Title: Problems with Zpateller on incoming external calls
I've setup the following in extensions.con:
exten = 2200,1,Ringing
exten = 2200,2,Wait(2)
exten = 2200,3,Answer
exten = 2200,4,Zapateller
exten = 2200,5,Macro(stdexten,2205,SIP/2205)
This works as expected if I dial from a
All,
I've almost got my Asterisk PBX setup, but I've having some problems with
the VoicePulse IAX trunk.
On outbound calls, when dialing a PSTN number through the IAX2 trunk,
music on hold (moh, using the m option in the dial command) does not work.
The console states that stop sound on IAX2
Hello,
I need a few access numbers in the UK and Germany. Does anyone have
this available right now? I need the incoming calls to be directed
through IP to one of my asterisk servers in Europe. Please contact me
off the list if you want.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
Has anyone else had trouble with the AGI command GET DATA on the latest
stable cvs?
I can't get it to work with asterisk-perl, or by using print statements
and reading stdin.
I get 200 result= (timeout). (this is from the print statements, and
asterisk-perl reports nothing).
But asterisk is
Hi all,
We've been experimenting with the app_queue application, and it works
quite well. The only problem we encountered was that outgoing calls (to
the operators) aren't logged in CDR.
Example,
* operators dial a specific number/extension, and AddQueueMember(..) runs
(they get added without
Finally, i will get back to a RedHat 9 distrib as I see that it works with that distribution...[EMAIL PROTECTED])fjåËbú?jË^®+$ºÇ«±:5%H$HJ+ºZµê)¶*'²ø¬Øm¶ÿ+-±Ø é¢oæj)fjåËbú?jË^®+$ºÇ«
Sent 12 hours ago and it never showed up (slightly reworded here).
Sorry if this is a duplicate:
-
Scenario: a person selects an Auto Attendant option that fires off the
Directory application (CVS circa 3/22). Three questions:
1) How do they escape
Use another DID for just voicemial so all users can call into it, enter
their extension and then then password to access their own voicemail. I
just created this today for a Production system. The extensions.conf looks
like this for anyone who call 963-4400:
exten = 4400,1,Voicemailmain()
You guys are making this way harder than it needs to be. Assume your
main number comes in on 4400, you want to give the receptionist an
opportunity to answer the call, but if s/he's away from the desk or on
another call you want to proceed to an auto-attendant to direct the call
as necessary. In
Hi Christian,
On Thu, Apr 08, 2004 at 06:13:50PM -0500, Christian Hoffmeyer wrote:
Here's my situation.
I have two receptionists that answer incoming lines. Each has a 7960G with
5 incoming lines each. I'm trying to set this up so each line on each phone
doesn't utilize call waiting. My
I need a few access numbers in the UK and Germany. Does anyone have
this available right now? I need the incoming calls to be directed
We do. I'll mail you off-list.
Linus
Magrathea
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Dear Willy
i notice the same problem with my E100P using the latest cvs zaptel driver i
have try every type of config in /etc/zaptel.conf to check if i have missed
something in timing conf but nothing... Digium help... :-)
thanks in advance
Dimitri
On Thursday 08 April 2004 23:07,
sipgate.de has DIDs in Germany and the UK.
-Alfred
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stephen
Karrington
Sent: Friday, April 09, 2004 4:08 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Who has access numbers in the UK and Germany?
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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I'm afraid I'm just out on a family 'outing', can you give me an overview
via email of what you are looking for ?
- Original Message -
From: Stephen Karrington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 09, 2004 12:07 PM
Subject: [Asterisk-Users] Who has access numbers in
Can anyone here help me in getting connected to Clearpath? They have
supposedly setup a DID and 800 number for me but not provided login
info. They're really hard to reach.
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist
massimo wrote:
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
Bye
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I am using the NetJet cards from www.traverse.com.au, it uses a TigerJet
320 chip on it. It works fine with isdn4linux, you need the netjet
driver under the passive cards. Also need the audio option in the kernel
to make it work with asterisk. It appears this is already in the 2.4.24
kernel, but
I can't read German. Can you outline the cost for me? Thanks.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice - 877-203-9308
Fax - 310-943-2606
Dreamtime
On Thursday 08 April 2004 22:41, Ryan Thrash wrote:
Scenario: a person selects an Auto Attendant option that fires off
the Directory application (CVS circa 3/22). Three questions:
1) How do they escape if they didn't mean to go there in the first
place (without having to hang up...)? Config
I am using the NetJet cards from www.traverse.com.au, it uses a TigerJet
320 chip on it. It works fine with isdn4linux, you need the netjet
driver under the passive cards. Also need the audio option in the kernel
to make it work with asterisk. It appears this is already in the 2.4.24
kernel, but
Let's say an unsuspecting soul accidently selects the Directory option
from an Auto Attendant (CVS circa 3/22). Three questions:
1) How do they escape if they didn't mean to go there in the first
place (without having to hang up...)?
exten = 1,1,Directory(vertex)
exten =
Try this:
exten = _0.,1,Dial(CAPI/xxx:b${EXTEN:1})
The :1 tells it to use everything except the first digit.
Robert Jackson
-Original Message-
From: massimo [mailto:[EMAIL PROTECTED]
Sent: Friday, April 09, 2004 6:59 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Ignorepat
Ryan,
Did you have to apply a patch to get this to work, or is it in CVS?
You mentioned posting some doc of what you did?
Thanks in advance,
Ed
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan R. Fligg
Sent: Thursday, April 08, 2004 3:10 PM
To: 'Dan B. Long';
On Thu, 2004-04-08 at 13:30, Jeremy Hall wrote:
I have no idea if that setting affects it or not. Is that a command
line switch when starting Asterisk?
Woah... I just moved the cards back to my old test box (dual cpu
athlon). Guest what? CallerID worked *every time*. I noticed on my
Hey all,
I am dialing a DID through VoicePulse Connect. The number is
answered by a main menu type of IVR. The configuration is as specified
in both the wiki and VoicePulses documentation. The call comes through
without a problem, but when the caller enter any keys they are either
not
On Fri, 2004-04-09 at 12:58, massimo wrote:
Hi to all,
I'm trying to make outside call in this way :
ignorepat = 0
exten = _0.,1,Dial(CAPI/xxx:b${exten})
But the first number 0 is not ignored.
I'm doing something wrong ?
I don't have CAPI but to get my analog to work I have
Well, I have a few things to discuss.
1. I need a few numbers for the UK and Germany to start. This is for one small project.
2. For my second project I need local access numbers in the major
markets. This is for our system at www.diamondcard.us. This is a VOIP
services MLM product.
Sincerely,
Stephen Karrington wrote:
I can't read German. Can you outline the cost for me? Thanks.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice - 877-203-9308
Fax -
On Fri, 9 Apr 2004, Stephen Karrington wrote:
I can't read German. Can you outline the cost for me? Thanks.
http://www02.sipgate.de/user/tarife.php
Tarife NATIONAL
Deutschland* - E 1,79Ct/Min (US$ 0.021637)
Cellular: E 22,90Ct/Min (US$ 0.276815)
They have a plan which includes 1000
Dimitri,
I just got off the phone with digium. Here's what I (from my
notes) the event codes mean
Event 4: Alarm detected
Event 5: Alarm cleared
Event 6: Abort HDLC Frams
Event 8: Bad HCS
The 6 8 which occur sporadically are possibly causing the
observed symptoms.
Now ... what causes 6 8 is the
Can someone send me a quick snippet of a dialplan for international
dialing via NuFone? I'm having a hard time getting any help from them
this week.
Scott
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Is there a way to set default caller id info to pass to * when the telco does not
provide it?
Regards,
Victor Perez
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On Fri, 2004-04-09 at 00:59, Jeb Campbell wrote:
Has anyone else had trouble with the AGI command GET DATA on the latest
stable cvs?
I can't get it to work with asterisk-perl, or by using print statements
and reading stdin.
I get 200 result= (timeout). (this is from the print statements,
On Fri, 2004-04-09 at 10:12, Robert Jackson wrote:
Hey all,
I am dialing a DID through VoicePulse Connect. The number is
answered by a main menu type of IVR. The configuration is as specified
in both the wiki and VoicePulses documentation. The call comes through
without a problem, but
Has anybody tried to install * in any of these minimalist linux distros like tinylinux?
Which linux distro would you use to run * in old P2, P3 boxes?
Regards,
Victor Perez
[EMAIL PROTECTED]
(469) 221-4189
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Hi Vic,
I hit that same problem! My SIP phones would sound okay when I made
changes to indications.conf but incoming calls in to my TE410P had their
own thing going on!
Have a look at the zaptel source files, there's one called zonedata.c.
You'll see the au settings... replace what's there with
Scott Laird wrote:
Can someone send me a quick snippet of a dialplan for international
dialing via NuFone? I'm having a hard time getting any help from them
this week.
exten = _3.,1,Dial,IAX2/[EMAIL PROTECTED]/011${EXTEN:1},60,tr
--
Best regards,
Duane
http://www.cacert.org - Free Security
Title: Latency and 'Scratchy' Voice...
Dear All,
I have move from the USA to Sydney, Australia. I have gone from a data center environment at work and cable at home to a 513k/128k ADSL line.
I'm experiencing two issues;
1) There is a latency of .5 - .8 seconds between me and the USA.
2)
Title: Asterisk Server Crashing with New Application
Dear All,
I have been running a successful and very stable call center PBX based on 0.7.1 release. I need to be on this release because of a number of features that I have complied from 3rd party patches, for the call center. I will not
Hello steve. Here is a patch I wrote for app_voicemail.c which does
exactly as you describe. When the outgoing message is playing, if the
listener hits the * key, they're prompted for a mailbox and password,
whereupon they can check their voicemail as if they were using the internal
Hello, just compiled zaptel in mandrake 9.2 and this is what I get when trying
modprobe wcfxo:
Apr 9 11:35:15 localhost kernel: PCI: No IRQ known for interrupt pin A of devic
e 00:05.0. Please try using pci=biosirq.
Apr 9 11:35:15 localhost kernel:
Victor Perez wrote:
Has anybody tried to install * in any of these minimalist linux distros like tinylinux?
Which linux distro would you use to run * in old P2, P3 boxes?
I have got it to install on Trustix (92MB min install) but I have moved
to Fedora now for other reasons..
Title: RE: [Asterisk-Users] small linux distro to run * in old boxes
I am using * on a RH9 380Mhz AMD K6 processor (With XP100 card), as well as Fedora Core 1 on a PII 333 Mhz machine for a couple of small SIP phone tests. One at work, and one at home. Things seems to be working just fine.
In /etc/asterisk/zapata.conf before the
channel=x
(where x is the channel assigned to the FXO port)
put:
callerid=PSTN Call 1234567
You will need to restart * for this change to take effect
Andy
*** REPLY SEPARATOR ***
On 09/04/2004 at 10:56 Victor Perez wrote:
Is
I made a custom fedora mini distro, something like
350 megs, including apache,php,mysql webmin
of course installable from a cd in 20 minutes, more or less :)
at the end you have a fully working asterisk installations,
along with some basic tools like webmin and
a full webserver
Matteo.
Il
On Apr 9, 2004, at 9:52 AM, Tilghman Lesher wrote:
On Thursday 08 April 2004 22:41, Ryan Thrash wrote:
Scenario: a person selects an Auto Attendant option that fires off
the Directory application (CVS circa 3/22). Three questions:
1) How do they escape if they didn't mean to go there in the first
I have the same problem, got this from VoicePulse today:
Chris,
Thank you for contacting VoicePulse.
Our engineers are aware of the DTMF problem and are working to have it
resolved as quickly as possible.
Please reply directly to this email if we can provide any additional
assistance.
Brancaleoni Matteo wrote:
I made a custom fedora mini distro, something like
350 megs, including apache,php,mysql webmin
of course installable from a cd in 20 minutes, more or less :)
at the end you have a fully working asterisk installations,
along with some basic tools like webmin and
a full
Brian Cuthie wrote:
I've setup the following in extensions.con:
exten = 2200,1,Ringing
exten = 2200,2,Wait(2)
exten = 2200,3,Answer
exten = 2200,4,Zapateller
exten = 2200,5,Macro(stdexten,2205,SIP/2205)
This works as expected if I dial from a SIP phone on my desk.
However, if I dial in
I've got following problem with manager api:
In my Asterisk installation when I connect two channels (IAX,SIP) I get
following sequence of events(these are events for *single* connection,
come one by one without any delay):
Event: Link
Channel1: [EMAIL PROTECTED]:5036]/3
Channel2: SIP/kamyk-9950
I know an email can be sent when a user get a voicemail message, but is
there a way to send a message to a SIP phone to say they have a message? Or
how hard would it be to write an app that could popup on a PC when there is
a message in the mail box?
Kyle
Hi!
4) Installed the Flash Operator Panel
5) Installed a modified version of Monastery to show me which agents were
[...]
I suspect the problem to be either caused by 4 or 5, in which case they
will be very easy to rectify. I would however like to know if anyone else
has had a) the same
Hi,
John Todd said:
9) Speech recognition support
Nothing towards this yet - sphinx keeps getting mentioned, though I
don't know anyone who has had it running in anything other than a
crippled test, or at least I don't remember anyone saying anything
about it.
Which features do Asterisk
Tried that, and no go. There's something wrong with Zapteller. It works fine
on internal calls, but the only way I can get it to work on external calls
(through a SIP/PSTN gateway, no Zap hw necessary) is to first play a
message. For instance, this works:
exten = 2200,1,Playback(ss-noservice)
Hi
I made a custom fedora mini distro, something like
350 megs, including apache,php,mysql webmin
of course installable from a cd in 20 minutes, more or less :)
at the end you have a fully working asterisk installations,
along with some basic tools like webmin and
a full webserver
So all you do is pop the
CD in and install it and Asterisk is ready to go ?
And it only takes up 350 MB
?
Is there any way I could get a copy of it
??
Thanks, Paul.
--- Original Message
---
From:
[EMAIL PROTECTED] on behalf of WipeOutPosted At:
Fri 09/04/2004 17:55Posted To:
Running straight from extensions.conf, for now. The dialplan looks like
this:
[voicepulse-incoming]
Exten = _NXXNXX,1,Goto(mainmenu,s,1)
Exten = _NXXNXX,2,Hangup
[mainmenu]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten =
Most sip phones have a message indicator. To use it, just specify
mailbox=1234 (or whatever the mailbox number is) is the phone's
definition in the sip.conf file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
Sent: Friday, April 09, 2004
Shad,
I don't remember how far in the past, but a while back at least one
person if not more reported instability in asterisk caused by more than
one manager client connecting to the Asterisk server at the same
time. Your monastery as well as the Flash Panel both access the
manager application
Hi Victor
I'm currently working in a Linux Distro, it is being internal alpha
testing by my self and a couple of me my colleagues, over the next
couple of weeks I'm hoping to release a beta version to the asterisk
community., I'll keep you posted via asterisk users, about its features
as it
Hi,
Does anyone know where I can get a telephone card that will fit into
the PCI slot on my PC and work with the UK telephone system (BT) ?
I would really like the retailer to be based in the UK if at all possible
?
Also, is there any way to set up asterisk so that only certain phones can
Paul Tyreman [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
Does anyone know where I can get a telephone card that will fit into the
PCI slot on my PC and work with the UK telephone system (BT) ?
I would really like the retailer to be based in
Robert Boardman [EMAIL PROTECTED] wrote:
The first to note is Its currently a 28Mb ISO for installation with
asterisk installed with zaptel, and lib pri
this includes apache perl PHP, and Mysql
That's impressive. My MySQL installation has munched its way through
48MB of disk space on its
Kevin Walsh wrote:
The nice people at TelAppliant will sell you an analogue FXO card,
and are based in London, England. See here:
http://www.voiptalk.org/
The Digium X100P (well, the X101P now) works in England with the notable
exception of support for BT's caller ID.
Some UK cable
Chris,
This does sound like my scenario. Do you
remember how they achieved this?
Now that I have removed these components Im
stable again.
Thanks for the feedback and help.
Warm Regards
Shad
From: Chris A. Icide
[mailto:[EMAIL PROTECTED]
Sent: Saturday, April
Hello all,
Does anyone know of a good IAX softphone for Pocket PC's?
Thanks!
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Kevin Walsh wrote:
Did you mean 280MB by any chance? :-)
He said that was the iso size, I managed to get debian installed down to
about 32megs, but this was minus apache, php, mysql... but you can
compress the installer files on the iso and then have it extract them as
it installs...
--
Best
Hello,
I have been running into a problem on my server (which I believe was
the cause of an O/S crash earlier today). I am consistently seeing the
following messages in /var/log/messages:
Apr 8 05:24:04 east insmod: /lib/modules/2.4.20-8/misc/torisa.o:
insmod char-major-196 failed
Apr 8
At 08:01 PM 4/9/2004 +0100, Robb wrote:
I'm currently working in a Linux Distro, it is being internal alpha
...
The first to note is Its currently a 28Mb ISO for installation with
asterisk installed with zaptel, and lib pri
this includes apache perl PHP, and Mysql
Is there anything in
Greetings folks,
I have updated our asterisk RPM repository with CVS builds for RH 7.3, 9
and FC1. The zaptel package is compiled against the supplied kerenel rpm
and SRPMS are supplied for those wishing to rebuild. Other than the CVS
update no other changes are made from previous releases...
Hi !! all !!
My MeetMe is moving by SIP.
Does Ztdummy load to the kernel?
- Original Message -
From: Jain, Sonal [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 09, 2004 4:53 AM
Subject: [Asterisk-Users] MeetMe conference
I am trying to setup MeetMe conference.
In my
Just disable (via the .conf file) and remove most of the apache modules --
its very small then.
Small enough to go on a LEAF/LRP to drive the control interface, anyway.
There are a lot of modules that only make sense on a full web serving
sitution (like mod_speling for example) but if you get
Yes !!
My Asterisk is working by the spec of the P2 average.
- Original Message -
From: Victor Perez [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, April 10, 2004 1:02 AM
Subject: [Asterisk-Users] small linux distro to run * in old boxes
Has anybody tried to install * in
On Sat, 10 Apr 2004, Benjamin Wakefield wrote:
Have a look at the zaptel source files, there's one called zonedata.c.
You'll see the au settings... replace what's there with this:
detail snipped
Benjamin, LEGEND! ;)
Don't know why I didn't see this sooner -- thanks indeed!
For my ear
After rebooting my
asteriks server, e-mail notifications are no longer being sent after a
voice-mail is left.
I can see the
messages in /var/spool/asterisk/vm.
has anybody had the
same experience? how was it resolved?
Uri
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