Hi guys,
I'm writing an AGI program and use EXEC DIAL to do the dialing. The result
reply from Asterisk doesn't come back until the dialing times out, or the
channel is hung up after the call is connected with remote party and finally
hung up. Is it possible to tell from the return code, or
In article [EMAIL PROTECTED],
Dima Pursanov [EMAIL PROTECTED] wrote:
How to limit the duration of the MeetMe conversation?
The easiest way is to set an absolute timeout on each participant
before they enter, by using AbsoluteTimeout(xxx) or the newer
Set(TIMEOUT(absolute)=xxx)
You will
I have this code which was taken from the phpagi project page along with the
following in extensions_conf and the output from the asterisk CLI. When I
call the 311 extension, I does nothing then hangs up. What am I doing
wrong??
php code
#!/usr/local/bin/php -q
?php
Marco,
Did you install the kernel sources? the messages that you wrote are telling
that you don't have the sources.
Regards,
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco Torrez
Sent: Tuesday, December 26, 2006 6:45 PM
To: asterisk-users@lists.digium.com
i have problem with dial-plan in php. I have 3 extention in dial plan, 555,551
and 551. the problem is that READ PIN||3 works in 555 and hangs in other
extentions. (after timeout asterisk writes thar user entered nothing). i
can't get what's wrong...
here is my dial-plan
[incoming]
exten =
There was a new format_g729.so but there was no codec_g729a.so, so I put the
right one for the x86_64 kernel.
I choose that on the make menuselect already.
After your question, I saw in the messages the report that codec_g729a.so
was not able to load, so I moved out. No messages about g729 after
Good morning,
I have a Polycom 601 with two side cars. I created a list of contacts in XML
and it shows up on the side cars exaclty how I set it up in the
-directory.xml file (in the order that I wanted it etc.). However
when I hit the directories button and then contact directory I
I don't think that's possible. We have the same issue.
-Original Message-
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 27, 2006 8:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom 601 Contacts List
Good morning,
I have a Polycom 601 with
You could put together a web page that talks to the Asterisk Manager.
-Original Message-
From: Rob Hillis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 26, 2006 11:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Agent presence
Hi
Not sure if this has anything to do with it but running the input.php script
directly from the command line gives this warning:
PHP Warning: Unknown(): Unable to load dynamic library
'/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file:
No such file or directory in Unknown
Ok,
I realize that by mistake I put the old codec_g729a.so instead of the one
belonging to the 1.4 version. (However it worked on the beta).
As soon as I changed it, now I got this on the /var/log/asterisk/messages
[Dec 27 10:24:25] WARNING[7512] translate.c: plc_samples 160 format 6
[Dec 27
Not quite the solution I was looking for - I was wanting the agent's
status to be reflected in it's presence hint. I'm somewhat inclined
to believe that 1.2 isn't going to do the job at this stage since I
don't think it supports SIP presence to the degree required.
Douglas Garstang wrote:
All,
I just installed latest Asterisk 1.4 release today and notice two problems
that I couldn’t figure it out how to resolve it.
SIP invitestate keep showing up in console. I am unable to stopping it and I
even tried to do core set verbose off and core set debug off. What else do I
have
Don't know if this will do it (see your full logs for details), but
the timeout lines in your 311 are at least depricated if removed --
use the set statement and functions like this
exten = s,n,Set(TIMEOUT(digit)=5)
Hope this helps.
on Wednesday 12/27/2006 blackwater dev([EMAIL PROTECTED]) wrote
If you want to install the php imap support it is usually doen according
to the linux distro you use. On debian the package would be php4-imap.
If you don't want to install it you need to make sure your php and
phpagi config don't require it.
blackwater dev wrote:
Not sure if this has anything
Wasn't Olle Johansen working on something that would allow (polycom phones at
least) to show the status of agents on the phone...
-Original Message-
From: Rob Hillis [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 27, 2006 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial
KC wrote:
I just installed latest Asterisk 1.4 release today and notice two
problems that I couldn’t figure it out how to resolve it.
SIP invitestate keep showing up in console. I am unable to stopping it
and I even tried to do core set verbose off and core set debug off. What
else do
Lol my mistake.. Thanks Kevin..
Sincerely,
KC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, December 27, 2006 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Verbose
Not sure if this has anything to do with it but running the input.php script
directly from the command line gives this warning:
PHP Warning: Unknown(): Unable to load dynamic library
'/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file:
No such file or directory in Unknown
Hints for agents work just fine insofar as showing when the agent is not
online (logged off), Ready (logged on) or on a call, but nothing when
the line is ringing or when the agent has been paused... at least not in
Asterisk 1.2.13. I'm not quite ready to take the somewhat significant
plunge
There is an index in the configuration file which I believe it will
obey. I'll try and find it later if you haven't found it by the time I
get to the office.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday,
Hey guys. I am new to the list and would like to know how to search it so
that I do not post any questions that have already been answered (like this
one)
- Mark
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To
The directory file has an sd Speed dial index tag. The phone honours this
index when displaying entries on the LCD screen and when the up arrow is
pressed. However, it does not honor this order, and instead displays entries in
alphabetical order, when you press the 'Directories' button.
Mark Greene wrote:
Hey guys. I am new to the list and would like to know how to search it
so that I do not post any questions that have already been answered
(like this one)
http://lists.digium.com/mailman/listinfo/
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty
Hi Mark,
I don't think there's a built in search (someone please correct me if I'm
mistaken here), but Google can filter results for you:
site:http://lists.digium.com/pipermail/asterisk-users/ searchterm
Alex
On 12/27/06, Mark Greene [EMAIL PROTECTED] wrote:
Hey guys. I am new to the list
I'm not sure if there is a more official method but Google has always
been my friend when searching the lists.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Greene
Sent: Wednesday, December 27, 2006 12:05 PM
To:
On Wed, Dec 27, 2006 at 11:05:29AM -0600, Mark Greene wrote:
Hey guys. I am new to the list and would like to know how to search it so
that I do not post any questions that have already been answered (like this
one)
http://gmane.org/find.php?list=asterisk
This one is
You can only search a month at a time... :(
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 27, 2006 10:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Searching the list
Mark Greene wrote:
Ed,
Thanks for the help. One more question, however. Everything is working
fine with the exception of sox joining the calls. I have sox installed
and monitor-join set to yes in both queues.conf and agents.conf
I installed sox after I installed Asterisk. Do I need to recompile
Asterisk
Thanks, I'll try. I'm using the trixbox and my 311 info was in
extensions_custom.conf if that means anything.
On 12/27/06, John covici [EMAIL PROTECTED] wrote:
Don't know if this will do it (see your full logs for details), but
the timeout lines in your 311 are at least depricated if removed
Ok, I did that and I can now do a locate and do now see:
/usr/lib/php4/imap.so
/usr/lib/httpd/modules/mod_imap.so
So I restarted apache, and tried to run my file from the command line and
get the same error. I have another one that doesn't use the phpagi class
and it works ok but throws the
On google, do your search terms site:lists.digium.com
- Original Message -
From: Mark Greene [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, December 27, 2006 11:05:29 AM GMT-0600 US/Central
Subject: [asterisk-users] Searching the list
Hey guys. I am new to
I'm looking for some doc's what is new in 1.4
in deep for the comand stun debug
neptun:/usr/src/asterik_pgk/asterisk-1.4.0/main # asterisk -r
Asterisk 1.4.0, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer [EMAIL PROTECTED]
Asterisk comes with ABSOLUTELY NO
I get the following warning when starting Asterisk 1.4. Does anyone
know what these mean, and/or how I can get rid of them?
[Dec 28 02:12:28] WARNING[3419]: translate.c:675
__ast_register_translator: plc_samples 160 format 6
[Dec 28 02:12:28] WARNING[3419]: translate.c:675
Is ztdummy still required with Asterisk 1.4 when no zaptel cards are
available to use for timing?
In all the beta releases I used to get a warning when Asterisk started
up, saying that no timing device was found. The warning seems to have
gone away with the full release of 1.4, which prompts the
Yes, timing (ie; ztdummy, if no hardware is installed) is still required for
things like meetme or iax2 trunking.
- David Thomas [EMAIL PROTECTED] wrote:
Is ztdummy still required with Asterisk 1.4 when no zaptel cards are
available to use for timing?
In all the beta releases I used to
thanks for the tips.
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I got the same messages but as far as I can see my problems are still with
translation codec problems between g726aal2, g726 and g729.
The problem with the G729 is refered to registration, since I cannot see
anything on the CLI referred to my license, however the message says
that is there.
Do you have a zap section on the CLI?
Just do ? And check if you have that. I have zaptel working on two machines
with wtc1xxp and ztdummy. The one with the card doesn't show zap section,
the other one with ztdummy does.
I thought that both should show the section on the CLI.
Carlos Alperin
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't
Re: [asterisk-users] Follow-me challengeWhat you can do is have the called
person press a digit to accept the call. If the user dosent then you can set
the h extension to update the call logs
- Original Message -
From: Eric Jacksch
To: Asterisk Users Mailing List - Non-Commercial
A PI that does asterisk on the side ?? WTF ??
- Original Message -
From: C F [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, December 24, 2006 2:22 PM
Subject: Fwd: [asterisk-users] The Good, Bad and Scam VoIP Providers
I Find It Funny, So I Decided To Let Others
Embarased like when you were caught wuth your pants down ?
- Original Message -
From: Al Bochter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 24, 2006 3:37 PM
Subject: Re: [asterisk-users] The
Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
On 12/27/06, Carlos Alperin [EMAIL PROTECTED] wrote:
Do you have a zap section on the CLI?
Just do ? And check if you have that. I have zaptel working on two machines
with wtc1xxp and ztdummy. The one with the card doesn't show zap section,
the other one with ztdummy does.
I thought that both
As an FYI, this has absolutely NOTHING AT ALL to do with Asterisk.
Asterisk is a PBX application. Dovcot is an email application. They have
nothing to do with each other. Asterisk is not a Linux distribution or
operating system.
I suggest that you ask your question on a more appropriate mailing
Dan, Please accept my sincerest appology. I had my head thoroughly up my
back orifice. I haven't kept up with the new IMAP feature in 1.4
I'll go back in my corner now :-)
On Wed, Dec 27, 2006 at 02:19:07PM -0500, Walt Reed said:
As an FYI, this has absolutely NOTHING AT ALL to do with
Mark wrote:
I've been attempting the same with Cyrus and get the same results.
The
interesting thing is if I take the same string (like
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX) and plug it
into the 'mtest' command from the c-client package, it works OK. I
have
not tried
On Wed, Dec 27, 2006 at 10:44:10AM -0800, Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
Not that I can be of much help, but:
what is your MAILBOX env. set
Did someone has an ATA Grandstream HT496 working on T.38 on Asterisk 1.4?
I'm trying different configurations but so far none has worked.
Thanks,
Carlos Alperin
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Well the addons from 1.4 are installed. This original Asterisk 1.2.x box
was created by my predecessor and he had the cdr_addon_mysql.so and
res_config_mysql.so files on a server that we copied to any new
installation. I'm not sure where he got these files. As far as I can
tell shouldn't the
Thanks for all the help. I finally got it all working. There was some
problems with the swift functions of phpagi as it couldn't find it so that
command was failing. I fixed it and all is well now.
Thanks!
On 12/27/06, blackwater dev [EMAIL PROTECTED] wrote:
Ok, I did that and I can now do
curious about what swift problem you saw and what you did to get the
swift problems resolved...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of blackwater
dev
Sent: Wednesday, December 27, 2006 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial
Dan,
I have IMAP support working now with Courier IMAP. Since Courier (and
probably Dovecot) do not support a single authuser connection that may
access any mailbox, you have to omit the 'authuser' and 'authpassword'
settings in voicemail.conf and then add the username/password login per
Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?
I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out,
Can anybody point me to a vendor that can provide a toll free number that
can be used in India to reach the united states? Verizon Business is
telling me they can't get one.
Thanks - Tom
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I printed out the exec statement in the swift function within the phpagi
script and noticed that it didn't have a swift location. For some reason,
it wasn't picking it so I just changed that to the location on my machine.
I also noticed that it was adding to many slashes to the temp file name.
Am Mittwoch, den 27.12.2006, 17:22 -0800 schrieb Tom Lynn:
Can anybody point me to a vendor that can provide a toll free number
that can be used in India to reach the united states? Verizon
Business is telling me they can't get one.
Looks like a -biz related question...
A quick google search
This is not true of PRI (ZAP channels). Asterisk gets status such as
ringing, answered, busy...
Thanks,
Steve
- Original Message -
*From:* Eric Jacksch mailto:[EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
Walt wrote:
Dan, Please accept my sincerest appology. I had my head thoroughly up
my
back orifice. I haven't kept up with the new IMAP feature in 1.4
Don't sweat it...
I saw your first post before going out for the day with the family, and
couldn't figure out how it wasn't related to
Tzafrir wrote:
On Wed, Dec 27, 2006 at 10:44:10AM -0800, Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
Not that I can be of much help, but:
what is your
Hi all,
I need to connect two asterisk server in same network and i'm using sip
user as my clients..
plz anyone suggest me
Regards,
Thiru
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Dovid B [EMAIL PROTECTED] wrote:
A PI that does asterisk on the side ?? WTF ??
Do you have a list of business types that are not allowed to use VoIP?
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/
I use an SPA3000 to connect to the PSTN (SIP/pstn). Since I only have one line,
if it is in use, and someone else tries to dial out, they get the
all-outgoing-lines-unavailable message played.
I'd like to find a way to instead tell them which extension is using the PSTN
line. I know that info
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