I'd like to use Asterisk to build a phonetree (www.phonetree.com) type
of application, like this:
1. Read a text-based name/phonenumber file.
2. Call every number and play a recorded message.
3. If a beep is detected, replay the message from scratch (to leave
messages on an answering
Hello,
While logger.conf indicates that you can log to a file I can't see _where_
the logging is sent to (which directory). Looking at the source it seems
that the directory used is specified in ast_config_AST_LOG_DIR but I can't
see where this variable is defined. Help?
I have seen on the
I'd like to use Asterisk to build a phonetree
(www.phonetree.com) type
of application, like this:
1. Read a text-based name/phonenumber file.
2. Call every number and play a recorded message.
3. If a beep is detected, replay the message from scratch (to leave
messages on an
Hi Edwin! (and everybody)
I have some questions about SIP, as I wrote in another mail. I have a SIP
Gateway and I have two phones conected to it.Also, I have two Dlink
dg102s with four phones conected to them. The main problems are two.
Calls between the phones conected to the SIP GW and the
hi all
I still can't use both BRIs on the AVM C2 with chan_capi. This is _annoying_
since people have started complaining about the number of available lines.
Have anyone else seen this?
thanks
roy
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356
On Wed, 11 Jun 2003 [EMAIL PROTECTED] wrote:
our Asterisk pbx is sitting behind a normal analog hardware pbx, we have
to dial 9
to take an outside call through the hardware pbx, our fxo interface is
also connected
to one of the extensions of it. we can make calls to internal hardware
pbx
Hello,
I'm testing Snom 100 with sip at Asterisk. Seems to work quite nicely so
far. I have one problem though:
If the other end is busy, I can't hear the busy indication -tone.
Asterisk seems to know, that the other end is not available:
-- Couldn't call [EMAIL PROTECTED]
-- Hungup
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[EMAIL PROTECTED]
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with an AVM C2 and chan_capi-0.2.1b and CVS version of asterisk
(gdb) bt
#0 0x407745b4 in pipe_msg (PLCI=770, CMSG=0x810f678) at chan_capi.c:1156
#1 0x40777f30 in capi_handle_msg (CMSG=0x302) at chan_capi.c:1914
#2 0x407772b9 in do_monitor (data=0x0) at chan_capi.c:1941
#3 0x40020d53 in
Hi,
I've had a search through the archives and didn't find much. Is anyone using the
Monitor application? I have it working but there is a really big drawback. The files
are always called the same thing, which means if I make 2 calls one after the other
the first recording is lost. I half
Hi everybody one more time!
I also have done a SIP debug and that's an extract of what I have found:
(...)
s=session
c=IN IP4 188.208.12.237
t=0 0
=audio 13532 RTP/AVP 0
a=rtpmap:0 PCMU/8000
to 229.159.241.112:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
Asterisk uses the routing table to determine ourip for any given IP
address. It should, in principle, always be right... Do you have a
specific case in which it isn't? If so contact me off-list first so we
can try to resolve it although i'm going to be a bit hard to reach for the
next week or
Hi all,
I'm a student (my thesis work consist in testing
interopearbility SIP/H.323) and I begin to work with
asterisk in this days.
I have to testing to SIP/H.323, since today I have used
Vocal system, but there are some problem for this
features.
In the asterisk mailing list,
On 12 Jun 2003, Jukka Tainio wrote:
If the other end is busy, I can't hear the busy indication -tone.
Asterisk seems to know, that the other end is not available:
-- Couldn't call [EMAIL PROTECTED]
-- Hungup 'H323:0'
== Everyone is busy at this time
You probably need the
Hi Siggi,
Thanks for your thorough report and test results.
You are right. The transmission of voice packets after
silence periods is done with incorrect timestamps,
causing slight voice drop-outs.
I 'll see how this can be fixed and let you know.
Regards,
Michael.
Siggi Langauf wrote:
After
Hi,
I just made a fresh install on a new box and at the end I got this message:
make: warning: Clock skew detected. Your build may be incomplete.
I had all the various libs added to a default install of RH 9. Though its
possible that I'm short on developer tools. Any clues anyone?
--
Steve
Steve wrote:
Hi,
I just made a fresh install on a new box and at the end I got this message:
make: warning: Clock skew detected. Your build may be incomplete.
I had all the various libs added to a default install of RH 9. Though its
possible that I'm short on developer tools. Any clues
I just downloaded the latetst CVS. A compile now complains about a missing
srv.c srv.h used in chan_sip.c. Can they be added?
--
Betel Consultancy
Abelenlaan 19
1185 RT Amstelveen
The Netherlands
http://www.betel.nl
tel. +31 621 858 469
___
Sorry, my mistake.
The point was that I had a message playing with Background() and a
couple of Setvar() after it. As I started to dial an extension before
the message had finished, the setvar() calls didn't get invoked.
PHM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Thursday 12 June 2003 08:57 am, you wrote:
I just made a fresh install on a new box and at the end I got this
message: make: warning: Clock skew detected. Your build may be
incomplete.
I had all the various libs added to a default install of RH 9. Though
its possible that I'm
On Thursday 12 June 2003 08:49 am, julian green wrote:
Steve wrote:
Hi,
I just made a fresh install on a new box and at the end I got this
message: make: warning: Clock skew detected. Your build may be
incomplete.
I had all the various libs added to a default install of RH 9. Though
use Flash and SendDTMF to talk to dumb pbxs, Flash'll only work on ZAP
channels though...
exten = 0,1,Flash
exten = 0,2,SendDTMF(2)
exten = 0,3,Wait(1)
exten = 0,4,SendDTMF(123123123)
exten = 0,5,Hangup
-wasim
On Thu, 12 Jun 2003, Steven Critchfield
hi all
as with the standard 'shutdown' command, it'd be nice to have a 'canceller' to
'die when convenient'. is this a heavy task to add?
roy
--
Roy Sigurd Karlsbakk, Datavaktmester
ProntoTV AS - http://www.pronto.tv/
Tel: +47 9801 3356
Computers are like air conditioners.
They stop working
Hi,
Zhone say yes, try:
http://www.zhone.com/support/technical_support/zplex/product_documents/ug_0011_10b.pdf
I try this and have stopped on
login and password. Zhone defaults are changed, answer
from seller - its not possible and I hold it. I cant
take risk to reset zplex-software
Ahh, wonderful thanks...
Andy
On 12/06/2003 at 13:35 Pertti Pikkarainen wrote:
Check
http://www.loligo.com/asterisk/current/extensions.conf
and find macro called macro-record-on
There is at least one way described ( author is John Todd ).
--Pertti
append=yes might help :)
On Thursday 12 June 2003 15:35, Dan wrote:
Hi all,
There is something special I must configure in order to get the voice
mssage by mail?
In voicemail.conf I have:
[EMAIL PROTECTED]
attach=yes
[default]
301 = 6535,Home Mailbox,[EMAIL PROTECTED]
I have tried to
On Thursday 12 June 2003 09:06 am, Steve wrote:
On Thursday 12 June 2003 08:57 am, you wrote:
I just made a fresh install on a new box and at the end I got this
message: make: warning: Clock skew detected. Your build may be
incomplete.
I had all the various libs added to a
hi guys,
I have a little problem maybe you can help ...
I have an asterisk setup, with an E100P, and an ISDN-PRI 30 channel
line from the telco going into it .. the E1 line is OK, because plugged into
a Lucent Portmaster 4 it works OK .. plugged into the asterisk box
I just get an engaged tone,
yes.. it is installed and I have tested that I can send an email using it
(with the telnet console).
I have done a search through the local messages and the problem seems to be
that the mail sent by Asterisk use as sender [EMAIL PROTECTED], but
this domain is not a registered one, so the
Ok, thanks for the clarification
Shame it still doesn;t work for me :( maybe it only works
with US phones... anyone in Europe got this working?
Andy
*** REPLY SEPARATOR ***
On 11/06/2003 at 21:55 Tilghman Lesher wrote:
On Wednesday 11 June 2003 19:10, Andy Powell wrote:
not really asterisk related this,
but is it normal for a mail to take so long
to be resent through the mailing list server?
i'm speaking about 20 minute + delays here ..
(or it it only me ?)
cheers
Dave
___
Asterisk-Users mailing list
[EMAIL
Hello everybody,
We are running into some strange problems on inbound calls from our
termination provider.
They are using a VegaStream 100 with H.323 and a couple of codecs available.
We are running Asterisk with the H.323 channel and the G.729 codec's from
digium (10 pack license so there
Try to start Asterisk with debug turned on. Make sure you have
in /etc/asterisk/logger.conf the following line:
console = debug,notice,warning,error
Then start Asterisk with -cd.
Send the screen log of the session.
Also a backtrace on the core would help.
Michael.
Makerere University wrote:
On Thursday 12 June 2003 09:24, Dan wrote:
yes.. it is installed and I have tested that I can send an email
using it (with the telnet console).
I have done a search through the local messages and the problem seems
to be that the mail sent by Asterisk use as sender
[EMAIL PROTECTED], but this
(I'm producing a writeup on all the things one run into building an * box.
Which after a couple of boxes is showing up aplenty.)
So this uses the working h/w and config's from another box. All I do is
build in a new computer.
Loading the modules I got:
For all thos Asterisk users not on the FWD list, it works for me!:
-Original Message-
From: Free World Dialup - The Future of Dialing
[mailto:[EMAIL PROTECTED] On Behalf Of Leonidas Piagkos
Sent: donderdag 12 juni 2003 0:58
To: [EMAIL PROTECTED]
Subject: Re: [FWD] FWD losing Registration
I have edited my zapata.conf file and I still cannot get voicemail
notification to work.
zapata.conf
...
mailbox=403
callerid=Bruce Wayne 403
channel=4
...
Shouldn't this create a stutter dial tone for channel 4?
Are there any other places that I must configure for this to work?
This is most obvious if you get a message while checking messages. This happens here all the time.
I've been complaining about this 'feature' for 3 months. It is more than a simple patch as it would
require rewriting the code that counts messages, the code that keeps trak of where in playback
In case anybody is wondering...
I am in Canada, which apparently makes a difference. I found this in
the archives:
And this does actually work. There is a stutter tone for about half a
second when you pick up the configured channel. This is apparently how
the Telco's do it in the US. I didn't
Hi,
If you set the 'serveremail' variable to a complete address, Asterisk
will use that as the Sender, otherwise it will append the machinename
to the name specified.
I have done it, but Asterisk still uses [EMAIL PROTECTED] as sender
address..:(
Because doing so is far more complex than
On Thu, 12 Jun 2003, Michael Manousos wrote:
Thanks for your thorough report and test results.
You are right. The transmission of voice packets after
silence periods is done with incorrect timestamps,
causing slight voice drop-outs.
I 'll see how this can be fixed and let you know.
Cool.
On Thu, 12 Jun 2003, Andy Powell wrote:
Shame it still doesn;t work for me :( maybe it only works
with US phones... anyone in Europe got this working?
We have the MWI (Message Waiting Indicator) working on the Snom 100/200
but we haven't tried the stuttered tone. We didn't manage to get the
Well after moving to a diff slot and rebooting twice I finally got:
Zapata Telephony Interface Registered on major 196
No ISA tormenta card found at 0d000
Zapata Telephony Interface Unloaded
CSLIP: code copyright 1989 Regents of the University of California
PPP generic driver version 2.4.2
Zapata
On Thursday 12 June 2003 10:48, K. C. Li wrote:
On Thu, 12 Jun 2003, Andy Powell wrote:
Shame it still doesn;t work for me :( maybe it only works
with US phones... anyone in Europe got this working?
We have the MWI (Message Waiting Indicator) working on the Snom
100/200 but we haven't
On Thursday 12 June 2003 10:30, Dan wrote:
Hi,
If you set the 'serveremail' variable to a complete address,
Asterisk will use that as the Sender, otherwise it will append the
machinename to the name specified.
I have done it, but Asterisk still uses [EMAIL PROTECTED] as
sender
Roy,
This is already available with the 'abort halt' command
James
On Thu, 12 Jun 2003, Roy Sigurd Karlsbakk wrote:
hi all
as with the standard 'shutdown' command, it'd be nice to have a 'canceller' to
'die when convenient'. is this a heavy task to add?
roy
--
Roy Sigurd Karlsbakk,
Mark did the commit so I guess he'll add it when he gets a chance.
Martin
On Thu, 12 Jun 2003, Michiel Betel wrote:
I just downloaded the latetst CVS. A compile now complains about a missing
srv.c srv.h used in chan_sip.c. Can they be added?
--
Betel Consultancy
Abelenlaan 19
1185 RT
Hi,
* Erik Lagerway wrote/schrieb:
There is a provider in the US - www.AddaLine.com, who just launched a
SIP service with some great rates for North America
I have been using their service for months and I am extremely happy with
the
service.
looks like Germany is again laggin
And you did that in the [general] section of voicemail.conf?
Yup!
Are you using Voicemail or Voicemail2?
Voicemail.
Thanks,
Dan
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Whenever I issue the reload command, asterisk crashes. Below is the output I get from
(gdb) bt
Any help is appreciated.
***
*CLI reload
== Parsing '/etc/asterisk/manager.conf':
Found
== Parsing
Check the line 118 of extensions.conf ???
Martin
On Thu, 12 Jun 2003, Derek Beaumont wrote:
Whenever I issue the reload command, asterisk crashes. Below is the
output I get from
(gdb) bt
Any help is appreciated.
***
*CLI
On Thursday 12 June 2003 13:05, Dan wrote:
And you did that in the [general] section of voicemail.conf?
Yup!
Are you using Voicemail or Voicemail2?
Voicemail.
In that case, your mailer is probably rewriting the headers. You'll
need to seek help in the forum related to whatever mailer
Iconnecthere seems to have better rates...
-Original Message-
From: Martin Dommermuth [EMAIL PROTECTED]
Date: Thu, 12 Jun 2003 19:48:43 +0200 (MEST)
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoIP Provider
Hi,
* Erik Lagerway wrote/schrieb:
There is a
Martin == Martin Dommermuth [EMAIL PROTECTED] writes:
Martin looks like Germany is again laggin behind all others in the
Martin communication field. Or I asked at the wrong place. There
Martin might not be to many people from Germany in this list.
One possibility is Pulver's LibrTel at
[EMAIL PROTECTED] (Dan) writes:
I have done a search through the local messages and the problem seems to be
that the mail sent by Asterisk use as sender [EMAIL PROTECTED], but
this domain is not a registered one, so the destination server does not
accept it.
I must have a registered
On 12 Jun 2003, James H. Cloos Jr. wrote:
One possibility is Pulver's Libr=C3=A9Tel at http://www.libretel.com.
Whenever I try any of their access numbers (at least the ones around me,
in the DC area), I get a recording The number you have reached is not in
service. This does not inspire great
[EMAIL PROTECTED] (Michael Manousos) writes:
While logger.conf indicates that you can log to a file I can't see
_where_
the logging is sent to (which directory). Looking at the source it seems
that the directory used is specified in ast_config_AST_LOG_DIR but I can't
see where this
[EMAIL PROTECTED] (Steven Critchfield) writes:
On Thu, 2003-06-12 at 09:31, Dave Alan Caruana wrote:
not really asterisk related this,
but is it normal for a mail to take so long
to be resent through the mailing list server?
i'm speaking about 20 minute + delays here ..
(or it it
Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues
Zaptel was compiled with -D__SMP__
We've installed irqbalance and the picture improved a lot
(thanks to Jared Smith). Do you still see problems in our /proc/interrupts?
The big issue for us now is that after 24+ hours of the
We are not having any luck with the E100p card here in Australia, it
will work with a crossover cable to another device but will not talk to
our Telco Telstra who probably have a weird implementation of an E1.
Any suggestions on a replacement?
Regards
Mark McKibbin
DCS Internet
64 Queen St
Hi
We can found a couple of ITSP at Jasomi networks's PR.
http://www.jasomi.com/pr_deployment.html
Does anyone try it?
mack
On Thu, 12 Jun 2003 19:48:43 +0200 (MEST)
Martin Dommermuth [EMAIL PROTECTED] wrote:
Hi,
* Erik Lagerway
does anyone have a recorder GSM file that emulates the Telco's if you are a
telemarketer please hangup now recording? I don't see one in the sounds dir. the
ZapATEller works great for computerized callers but if a human hears this message
asking them to go away they have to. Isn't that
I had some problem with this also. If you are using
sendmail, edit sendmail.cf and input a couple of
parms:
-Find a line that starts with DS. Put in your
provider's mailserver address right after the DS.
Example: DSmail.swbell.net
This will allow your provider to relay the mail for
you.
Then
Visit www.aislecom.com for FXS to FXO
converters.
Product: ACOM300
Dear Folks,
Before using my precious time trying to configure what I'm imagining, I
would like to confirm if it's possible with asterisk.
I'm planning to buy several E100 cards to plug in my PSTN network here in
Japan and configure another H323 device in Brazil.
I would receive all calls here
1) I'm working on a quick replacement for DISA, and I ran into the
following snag: When I specify Playtones(dial) I can only get
around 7 seconds of wait time before the dialtone stops, and the
context goes to the h extension. Is there a way around this fixed
timeout? The DigitTimeout
On Fri, Jun 13, 2003 at 01:48:13PM +1200, Peter Armstrong wrote:
You need to get the ETSI or Euro version of PRA from Telstra and then it
will work, they offer it as well as their quaint version of PRA access.
Peter
Are the E100p cards Austel approved? Or does Telstra give you written
You cant get the old in australia anymore ;-)
And no his problem is with etsi...
On Fri, 13 Jun 2003 13:48:13 +1200, Peter Armstrong wrote:
You need to get the ETSI or Euro version of PRA from Telstra and then it
will work, they offer it as well as their quaint version of PRA access.
Peter
And you expect him to answer on an open list ?
On Fri, 13 Jun 2003 12:41:02 +1000, Anthony Wood wrote:
On Fri, Jun 13, 2003 at 01:48:13PM +1200, Peter Armstrong wrote:
You need to get the ETSI or Euro version of PRA from Telstra and then it
will work, they offer it as well as their quaint
I think winging it is too strong a word.so of course I have not
plugged it in yet, just a hypothetical question, anyway Telstra rekon
they are ETSI or at least their version of it.
Regards
Mark McKibbin
-Original Message-
From: Anthony Wood [mailto:[EMAIL PROTECTED]
Sent: Friday,
Your Verion Rep is way out in left filed.
DID is about the only way to go on PRI's. And doing it is about 3 lines in
the CO switch.
DMS 500
Table DN Route: Numbers to OFRT 223
table OFRT 223 Trunk Group Name
table trkgrp Trunk Definition.
That is all there is to it.
Routing a single number
Has anyone tried a Dialogic DM/IP3031A-E1 with Asterisk? We are getting
desparate.
Regards
Mark McKibbin
DCS Internet
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Sorry about the delay for synchronised monitoring guys. It is actually quite
easy to implement, no major risks that I can see; the time is the only issue
at the moment. The project I was working on being just finished, I am
planning to catch up with non-work stuff such as this patch pretty soon.
Yeah thats a good one but it does not give them the message this number does not
accept telemarketing calls, if you are a telemarketer then please add us to your do
not call list and hang up now. if your not a telemarketer then press one to continue
yada yada yada... This is the message
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