This should do what you asked:
http://voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message
bp
On 2/8/07, Forrest Beck [EMAIL PROTECTED] wrote:
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
It is called exec.
http://www.google.com/search?hl=enq=asterisk+agi+exec
On 1/18/07, Thomas Hecker [EMAIL PROTECTED] wrote:
Hi everyone,
Is it possible to call an asterisk function out an AGI script? How do I do
this?
Thank you,
Thomas
___
Try www.asterisk2billing.org
On 1/11/07, Pablo Bullian [EMAIL PROTECTED] wrote:
Hi,
I have an issue with the authentication for the outgoing calls.
What I want is to give every user a different password, that they must
enter everytime they make an outgoing call.
What are my possibilities?
Try adding a forkcdr in just before your dial command.
bp
On 1/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
I have a little call recording script that I am running and it works
fine, but I have one problem. I get CDR when a user calls into the
extension, but I do not get CDR for
I am running console. I'm a newbie for AGI's but not that new.
Thanks,
bp
On 12/19/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Dec 19, 2006 at 09:35:56AM +0200, yusuf wrote:
to see debug output for AGI's, you *must* be connected to the first Ast
terminal. So start Asterisk like
.
Also, any output to stdout is interpreted by asterisk as a command, so
those fputs statements would be a problem -- do
fputs($stdout,VERBOSE \There have been\\n);
fputs($stdout,VERBOSE \$row_count calls made\\n);
instead.
William Piper wrote:
List,
I finally decided to break down start
VERBOSE125 calls made
AGI Tx 510 Invalid or unknown command
-- AGI Script test.php completed, returning 0
-- Executing Hangup(SIP/216-e866, ) in new stack
On 12/19/06, William Piper [EMAIL PROTECTED] wrote:
Jay,
I just tried the suggested changes... same response.
I tested the script
List,
I finally decided to break down start playing with AGI scripts, but for
the life of me, I can't figure out what I am doing wrong.
Below is a super simple script to run a query in mysql to see how many call
records there are for the extension calling in, then print the total in the
CLI.
I second that Luki.
We at www.cyberdyne-ip.com (yes shameless plug) only use ulaw for
termination. Of course we have to offer g729, GSM, etc. to our customers...
but for best quality, we transcode to ulaw if we send the call to another
carrier for termination. 729 may use less bandwidth and in
Check out www.cyberdyne-ip.com. Great rates, great quality, unlimited
channels, and an easy to use GUI to manage your account.
FYI, You may have more responses if you ask the -biz list.
bp
On 12/15/06, Paul Connolly [EMAIL PROTECTED] wrote:
We currently have an Asterisk system with a PRI
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo
bp
On 12/17/06, Damon Estep [EMAIL PROTECTED] wrote:
Started out looking for what I thought was going to be a simple variable
name, have not found it.
Does anyone know of a variable that would
://www.bochterservices.com/?t=TFdidt=email
For new and used security itemshttp://www.bochterservices.com/?j=storet=email
BUY Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=email
William Piper wrote:
Check out www.cyberdyne-ip.com. Great rates, great quality, unlimited
channels
I've also had these problems. If the call is going between two Asterisk
servers, connect them with dtmf=info. That solved my problems.
bp
On 12/10/06, Forrest Beck [EMAIL PROTECTED] wrote:
I too have seen this. I have to press the digits just right. I have
tried RFC2833, and Inband to send
In sip.conf set dtmfmode=INFO
On 12/7/06, CheungJenny [EMAIL PROTECTED] wrote:
Hi all,
I have a question: how to configure Asterisk to support SIP INFO method?
I encountered this problem when I find my UA don't send INFO message to
another UA, actually it should. Asterisk was used as a SIP
Something like this should work in your iptables:
iptables -A PREROUTING -t nat -p tcp --dport 1234 -i eth0 -j DNAT
--to-destination 127.0.0.1:4569
iptables -I FORWARD 1 -d 127.0.0.1 -p tcp --dport 4569 -j ACCEPT
This would forward port 1234 to port 4569.
bp
On 11/23/06, Zeeshan Zakaria
To hide the caller ID, do this:
exten = _9NXXNXX.,1,Set(CALLERID(all)=Unknown00)
exten = _9NXXNXX.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
bp
On 11/7/06, Nik Engel [EMAIL PROTECTED] wrote:
Hi all !I have a question regarding flexible callerid settingusing the misdnI want to acheive
That won't check to see if it is a valid extension... it will only make sure that it is 4 digits.
If you want to send everything just do:exten = _X.,n,Dial(SIP/[EMAIL PROTECTED],60,tr) That will send everything over 1 digit to ${SERADDRESS}
bp
On 11/6/06, Evert [EMAIL PROTECTED] wrote:
Hi!
Does your variable ${SERADDRESS} have the port number defined? If you replace it with localhost, I believe port 5060 will pick up the call by default.
Try making sure that your ${SERADDRESS} has the IP like the following: 127.0.0.1:5070 (replace 5070 with whatever port your SER is listening on).
It means that the number you are calling does not exist in the dialplan of wherever you are sending it.
FYI, next time start a new email instead of hijacking someone elses thread.
bp
On 11/6/06, Angel Heart [EMAIL PROTECTED] wrote:
Hi,
Could anyone knows what this error codes means;
-- Got
The transfer function will doa 302 redirect...
http://voip-info.org/wiki/index.php?page=Asterisk+cmd+Transferbp
On 11/3/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi guys,I've been looking on wiki, but i could find it only for chan_capi:
http://www.voip-info.org/wiki/view/Asterisk+PBX+functionsIn
You're on the right track.
Try the following:
[whatever]
exten = 988,1,Answer
exten = 998,2,Background(agent-newlocation)
exten = 998,3,DigitTimeout,5exten = 998,4,ResponseTime,10
exten = _,1,Dial(SIP/[EMAIL PROTECTED],60,tr)
exten = i,1,Playback(pbx-invalid)exten = i,2,goto(whatever,988,2)
Steve,
I would suggest doing Set(__CALLERID(name)=${ANI}) without the __underscore.
I would think that Set(__CALLERID(name)=${ANI}) is not the same as Set(CALLERID(name)=${ANI}) so when you are doing the NoOP(${CALLERID}) it is looking for $CALLERID(name) $CALLERID(number)not $__CALLERID(name)
I think this page will get you on the right track:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
bp
On 11/1/06, Zak Kinion [EMAIL PROTECTED] wrote:
Hello all,This is my first message to the mailing list.I am seeking advice as tohow to proceed/what to get for my current
one more question.Can I do the same add fieldname=1 if I adda field fieldname in the cdr table to perform the same action?
On 10/19/06, William Piper [EMAIL PROTECTED] wrote: In cdr_mysql.conf add userfield=1 under the globals setting. bp
On 10/18/06, unplug [EMAIL PROTECTED] wrote: I want to set
In cdr_mysql.conf add userfield=1 under the globals setting.
bp
On 10/18/06, unplug [EMAIL PROTECTED] wrote:
I want to set some custom data in the field of userfield in table CDRas following.exten = s,19,Set(CDR(userfield)=1234)
exten = s,20,Dial(SIP/1234)However, the userfield doesn't get update
You may have better luck asking the a2billing list.
Try here: http://forum.asterisk2billing.org/
bp
On 10/12/06, Khaled Chehab [EMAIL PROTECTED] wrote:
Dear
I am using a2billing accounting software, how can I charge on the destination target not at the caller side
Ex: if user A have 10$ and
than zero, wheather they are in use or
not...anywayThanksI'll keep diggingGeorgeOn 10/9/06, William Piper [EMAIL PROTECTED] wrote: A2B already shows this in the DB. If you have any php/perl skills just run a
query like the following: select * from cc_card where inuse 0 Then write a js to refresh
No idea, I've never used Trixbox.
I believe they have a support forum though...
bp
On 10/9/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi William,Thank you for response. Sorry. I forgot to say that am configuring using Trixbox. Can you tell me the solution? Thank you.
Regards,Chandra,
William Piper
A2B already shows this in the DB. If you have any php/perl skills just run a query like the following:
select * from cc_card where inuse 0
Then write a js to refresh the script everyfew seconds. I believe this will give you the reporting that you'd need.
bp
On 10/9/06, George Masgras [EMAIL
Your server seems to be doing exactly what you are telling it to do:
-- Executing Playback(SIP/216.89.79.2-09e1d020, ss-noservice) in new stack-- Playing 'ss-noservice' (language 'en')
Read the extensions.conf directions on the wiki site:
On 10/5/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I have an [EMAIL PROTECTED], using Broadvoice, with 4 Grandstream Budgetone SIP phones. After a cool boot the system works well, all extensions work well, I can place and receive calls from all extensions, and everything looks normal.
Sure sounds like a firewall issue... if you pinging port 4069 and it is not coming back, that sounds like a firewall problem. Try taking down your iptables and then try see what happens.
bp
On 9/28/06, Wolfgang_Borgon [EMAIL PROTECTED] wrote:
David,Yes, I've also forwarded port 4569 to the
Start up a sip debugip x.x.x.x (x=ip address of thedevice)and see if it is trying to register.You should be able to tell from that if you needto mess with the NAT settings. If nothing is coming in at all, it sounds like a networking issue.
bp
On 9/19/06, Allan Kamau [EMAIL PROTECTED] wrote:
I
Try changing your password to all alpha or all numeric. I've seen problems, ongrandstream in particular, where alphanumeric passwords did not pass correctly. Perhaps your softphone has the same type of problem.
bp
On 9/12/06, David R. [EMAIL PROTECTED] wrote:
Okay. I'm setting up my first
I've used Voxee for about a year now (as a backup carrier) and I've needed to contact their support only once. Pretty good, I think. Support was fast and fixed my problem within about an hour after I reported it.
I know only 1 support experience isn't really enough to warrent a good service
Both trixbox and asterisk2billing have their own lists... you may have better luck searching there.
bp
On 9/11/06, Steve Totaro [EMAIL PROTECTED] wrote:
Sharon Lim wrote: Hi all, I have tried to install freepbx and a2billing application. Now see
both application is not integrated special on cdr
It's still beta... here is the link:
http://www.grandstream.com/BETATEST/GXP2000_BT200/
bp
On 9/10/06, Barry D. Hassler [EMAIL PROTECTED] wrote:
Firmware 1.1.1.9? Where's that? most recent on Grandstream's site that I see is
1.1.0.16, which I have loaded.
On Fri, 2006-09-08 at 20:20 -0500,
I've noticed that Grandstream works better using stun and not port forwarding your router.
Try setting stun.xten.com or stun.fwdnet.net in your GS2000 and make sure sip.conf has nat=yes. It should work fine.
Also, i've noticed that Linksys wireless with speed booster has something in it that is
whatever the did is needs to be put in the extensions.conf told to dial your cellphone.
Example:
exten = _011123445566,1,Dial,SIP/[EMAIL PROTECTED]
assuming that your using a SIP carrier, replace 1234567890 with your cellphone 1.2.3.4 with the carrier's IP or carriers context name in
Check out http://www.grandstream.com/GAPSLITE/
It's a bit of a pain to setup but it works. You will need to create a script that changes the info on the template for each ATA that you want to configure.
bp
On 9/8/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
The configuration tool downloaded from
This is what I do:
[cf]exten = _*72XXX,1,DBput(CF/${CALLERIDNUM}=${CALLERIDNUM:-10:3}${EXTEN:3})exten = _*72XXX,2,Answerexten = _*72XXX,3,Playback(call-fwd-unconditional)exten = _*72XXX,4,Playback(is-set-to)
exten = _*72XXX,5,SayDigits(${EXTEN:3})exten =
Does the phone have stun settings? If so, try using stun.fwdnet.net and take out the port forwards and see if it works.
bp
On 9/7/06, Noc Phibee [EMAIL PROTECTED] wrote:
yusuf a écrit : Hi, you dont have to/should'nt be using different SIP ports for each
phone.Its completely not needed.Also, you
This is not the list for [EMAIL PROTECTED]. For questions about [EMAIL PROTECTED] functionality, they have their own mailing list.
bp
On 9/3/06, George A. Roberts IV [EMAIL PROTECTED] wrote:
No one has any ideas?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of George A.
The correct syntax would be sip show agents or sip show agent (agentname)
bp
On 8/31/06, Delca [EMAIL PROTECTED] wrote:
Hi, i'm using Asterisk 1.2.9.1 and i'm needing the AGENT functionsince i need something to offer the agents a way to check if they are
logged in or not. i was specting to use
Yea, but even with that you can't run a macro from the CLI. You'd need the manager API to do this.
bp
On 8/30/06, Tim St. Pierre [EMAIL PROTECTED] wrote:
Sort of.There is a command line argument to the asterisk process that runsit's arguments as CLI commands.You could write a shell script that
exten = 0.,1,Dial(SIP/Voip1/${EXTEN:1}SIP/voip2/${EXTEN:1},90,rt)
exten = 0.,2,hangup
bp
On 9/1/06, Noc Phibee [EMAIL PROTECTED] wrote:
Hiit's possible to create a group of outgoing dial ?For exemple:exten = _0.,1,Dial(SIP/voip1/${EXTEN:1},90,rt)
exten = _0.,2,Hangupexten =
on macros. Larry William Piper wrote: I don't know then, I do the same exact thing:
exten = _352688,3,Dial,SIP/202SIP/214|20 Perhaps try sending everything in that context exactly as it is typed let us look at it.
I'm pretty sure you have something configured incorrectly. Thanks, bp
= _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there.
Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp
___
--Bandwidth and Colocation provided by Easynews.com
Google is your friend:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState
bp
On 8/30/06, Juraj Bednar [EMAIL PROTECTED] wrote:
Hello,I would like to somehow get the presence of IAX2 and SIP users fromAsterisk Manager API or using any other means.
I tried
Did you try a combination of qualify=yes in sip.conf and then try the ExtensionState in the manager?
Seems like if qualify=yes or 2000... whatever, is not set then asterisk will not always know the state of the phone if it looses registration. That would seem to explain the problem you have with
I don't believe that the CLI was ever designed to do what you are asking. You should check out the manager API to do this type of stuff.
bp
On 8/30/06, unplug [EMAIL PROTECTED] wrote:
Hi,In CLI, I can issue a dial command.How can I run a macro in CLI?Is it possibe?Thanks.
Sounds like you need to invoke the asterisk -rx comand or do it via the manager api.
I personally prefer doing it via php. You could write a php scriptlike the following:
shell_exec(/usr/sbin/asterisk -rx 'database put cidname 18005551212 Char String');
bp
On 8/30/06, Nilesh Londhe [EMAIL
of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry
William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120)
bp
Sounds like you still have the old exten still there.
Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120)
bp
On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote:
This is a reply to a fairly old thread.My EXTEN string is meant to ring 3 phones (will increase to 12) thus:
old: exten
be deleted?Larry
William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff
[EMAIL PROTECTED] wrote: This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase
Perhaps a stupid suggestion... but did you make sure that the ATA had the T38 selected in the GUI?
bp
On 8/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Hi,I've installed Asterisk t38passthrough branch and I'm using oneGrandstream ATA to connect Asterisk to a Fax machine. Every time I send
a
That did the trick. Thanks for the tip.
Interesting though. Although technically it is behind a NAT, it is also connecting with the server who is also behind the NAT, I figured that in the eyes of the server... it would need NAT=no because neither device is connecting to it *through* the NAT.
List,
I have been using asterisk for a while now finally ran into the problem that I hear so often... one way audio.
We were using our asterisk server with 1 NIC and a Public IP with multiple ATA's all in different locations with no problems at all. We have now decided to migrate our existing
Yea, I've already tried that. I've tried canreinvite=yes and no.
Thanks though. Keep 'em coming.
bp
On 8/14/06, Alexander Lopez [EMAIL PROTECTED] wrote:
This may not solve your problem but try adding
canreinvite=no
to your sip.conf definitions.
Snip
The command is rtcachefriends=yes not rtpcachefrends
Check out this page: http://www.voip-info.org/wiki-Asterisk+RealTime
bp
On 8/4/06, Dovid Bender [EMAIL PROTECTED] wrote:
Yes.- Original Message -From: Peder @ NetworkOblivion
[EMAIL PROTECTED]To: Asterisk Users Mailing List -
You guys are making this morecomplicated than it needs to be.
Set your context to dialout do this:
[dialout]
include = blocked
include = notblocked
[blocked]
exten = _003763.,1,Congestionexten = _003764.,1,Congestionexten = _003765.,1,Congestion
[notblocked]
exten = _00376.,1,Dial(my iax
On 8/2/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Carlos Chavez [EMAIL PROTECTED] wrote:
The only thing you can really do from the dialplan is to use the CDR(userfield) function to add custom information, but I do not think there is a way to put additional fields into the CDR database
Sure it's possible. The install is no different than if you were to install it on a standard Asterisk server.
bp
On 7/31/06, Giedrius Augys [EMAIL PROTECTED] wrote:
Hi,With freepbx I have created sip users and sip trunks. But I need to charge calls and I want to use a2billing. But in a2billing I
Set maxmsg=10
Check out this site: http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
bp
On 7/31/06, Khaled Chehab [EMAIL PROTECTED] wrote:
Hi,
Dear
How can I limit the number voicemail messages for a user by 10 messages only,
I am using [EMAIL PROTECTED]
Regards
On 7/31/06, Joshua Colp [EMAIL PROTECTED] wrote:
I tried to edit the cdr import function but I didn't know where it placed or what function to edit ,
Please can you tell me where to place thisfunction exten = s,1,Set(CDR(userfield)=${SIPCHANINFO(recvip)}) to have it stored in the mysql record .
On 7/31/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote:
On Monday, July 31, 2006 3:12 PM Kai Ober wrote: (How do you get to the dial command, can you send the extension for
this?) the idea is toto use $EXTEN.call a macro with $EXETN as an argument ...The problem is this:exten =
On 7/26/06, Randy Paries [EMAIL PROTECTED] wrote:
On 7/25/06, William Piper [EMAIL PROTECTED] wrote:
On 7/25/06, Randy Paries [EMAIL PROTECTED] wrote: Hello, I just got my Asterisk up and running, and everything is great
What i can not seem to find is a doc that describes any of the user
You'll need to create a php or perl script to load your sip table into a txt file before doing a reload. You can create your own or just steal the code from AMP and modify it a tad. I believe the file name is retrieve_sip_conf_from_mysql.pl
bp
On 7/25/06, marek cervenka [EMAIL PROTECTED] wrote:
On 7/25/06, Randy Paries [EMAIL PROTECTED] wrote:
Hello,I just got my Asterisk up and running, and everything is greatWhat i can not seem to find is a doc that describes any of the user commands
Like is there things like, end message or listen to the message i amleaving , or anything like
On 7/24/06, Khaled Chehab [EMAIL PROTECTED] wrote:
Hi
Please how can I get the user
register ip address and put it at cdr ,its too important
Thanks
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer
I believe that some variation of that command and the
List,
I'd like to create custom cdr columns in the database.
Here is what I'm trying to do:
exten = s,2,Set(CDR(ipaddress)=${ipaddress})
I'm able to get the ipaddress and set it toa variablebut I'mnot sure how to record the IP addressto the CDR's.I know it has something to do with
own table and write whatever you want in it via AGI.
- Original Message -
From:
William Piper
To:
Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, July 24, 2006 4:40 PM
Subject: [asterisk-users] create custom cdr's
List,
I'd like to create custom cdr columns
On 7/18/06, Jerry Bonner [EMAIL PROTECTED] wrote:
Hi all,
Does anyone have any tips on how I would accomplish a plan where if a user dials 4 digits, then prefix 6 digits, then if there is a local extension configured for that number dial it, otherwise send it out another sip gateway ( my pstn
Why not use mysql?Do something like this: exten = s,1,MYSQL(SELECT * FROM whatever)bpOn 7/13/06,
Tomislav Parčina [EMAIL PROTECTED] wrote:
Hi list!I'm planning do use LookupCIDName application. TO use it I need to input CID data to internal asterisk DB. Question is, how much data can I store to
Assuming I understand what you are trying to do, just put accountcode=whatever in your iax.conf for each user.bpOn 7/13/06, Matt
[EMAIL PROTECTED] wrote:Ok,Here is my scenario
I have one (1) server that does my termination. (SERVER A)I have three (3) call centers that I want to terminate
butwondered if it would work since you aren't pushing anything back to
them).At any rate, I did just go with the accountcode=blah for each account.On 7/13/06, William Piper [EMAIL PROTECTED] wrote: Assuming I understand what you are trying to do, just put
accountcode=whatever in your iax.conf for each
Man, with a thread like this... who needs a soap opera? ;-)
bp
On 7/12/06, Steve Totaro [EMAIL PROTECTED] wrote:
Steve Totaro wrote: Jeremy McNamara wrote: Gonzalo Servat wrote: He is not doing a very good job, Jeremy. For starters I sincerely
doubt he actually paid anyone $3,000. He sounds like
On 7/5/06, [EMAIL PROTECTED] [EMAIL PROTECTED]
wrote:
so when i dial sip:[EMAIL PROTECTED] from an ip phoneasterisk reply 404 not found .
This means that the number that you dialed does not exist in voiptalk's list of phone numbers.
when i dial from asterisk console Dial [EMAIL PROTECTED]
Did you do a logger reload command from the CLI?
bp
On 7/5/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I want to have the output of 'sip debug' go to /var/log/asterisk/messages.My logger.conf has:console = notice,warning,error
messages = notice,warning,error,verbose,debugHowever, output from
Message-From: William Piper [mailto:
[EMAIL PROTECTED]]Sent: Wednesday, July 05, 2006 11:58 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] 'sip debug'
Did you do a logger reload command from the CLI?
bp
On 7/5/06, Douglas Garstang [EMAIL PROTECTED]
wrote
, and typed 'sip debug' on the console. All sip debugging went to the console ONLY. That's the opposite of what I am trying to do. I'd like to get the output from 'sip debug' go to to the messages file only.
-Original Message-From: William Piper [mailto:
[EMAIL PROTECTED]]Sent: Wednesday, July
If you are using putty as your ssh client...create the ssh tunnel to the * box then go tosession- logging- log printable output only in your putty configuration save.
To do it on the asterisk box only, I only know of the logger:
1. In logger.conf set full = warning,error,verbose,debug then
The same thing happened to me... I had to get a linux expert to take care of it for me.
I believe the files were either libpamor libss. They were telling asterisk to shutdown. I believe they deleted the files it that fixed it.
Tighten down your firewall.
bp
On 6/28/06, Doug Lytle [EMAIL
On 6/28/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jun 28, 2006 at 04:39:29PM -0400, William Piper wrote: The same thing happened to me... I had to get a linux expert to take care of
it for me. I believe the files were either libpam or libss. They were telling asterisk to shutdown. I
Why not add the g parameter and make your deadAGI as the next priority?
I think that would accomplish what you are trying to do.
Example:
exten = x,1,Dial(Sip/|30|gL(6:3:1))exten = x,2,DeadAGI()bp
On 6/27/06, El Flynn [EMAIL PROTECTED] wrote:
Andrew Nowrot wrote: Hi I
Although I've never tried it along withthe L option, you couldtry absolutetimeout:
exten = x,1,AbsoluteTimeout(6)
exten = x,2,Dial(Sip/|30|L(6:3:1))bp
On 6/27/06, Andrew Nowrot [EMAIL PROTECTED] wrote:
Thanks for all repliesI noticed that L option does not hangup the
()
exten = x,10,hangup()
This should send anything that did not clear normally (I.e. the timeout) to priority 4. Normal clearing will be sent to 10.
Again, not tested, but some variation should work for you.
bp
On 6/27/06, Andrew Nowrot [EMAIL PROTECTED] wrote:
On 6/27/06, William Piper
[EMAIL
Here is a bash script that will increase volume of all wav files in a directory:---#!/bin/bash for i in *.wav; do val=${i%.wav} echo converting $val.wav sox $i -v 2 $I $val.wav
done
-v 2 will increase the volume 2 times.
Here is a patch that someone
John,
I'm glad it worked for you.
Correct me if I'm wrong, but I believe that s will only workfor a macro.
Good luck with the outbound, if all else fails... givebroadvoice a call. They **may** have example conf files for you to look at for interconnecting with them.
bp
On 6/21/06, John
Close but not quite. Try below:
1. Setup sip.conf in theremote server to direct the call to the correct context
[incoming]
host=(xxx.yyy.zzz.xxx)IP of the sending servertype=friend context=(context that is holding theexten for the user) allow=ulaw
2.Setup extensions.conf on theremote serverlike
If your phone number is 5703380128, then you need to put it in your extensions.conf.
exten = _5703380128,1,Answer()
bp
On 6/21/06, John Klimek [EMAIL PROTECTED] wrote:
Anybody else able to help...?On 6/19/06, John Klimek [EMAIL PROTECTED]
wrote: Ahh, good catch.I've changed the context to be
List,
Does anyone know how to add the dst Country to the CDR's via Macro (preferably).
For example, I will add a column in the cdr DB table andwhen someone dials 01158212XXX. I want the CDR's to show Caracas as the destination in this new column.
I have all of the International destinations in
Thanks Bret, but how about an example or webpage?
I'm not finding anything on google about this command for asterisk.
What about AppendCDRUserField()... would this work?
bp
On 6/20/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Tue, 2006-06-20 at 10:49 -0400, William Piper wrote: List
What version of * are you using? I am running 1.2.7.1 with call-limit= and it works fine.
bp
On 6/19/06, Eric Bishop [EMAIL PROTECTED] wrote:
Hi all,We have a SIP trunk with * and even when there are calls in progress sip show inuse always shows 0 calls in progress. I have outgoinglimit and
You don't need a T1 card for a data T1. Just run it through your Cisco box send it over to your NIC on the asterisk box.
bp
On 6/19/06, Warren [EMAIL PROTECTED] wrote:
I have a data T-1 available to me to do some testing of a new asterisksystemthat I am putting together.Do I just leave this T
Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1.
Remember, you have 1544Kbs of bandwidth.
g279=8Kbs per call
uLaw=64Kbs per call
Just do the math.
bp
On 6/19/06, Warren [EMAIL
Check out www.plainvoip.com, they are about a 3-4 second setup time for me.
bp
On 6/19/06, Arnaud [EMAIL PROTECTED] wrote:
Did you check out their call setup time ?thanks - ArnaudOn 6/19/06, Cullin J. Wible
[EMAIL PROTECTED] wrote: Use Teliax - http://www.teliax.com/ Cullin J. Wible Co-Founder
Just put your carriers in static sip.conf and put your users in the realtime DB.
bp
On 6/15/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Kevin Fleming has said on numerous ocassions that this is known not to work, and is not supported.
-Original Message-From: Benjamin Stocker
You can do this without needing an AGI.
Look here http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail
bp
On 6/16/06, Miles Scruggs [EMAIL PROTECTED] wrote:
I want to setup a fragment of my dialplan to dial an ext at the sametime as another, but only if the other is avalible for
13Kbs
Google is a wonderful tool. Learn to use it!
bp
On 6/14/06, Pablo Allietti [EMAIL PROTECTED] wrote:
On Fri, Jun 09, 2006 at 04:45:51PM -0400, William Piper wrote:GSMand what is the size in KB that gsm spent?
bpOn 6/9/06, Pablo Allietti [1]pablo@lacnic.net wrote:hi all, i saw in digium
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