nik600 wrote:
In the last months i've developed a web application for the use of an
asterisk call center.
Yuo can
- make calls from a web interface
- login/logout in queue
- view members logged in a queue
- view callers queued in a queue
- pickup a callers from a queue
What is license of this
Hello all,
We had an experimental system which works on OpenLine4 telephony card
and Asterisk 1.0.9. Customer
asked to upgrade Asterisk to 1.4, then we found our problem:
At first Asterisk 1.4 does not compile chan_vpb.so. The problem is it
tries to compile chan_vpb.cpp to chan_vpb.o and
Hello.
I have a TE212 configured in E1 mode.
This is shown in a cat /proc/zaptel/2 and 3 (where the card is configured):
cat /proc/zaptel/2
Span 2: TE2/0/1 T2XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 RED NOTOPEN
25 TE2/0/1/1 Clear
26 TE2/0/1/2 Clear
27 TE2/0/1/3 Clear
Thank you, that is exactly what I needed.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp
Sent: Tuesday, February 27, 2007 11:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Do I understand
thank you all,
temporarly the problem is solved
i v set my zaptel.conf by modifing span line
span=1,0,0,ccs,ami
the yel/ok alarm was caused by ',crc4'
now when i m running zttool i get OK and the led comes green :)
i run cat /proc/zaptel/1 i get:
*
Span 1: WCT1/0 Digium Wildcard TE110P
I have a client intersted in a system, but they have an ISDN30 line - the
down side is that I've not done any before...
Now, I've no reason to think it won't work, but as going on-site with a
new card and no first-hand knowledge isn't particularly wise, I'm
wondering about the best way to
Dears
Please how can create an independent group of users on asterisk ,in which
user on group A cant dial user on group B.
Thanks
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
Postal Code: 1102-2080
Tel: (961) 1- 868 686
Fax :(961)
On 2/28/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
nik600 wrote:
In the last months i've developed a web application for the use of an
asterisk call center.
Yuo can
- make calls from a web interface
- login/logout in queue
- view members logged in a queue
- view callers queued in a
Hi Gordon
'Fraid I don't have a line you can 'play with' so to speak! However,
firstly, I have installed several E1 based Asterisk systems both in UK
and elsewhere, and apart from a few telco issues in a Latin American
country, it just works.
You can do as you suggest, i.e. have two
Dear Khaled,
The way I would go to do so is to put the group of people you want to
call each other in one context and the other people in an another
context. That's one way to do so.
Thx
MAG
Khaled wrote:
Dears
Please how can create an independent group of users on asterisk ,in
which user
Hmm, I am in England too on the East London corner. Tell me what you are about
to do with the
ISDN30 in relation with your TE110p? It is not clear how you would set this up
based on your
e-mail. Be specific, I might be able to help you. Explain more how your client
wished to have you
work on.
Hi,
What is the main purpose of this setup by the way?
Hi Gordon
'Fraid I don't have a line you can 'play with' so to speak! However,
firstly, I have installed several E1 based Asterisk systems both in UK
and elsewhere, and apart from a few telco issues in a Latin American
country, it just
On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:
Hi,
What is the main purpose of this setup by the way?
For me? To provde a client with a VoIP capable PBX in their office to
replace their current steam driven PBX...
(And hopefully to earn a few ££in the process!)
I have a lot of
On Wed, 2007-02-28 at 13:05 +0100, nik600 wrote:
On 2/28/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
nik600 wrote:
In the last months i've developed a web application for the use of an
asterisk call center.
Yuo can
- make calls from a web interface
- login/logout in queue
On Wed, 28 Feb 2007, tim robinson wrote:
Hi Gordon
'Fraid I don't have a line you can 'play with' so to speak! However, firstly,
I have installed several E1 based Asterisk systems both in UK and elsewhere,
and apart from a few telco issues in a Latin American country, it just works.
On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:
Hmm, I am in England too on the East London corner. Tell me what you are
about to do with the ISDN30 in relation with your TE110p? It is not
clear how you would set this up based on your e-mail. Be specific, I
might be able to help you. Explain
Cant take the credit. I didnt create it. as far as a phone you can go with 2
things. either use chan_cellphone and use bluetooth or you can go with a cell
phone dock (as some one mentioned earlier). if you are using the cellular
docking station that you dont need to worry about chan_cellphone.
And how would you be able to make a test telephone call with ISDN30 when you
don't have an E1
link? You gonna have two asterisk box connected peer to peer and have the other
as the master and
the other as a slave. Or the other way of saying that your other asterisk box
generates the
This is perhaps an architectural issue. I suppose you are planning to interface
the shining
asterisk-based VOIP box with their millenium old pabx? What is the brand
name of their old PABX
machine though?
In my humble opinion, your setup to connect two asterisk box peer to peer using
two TE110p
Gordon Henderson wrote:
On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:
Hi,
What is the main purpose of this setup by the way?
For me? To provde a client with a VoIP capable PBX in their office to
replace their current steam driven PBX...
(And hopefully to earn a few ££in the process!)
Hi Tim,
What is the brand name of your existing PABX?
Hi Gordon
'Fraid I don't have a line you can 'play with' so to speak! However,
firstly, I have installed several E1 based Asterisk systems both in UK
and elsewhere, and apart from a few telco issues in a Latin American
country, it just
Hi Jake,
Perhaps you can add
NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI
Hope that helps.
Best Regards,
Joanna
On 2/28/07, Kuba [EMAIL PROTECTED] wrote:
After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very
strange
This guy could save his brain cells by just getting his shining good 'ol voip
pabx box interface
directly with the existing pabx of his client.
I just wonder what is the brand name of that existing pabx?
Gordon Henderson wrote:
On Wed, 28 Feb 2007, [EMAIL PROTECTED] wrote:
Hi,
What is the
Hi Cameron,
Why not automatically set the language that should be use at the beginning.
Set(LANGUAGE()=nz)
Hope that helps.
Best Regards,
Joanna
On 2/28/07, Moises Silva [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage
There you can found how
What is the make of the existing pabx? Be aware that if it is an older pabx
the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jake,
Check to make sure you have the sound files for whatever audio format
(gsm.wav, etc) that you are using. I don't remember the details, but Asterisk
quit including the sound files in the base distribution to minimize the size of
the download. Then, in a later version, they have a
If the temp message exists then that will play. The user has to log
into the mailbox (app_voicemailmain) and select 0 for mailbox options,
and delete the temp message. Or you could do it using the shell.
On 2/27/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi:
This should be easy. I'm running
[EMAIL PROTECTED] wrote:
And how would you be able to make a test telephone call with ISDN30 when you
don't have an E1
link? You gonna have two asterisk box connected peer to peer and have the other
as the master and
the other as a slave. Or the other way of saying that your other asterisk box
On Wed, 28 Feb 2007, asterisk wrote:
What is the make of the existing pabx? Be aware that if it is an older pabx
the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.
This is another issue I didn't mention (saw no need!)
Gordon Henderson wrote:
On Wed, 28 Feb 2007, asterisk wrote:
What is the make of the existing pabx? Be aware that if it is an older
pabx
the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather than true isdn30e.
This is another issue I
The nature of the existing PABX to be interface is of prime importance. Before
you say it will
works, please check the latest postings.
[EMAIL PROTECTED] wrote:
And how would you be able to make a test telephone call with ISDN30 when you
don't have an E1
link? You gonna have two asterisk
On Wed, 28 Feb 2007, Julian Lyndon-Smith wrote:
Gordon Henderson wrote:
On Wed, 28 Feb 2007, asterisk wrote:
What is the make of the existing pabx? Be aware that if it is an older
pabx
the signaling over the isdn30 could actually be dass2 which is not
compatable with asterisk cards, rather
[EMAIL PROTECTED] wrote:
The nature of the existing PABX to be interface is of prime importance. Before
you say it will
works, please check the latest postings.
But it isn't because they are changing protocols/signaling to something
that is Asterisk compatible. Please check the latest
a disable mode for reading general stuff and an enable mode for
configuration related tasks I think would be a very nice feature fro
asterisk to have. especially in this situation, some type of copy
running-config startup-config would have proven useful. lucky for me my
screw up wasn't on a
On 28 Feb 2007, at 06:53, Gordon Henderson wrote:
On Wed, 28 Feb 2007, Julian Lyndon-Smith wrote:
Gordon Henderson wrote:
On Wed, 28 Feb 2007, asterisk wrote:
What is the make of the existing pabx? Be aware that if it is an
older pabx
the signaling over the isdn30 could actually be dass2
Thank you all. Was a signaling issue.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Wednesday, February 28, 2007 12:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TE110P: Error == Asterisk died with code
1.
Joanna Liza Mariazeta wrote:
Perhaps you can add
NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI
Hi Joanna,
I added that, but it looks like it does nothing :(. I don't see any
status after Playback in the CLI.
All I get is:
-- Executing
This can happen if you have a Digium card (maybe Sangoma too) in the
system that is configured, but has no actual line plugged into it. I
don't know if this applies to analog, but I know it applies to T-1/PRI/E-1
Kuba wrote:
Joanna Liza Mariazeta wrote:
Perhaps you can add
Hello, we are setting up another system that will run either 1.2.4,
the latest version of 1.2 or 1.4. We have not yet decided on the
version.
Anyhow, this is a higher volume system (dual processor) which will
handle 30-50 simultaneous calls with 60 to 100 simultaneous channels
lit up. Most calls
Hi all,
I have some questions about h323. Is it mandatory to install a oh323 or I can
do h323 calls without patching or adding any new drivers ti asterisk?
I did compile the asterisk with channel driver chan_h323 but it still gives me
this error:
[Feb 28 18:12:58] WARNING[1902]: app_dial.c:1081
Dear all,
I've noticed that when I have a phone registered in Asterisk, and then I
register another phone with the same user, the sip show peers in the
CLI shows that Asterisk replaced the IP of the first phone by the IP of
the last one registered for that user. Consequently, if someone calls
28th February
I am working with Asterisk 1.2.15. I have configured sip.conf for two soft
phones (I am using Xlite).I have installed the Bluez stack and so far, i
manage to make a phone call from a soft phone to a GSM network. However, i
have an audio problem. The soft phone can be heart by
Create a different user for each phone and create a ring group with the
phones that you want to ring.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Wednesday, February 28, 2007 9:15 AM
To: Asterisk Users Mailing List -
Ricardo Carvalho wrote:
Dear all,
I've noticed that when I have a phone registered in Asterisk, and then I
register another phone with the same user, the sip show peers in the
CLI shows that Asterisk replaced the IP of the first phone by the IP of
the last one registered for that user.
Florea Igor wrote:
Hi all,
I have some questions about h323. Is it mandatory to install a oh323 or I can
do h323 calls without patching or adding any new drivers ti asterisk?
I did compile the asterisk with channel driver chan_h323 but it still gives me
this error:
[Feb 28 18:12:58]
Can't I register multiple phones with the same user/password? That's
what I pretend to do, not ring groups...
Thanks,
Ricardo.
Azfhasterisk wrote:
Create a different user for each phone and create a ring group with the
phones that you want to ring.
-Original Message-
From: [EMAIL
Anyone have any idea if there is some sort of limitation to the number
of SIP or IAX end points which can register to an Asterisk system
(2.8Ghz dual processor, 2GB ram) while also handling 30-50
simultaneous calls without getting into trouble?
Of course the 30-50 simultaneous calls end up being
I'm having a strange issue. My voicemail is working fine, however,
any time I try to access it via one of my analog phones that are
connecting to Asterisk via a Mediatrix 1124... the voicemail system
complains I've entered the wrong password.
There is about a 15 second pause between when I
Ricardo Carvalho wrote:
Can't I register multiple phones with the same user/password? That's
what I pretend to do, not ring groups...
No, you cannot register multiple phones with the same user/password.
___
--Bandwidth and Colocation provided by
On Wed, Feb 28, 2007 at 10:47:48AM -0500, Jeronimo Romero wrote:
Thank you all. Was a signaling issue.
And for the benefit of those who will read the archive: how have you
debugged it? how have you resolved it?
--
Tzafrir Cohen
icq#16849755
On Wed, Feb 28, 2007 at 10:00:10AM -0600, voiplist wrote:
Hello, we are setting up another system that will run either 1.2.4,
the latest version of 1.2 or 1.4. We have not yet decided on the
version.
Anyhow, this is a higher volume system (dual processor) which will
handle 30-50
Iban Lopetegi Zinkunegi wrote:
28th February
I am working with Asterisk 1.2.15. I have configured sip.conf for two
soft phones (I am using Xlite).I have installed the Bluez stack and so
far, i manage to make a phone call from a soft phone to a GSM network.
However, i have an audio problem.
Hi List,
I put this in to my voicemail.conf as per the wikki and the users are still
getting the emails from the root account. Any ideas on what it can be ?
I have: mailcmd=/usr/sbin/sendmail -v -t -f [EMAIL PROTECTED]
___
--Bandwidth and Colocation
voiplist wrote:
Anyone have any idea if there is some sort of limitation to the number
of SIP or IAX end points which can register to an Asterisk system
(2.8Ghz dual processor, 2GB ram) while also handling 30-50
simultaneous calls without getting into trouble?
Of course the 30-50 simultaneous
Too bad... Thanks for all replays.
Regards,
Ricardo.
Eric ManxPower Wieling wrote:
Ricardo Carvalho wrote:
Can't I register multiple phones with the same user/password? That's
what I pretend to do, not ring groups...
No, you cannot register multiple phones with the same user/password.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
; Who the e-mail notification should appear to come from
;[EMAIL PROTECTED]
Dovid B wrote:
Hi List,
I put this in to my voicemail.conf as per the wikki and the users are
still getting the emails from the root account. Any ideas on what it can
be
I goofed that up on my dCAP exam. Spent 20 valuable minutes trying to
fix it!
Eric ManxPower Wieling wrote:
This can happen if you have a Digium card (maybe Sangoma too) in the
system that is configured, but has no actual line plugged into it. I
don't know if this applies to analog, but I
Yuan,
It looks like you are getting 202 for SIP Request method MESSAGE. The 202
response is processed properly. Need to see the message fully. You can
capture sip debug if you don't have ethereal. This will provide more detail
call flow.
.{N/MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
That only helps on the SIP side.. :(
Although it would help some..
Before we go making changes, we are really just trying to determine
what the cause is.
On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
voiplist wrote:
Anyone have any idea if there is some sort of limitation to the number
From: voiplist [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 10:54:30 -0600
Anyone have any idea if there is some sort of limitation to the number
of SIP or IAX end points which can register to an Asterisk system
(2.8Ghz dual processor, 2GB ram) while also handling 30-50
simultaneous calls without
After testing some AGI's, I noticed several extra Asterisk processes. They
are not zombies, but can't be killed by safe_asterisk. Nor will they die
when CLI issues stop now. Then I read that each AGI spawns a separate
Asterisk process. But all my AGI calls have apparently completed
Yuan LIU wrote:
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 10:57:43 -0600
Ricardo Carvalho wrote:
Can't I register multiple phones with the same user/password? That's
what I pretend to do, not ring groups...
Ricardo,
Any particular reason for not using ring
you using dynamic dns?
On 2/28/07, voiplist [EMAIL PROTECTED] wrote:
That only helps on the SIP side.. :(
Although it would help some..
Before we go making changes, we are really just trying to determine
what the cause is.
On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
voiplist wrote:
any dns in the sip channel could do this not only dynamic
On 2/28/07, voiplist [EMAIL PROTECTED] wrote:
That only helps on the SIP side.. :(
Although it would help some..
Before we go making changes, we are really just trying to determine
what the cause is.
On 2/28/07, Steve Totaro [EMAIL
When we do SIP - SIP with asterisk 1.2, we do NOT experience this.
Polycom 501s, the instant you hit Send on the phone or the digit map
times out, the target phone rings AND you hear ringback. it's instant,
so I would guess this would be configuration on your end. back to the
digit map
Yuan LIU wrote:
Doesn't seem to happen in TDM400P and X100P cards, though. Could it be
some feature configured in your particular card?
Notice just after my name:
P.S. Polycom Soundpoint 501, *TDM* w/ 4xFXO, Asterisk 1.2.13
As El ManxPower mentioned, have you tried using ZapBarge to detect
As far as I can tell, the only way to do this using Polycom soundpoint
phones and NOT asterisk's built-in blindxfer function, is to hit their
Transfer button first, and then the Blind softkey that appears on the
screen. Then continue as normal; dial the number and hit Send I
believe. If you
But just to handle 10 simultaneous calls, you probably don't even need 1
GHz!
Matt Richards wrote:
I don't see any reason why a single server wont handle 700 phones as
long as its powerful enough.
I would think that anything over 1GHz should be fine maybe less :)
Matty.
Jerry Geis wrote:
I
Linksys SPA400 is a 4 port FXO gateway.
Cameron
___
New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at
the Yahoo! Mail Championships. Plus: play games and win prizes.
Eric ManxPower Wieling a écrit :
Yuan LIU wrote:
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Date: Wed, 28 Feb 2007 10:57:43 -0600
Ricardo Carvalho wrote:
Can't I register multiple phones with the same user/password? That's
what I pretend to do, not ring groups...
Ricardo,
Any
Sorry for disturbing, but I sent some messages today and I am not seeing
them on this list.
Can sombody tell me, in case this message appear on the list.
Thank you
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
On Wed, Feb 28, 2007 at 10:56:14AM -0800, Yuan LIU wrote:
After testing some AGI's, I noticed several extra Asterisk processes.
An agi script is run by the same user running asterisk, but is not
asterisk: it is a different program. What is the command name on those
scripts?
They
are not
Outbound calls on my Telus PRI aren't taking the Name portion of the
callerID. I've looked at the logs, and it is being set (see below), but
the PRI debug output doesn't show the name being sent anywhere. As a
result, received calls always display from Unknown (or just the number).
Is there some
Excuse the ASCII diagramme - you will need a fixed width font to
understand it.
-- --- --- -
| A | == | NAT | === === | NAT | == | B |
-- ---| |--- -
I tried that as well and I get the same problem. Can it be an issue with
sendmail ?
- Original Message -
From: Jacob Helwig [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 28, 2007 7:37 PM
Bayrouni wrote:
Sorry for disturbing, but I sent some messages today and I am not seeing
them on this list.
Can sombody tell me, in case this message appear on the list.
Thank you
___
--Bandwidth and Colocation provided by Easynews.com --
I am running asterisk 1.4. I have 2 NICS in the my server.
Over the last couple days I have lost internet connection a couple times
(lets not go there)...
Anyway everytime I loose internet my internal phones loose registration.
The phones are not using DNS they are coded to the servers IP.
Is there a way to send DTMF's to a channel before the call is answered?
For example, send DTMF's to a SIP channel after the 180 Ringing or 183
Session Progress have been received from it, but before the 200 OK, or
in the E1 side, after the Q931_ALERTING is received, but before the
a) to what extent Asterisk can manage everything necessary to allow
machines A and B to communicate if they were SIP phones. Is it
possible to go for a setup with the firewalls/NAT devices as shown
If the asterisk machine isn't NATed you shouldn't have a problem at all.
If you're using
On 2/28/07, Dovid B [EMAIL PROTECTED] wrote:
Cant take the credit. I didnt create it. as far as a phone you can go
with 2 things. either use chan_cellphone and use bluetooth or you can go
with a cell phone dock (as some one mentioned earlier). if you are using the
cellular docking station that
Try latest zaptel 1.2 from svn. I made a fix that should reduce stack
usage.
Matthew Fredrickson
On Feb 27, 2007, at 4:49 PM, Marco Parisotto wrote:
Hi Michelle,
actually, I didn't try it...
The server is a HP Proliant ML150T G3.
Currently I'm not in the condition to follow your
Hi,
I am using asterisk's SMS functionality for sending messages. Most of
the time it works without problems (as in situation 1) but sometimes
something seems to go wrong with the transmission (as in situation 2). I
am using Deutsche Telekom, Germany's main TELCO, so I suppose the
problem
On Wednesday 28 February 2007 3:45 pm, Alan Chandler wrote:
I am trying to setup an arrangement whereby clients on machines A, B, C
and D can talk to each other on Softphones. A,B,C are are all Windows
XP machines, machines D and S are linux. This has to include A talking
to B and ultimately
Asterisk gets very upset when DNS is down. You might want to confirm
that /etc/hosts has entries for ALL interfaces in that system. That
should cause the system to not issue a DNS request to resolve local IPs.
Jerry Geis wrote:
I am running asterisk 1.4. I have 2 NICS in the my server.
Over
I there,
I have some trouble to do working caller id name for outgoing calls on
the PRI we just installed. Caller id name work on incoming calls only.
Caller id number work on incoming and outgoing calls.
The provider, Goup Telecom, said that's in what i'm sending. They said
that I send the
I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf
exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)
I understand what is going on with this line but I don't know where
in the extensions.conf file to put
Hello @List,
i'm using a Cisco 7970 / 7914 phone with Asterisk 1.4 sccp
part of my sccp.conf
type= 7914 (Cisco 7970 with 7914 phone extension)
autologin = 117
description = Test
speeddial = 10,Test (10),[EMAIL PROTECTED]
my speeddial line is for internal call
Hey everyone,
I came across a situation where I needed to use CDP to advertise a
voice vlan to Polycom/Cisco (and other CDP capable phones) without a
Cisco switch.
Sure I have Cisco switches in places but I like my Polycoms to work
out of the box and it isn't always practical to purchase a
Hi,
I'm having a similar problem, but the name isn't even appearing in debug
output of PRI (see my other post about this). My PRI is with Telus, and
they told me that NI-2 doesn't support CallerID name function, only
NI-1.
Andrew
-Original Message-
From: [EMAIL PROTECTED]
nice one.. we have rogers and primus.. ni'2 and same..
let me know if this ni2 and ni1 thing is crap or not
On 2/28/07, Webster, Andrew [EMAIL PROTECTED] wrote:
Outbound calls on my Telus PRI aren't taking the Name portion of the
callerID. I've looked at the logs, and it is being set (see
Hello,
I created a new field named pre_dst of type varchar(80) exactly like dst
field in cdr table.
In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
and when I call, all goes fine except that pre_dst has always NULL value
in cdr.
Do you know why?
Is something wrong I did?
We have decided to allow our tech's to do support for non-clients of
voicemeup.com
You can head to http://support.voicemeup.com/ and one will be in touch 8 to
6pm business hours.
3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc.
--
Mike
Sales Manager
Answers in-line...
Hope this helps!
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alan
Chandler
Sent: Wednesday, February 28, 2007 3:46 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie Planning Help
snip
--
On Wednesday 28 February 2007 21:26, Andrew Kohlsmith wrote:
On Wednesday 28 February 2007 3:45 pm, Alan Chandler wrote:
I am trying to setup an arrangement whereby clients on machines A,
B, C and D can talk to each other on Softphones. A,B,C are are all
Windows XP machines, machines D and
On Wednesday 28 February 2007 21:08, mail-lists wrote:
a) to what extent Asterisk can manage everything necessary to allow
machines A and B to communicate if they were SIP phones. Is it
possible to go for a setup with the firewalls/NAT devices as shown
If the asterisk machine isn't NATed
Hi,
On Wed, 2007-02-28 at 23:43 +0100, Bayrouni wrote:
Hello,
In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})
and when I call, all goes fine except that pre_dst has always NULL value
in cdr.
Do you know why?
Is something wrong I did?
As far as I know, custom fields
try not using dst.. maybe its a regex on te fieldname that matches for
reserved keywords..
try pre_dest instead
On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote:
Hello,
I created a new field named pre_dst of type varchar(80) exactly like dst
field in cdr table.
In the dialplan I put:
exten =
Does the ${BLACKLIST()} function allow for values other than 1 to be
returned and if so how can I use that is the AEL? Can I use the
function in a switch statement?
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Linux Home Automation Neil Cherry [EMAIL PROTECTED]
http://www.linuxha.com/ Main site
try putting near the exten = 1000,1,dial stuff
On 2/28/07, Chris Griffin [EMAIL PROTECTED] wrote:
I've installed Sven Slezak's Notify module. He gives the follow as an
example line to put into extensions.conf
exten = s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/
sunnybook)
I
You could try Fast agi.. then i think master agi deamon runs from services
and replies to requests by including sub scripts.
however i do see some connect failures sometimes...
On 2/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Feb 28, 2007 at 10:56:14AM -0800, Yuan LIU wrote:
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