled.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ??
Sent: Thursday, February 01,
Hi this was on the mailing list. Some one posted, I didn't tested my
self but I believe it should work.
I believe that you can set a systemname=blah in asterisk.conf and that
will be pre-pended with a dash to the uniqueid.
For example:
systemname=node1
the uniqueid might look like
Benko,
You can put multiple files in the MOH directory giving your listener a good
chance of getting a new piece of music each time he is on hold. Asterisk
picks one of your files randomly.
Regards,
Brian
- Original Message -
From: Benko [EMAIL PROTECTED]
To:
Rilawich Ango wrote:
Hi all,
I am using asterisk1.2.14,realtime and I find there is a strange
case in the receiver's display. I have a dial plan to route a call
to the destination. I haven't set the callerid(num) for the caller.
In the receive ends, it's display shows asterisk when I make a
I have an F3000 phone utstarcom and sending a text message to it.
All is working but there is a line of sender: asterisk.
How do I control what this line says?
THanks,
Jerry
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asterisk-users
Hi,
I'm looking for an hardware platform for an * installation that should
have at least 3 PCI slot with no irq sharing whatsoever.
Hardware raid 1 with hot swap is a premium, but not mandatory ...
What would you choose? compaq/hp ? Dell ? Ibm ?
Tnx for any advice on this matter!
--
I
From: Jerry Geis [EMAIL PROTECTED]
I have an F3000 phone utstarcom and sending a text message to it.
All is working but there is a line of sender: asterisk.
How do I control what this line says?
Try sip.conf, callerid=...
Yuan Liu
THanks,
Jerry
Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can
pick up CallerID. How can I redirect the log output of asterisk to
/dev/ttyS0 or /dev/console?
Many thanks,
Neil
safeharbour IT Ltd
Your IT Department
fax: 0845 867 2891
mob: 07812 114784
voip: [EMAIL PROTECTED]
It needed BOTH the text callerid and numerid callerid to display
the text form. Callerid: Some name number
At first I was only supplying the name. Works fine. Thanks,
Jerry
/From: Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users
//
//I have an
Hi,
I asked some questions here about G.729 earlier in the week, and it
looks like it would fit the bill for compressing audio between my *
server in colocation and sip phone at home.
This is what I want my setup to look like.
(Wont make sense unless you are using a fixed width font)
Any ideas? It should be simple...
Cosmin Prund wrote:
Hello everyone:
using chan_capi 0.7 and asterisk 1.4
Quick question:
How can I detect the number of voice channels (B channels) in use at
a given time. I'd like to call Busy if two B channels are used on my
BRI to give the calling
Hello!
Is there any easy way to use the caller ID display info
(CALLERID(name) in Asterisk) in dialplan just as we could use the number in:
exten = _X./67803287, 1, action
I have a SIP GSM device, and when a call comes in, it passes me the
caller ID like so:
-- Sip message Header:
From:
Are there any scripts out there that would help me stress test two boxes
that are setup back to back with 4 PRI connections? We're having
problems with Sangoma cards w/ PCI-e on HP DL385 G2 hardware and I'm
tired of testing them in a production environment. As Sangoma
provides firmware updates
Rich Doughty wrote:
Richard Lyman wrote:
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set
Use auto dial. You can have as many calls as you wish.
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
Porier, Jeremy M. wrote:
Are there any scripts out there that would help me stress test two boxes
that are setup back to back with 4 PRI connections? We're having
problems with
On 31 Jan 2007, at 14:32, Rob Schall wrote:
Perfect. Here's a quick and hopefully doable followup question. We
have
Polycom Soundpoint 501 phones. Is there a way to have a phone check 2
voicemail boxes? If we have a queue, and we want the MWI to show
for say
that users's extension 1000
From: Jerry Geis [EMAIL PROTECTED]
It needed BOTH the text callerid and numerid callerid to display
the text form. Callerid: Some name number
Related to callerid: I can't get text ID to work in an analog phone on FXS.
I tried the above format, it simply displays the entire string in both
On Thu, 1 Feb 2007, Cosmin Prund wrote:
Any ideas? It should be simple...
It is easy: read the README in chan-capi.org package ;-)
Just look into the variable BCHANNELINFO and you will know if it is a call
without b-channel (the third call).
Armin
Cosmin Prund wrote:
Hello everyone:
Richard Lyman wrote:
Rich Doughty wrote:
Richard Lyman wrote:
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't
*snipped
ast_set_variables(chan, vars);
insert pbx_builtin_var here --
ast_pbx_run(chan);
since DIALSTATUS and HANGUPCAUSE are both protected, you will probably
have to create another such as FAILEDCODE.
i hope
take a look on Originate command for Asterisk manager interface to get web
page generating calls between the two boxes.
Easier I believe is to use SIPp to be used as an UAC that starts dialing to
your box1 and in the dialplan of this box make a dial for a Zap channel on
Box2.
You need to
On 2/1/07, Andy Davidson [EMAIL PROTECTED] wrote:
What I would expect to happen, is that Asterisk would transcode
between the ulaw/alaw party, and me, wanting to listen via g729. Is
this what *should* happen ? Worth noting that my provider does not
support G.729. Is what is happening a bug
Hi,
I was planning on getting a Dell PowerEdge 2950 for our new Asterisk
configuration.
But while searching for documentation about it and/or reported issues, I
found this:
http://www.voip-info.org/wiki/view/Asterisk+hardware
WARNING - many Dell motherboards use the e1000 gigabit ethernet
Hello,
I have installed Asterisk on several of them and there can be issues.
Will you be installing a telco interface card on this server?(If so, which one)
Will this server have PCI or PCIexpress expansion ports?
MATT---
On 2/1/07, Eric Rousse [EMAIL PROTECTED] wrote:
Hi,
I was planning
Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can
pick up CallerID. How can I redirect the log output of asterisk to
/dev/ttyS0 or /dev/console?
I think you might be better off with a System() call in your dial plan such as:
System(echo ${CALLERIDNUM} /dev/ttyS0)
Has anyone used the G option with the Dial app? I'm looking for a way
to control the called party leg. Specifically, I'd like to pass a few
variables to the called side for some call control. Here's a synopsis
of what I'm doing:
Make outbound call w/ AMI Originate action.
Called party answers
On Thu, 1 Feb 2007, Eric Rousse wrote:
Hi,
I was planning on getting a Dell PowerEdge 2950 for our new Asterisk
configuration.
But while searching for documentation about it and/or reported issues, I
found this:
http://www.voip-info.org/wiki/view/Asterisk+hardware
WARNING - many Dell
Hello,
Well we're planning either use some PRI lines or IP Trunk, where not
sure yet.
For the PRI lines we will probably use a Wildcard TE412P, so PCIe
For the IP Trunk, not sure yet I don't have a lot of info in that regards.
I'm also planning to put an extra server with some cards to
Hello all,
I'm trying to see if I can finally get rid of a talkoff problem that
I've been having with my Asterisk server since I started messing with it
over a year ago. Currently, I'm running it on SuSE 10.1 box with
Asterisk 1.4. I'm using Snom 360s with the set. My setup is one where
the
We have a 2850 in a productive environment with a BNE1 performing well
(OpenSuSE 10) and a 2950 with BNE1 and BN8S0 also performing OK (on Ubuntu
Edgy). You only have to blacklist some hotplug kernel modules and yes, we do
have very long pings (1 ping per week with a check rate of 10min per
On Thursday 01 February 2007 4:01 pm, McGhee, Stefano wrote:
I'm trying to see if I can finally get rid of a talkoff problem that
I've been having with my Asterisk server since I started messing with it
over a year ago. Currently, I'm running it on SuSE 10.1 box with
Asterisk 1.4. I'm using
We have the 2950. It came with only 2pcix ports. And if you need to
power an fxs card, you need to route wires around. It wasn't easy to
work with.
==
Jeronimo Romero
EUS Networks
Email: [EMAIL PROTECTED]
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
We built some systems based on Dell 2950's and they ran fine.
We put a TE110P in a Dell 860 last week, and it makes a noise on the
outbound part of the call. Not on the inbound, which is really odd.
The 860 works perfectly with a TE210P in it though.
(This fault has been logged with Digium.)
What is the manufacturer and model of the 4-port T1 card? I
have had talkoff
with the TE406 (1st gen echo canceller), and have heard of
talkoff occurring
with relaxdtmf=yes in zapata.conf.
Hey there. I do believe it it a Digium TE406 with Echo Canceller. I
can't remember how many
Any mid-level server (kinda 3ghz 2GB ram) you have been wonderfully
happy with?
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I wish to have my Grandstream GXP-2000 phones make a different
distinctive ring for internal calls ( Internal ) or if the incoming call
has no caller id 'NOCID'.
The Grandstream phones calls allow 3 distinctive rings depending on the
caller id. I have one set up and working for 'Internal'
I have been trying to get a DIALSTATUS output from a call started with
originate. I searched a fair bit and have found several references to using
local channels to do this. However, I could not find enough of the specifics
to get it working myself.
What I need to do is dial a zap channel and
Am Donnerstag, den 01.02.2007, 16:15 -0600 schrieb Larry Alkoff:
I wish to have my Grandstream GXP-2000 phones make a different
distinctive ring for internal calls ( Internal ) or if the incoming call
has no caller id 'NOCID'.
The Grandstream phones calls allow 3 distinctive rings
Alessio Focardi wrote:
Hi,
I'm looking for an hardware platform for an * installation that should
have at least 3 PCI slot with no irq sharing whatsoever.
Use an industrial PC with a backplane bus. You can easily get 3-4 usable
slots in a 2U and 10-14 slots if you use a 4U.
Leo
Yuan LIU wrote:
Related to callerid: I can't get text ID to work in an analog phone on
FXS. I tried the above format, it simply displays the entire string
in both numeric and text field (i.e., displays the same string
twice). Tried a few other ways, got varied results (some resulting in
Trevor Peirce wrote:
Mitch Thompson wrote:
[SATX_555_Extensions]
exten = 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at
the beginning of [from-trunk-custom] to the full dialed digits in
${EXTEN}, before I break it down.
exten = 1212,n,Busy(); if the file exists, someone else has
Hi There,
I am interested to know the solutions available for a large call centre
with more than 200 seats running asterisk and iax soft clients(eg .idefisk).
All the calls are through soft clients , so there is no PSTN
requirement/connectivity.This is a pure iax implementation.
1. How to
hi all
i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0
any suggestions ?
make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect'
make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect'
Generating input for menuselect ...
I've discovered that when dialing out using API's Originate action, a no
answer is considered a failed attempt, while a busy is considered a
successful attempt. The problem I'm having is that when I dial an
invalid number, say a disconnected number that gives a fast busy, my
CDRs are identical to
Bill Gibbs,hello
Thank you so much. According to this method , I get the app_meetme.so .
=== 2007-02-01 22:49:43 您在来信中写道：===
Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in
menuselect.makeopts I removed the DEPSFAILED line that had meetme in
Michelle Dupuis wrote:
I am trying to set callerid from a PHP script, using one of two
functions as shown below (setid1 and setid2). The first function
works great with regular names and numbers, BUT, if I call the
function with (Test,UnknownNumber), the cid number gets set to
asterisk. Why
On 2/1/07, Michael Collins [EMAIL PROTECTED] wrote:
Is there a way to distinguish between a no answer and an invalid? For
me, a 'failed' attempt is dialing an invalid number, and I'd like the
CDRs to reflect that. I'd like a no answer to be as 'successful' as a
busy.
The ${DIALSTATUS}
Leo Ann Boon wrote:
Yuan LIU wrote:
Related to callerid: I can't get text ID to work in an analog phone on
FXS. I tried the above format, it simply displays the entire string
in both numeric and text field (i.e., displays the same string
twice). Tried a few other ways, got varied results
hi all
what do must win32 clients use as a free or OSS sip softphone ?
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Dennis Kavadas wrote:
hi all
what do must win32 clients use as a free or OSS sip softphone ?
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Hi
Try that one http://www.counterpath.com/index.php?menu=Productssmenu=xlite
Bruno C. Branco
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis Kavadas
Sent: February 02, 2007 10:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Eric ManxPower Wieling wrote:
You should not have quotes in Caller*ID info. MOST devices will just
ignore the quotes, but a few will refuse to accept Caller*ID with
quotes in it. At least one revision of SIP firmware for Cisco phones
does this.
Thanks for the heads up. On the other hand,
On Thu, 2007-02-01 at 08:47 -0500, Matthew Rubenstein wrote:
The point is that the SIP carrier side gets the abort *before the SIP
carrier can complete the connection*. That doesn't take 45s. It takes
something like a few seconds. What is causing my (Asterisk) side to
abort right after
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roi Stork
Sent: Thursday, February 01, 2007 6:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] API Originate Action - distinguishing between
From: Leo Ann Boon [EMAIL PROTECTED]
Yuan LIU wrote:
Related to callerid: I can't get text ID to work in an analog phone on
FXS. I tried the above format, it simply displays the entire string in
both numeric and text field (i.e., displays the same string twice). Tried
a few other ways, got
From: Dennis Kavadas [EMAIL PROTECTED]
hi all
i'm getting the below error when trying to compile asterisk-1.4 on
redhat-9.0
any suggestions ?
Likely your make version. See this thread
http://forums.digium.com/viewtopic.php?t=12707 in forum.
Yuan Liu
make[2]: Leaving directory
Leo Ann Boon wrote:
Eric ManxPower Wieling wrote:
You should not have quotes in Caller*ID info. MOST devices will just
ignore the quotes, but a few will refuse to accept Caller*ID with
quotes in it. At least one revision of SIP firmware for Cisco phones
does this.
Thanks for the heads up.
I want to essentially transplant my existing Asterisk server to a new
machine, and take the old sever out of service.
Assuming I install Asterisk on the new machine, does anyone know what files
I would have to copy over? What comes to mind are the *.conf files in
/etc/asterisk, as well as the
I have found that if you don't have the minimum balance required for the
voipjet premium server, you get the circuits busy message, you might
want to check your balance.
On 1/30/07, Alejandro Lengua [EMAIL PROTECTED] wrote:
Hello, we have this problem with Trixbox 1.23
I have created an
Thanks for your reply Anselm. I'll play it with tomorrow.
Let me ask a related question. I also have to assign a calleridnum
(number) of 'Internal' to each extension dialed on an internal to
internal call. They would all have 4 digit calleridnum in the range 4??
( or _4xx in dial plan form
Hi All,
I use the Asterisk Manager Interface to redirect the channels.
There have two channels :
SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456)
SIP/612-5456 [EMAIL PROTECTED]:10 Up Dial(SIP/[EMAIL
PROTECTED]
Then I send a
Assuming some defaults... your results may vary.
/etc/asterisk = Configs
/var/spool/asterisk = Voicemail, other spool files
/var/lib/asterisk = Licenses (G729 for example), stock sounds, astdb,
etc
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Thu, Feb 01, 2007 at 03:32:22PM -, Neil Tancock wrote:
Hi, I have /dev/ttyS0 set up with a serial cable so a call centre system can
pick up CallerID. How can I redirect the log output of asterisk to
/dev/ttyS0 or /dev/console?
If you can't simply put /dev/ttyS0 or /dev/console as a log
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