Kristian Kielhofner wrote:
Hello everyone,
As I promised at eTel last week, I have finished up work on my
Asterisk Native Sounds project. Here's a little diddy from
astlinux.org:
---
Asterisk Native Sounds are a collection of audio prompts for
Jerry Geis wrote:
[Switching to Thread -1208027456 (LWP 24132)]
0x080d8471 in el_init (prog=0x8103615 asterisk, fin=0x995740,
fout=0x407c, ferr=0x407c) at el.c:67
67 if (el == NULL)
(gdb) where
#0 0x080d8471 in el_init (prog=0x8103615 asterisk, fin=0x995740,
Hi,
i was reading about connecting a cellular phone over chan_bluetooth.
I was wondering, if one is able then to make/receive concurrent calls or if
you can make just one at time?
Peter
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On 2/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
linksys 942 doesnt look very competetive anyway.
(2 10mbit ethernet ports? who is linksys kidding?)
Ouch, indeed, that's really working against the other features this
phone has. So the advantage over a 941 is also gone now.. :(
Thanks for
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner
Sent: 06 February 2006 17:48
To: Discussion of AstLinux - Asterisk on Compact Flash; Asterisk-
[EMAIL PROTECTED]; [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject:
Hello all,
if I try to call from one phone on the internal S0 to another on the same S0
using zaphfc, the bus is hung up. The called phone is ringing, but I can't
talk from one phone to the other. The error I get is:
-- Executing Dial(Zap/2-1, ZAP/1/55|15|tr) in new stack
-- Requested
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
[internal]
exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD)
exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx)
Hi,
I'm using asterisk 1.2.1 with 3 phones: A,B and C
When I call B from A and then I transfer the call from B to C, I get
only a cdr record from A to C.
Is this the right Asterisk behaviour? Shouldn't be 2 records inside cdr,
one from A to B and then one from B to C?
TIA
Giorgio Incantalupo
--
Bayrouni
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Am Dienstag, 7. Februar 2006 09:38 schrieb Sven Fischer:
Hello all,
if I try to call from one phone on the internal S0 to another on the same
S0 using zaphfc, the bus is hung up. The called phone is ringing, but I
can't talk from one phone to the other. The error I get is:
-- Executing
Greetings all,
I'd like to start implementing a private DUNDi peering group between one of
our asterisk servers hosted at a datacentre and the various asterisk boxes
sitting at clients' premises.
On most of the clients' boxes the dialplan will have an [in-pstn] section
containing the various
The problem is not about * and verbosity and * logs;
The problem is only abot AGI scripts written in php.
The log affected is /var/log/messages, not an * log file.
I think that the problem could arise due to some configuration of my
operating system AND php configuration:
I am using openSuse 10.0,
Hello,
I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring
but I don't hear the caller and the caller doesn't hear me (all IP Phones
have the same problem).
This problem appear also if the call is directly send to the second E1 of
the digium card who is connected to an
I solved my problem.
I modified
/var/lib/asterisk/agi-bin/phpagi.php
on lines 218
// foreach(explode(\n, print_r($this, true)) as $line)
syslog(LOG_WARNING, $line);
and 716;
//@syslog(LOG_WARNING, $msg);
I left the syslog on the line 1697, which is part of the error handler for
phpagi.
Francesco Peeters (Asterisk) schrieb:
They have several ISDN BRI connections, most of which will be dropped.
Only one will be retained, for 2 reasons:
1) It has the ADSL link
2) The number has been the main contact number for over 20 years.
In germany you could move that number to a VoIP
Hi,
Have you try to search on eBay? I found some welltech devices for sale.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph
Sent: Tuesday, February 07, 2006 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Not sure answer your question? Try to write some html code and let user
register the username password online.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
Sent: Tuesday, February 07, 2006 7:31 AM
To: asterisk-users@lists.digium.com
Hi,
In my remember. The AAH will automatic config the TDM04B
card. try the CLI command"zap show channels"
regards,
kevin
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nelson
GranadosSent: Tuesday, February 07, 2006 7:18 AMTo:
asterisk-users@lists.digium.comSubject:
Hmm see if this works,
Extensions.conf
[ring-30]
exten = s,1,ringing();
exten = s,2,wait(30);
exten = s,3,hangup();
...
then in your Dial(SIP/1SIP/2Local/[EMAIL PROTECTED]);
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian J.
Hi All
Is there any special configuration needed to send and receive faxes on
an ATA device?
I am using G711.a with a Grandstream Handytone 486. I can send faxes
from a fax machine on the ATA, but receiving doesn't work. I get the
fax signal, but it just doesn't continue. The LAN is used
On Tue, February 7, 2006 9:53, Sven Fischer said:
Am Dienstag, 7. Februar 2006 09:38 schrieb Sven Fischer:
Hello all,
if I try to call from one phone on the internal S0 to another on the
same
S0 using zaphfc, the bus is hung up. The called phone is ringing, but I
can't talk from one phone
On Tue, February 7, 2006 11:16, Peer Oliver Schmidt said:
Francesco Peeters (Asterisk) schrieb:
They have several ISDN BRI connections, most of which will be dropped.
Only one will be retained, for 2 reasons:
1) It has the ADSL link
2) The number has been the main contact number for over 20
Hi,
The magical incarnation is:
0. Fix /etc/udev./rules.d/50-udev.rules if it has been banged up by
yum.
1. rebuild_zaptel
2. reboot
3. genzaptelconf
4. reboot
That should do the job. There are more elegant ways than reboot'ing ...
but wtf.
BTW. [EMAIL PROTECTED] 2.5 is out now. It includes some
Martin Joseph wrote:
Any feedback on this brand and in particular on doing business with
WelltechUSA?
Don't worry they're OK.
I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I
am hoping to replace the near worthless Grandstream HT-488.
I'd personally recommend to get
Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config.
Hans
Garth van Sittert schrieb:
Hi All
Is there any special configuration needed to send and receive faxes on
an ATA device?
I am using G711.a with a Grandstream Handytone 486. I can send faxes
from a fax machine on
Jerry Geis wrote:
/ [Switching to Thread -1208027456 (LWP 24132)]
// 0x080d8471 in el_init (prog=0x8103615 asterisk, fin=0x995740,
// fout=0x407c, ferr=0x407c) at el.c:67
// 67 if (el == NULL)
// (gdb) where
// #0 0x080d8471 in el_init (prog=0x8103615 asterisk,
I am using alaw and I have already enabled the pass through. Does alaw
and ulaw work?
I can fax out, but not receive faxes.
Garth
Johann Steinwendtner wrote:
Enable pass thru fax mode on the HT486, or enable ulaw in your SIP
config.
Hans
Garth van Sittert schrieb:
Hi All
Is there any
On Mon, 2006-02-06 at 11:48 -0600, Kristian Kielhofner wrote:
Hello everyone,
As I promised at eTel last week, I have finished up work on my
Asterisk Native Sounds project. Here's a little diddy from astlinux.org:
Which format would be best/cpu-easiest on an analog channel like the
Garth,
this is my sip-configuration for a fax machine at a AT386
; SIP Accounts Analog devices like Faxmachines
[analogdefaults](!)
type=friend
host=dynamic
dtmfmode=info
disallow=all
allow=gsm
allow=alaw
allow=ulaw
[222](analogdefaults)
context=sip-ol
callerid=Fax 222
username=222
On 2/6/06, Mark Phillips [EMAIL PROTECTED] wrote:
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get the system prompts to then speak
the selected langauge. For example, a caller has selected Spanish and so
is
ulaw was neccessary when pass through was disabled. What does a sip
debug tell you ?
Hans
Garth van Sittert schrieb:
I am using alaw and I have already enabled the pass through. Does alaw
and ulaw work?
I can fax out, but not receive faxes.
Garth
Johann Steinwendtner wrote:
Enable pass
Hello,I would like to know if it's possible to configure asterisk to play something nice to a person calling me during week-ends when there is noone available at the phone and switch back to normal calls receiving on Monday morning. Please help.
Thanks.
Any More news on this from Kevin ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hulber
Sent: Saturday, February 04, 2006 8:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DTMF Sporadicaly Being
When I doing hangup the zaptel channel, Asterisk show this message: Feb 7 12:14:44 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 4: Red Alarm Feb 7 12:14:44 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 4 Feb 7 12:14:44
Hi List,
Do you know if there are any plans to improve i18n for Asterisk? The
current i18n way of doing it with asterisk is very limited and most of
the time does not work.
For example, take voicemail:
message received at seven 30 am might sound good in English.
But:
message recu a sept
Yes. Google GotoIfTime. I use this to not ring our phones during
the day (we're night people), you can just as easily set it up to play
a message during times that you're closed and send directly to
voicemail (you can specify certain times of the day on certain days,
or whole days such as
Hello,
I'm setting up a fax transmition using automated dial out
(http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out).
When nobody is talking in the moment of faxing, everything goes ok, fax
is successfully delivered (even multipaged). But when there is a
concurrent call on
I thought more about this, perhaps you were instead asking if you
could have multiple phones setup, and have multiple calls processed?
It should work, I haven't attempted it, so I'm not sure if
chan_bluetooth will handle multiple phones at the same time, but
otherwise it should be fine. Just
Richard Amerman wrote:
Doug,
Can you provide any information on how you deployed that card
into your setup? If it works for you we could put up a page on voip-info.org
Richard
I currently have 13 incoming analog centrex lines plugged into an Adit
600 Channel bank. The
Hello,
Thoughts inline
On Tue, Feb 07, 2006 at 07:43:59AM -0600, Joseph Tanner wrote:
I thought more about this, perhaps you were instead asking if you
could have multiple phones setup, and have multiple calls processed?
It should work, I haven't attempted it, so I'm not sure if
Kevin Collins wrote:
Any More news on this from Kevin ?
The only news is that I have not had time to work on it since last week.
However, this is the development trunk. You should _not_ be running it
in production, and realistically there is no reason to be discussing
issues with it on this
Hi,
People here often work on 2-3 places (office 1, office 2 and home).
I would like to give them 1 extension (XXX) and to ask them to
'register' the phone they use at a certain moment.
The idea is that, when you need someone, just dial XXX and the
phone near him (in Office 1, Office 2 or at
I have brought down the server to single user mode.
deleted the source for asterisk,libpri and zaptel.
recompiled everything
rebooted the machine...
Things seems to be behaving better now.
Lets hope it stays that way.
Jerry
Jerry Geis wrote:
/ [Switching to Thread -1208027456 (LWP 24132)]
//
More than that, in their fine print some only claim to pass maybe two or
three of the tests. There is nothing that defines what you must achieve
before you can claim G.168-2002 compliance.
Well, isn't that just wonderful :-) Standards are amazing things, from a
marketing perspective
Hi,
i use in my
extensions.conf a testline for an internal test :
exten =
10,1,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3)
When i call 10,
Asterisk answer and i see in the CLI, that MP3player works without problems -
but i can't hearthe soundat the phone
?
Where is the
Hi all,
I hope you can help me.
I have connected a E1 controller to a PBX, the E1 is only using timeslots
1-4. What I want to do is to pass those timeslots over a HDLC (Clear-Channel)
link to a remote location.
Of course, at the remote location I want to do exactly the same, receive
Hello everybody! I've seen that you can connect your cellphone via
bluetooth, but I've a Motorola V300 and it doesn't have that feature,
so I wish to connect it via USB cable, is it pissible con use my
cellphone with asterisk like that? I 've not been able to find
information on how to do this,
Rich Adamson wrote:
More than that, in their fine print some only claim to pass maybe two or
three of the tests. There is nothing that defines what you must achieve
before you can claim G.168-2002 compliance.
Well, isn't that just wonderful :-) Standards are amazing things, from a
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get the system prompts to then speak
the selected langauge. For example, a caller has selected Spanish and so
is routed to the Spanish part of the IVR. At some point
exten =s,1,Gotoiftime(*|sat|*?closed,1)exten =s,2,Gotoiftime(*|sun|*?closed,1)exten =s,3,Gotoiftime(8:00-17:00|mon-fri|*|*?open,1)exten =s,4,Goto(closed,1)
exten =closed,1,Playback(we-are-closed)
exten =closed,2,Voicemail(1000)
exten =open,1,Dial(technology/resource,25,T)
exten
But, AFAIK, when they get to voicemail, the greeting is not based on
the language setting, so you have to record it in those 3 languages,
which makes a pretty long greeting
This is common in Canada which has 2 official languages. The convention here
is to intersperse the secondary language with
You can use the agent channel to have the calls follow the person
around.
Use AgentLogin or AgentCallBackLogin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alex Ongena
Sent: Tuesday, February 07, 2006 9:26 AM
To: Asterisk
Subject:
Also if you have a nice linux script to take out some of the effort that
would be fantastic but if not I am sure the sox man page will help me
out.
Prep your WAV's as 8Khz mono. In a pinch, Windows sound recorder will do.
Then:
GSM:
#/bin/sh
for I in *.wav
do sox $I `basename $I .wav `.gsm
On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote:
I just implemented a system using a TE411P hardware echo cancellation
card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as
I always have. To my surprise calls out to the PSTN had a terrible
echo. 1 - 2 second delay, and quite
On Feb 6, 2006, at 3:57 PM, Tim Connolly wrote:
I wonder if Digium has any intentions of fixing this. I brought this to
their attention shortly after purchasing a pair of TE411's. You can
issue a
loopup on span 2 only to get a message saying looping span1 which is
to
say, a bit scary when
Colin Anderson wrote:
But, AFAIK, when they get to voicemail, the greeting is not based on
the language setting, so you have to record it in those 3 languages,
which makes a pretty long greeting
This is common in Canada which has 2 official languages. The convention here
is to intersperse
I don´t know if the last message was with content. So, I sent again. I have
installed a Digium card TE210P and unicall for use MFC/R2. I think that it´s
all right but I can´t make and receive calls. I´m using asterisk 2.1 with
the patch made by José P. Leitão and the follow libs:
I've read something on connecting a cellphone to asterisk with bluetooth, I'm not really sure about connecting to a usb phone.I think Joseph Tanner can help us out, as he did it with bluetooth.
Truely/
Joe
From: Facundo Ameal [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List -
Thanks for the info! Fortunately, we have something closer to the
latter configuration you described. The PSTN goes directly into an
MX250 as a SIP gateway, and our Asterisk server connects to that. The
MX has a few FXO ports, but we don't want to use them. It doesn't
seem very clean to
Am Dienstag, 7. Februar 2006 12:55 schrieb Francesco Peeters (Asterisk):
On Tue, February 7, 2006 9:53, Sven Fischer said:
Am Dienstag, 7. Februar 2006 09:38 schrieb Sven Fischer:
Hello all,
if I try to call from one phone on the internal S0 to another on the
same
S0 using zaphfc, the
when exactly would you like to stream this "register me" thingy? whenever an employee picks up the phone to dial? or when?
Please specify more.
Truely/
Joe
From: Alex Ongena [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo:
I've used Voicetronix FXO/FXS ports and noted pretty heavy echo on both
short and long runs to other switches. We went through some steps to try to
tune the echo out using some settings on the card, and it helped with some
of the higher frequencies, but the problem still remains for many users.
It's more helpful to learn more about pre-defined variables in asterisk, then you'll be able to develope more complicated agi scrips or dialplan checks, follow the below link:
http://www.voip-info.org/wiki-Asterisk+variables
Truely/
Joe
From: Joseph Tanner [EMAIL PROTECTED]Reply-To: Asterisk
I badly need to get the callerid of the person who hanged up along with the extension dialed, and I need to do it with DeadAGI where channel variables are destroyed,
any ideas?
Or at least someone tells me why my * does not take PGSQL or MYSQL when I try to insert or retrieve data from a DB, as
Not really sure, but once I had a problem when I changed the txgain and rxgain, so set them again to 0.0 and see how it will work.
Truely/
Ammar
From: "Jerome SOUCANY" [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo:
I have the cards set to auto address assignment, but changed it to shelf255d
setting (option 31) still get the same behaviour... is there someplace else
that this can be set?
Thx!
Dan Elder wrote:
Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting
assembly? I can
You know, I'm still a little confused. Kristian, the original poster, said...
I had Allison Smith (the voice of Asterisk) re-record all of the sound prompts
present in Asterisk 1.2.
Was there really an extra 1400 sound files added from Asterisk 1.2 to Asterisk
1.2.4? Sorry, but I'm just not
certainly on his first call, but it should be possible for him to explicitly
'register' and 'unregister'
On Tuesday 07 February 2006 17:06, Joe Tahan wrote:
when exactly would you like to stream this register me thingy? whenever
an employee picks up the phone to dial? or when? Please specify
On 2/6/06, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
I have tried both the stable version ARI-00.04.006 and the development
version ARI-00.05.018 with the same results. I can see call detail
records just fine but I cannot see any voicemail. I am using the
voicemail extension and password to
Dan Elder wrote:
I have the cards set to auto address assignment, but changed it to shelf255d
setting (option 31) still get the same behaviour... is there someplace else
that this can be set?
Option 30 allows to set Module Shelf Address/ID.
Doug
No, what was rerecorded was the sounds that come with the asterisk
package. Digium has another package called asterisk-sounds that has
many additional sounds - that package was not rerecorded.
Douglas Garstang wrote:
You know, I'm still a little confused. Kristian, the original poster,
AnyOne? any help?
As I'm looking at your zapata.conf I recall a problem in receiving dial-outs from a non-asterisk IVR to an * server1 and server1 routs the call to server2 with IAX2 in order to make a final dial command to a ZAP channel, but in server2 cli console I get the error (UNABLE TO
This can easily be accomplished with AMP using the Users and Devices mode.
http://voipspeak.net/index.php?/content/view/49/28/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alex Ongena
Sent: Tuesday, February 07, 2006 8:55 AM
To:
I'm not receving anything from the list, is this a Gmail problem? or
just my account?
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What do you do with the other 15 channels?
your zapata.conf says:
channel = 1-15 ;,17-31 = only 15 first channels on PRI
but your zaptel.conf says:
span=1,1,0,ccs,hdb3
bchan = 1-15, 17-31
You use all 30 channels in Zaptel.conf but only 15 in zapta.conf
I never configured Zap on asterisk and
Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk?
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Kevin,
Sorry for the interruption but I was replying here because the message
thread was on this list. Thanks for being gentle ;-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, February 07, 2006 9:24 AM
To: Asterisk
911 *should*
work on a PRI. If you are getting a hangup and you dont see a valid
hangupcause, it might be best to get your carrier on the line and have them
monitor the circuit while you dial 911. They might be able to tell you what
the problem is.
-MC
From: [EMAIL PROTECTED]
I upgraded to 1.2.4 today and am having issues and can't figure this
out. Here's the bottom part of a gdb and a backtrace. Any
thoughts? May roll back to 1.2.3?
Mark
Reading symbols from /usr/lib/asterisk/modules/app_saycountpl.so...done.
Loaded symbols for
Douglas Garstang wrote:
You know, I'm still a little confused. Kristian, the original poster, said...
I had Allison Smith (the voice of Asterisk) re-record all of the sound prompts
present in Asterisk 1.2.
Was there really an extra 1400 sound files added from Asterisk 1.2 to Asterisk
1.2.4?
Colin Anderson wrote:
Also if you have a nice linux script to take out some of the effort that
would be fantastic but if not I am sure the sox man page will help me
out.
Prep your WAV's as 8Khz mono. In a pinch, Windows sound recorder will do.
Then:
GSM:
#/bin/sh
for I in *.wav
do sox $I
Brian J. Murrell wrote:
On Mon, 2006-02-06 at 11:48 -0600, Kristian Kielhofner wrote:
Hello everyone,
As I promised at eTel last week, I have finished up work on my
Asterisk Native Sounds project. Here's a little diddy from astlinux.org:
Which format would be best/cpu-easiest on an
Alex Barnes wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner
Sent: 06 February 2006 17:48
To: Discussion of AstLinux - Asterisk on Compact Flash; Asterisk-
[EMAIL PROTECTED]; [EMAIL PROTECTED]
Cc: [EMAIL
I have used 911 with PRI with nothing else
configured. Telco had to make changes to their router for DID numbers to call
through.
Adam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail
Sent: Tuesday, February 07, 2006
12:10 PM
To: Asterisk Users
Benoît Mérouze wrote:
Kristian Kielhofner wrote:
Hello everyone,
As I promised at eTel last week, I have finished up work on my
Asterisk Native Sounds project. Here's a little diddy from
astlinux.org:
---
Asterisk Native Sounds are a collection of
I have a
fresh install of AMP. In the AMPortal, Setup, Devices or Users, I get:
Cannot connect to Asterisk Manager with user/password (set respectively)
This module requires access to the Asterisk Manager. Please ensure Asterisk is
running and access to the manager is available.
I
Kirs et al,
I did this already. It's on my website. Your most welcome to use them
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Kristian Kielhofner wrote:
Alex Barnes wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
And (as GSM Restriction) one can do only one call per phone (conferences
and onHold are managed by the GSM-AP).
This was what i was actualy interested in. My idea was, when conferecnces
work, it should be possible to make 2 calls over 1 GSM phone at a time. But
apparently this wont work.
Hi,I am forcing caller ID to be sent to our VoIP provider using the SetCallerID app:exten = _91.,1,SetCallerPres(allowed)exten = _91.,2,SetCallerID(Company Name 5)
exten = _91.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])Ever since I started doing this however, the CDR gets overwritten with this
I dunno about your provider but I know that 2 of my 3 MCI PRI circuits
have no 911 abilities. MCI tells me this is becasue I have no local
dialing plan on them.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Michael Collins wrote:
911 **should** work on a PRI. If you are getting a
One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register = user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
Erm ... sorry. That should read Kris et al
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Mark Phillips wrote:
Kirs et al,
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Signate runs asterisk on a SGI box.
Nothing special, do yourself a favor and just buy the SGI box yourself. In
fact I have 3 SGI boxes for sale. Ill rip off the Signate labels and
sell them to you.
I worked out an asterisk load
balance solution, so I dont need one all powerful PC. I
The same 7 sound file is used to indicate both time and quantity. The
sound file could be easily recorded to say sept heure but then every
time the VM system tells a user that they have 7 messages they'll hear
something like vous avez sept heure notification (excuse my schoolboy
French).
Hi,
I made a simple menu using the Background
application and some wav files. I converted the wav files using
forain*.wav;dosox"$a"-r8000-c1"`echo$a|sed-es/wav//`gsm";done
(from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk)
The first
I've come across this in my dealings with my customers in Toronto. As an
Englishman I find it most infuriating. French is after all, the most
hated language in the world from an Englishmans perspective ;-}
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Derek Whitten wrote:
Colin
Aha!! why didn't I think of that.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Gonzalo Servat wrote:
On 2/6/06, Mark Phillips [EMAIL PROTECTED] wrote:
A customer of mine wants an IVR where the first 3 choices are
1 English
2 Spanish
3 French
I can build the IVR but how do I get
Try adding insecure=very to the guest user account in iax.conf. This
should not do a user/pass challenge on the incoming call.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
kevin ling wrote:
Not sure answer your question? Try to write some html code and let user
register the username
30 says it's view only in the docs I can't seem to change it, any other
options?
Option 30 allows to set Module Shelf Address/ID.
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As evident in the SuperDial script and others based
upon groups, you can place a call into a group, which can have a limit on the
number of concurrent calls. Can a call belong to multiple groups?
IE: I have only a limited number of channels to upstream X.
Downstream Y is only paying me for
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