Mac OS X, Asterisk 1.4 beta
--- Martin Joseph [EMAIL PROTECTED] wrote:
On 2006-11-12 23:08:05 -0800, zen Perry
[EMAIL PROTECTED] said:
I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and
stops
immediately and as a result I dont hear the
On 2006-11-12 14:48:13 -0800, joe a. ([EMAIL PROTECTED])
[EMAIL PROTECTED] said:
Experiencing one way audio using IAX2.
I did see some other posts on this, and see there may be some internal
issues with asterisk and one way audio. Can this be a widespread
problem? So many seem to be
On Mon, Nov 13, 2006 at 12:45:40PM +0530, Sri Keerthy wrote:
Can two versions of asterisk run on same PC??
Basically, yes: use a custom asterisk.conf (or custom compiled defaults)
to use different pathes for just about anything.
Note, however, that only one program can listen on the same port.
Could this be considered spam ? I believe this is second threas realted to that pbx .On 13/11/06, Jordi Nelissen
[EMAIL PROTECTED] wrote:
Check out the ESCAUX net.PBX operator console. In use in variouscompanies with 200+ extensions. Powerfull and convenient.
I am not sure if it will help but try to put notansfer=yes in ur iax2 extension (just experiment a bit ;) ).On 12 Nov 2006 17:48:13 -0500, joe a. (
[EMAIL PROTECTED]) [EMAIL PROTECTED] wrote:
Experiencing one way audio using IAX2.I did see some other posts on this, and see there may be some
Hi!
I have a working asterisk-setup with four sip-clients. Everything works
great but when the users call someone the phonenumber shows up on the
receiving ends callerid-display.
To correct this my provider told me to send #31# before the phonenumber,
tried this with: Dial(SIP/[EMAIL
Also, where can I get
information on provisioning? These phones will be out of my hands soon
and I'd like to be able to update the configs. I saw a few utilities for
generating the configs, but I'd like more specific info - I don't mind editing
files by hand but want to know how
On 2006-11-12 23:08:05 -0800, zen Perry [EMAIL PROTECTED] said:
I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and stops
immediately and as a result I dont hear the audio.
-- Started music on hold, class 'default', on
channel 'SIP/XXX'
--
Tzafrir Cohen wrote:
Note, however, that only one program can listen on the same port. So if
you want both to e.g., listen on IAX, one, at least, has to listen on a
custom port.
...or you need to run two IP addresses on the machine,
and configure each Asterisk installation to use the
Is there any way to let asterisk listen to two different
ports 5060 and 5061 for example , or this can be done from iptables firewall
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium
What is the length of music on old mp3 file ? Maybe file is very short .On 13/11/06, zen Perry [EMAIL PROTECTED]
wrote:
Mac OS X, Asterisk 1.4 beta--- Martin Joseph [EMAIL PROTECTED] wrote: On 2006-11-12 23:08:05 -0800, zen Perry
[EMAIL PROTECTED] said: I'm trying to set up the Music on Hold
You can also use waitexten = X,1,Wait(3)(for3secs) On 13/11/06, Jim Archer [EMAIL PROTECTED] wrote:
--On Sunday, November 12, 2006 11:53 PM -0500 John Novack[EMAIL PROTECTED] wrote: Dovid B wrote: snip How hard would it be to have asterisk detect a dial tone ?
I really can't say. I am not a C
Dear all,
I've just a little question ...
I've configured asterisk to run with voicemail realtime in the
extconfig.conf like this :
voicemail = mysql,database,voicemail
I just want to have a row, in the voicemail table, like emailbody, which
is capable to give me a body to the mail. Therefor
Hi,
I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a
user via SIP and playing a reminder file when the user picks the phone.
I use Gizmo service for SIP and I'm able to call through it.
However, when asterisk dials a number, Gizmo first answers then tries
Put canreinvite=no in asterisk sip user extension.Someprovidersdonotsupportreinvitesandhenceyougetsilenceiguess. On 13/11/06, Yuri Veremeyenko
[EMAIL PROTECTED] wrote:
Hi, I am using Asterisk to set up a reminder-like system, with asterisk auto-dialing a user via SIP and playing a reminder file
Hi all,
I am interested in integrating my telephone system (I am using
hardphones and Asterisk) with my desktop - something like this:
1. someone sends me his/her phone number via email/icq
2. I cut/paste the number in some application/web page (?)
3. my phone starts ringing and when I pick
Hi
i have an application developed with bayonne.
Recentely i'm experiencing some problems and i am planning to migrate
to asterisk.
I would like to know if i can do these things whit asterisk:
- IVR integration with database (mysql, insert,delete,update,select)
- TTS
- record exploration (for
Yes, it is possible also if you use chroots under linux (
http://en.wikipedia.org/wiki/Chroot ). But you'll need to change the
listening ports.
Regards,
Pavel Siderov
Tzafrir Cohen wrote:
On Mon, Nov 13, 2006 at 12:45:40PM +0530, Sri Keerthy wrote:
Can two versions of asterisk
Basically here is what the application you need has to do:1. take the number you paste2. make a call file 3. drop it in the outgoing spool directory of asteriskThis could be easily done in php - just one page. Donno if any app already exists (have heard of many, but not sure if they come alone or
Ondrej Valousek napisał(a):
Hi all,
I am interested in integrating my telephone system (I am using
hardphones and Asterisk) with my desktop - something like this:
1. someone sends me his/her phone number via email/icq
2. I cut/paste the number in some application/web page (?)
3. my phone
Yes, you can but you need to use external software for TTS.
nik600 wrote:
Hi
i have an application developed with bayonne.
Recentely i'm experiencing some problems and i am planning to migrate
to asterisk.
I would like to know if i can do these things whit asterisk:
- IVR integration with
On 11/13/06, nik600 [EMAIL PROTECTED] wrote:
i have an application developed with bayonne.
I would like to know if i can do these things whit asterisk:
- IVR integration with database (mysql, insert,delete,update,select)
Yes, you have to write AGI scripts to do this.
- TTS
No idea.
-
On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote:
Hi
i have an application developed with bayonne.
Recentely i'm experiencing some problems and i am planning to migrate
to asterisk.
I would like to know if i can do these things whit asterisk:
- IVR integration with database
Vicky,
my other post related to a Web GUI for asterisk. This post is related to
an Operator Console. I am simply answering the user's question, so I
don't see why you would consider this to be spam, and I never read you
can not send two mails to the list on the same day.
Jordi
Vicky wrote:
On 11/13/06, Pavel Siderov [EMAIL PROTECTED] wrote:
Yes, you can but you need to use external software for TTS.
ok, thanks
can you suggest me some example for the database interaction?
for example, how can i connect to the database?
how can i make a query and list the records found?
thanks
On 11/13/06, Brian Rogan [EMAIL PROTECTED] wrote:
On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote:
Hi
i have an application developed with bayonne.
Recentely i'm experiencing some problems and i am planning to migrate
to asterisk.
I would like to know if i can do these things whit
Hi,
I havent tested this yet, but I am just wondering what are the advantages of
using this GUI?
Does it help you with creating extensions or what?
Many thanks,
Christian
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Hello Michal,
Thank you for the hint!
Can I ask you for your script so I have some idea how it works?
I have apache already running on my * box.
Thank you,
Ondrej
Michał Niklas wrote:
Ondrej
Valousek napisał(a):
Hi all,
I am interested in integrating my telephone system (I am
On 11/12/06, nik600 [EMAIL PROTECTED] wrote:
On 11/12/06, Vicky [EMAIL PROTECTED] wrote:
Yep make the server with dynamic ip register to server with static ip ( sip
or iax both will do but in sip keep nat=yes while making extension )
the problem is that the server with dynamic ip can't
Hi,How would you monitor screensaver activity ?Cheers
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Hi Dean,
I will check that site - thanks for the hint.
The biggest problem I see with authentication and I do not think mexuar
could help me here (and I am definitely going to pay $2000 for it :-)
But it is another story...
Thank you!
Ondrej
Dean Collins wrote:
Ondrej,
Why is it awful ?Regards
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Hello peoples,
I have a grave problem. In my work i have an asterisk functioning
perfect. However whenever the link of internet falls the even for of
function. For that everything come back to the normal necessary one
remove the trunk sip. Someone knows say me as contour that situation?
What are you using for your Internet connection?
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.
Email for information: [EMAIL
We have a link dedicated of radio. But that presents problems the times!
Al Bochter escreveu:
What are you using for your Internet connection?
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free
Well if you want to use VOIP you will have to get a better Internet
connection.
You can't do anything to the PBX Server to fix this.
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Al Bochter
Sent: 13 November 2006 14:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with internet down
Well if you want to use VOIP you will
Snap will do this for you. (Check my signature)
On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote:
Hi Dean,
I will check that site - thanks for the hint.
The biggest problem I see with authentication and I do not think mexuar
could help me here (and I am definitely going to pay $2000
asteriskextras.com has a FREE click to call script that is very popular.
On 11/13/06, mitcheloc [EMAIL PROTECTED] wrote:
Snap will do this for you. (Check my signature)
On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote:
Hi Dean,
I will check that site - thanks for the hint.
The
Thanks for the response!I've been reading and trying things and I cannot find a way to do a supervised transfer using this topology:pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk
Because if I do a flash() and a SendDTMF() to transfer the extension I have to Hungup(), otherwise it never
Hi Mitchel,
this looked Very good to me at the first glimpse - then I realized the
client is Windows only :-(
We have Linux desktops here...
Thanks anyway...
Ondrej
mitcheloc wrote:
Snap will do this for you. (Check my signature)
On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote:
Its pretty easy . If you have mysql records enabled via a patch just do sql queryuse asteriskcdrdb;select * from `cdr` where billsec 0 ( if answered then billsec always greater than 0 or you cna also use disposition = 'ANSWERED' )
On 13/11/06, Olivier [EMAIL PROTECTED] wrote:
Why is it awful
Hello,
We are running Asterisk-1.0.12 in a CentOS 4-4.2 system, kernel
2.6.9-42.0.3.ELsmp.
We have some custom AGI, and when we launch Asterisk the system works fine.
But **after some time**, each AGI execution generates a zombie defunct
process.
We believe that it's not a problem in the AGI
I believe that the problem really is fault of DNS lookups, but as I
should proceed for resolve that??
Steve Langstaff escreveu:
-Original Message-
I don't think that's necessarily true on a couple of counts:
1) You can use VoIP in-house and something else out to the rest of the
We use Asterisk Desktop Manager http://adm.hamnett.org/ very
successfully with both debian and windows desktops.
Bails
Ondrej Valousek wrote:
Hi Mitchel,
this looked Very good to me at the first glimpse - then I realized the
client is Windows only :-(
We have Linux desktops here...
Thanks
Making custom voicemail greetings seems fairly straight forward, and I've
done it.
However, I'm looking for a way to make the actual extension answer with You've
reached my Jim Dandy voice mailbox, go take a flying . . .. (OK, so maybe
not), instead of The person at extension , is
oops sorry i didnt saw quoted text of other user and it showed as first post in gmail draft so i thought u made a topic for that pbx ( so considered spam :P ) . Sorry again :)On 13/11/06,
Jordi Nelissen [EMAIL PROTECTED] wrote:
Vicky,my other post related to a Web GUI for asterisk. This post is
I have been using http://www.snapanumber.com/ 's Windows tray utility, and it
works great.
--
--
Steven
http://www.glimasoutheast.org
Ondrej Valousek [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi all,
I am interested in integrating my telephone system (I am using hardphones
On 15:34, Mon 13 Nov 06, bails wrote:
We use Asterisk Desktop Manager http://adm.hamnett.org/ very
successfully with both debian and windows desktops.
Firefox can't find the server at adm.hamnett.org.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key:
On 11/10/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Steve Davies wrote:
*bump*
No suggestions at-all? Does anyone use this facility in a similar way
and NOT have problems?
Check the gain on your ISDN interface. The monitor command doesn't
modify the volume by default. Have you tested calls
I believe that the problem really is fault of DNS lookups, but as I
should proceed for resolve that??
see the first point at
http://www.voip-info.org/wiki/view/Asterisk+administration
The best solution for now is probably to have a caching dns server on
your Asterisk box or in your LAN
Dear all,
My architecture is having some problems with redirects. In the following
diagram is shown a simple erroneous test. When someone dials from the
PSTN, signalling of the incoming call is passed to Asterisk which routes
to SIP Express Route (Ser), and then Ser routes to the phone. The
A search of google should turn up some recommendations about running a
local cacheing DNS proxy, or similar.
I've never done it myself (the cacheing proxy, not the searching on
google) so I don't know the specifics.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I've figured out the problem, alaredy, and posted in another thread,
but no one seems to have an answer yet.
It is a problem in the IAX trunk. If I turn the jitterbuffer on, I
get one-way-audio when I put someone on hold. If I turn the
jitterbuffer off... I still have two way audio. THis is
On Mon, Nov 13, 2006 at 07:10:12AM -0500, Brian Rogan wrote:
On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote:
Hi
i have an application developed with bayonne.
Recentely i'm experiencing some problems and i am planning to migrate
to asterisk.
I would like to know if i can
On Mon, 2006-11-13 at 17:00 +0100, Michiel van Baak wrote:
On 15:34, Mon 13 Nov 06, bails wrote:
We use Asterisk Desktop Manager http://adm.hamnett.org/ very
successfully with both debian and windows desktops.
Firefox can't find the server at adm.hamnett.org.
Just downloaded it with
On 11/13/06, Steven [EMAIL PROTECTED] wrote:
I have been using http://www.snapanumber.com/ 's Windows tray utility, and it
works great.
--
--
Steven
http://www.glimasoutheast.org
Ondrej Valousek [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi all,
I am interested in
List members,
Is it possible to record outbound analog calls using an X100P?
I was asked if I knew how to record all calls for a shop with 4 analog
phones transparently to the end users. I thought Asterisk was a good
fit for this and I envisioned using either Digium TDM400Ps or Sangoma
Hi:
I'm having audio dropouts in ChanSpy when the call is originated in a
Unicall (E1/MFCR2) channel and the destination is an Agent using a SIP
phone. If the agent is using a traditional phone (going from the PBX to
asterisk via another Unicall line) no dropouts are present. The dropouts
Am Montag, den 13.11.2006, 10:37 -0500 schrieb joe a.:
Making custom voicemail greetings seems fairly straight forward, and I've
done it.
However, I'm looking for a way to make the actual extension answer with
You've reached my Jim Dandy voice mailbox, go take a flying . . .. (OK, so
I have this very
weird situation where some callers hear the playback of sound prompts on half
speed. It only lasts a few second but it can happen at any time during playback.
My server is a 3.4
Ghz Xeon with 1 GB RAM and 80 GB SATA disk. I run Asterisk 1.2.13 on FreeBSD
6.1
Anyone who
joe a.[EMAIL PROTECTED] Wrote on: 11/13/2006 10:37 AM:
Making custom voicemail greetings seems fairly straight forward, and
I've done it.
However, I'm looking for a way to make the actual extension answer with
You've reached my Jim Dandy voice mailbox, go take a flying . . ..
(OK, so
Hi all,I've 2 * servers with static IP, and i notice that if i set both sip peers with host=server_ip and qualify=yes it presents UNREACHABLE on asterisk CLI.When i changed the host parameter to host=dynamic and set the register string in the [general] of
sip.conf on both servers, the connection
On Monday 13 November 2006 11:49, Yu Safin wrote:
when you use snap, does the call go to your iax hardphone connected to
asterisk or do you need a softphone on your PC?
Please trim your posts, you don't need to keep the headers and signature lines
of the entire thread to ask one sentence, now
Hi Moises,
Coul you give more details about how to use Cacti for CDR analysis,
there is some special pluggin, additional conf?
Your help will be appreciated.
Rgds.
On 10/31/06, Moises Silva [EMAIL PROTECTED] wrote:
of course you can always use http://cacti.net/download_cacti.php
On
Hi Eric,
Your answer solved my problem. I did a uname -r which= 1.6.12-27mdksmp, the Makefile had -27mdkblahblahblah. Once I switchedthe makefileeverything worked. Thanks for your help.
Julian
Date: Fri, 10 Nov 2006 12:05:37 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com
Hello,
I try to configure Asterisk to send voicemail in the language of the user's
mailbox.
But the only way I see is to modify the app_voicemail.c anybody has an
alternative idea for me ?
Lot of thanks
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On 17:27, Mon 13 Nov 06, Dave Cotton wrote:
On Mon, 2006-11-13 at 17:00 +0100, Michiel van Baak wrote:
On 15:34, Mon 13 Nov 06, bails wrote:
We use Asterisk Desktop Manager http://adm.hamnett.org/ very
successfully with both debian and windows desktops.
Firefox can't find the server
Hello,I know little to nothing about asterisk. I am currently reading info online but was looking for other links as to good places to start. I basically just need to use asterisk to create a prototype. I need to demo a system that will allow the user to call a number, enter in some data, the
OK, to simplify the reading I'll resume my problem...
Is there a way to make Asterisk send a call to Ser witch reroutes it
back to the same asterisk server ,without resulting in a loop detected
error in Asterisk?
Thanks,
Ricardo.
___
Setup overview:
We have an asterisk server serving a small number of SIP phones. The
asterisk server is connected to an old phone system via a T1. The
asterisk server is also connected to a second T1 used for
inbound/outbound calls.
Scenario:
We are using a call file to do auto-dialing.
Hi all,
Using the latest 1.4 of Asterisk. I have noticed that the music on hold files
are in wav, isn't mp3 supported anymore?
Many thanks,
Christian
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To
Hi
I have 2 asterisk servers
running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are
using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server
and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch
site and all calls go to
This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql properly with billsec field containing number of billed seconds . I want to know some function to round this to 6 seconds ( or any custom valud
I am planning to use asterisk with Digium TDM2404E card as a media
gateway to terminate traffic to Cell phones. Anyone got this working
before with no problmes, specially with Answer/Disconnect supervision?
Thanks
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Why not directly use ip address in host= lineinextensions instead of dynamic address like sip.voipprovider.com .. temporary fix but it may work .
On 13/11/06, Steve Langstaff [EMAIL PROTECTED] wrote:
A search of google should turn up some recommendations about running alocal cacheing DNS proxy, or
Please pardon the absolute noob questions. Someone has asked me to
interface with Asterisk and have it dial 4 numbers in succession to have
it track down an on-call person.
My initial reaction was to write an AGI program and return all 4 numbers
and have Asterisk hunt them - can Asterisk do
IF your asterisk server is behind NAT and no port forwarding is done then how can that static ip user/device reach it .Youwillhavetokeepasteriskserverinstaticipordoportforwardingtoacceptconnectionsfromoutside.
ORmaybeididntunderstandsenarioproperlyhere.Isitlikeyour Server with SIP application
Does it happen when you make more than one call from you main voip server alone ? Or it happens when there are more than 1 call on your branch server ? Pin the problem is in which server first , If main server can handle 2-3 calls with no lag then its probably problem in branch server .
On
sum(duration+(6-mod(duration,6) for summary of seconds divisible by 6, /60 for minutes
On Tue, 2006-11-14 at 00:07 +0530, Vicky wrote:
This is more of mysql question then asterisk :D . Most voip providers use 6 second rounding for costing . My asterisk server stores call cdr's in mysql
Sorry meant 2.6.12-27..
From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSubject: RE: [asterisk-users] Modprobe ZaptelDate: Mon, 13 Nov 2006 17:31:24 +
Hi Eric,Your answer solved my problem. I did a uname -r which= 1.6.12-27mdksmp, the Makefile had -27mdkblahblahblah. Once I
Hello All, as good?It would like to know if somebody has experience in
asterisk with ss7 protocol for isdn and asterisk with support to the
protocol sip-t.
Best RegardsJosué
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Dear all,
I'm trying to enable Asterisk to work with FAX using T38. I've tried
Asterisk 1.2.4 with the available patch found at URL
http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3
that is announced to support it too.
With both Asterisk versions, I've sent with
I'm trying to implement a voicemail distribution list using asterisk and I've
hit a bump. I've got an agi script that parses voicemail.conf and generates a
list of voicemail boxes to use as an argument to the voicemail() function.
The problem is that the argument exceeds 256 characters (100
Most usage charges are stored in various
billing databases as per MINUTE of use, not per 6 seconds of use.
6 second billing simply means that you
bill in decimal fractions of a minute, 66 seconds becomes 1.1 minutes.
1. Divide your billsec value by 60 and
round to 1 decimal place. Add
All,
I'm starting to tinker with Asterisk for use in my home. Here's my
current setup:
Cox broadband telephone -- spa3k-fxo
analog phones + answering machine (all on one line) -- spa3k-fxs
I can pick up a phone in my house, dial a certain extension, and the
spa3k will connect me to
Thx and what would the sql query be ?.Iplantoputadditionalfieldas6second.Howcanimakebillsecofvaluesofwholetablegetroundedandfilledinfield6second Sorry i am a noob with mysql :D
On 14/11/06, James Coberly [EMAIL PROTECTED] wrote:
sum(duration+(6-mod(duration,6) for summary of seconds divisible by
Hi
It's defiantly the branch
server. My main server handles 30 to 40 calls at a time on a regular basis. It
is only happening on the branch server and it acts like it is using up all the
bandwidth of the DSL. It is a 1.5 meg down and 512 up DSL line. I would think it
could handle 2
I have a two port TE205P Digium card. I have set everything up to
create a native zap bridge between the two spans. Everything works
perfectly except one thing. Our telco has a password that has to be
entered as soon as a long distance call is made. So if I dial a long
distance call from my
Donald Stahl wrote:
I'm trying to implement a voicemail distribution list using asterisk
and I've hit a bump. I've got an agi script that parses voicemail.conf
and generates a list of voicemail boxes to use as an argument to the
voicemail() function.
Generate in groups of 50 and loop it
Supposing you have an extra column called 6second:
UPDATE cdr SET 6second=billsec+(6-mod(billsec,6) where 6second=0
if you want a decimal minutes column called billmin
UPDATE cdr SET billmin=round((billsec/60)+0.5),1) where billmin=0
Vicky wrote:
Thx and what would the sql query be
? . I
Before I open a bug I'll ask again if anyone else is having trouble with
receiving MWI on SIP devices in 1.4. My configuration was working fine
in 1.2 but as soon as I change to any build of 1.4 I don't get
notification on any of several SIP devices. I can post my configuration
but since it
I initiate a call with a callfile, specifying the From phone# as the
channel Dial(), and the To phone# as the Extension Dial(). I announce
the To phone# to the From listener with the A() option to the Dial()
command. It seems that the A() app plays audio while blocking return
from the From
I'm trying to implement a voicemail distribution list using asterisk and
I've hit a bump. I've got an agi script that parses voicemail.conf and
generates a list of voicemail boxes to use as an argument to the
voicemail() function.
Generate in groups of 50 and loop it until you have them all?
I was reading the posts and someone said about the default 1000 to 2000
I see in the .conf the default is 1 to 2
I found a service that gives inbound DID's in the firewall 5060 and
1 - 2 is setup
no workie on the DID
But when I set 5060 , 1 - 2 and (Unblocked) 1000 -
Hey thanx for that Marnus . Thatsworking just exactly how i wanted :).Damoniactuallycameupwithsamerow/60+0.5thenrounduptrickwheniwasdoingsomethingsameinexcelsheets:)anditsusefulforbillingin1minuteroundup(60secpulse)butifailedtogetitworkingfor6secondpulse.
Donald Stahl wrote:
Define Generate.
Right now I Answer and then send them into the voicemail function
with the list of mailboxes. How would I go about first recording the
message and _then_ sending it to voicemail in a loop, 50 at a time? If
there is a function I missed or am unaware of
actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed .
On 14/11/06, Al Bochter [EMAIL
Right now I Answer and then send them into the voicemail function with
the list of mailboxes. How would I go about first recording the message and
_then_ sending it to voicemail in a loop, 50 at a time? If there is a
function I missed or am unaware of please let me know.
I was not clear on
Donald Stahl wrote:
Right now I Answer and then send them into the voicemail function
with the list of mailboxes. How would I go about first recording the
message and _then_ sending it to voicemail in a loop, 50 at a time?
If there is a function I missed or am unaware of please let me know.
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