On Sunday 11 March 2007 20:04, Ira wrote:
At 01:36 AM 3/11/2007, you wrote:
My servers don't run anything more than they need to and don't have
packages loaded that they don't need. I could rant on all day about
the bloat I see in modern RH/Fedora/SuSe, even my favourite Debian
systems, but
How did you install these packages -- make sure you do ./configure and
if needed make menuselect in each one of these before the make and
make install. This is the only thing I can think of -- check whether
there are any built-in modules as well.
on Monday 03/12/2007 Asterisk Asterisk([EMAIL
No one...?
This problem is really bugging me... :-/
Regards,
Evert
Evert wrote:
Hi all!
Working on the following brain-scratcher. I am setting up a Trixbox
system for someone who uses 'provider A'. Everything works fine, except
for the IVR: keypresses by callers are not being
Hi list,
I have an application which has to automatically dial and send out a voice
message to 50 different phone numbers at the same time. Does it mean that I
need to sign up 50 phone lines or voip accounts in order to achieve this
purpose? Is there a provider(voip prefer) who offer a special
On Mon, 2007-03-12 at 09:07 +, Kurt Kuo wrote:
Hi list,
I have an application which has to automatically dial and send out a voice
message to 50 different phone numbers at the same time. Does it mean that I
need to sign up 50 phone lines or voip accounts in order to achieve this
Dears
Please can you inform me by how to make a pickup group ?since all users can
pick up any line ?
Regards
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
Postal Code: 1102-2080
Tel: (961) 1- 868 686
Fax :(961) 1-808 810
GSM: (961) 3-979 343
Friends,
This week I'll be in Lissabon speeking at a Voip Conference on
Wednesday. I'm not aware if there's
an Asterisk Users group in Lissabon, but if there is maybe there
would be a chance to meet.
Next week, I'll be at Cebit, in the Digium stand. If you want to meet
me, I'll be in the
That sounds like not quite right maths...
More importantly, how many calls per day and how long per call.
Then you can figure out the other bits.
PaulH
On Mon, 2007-03-12 at 09:07 +, Kurt Kuo wrote:
Hi list,
I have an application which has to automatically dial and send out a voice
Some VOIP providers don't pass DTMF very welland sadly it's pretty
common.
relaxdtmf=yes (I have never used this function)
PaulH
On Mon, 2007-03-12 at 09:48 +0100, Evert wrote:
No one...?
This problem is really bugging me... :-/
Regards,
Evert
Evert wrote:
Hi all!
On Mon, 2007-03-12 at 20:52 +1100, Paul Hales wrote:
But top posted
That sounds like not quite right maths...
What maths was involved? He wants to make 50 simultaneous calls.
More importantly, how many calls per day and how long per call.
Then you can figure out the other bits.
He wants
Hey i installed zaptel and when i tried to install
asterisk and ran command menuselect it showed me that
there are some discrepencies that are not being
fullfilled for meetme application, but i have also
installed ztdummy when i installed zaptel. I am
totally stuck and nowhere to go what should i
We would need your exact steps in both installing zaptel and asterisk
in order to help, and this is a series of steps which is quite long,
so you would have to keep exact logs of what you did to both
configure, make and install both zzaptel and asterisk and you would
need to tell which zaptel
Hi All
I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with
with one of my ATAs not ringing.
Basically, when I execute the Dial command, an error occurs: Got SIP
response 400 In alert-info header: Empty value expected
Now in 1.2, I just issued the following command to
It has been my experience when working with PRI's that you have very limited
options when dealing with outbound CID. Due to restrictions of 911 most
Telco's will have to have the PRI split into Trunk Groups for proper CID
delivery. This would work for a situation of sharing one asterisk server
hymy server under heavy traffic give me the folowing error then
restarts asterisk:
Mar 8 21:35:39 ERROR[514]: ast_h323.cxx:169 void PAssertFunc(const
char*): Assertion fail: Operating System error, file tlibthrd.cxx, line
743, Error=24
edit..and again this one...
Mar 8 21:40:59
I'm using AMI on Asterisk 1.2.15 and I'm having problems with DBPut action.
If I execute this:
Action: DBPut
Family: checkin
Key: 316
Val: yes
Response: Error
Message: Missing action in request
I don't put anything in Asterisk DB.
If I execute this:
Action: DBPut
Family: checkin
Key: 316
Val:
FYI,
If you are using a Citel Handset Gateway, here is a working Time Zone rule to
fix the US DST change.
rule mar sun GTEQ 2 0200 -0400 nov sun GTEQ 1 0200 -0500
--
--
Steven
http://www.glimasoutheast.org
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Hi,
I know that if you have Polycom phones properly configured, you can use sip
notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download
the new configuration from the provisioning server and reboot.
Is there anyway to send the same command to all peers (let's say I had 50
Hello List
I am currently testing, using a shoutcast server as source for MOH.
Here is the command im using:
/usr/bin/wget -q -O - http://listen.coolfm.dk:80/ | /usr/local/bin/madplay -d
-Q -z -o raw:- --mono -R 44100 - | sox -r 44100 -w -s -t raw - -r 8000 -c 1 -t
raw - resample vol
Hi all,
Does anyone have any experience with creating a Single sign on (SSO)
concept where if someone logs in on their PC the phone next to that PC
is also automatically assigned to that user?
TIA,
Patrick
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Hi
Yes, with AGI you can do all what you need.
If you know php, i suggest you phpagi
http://phpagi.sourceforge.net/
But take a look at this page, you can interface with AGI with many languages
http://www.voip-info.org/wiki-Asterisk+AGI
Bye
On 3/12/07, [EMAIL PROTECTED] [EMAIL PROTECTED]
for i in `seq 100 150` ; do asterisk -rx sip notify polycom-check-cfg
$i ; done
Julian.
On 3/12/07, Mike [EMAIL PROTECTED] wrote:
Hi,
I know that if you have Polycom phones properly configured, you can use sip
notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download
the new
Hi everyone,
I'm completely new to Asterisk and before I buy any card, I would like to ask
for some information.
1. We'll be using analog PSTN phone lines. Is there anything that I should ask
the telecom company before I buy the card? What I mean is whether the card will
be compatible with
I've been running Digium TDM400P with 2 FXO and now I run a Sangoma A200 with 2
FXO and no hwec, both cards have suffered echo, one of my lines is much worse
than the other. I messed about for a year using the software EC in Zaptel and
whilst I could remove the echo on one line the other would
nik600 wrote:
new update
11/03/2006
- added the module stats
- updated the file db.sql with sql instructions for the creation of
queue_stats table
- added the files view.sql
I'm in no position to test your product now. Hopefully I will find some
time soon. Please keep group informed about
SNTP tcpIpApp.sntp.resyncPeriod=86400
tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset=
tcpIpApp.sntp.daylightSavings.enable=1
tcpIpApp.sntp.daylightSavings.fixedDayEnable=0
tcpIpApp.sntp.daylightSavings.start.month=3
tcpIpApp.sntp.daylightSavings.start.date=8
C F wrote:
Tomislav, really? and how does it show up on my POTS line?
It only can be seen if other end is also on Optima provider. Ant it is
shown exactly as originator has define it. It's strange when you, for
the first time, get the phone call from unknown number and you see his
name at
Best way I found to do this I wrote a quick bash script that takes
an ip address and runs that command. Then if your phones are in an ip
range, you can say something like
for i in `seq 194 197`; do /usr/sbin/sipReboot 192.168.101.$i; done
That will reboot 192.168.194 thru 197.
Rob
Hi Peder,
I think that CF was correct in his original post. From the Polycom SP IP
admin guide:
Attribuite: tcpIpApp.sntp.daylightSavings.start.date
Values permitted: 1-31
Default: 1
Description: Day of the month to start DST.
What the start.date=8 does is tell the phone to start DST on the
Peder @ NetworkOblivion wrote:
I'm pretty sure this is wrong:
tcpIpApp.sntp.daylightSavings.start.date=8
Should be:
tcpIpApp.sntp.daylightSavings.start.date=2
This is what I set it to as well.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Peder @ NetworkOblivion wrote:
SNTP tcpIpApp.sntp.resyncPeriod=86400
tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset=
tcpIpApp.sntp.daylightSavings.enable=1
tcpIpApp.sntp.daylightSavings.fixedDayEnable=0
tcpIpApp.sntp.daylightSavings.start.month=3
tcpIpApp.sntp.daylightSavings.start.date=8
Mike wrote:
Hi,
I know that if you have Polycom phones properly configured, you can use sip
notify polycom-check-cfg SIP_REGISTRATION_ID to have the phones download
the new configuration from the provisioning server and reboot.
Is there anyway to send the same command to all peers (let's
Alex Robar wrote:
Hi Peder,
I think that CF was correct in his original post. From the Polycom SP
IP admin guide:
What the start.date=8 does is tell the phone to start DST on the
first start.dayOfWeek it finds after the start.date. So in this case,
we're telling it
Not according to the
This all depends on the setting before it:
tcpIpApp.sntp.daylightSavings.fixedDayEnable=0
Since this isn't a fixed date, it isn't used the same way. It doesn't
understand 'second week of the month', so if you use the 8th, it will
use the next weekday of
Now, the H323 Channels is updated and your bugs fixed.
But Digium still develop your OOH323 Channel. My question is why?
What is the better in Asterisk 1.4.x.? I know that in Asterisk 1.2.x OOH323
is very better than H323 or OH323.
Thanks in advanced.
Thiago.
--
Is there a command in Asterisk that will cause all Aastra phones to reboot
and/or recheck for new firmware?
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as I know, ooh323 is external project from objective systems,
anyway, for 1.4 I prefer chan_h323 from asterisk tree.
Thiago Maluf wrote:
Now, the H323 Channels is updated and your bugs fixed.
But Digium still develop your OOH323 Channel. My question is why?
What is the better in Asterisk
On Mon, 2007-03-12 at 10:50 -0400, Matt wrote:
Is there a command in Asterisk that will cause all Aastra phones to
reboot and/or recheck for new firmware?
In his answer for the same question on Polycom phones Julian wrote
for i in `seq 100 150` ; do asterisk -rx sip notify polycom-check-cfg
$i
There does not seem to be an 'aastra' option in Asterisk.. that's why I'm
asking if there is another way.
On 3/12/07, Dave Cotton [EMAIL PROTECTED] wrote:
On Mon, 2007-03-12 at 10:50 -0400, Matt wrote:
Is there a command in Asterisk that will cause all Aastra phones to
reboot and/or recheck
1. We'll be using analog PSTN phone lines. Is there anything that I should
ask the telecom company before I buy the card? What I mean is whether the
card will be compatible with the line?
Just regular lines... not T1 or PRI? You shouldn't have any issues.
2. What about the hardware on
Anyone have a suitable configuration that takes into account the new
DST changes for a Cisco 7970 (XML format)
^gtg
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On Mon, 2007-03-12 at 11:29 -0400, Matt wrote:
There does not seem to be an 'aastra' option in Asterisk.. that's why
I'm asking if there is another way.
Then read the excellent Aastra documentation like I and probably many
others did and add the required code to sip_notify.conf
substituting
Hi all,
I just changed router at my clients office and installed a Linksys router
with latest firmware, which gives an option Filter IDENT(Port 113) in its
firewall. If it is checked, remote SIP phones do register but audio goes one
way. If I uncheck it, everything works fine. What I read on the
I can telnet to the ACM on the local machine but I can't get to it from
another machine I've been over the information about ACM at voip-info.org
and haven't been able to figure out what I'm missing. I've included my
manager.conf file and the error I'm getting from the other machine. Can
anyone
Hi!
Is there unicode support in Asterisk for SIP? E.g. How can I have a
displayname with special characters?
E.g. if I want to have the Umlaut ä in the display name:
callerid=Jeff Gräser 11
AFAIK SIP requires that the ä must be encoded using UTF-8. Thus, the ä
must be encoded as 2 bytes:
On Sun, 2007-03-11 at 22:29 -0800, Yuan LIU wrote:
From: Thomas Patterson [EMAIL PROTECTED]
Date: Mon, 12 Mar 2007 19:03:12 +1300
I have setup my Asterisk server to have 3 outbound routes
1 being for local calls
2 being for toll calls
3 being international call
What I am wanting to do
Thanks Marc, hadn't seen that one. I'm currently at 1.2.14, looks like this
is in .15.
Thank you very much!
On 3/12/07, Marc Archer [EMAIL PROTECTED] wrote:
Barry,
Have a look at http://bugs.digium.com/view.php?id=8804
I am assuming that you are trying to transfer using the # key (or
On Mon, 12 Mar 2007, David Ruggles wrote:
I can telnet to the ACM on the local machine but I can't get to it from
another machine I've been over the information about ACM at voip-info.org
and haven't been able to figure out what I'm missing. I've included my
manager.conf file and the error I'm
Does SIPAddHeader(Alert-Info:) not do it?
On 3/12/07, Nikhil Jogia [EMAIL PROTECTED] wrote:
Hi All
I have just upgraded from Asterisk 1.2 to 1.4 and am having trouble with
with one of my ATAs not ringing.
Basically, when I execute the Dial command, an error occurs: Got SIP
response 400 In
At 07:50 AM 3/12/2007, you wrote:
Is there a command in Asterisk that will cause all Aastra phones to
reboot and/or recheck for new firmware?
You can tell the phones to check ever morning at 3:00AM which is what
I do or if I want it now, I just unplug the switch for a few seconds.
But I only
Hi all,
Anyone know how to dynamically create meetme conference rooms on the
flight? I remembered a while ago there was a switch that tell meetme to
create the conference room is the room is not defined in the
meetme.conf. It doen't seem to be working for me anymore.
Thnx
In case it hasn't been posted before, here's instructions to get the
correct time to show up on your Grandstream GXP-2000's:
1. Login to phone
2. Go to Basic Settings tab
3. Change Daylight Savings Time to yes
4. Change Optional Rule to 3,2,7,2,0;11,1,7,2,0;60 (this means change
clocks the
Thanks Dave, good info!
And thanks to those who confirmed I needed to write a script because there
were no built in functions, I appreciate that info too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: Monday, March 12, 2007
Yup.. got mine set to reboot every morning at 3am and check for updates...
just was curious :)
On 3/12/07, Ira [EMAIL PROTECTED] wrote:
At 07:50 AM 3/12/2007, you wrote:
Is there a command in Asterisk that will cause all Aastra phones to
reboot and/or recheck for new firmware?
You can tell
Thanks! That was the problem.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
Sent: Monday, March 12, 2007 12:35 PM
To:
Barry D. Hassler wrote:
Thanks Marc, hadn't seen that one. I'm currently at 1.2.14, looks like
this is in .15.
Actually,
It didn't start working correctly for me until 1.2.16
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
There is an aastra option but only for individual extensions which you may
already know about. Perhaps if you try the same wildcards as suggested for
the polycom it may work?
nano /etc/asterisk/sip_notify.conf
[aastra-check-cfg]
Event=check-sync
Content-Length=0
Then from the asterisk console
Hello
I have a brief and a long question about a possible Asterisk deployment
I am planning.
Long Story Short:
I have four total offices, one main and three remote. All offices are
connected using dedicated network T1 lines creating one unified network
across offices. I would like to
Since we've had discussion about DST on polycom I thought I'd pass along
the rule I used to configure DST on my sipura units as well (This way
the date and time passed in caller ID will be correct).
Under the admin view go to the regional tab. At the bottom under
miscellaneous enter this in
You should take a look at the d or the D option of Meetme application ;)
Regards,
Tristan Mahé
Wai Wu a écrit :
Hi all,
Anyone know how to dynamically create meetme conference rooms on the
flight? I remembered a while ago there was a switch that tell meetme to
create the conference room
12 mar 2007 kl. 17.05 skrev Klaus Darilion:
Hi!
Is there unicode support in Asterisk for SIP? E.g. How can I have a
displayname with special characters?
No. I made a proposal for it a long time ago, without much comments.
We need to fix this, since IAX2 now by spec is UTF8 too. In order
How many simultaneous calls will this device support and with which
codecs/transcoding?
Do you sell the hardware stand-alone without your software so we can
load our own version of Asterisk/Gui?
On 3/12/07, Ioan Biris [EMAIL PROTECTED] wrote:
Hi ,
We have done exactly that … fan
Hi Folks,
I am in need of someone who can provide me with a LIDB / CNAM storage
database.I will set the pointcode on my numbers to point to your
database, and then I need to be able to update my numbers in your
database. If you are able to offer these services, please contact me.
Matt
I have about 4 GrandStream GXV 3000 video phones. I have been trying to
integrate them into our asterisk environment. All works fine until
someone trys to use the speakphone function or the conference call
function (conferencing to inbound call together). As soon as they try
either function
add this to your config
dst: 3600 03.02.07 02:00:00 11.01.07 02:00:00
--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
Thanks a million! Just verified after putting it in my encrypted configs and
it works like a charm! :)
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: Monday, March 12, 2007 12:15 PM
To: Asterisk Users Mailing List -
Since 1.4 ALERT_INFO variable has been deprecated. I used to send this
via AMI:
Action: Originate
Channel: Sip/1234
Application: AgentLogin
Data: 1234
Variable: _ALERT_INFO=info=alert-autoanswer
Callerid: AutoLogin[1234]
In order to send an autologin and autoanswer call to the agent 1234 on an
One solution I read about it is to forward the port 113 to an unused IP
address. I did so and now phones seems to be working again.
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Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble
on registration(? -- maybe it's on acquiring an IP?) has started again.
I still have the old sip.cfg, but can't figure out which option it is.
Any help?
Thanks!
-Ken D'Ambrosio
--
This message has been scanned for
Hi everyone,
I'm completely new to Asterisk and before I buy any card, I would like to
ask for some information.
1. We'll be using analog PSTN phone lines. Is there anything that I should
ask the telecom company before I buy the card? What I mean is whether the
card will be compatible with
Thanks for the info, Ken. I was about to research this tonight.
Todd
On Mar 12, 2007, at 12:53 PM, Ken Williams wrote:
In case it hasn't been posted before, here's instructions to get
the correct time to show up on your Grandstream GXP-2000's:
1. Login to phone
2. Go to Basic Settings
12 mar 2007 kl. 19.16 skrev Octavio Ruiz (Ta^3):
Since 1.4 ALERT_INFO variable has been deprecated. I used to send this
via AMI:
Action: Originate
Channel: Sip/1234
Application: AgentLogin
Data: 1234
Variable: _ALERT_INFO=info=alert-autoanswer
Callerid: AutoLogin[1234]
In order to send an
Evert wrote:
No one...?
This problem is really bugging me... :-/
Regards,
Evert
Evert wrote:
Hi all!
Working on the following brain-scratcher. I am setting up a Trixbox
system for someone who uses 'provider A'. Everything works fine, except
for the IVR: keypresses by
Ken D'Ambrosio wrote:
Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble
on registration(? -- maybe it's on acquiring an IP?) has started again.
I still have the old sip.cfg, but can't figure out which option it is.
Any help?
Ken: It's really smart to follow
OK, this makes sense, as I have seen this here in the US as well.
On 3/12/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
C F wrote:
Tomislav, really? and how does it show up on my POTS line?
It only can be seen if other end is also on Optima provider. Ant it is
shown exactly as originator has
Hi Matt,
Probably you already found it - but I think it could help others:
http://www.voip-info.org/wiki/view/Aastra+Failsafe+Reboot+Script
You have to give the password - but for us it was OK.
Best regards,
## nini @ www.modulo.ro ##
Matt wrote:
Is there a command in Asterisk that will
Brandon,
You're certainly inviting a challenge.
All you describe is possible with PBXware from www.bicomsystems.com a lot, lot
more of course and some helping hands.
Please contact me offline if you prefer for more detail.
Regards,
Steve
steve 'at' bicomsystems {dot} com
- Original
If you're up for it, I've done this a few times before , and with asterisk.
Contact me offlist, and I can help.
- Original Message -
From: Brandon mailto:[EMAIL PROTECTED] Comouche
To: asterisk-users@lists.digium.com
Sent: Monday, March 12, 2007 6:11 PM
Subject: [asterisk-users]
Brandon
Your on the right track with what is can do. It will also be good to
look into what kind of QOS you can do on the T-1 connections between
offices. I have an 8 office setup similar to this and many of your
goals I have achieved and would be glad to offer ideas and such if you
want to
Bruce Reeves wrote:
Does SIPAddHeader(Alert-Info:) not do it?
No, but from another thread, setting the _SIPADDHEADER variable works.
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Somebody could help me with a call back implementation, please?
I mean, I just want call to my Asterisk, hung up the phone, and wait it
calls me back... Somebody ever did that for local or international calls?
Thanks.
LC.
___
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Somebody could help me with a call back implementation, please?
I mean, I just want call to my Asterisk, hung up the phone, and wait it
calls me back... Somebody ever did that for local or international calls?
Thanks.
LC.
___
--Bandwidth and
Tomislav Parcina wrote:
I'm using AMI on Asterisk 1.2.15 and I'm having problems with DBPut action.
If I execute this:
Action: DBPut
Family: checkin
Key: 316
Val: yes
Try putting quotes around the value. I played with it a while back only
a little, but I can't remember if quotes did it or
Please start new threads for new questions.
Alex
On 3/12/07, Luis Claudio Santos [EMAIL PROTECTED] wrote:
Somebody could help me with a call back implementation, please?
I mean, I just want call to my Asterisk, hung up the phone, and wait it
calls me back... Somebody ever did that for local
Just wanted to update the group
I updated asterisk to SVN-branch-1.4-r58833M and page no longer crashes
Asterisk. My below example works great.
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Friday, March 02, 2007 3:27 PM
To:
Why does everyone want to go off-list? Is this not information that could
benefit others?
On 3/12/07, Bruce Reeves [EMAIL PROTECTED] wrote:
Brandon
Your on the right track with what is can do. It will also be good to
look into what kind of QOS you can do on the T-1 connections between
I take it back. It will not work if you hang up the calling phone first.
Still crashes
-- Executing [EMAIL PROTECTED]:1] SIPAddHeader(SIP/36651-b7d1cf48,
Call-Info: answer-after=0) in new stack
-- Executing [EMAIL PROTECTED]:2] Page(SIP/36651-b7d1cf48,
Wai Wu wrote:
Hi all,
Anyone know how to dynamically create meetme conference rooms on the
flight? I remembered a while ago there was a switch that tell meetme to
create the conference room is the room is not defined in the
meetme.conf. It doen't seem to be working for me anymore.
Thnx
Greetings,
Very occasionally, we have a complaint from a user that a portion of
a voicemail message is very speeded up - like when you press the fast-
forward button on an old-fashioned tape dictaphone. This affects both
the server-stored and emailed copies of the message. I have a sample
Hi:
I am working on a new system with a TDM-400P card with three FXO modules
and one FXS module.
The system has been in place for a week. Users are complaining of echo
problems. I have noticed this echo myself. It varies in severity. It is
sometimes bad enough to make it difficult to converse,
Luis Claudio Santos wrote:
Somebody could help me with a call back implementation, please?
I mean, I just want call to my Asterisk, hung up the phone, and wait it
calls me back... Somebody ever did that for local or international calls?
Luis,
Check out this article:
Anthony Rodgers wrote:
Very occasionally, we have a complaint from a user that a portion of a
voicemail message is very speeded up - like when you press the
fast-forward button on an old-fashioned tape dictaphone. This affects
both the server-stored and emailed copies of the message. I have a
Hi All
I have a problem with IVR scripts which consist mainly of Playback of
audio files, driven from an AGI application. There are clicks every few
seconds or more frequently that is audible on the remote end (PSTN), but
not on the Asterisk recording of the call. If I record the remote end
and
Just checked my figures, and I mean 0.5%-0.7%. Anyway, it is the
resulting
clicks that are the problem.
Any help still appreciated.
David
-Original Message-
From: David Brazier
Sent: 13 March 2007 00:33
To: asterisk-users@lists.digium.com
Subject: Playback 5% Too Fast?
Hi All
I
On Tue, Mar 06, 2007 at 11:14:15PM -0500, Steve Totaro wrote:
Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I
am totally floored with how cool it is!
Thanks,
Steve Totaro
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--Bandwidth and Colocation provided by
In general how I implemented is as follows:
- Caller calls asterisk.
- From AGI, asterisk gets the caller ID.
- Without answering, play back beeps, to simulate busy.
- When the user hangs up, asterisk detects broken connection, cannot
send beeps, and Originates a call back to the caller.
-
Can you provide some specific details as I would like to implement
something like this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov
Sent: Tuesday, 13 March 2007 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
All the code is in AGI. Take a look at the Originate application.
(http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Originate)
Klaverstyn, David C wrote:
Can you provide some specific details as I would like to implement
something like this.
-Original Message-
Ivo Zivkov wrote:
All the code is in AGI. Take a look at the Originate application.
(http://www.voip-info.org/tiki-index.php?page=Asterisk+Manager+API+Action+Originate)
I take it this is to make ultra-cheap calls from anywhere, right?
-Stephen-
___
This doesn't make sense to me. Are you able to give some example dial
plans?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ivo Zivkov
Sent: Tuesday, 13 March 2007 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
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