Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
you are using the attended transfer feature..
ist it already possible to hang up before the other person lifts the handset
without loosing the caller when you are doing an attendet transfer?
(person A takes an
Hi,
Somebody has
someinfos forasterisk and swyx connected via
DDI?
Somebody has a
example config for ddi wiith asterisk?
regards
rene
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Hello,
I am a newbie, so I apologize for this maybe simple question.
I want to connect two Asterisk machines with IAX.
>From one machine I want to call to the other Asterisk,but sometimes I
want to place the call on one context and sometimes in another one.
I how can I do this?? When dialing on
On Mon, 2006-03-20 at 11:38 +0530, ram wrote:
Hi
what is mtr ?
where can i find that
http://www.google.com/linux?hl=enlr=q=mtrbtnG=Search
Pete
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Unfortunately in Italy doesn't work: Italy and Spain uses Protocol Type2 and
app_SMS doesn't support it (to my knowledge).
http://www.rtx.dk/Files/Filer/tekniske%20artikler/SMStransmissionwithinthePS
TN.pdf
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
Hello,
Thought people might be interested in this.
I want my voicemails emailed to a person and not stored on my asterisk
server. However, I want them to have a sequential number. I found that
if I set the option delete=1 in my voicemail.conf file for the mailbox,
then the numbering would keep
Hello,
I am wondering if someone has got any ideas that can help solve this
problem.
I have a dial plan that you call into, and depending on certain conditions
it calls out on a number grabbed from a database.
Something like this :
exten = s,n,Do something
exten = s,n,Do something
Hi.
I am having troubles loading the res_ and cdr_odbc modules, they fail
because they cannot find libodbc.so.1
I have unixODBC properly installed and the needed DNS setup correctly.
Any ideas why I am having this troubles?
Where is asterisk looking for the libodbc.so.1 file?
And were can I
On Mon, 2006-03-20 at 09:32 +, David Waugh wrote:
NOTE: This is my first shell script so I'm sure it can be improved!
noted, in that spirit see notes below ...
***
[EMAIL PROTECTED] INBOX]# more /etc/asterisk/voicemail-clean
cd
Hello,
I recently bought a Junghanns Octobri Card. I have some problems with
this card to make outbound calls but I can receive calls.
I have 3 lines to PSTN and 3 lines to my existing PBX
FRANCE TELECOM -- OctoBRI -- Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h
-- OctoBRI -- PABX e-Generis
Hello!
I'm trying to setup a perl-deadagi, but my perl skills lack. can
someone tell me why the following code doesn't work:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$dialstring = $AGI-get_variable(DIALSTRING);
$res = $AGI-exec(DIAL $dialstring);
the asterisk output says:
Hello all,
I want to use mysql for to save the users of my
asterisk PBX. I use the realtime solution with mysql but when I made the
sip show peers command doesnt appear my users. My
configurations are:
res_mysql.conf
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser =
In article [EMAIL PROTECTED],
Darren Wiebe [EMAIL PROTECTED] wrote:
I'm using the Local channel in an app of mine and I'm finding that
the app is being cut out of the call path. You used to be able to
avoid this using the \n command but that doesn't seem to work any
more. This is on a
Thanks Bret for the input. Your solution seems a lot neater=)
I had problems with globbing I think it is called.
I kept getting files name being created called msg*.txt which caused
me problems later.
I think your way removes this.
The reason I was doing this was for testing purposes. I was
What about setting up DYNAMIC_FEATURES=pickupexten inside your
[globals] ?
This is needed for, as the variable name says, dynamic features. And
don't forget to set callgroup/pickupgroup to each one in your sip.conf
Does anyone tested the new application Pickup()?
[]'s
MM
On Mon, 2006-03-20
Hi;
I'm trying to record all inbound and outbound calls at a site, and I
have a problem with inbound calls that are transferred by a receptionist
using Snom's handset buttons (i.e. SIP transfer rather than using the
key sequences defined in features.conf).
The first leg of the call is recorded
Try setting it to sth like SIP/200 instead of a single number.
l.
On Mon, 20 Mar 2006 11:56:50 +0100, Christian B [EMAIL PROTECTED]
wrote:
Hello!
I'm trying to setup a perl-deadagi, but my perl skills lack. can
someone tell me why the following code doesn't work:
#!/usr/bin/perl
use
Tried:
$DIALSTRING???
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Lenz
Gesendet: Montag, 20. März 2006 12:56
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [Asterisk-Users] simple perl-agi - where's the error?
Hi,
Has anyone had experience installing AMP/FreePBX on Asterisk Business
Edition?
The main issue we have come across is FreePBX requires a dependency
PHP-PEAR PHP-GD which is not available on RedHat RHEL3 (ES)
Thanks
James
___
--Bandwidth and
snip
Anyway, so I went back to a plain text file for
sip.conf. What a dissapointment.
/snip
This is kind of backwards but you can make a script
that will pull all the info from the DB and save it as
sip.conf.
__
Do You Yahoo!?
Tired of spam?
of course, but this doesn't make the difference(i just simplified the
input-variable to verify it's not a regexp-issue). It should at least
try to use to dial the single number i've set, but it looks like the
variable is empty...
On Mon, 20 Mar 2006 12:55:38 +0100
Lenz [EMAIL PROTECTED] wrote:
Hi All
I had successfully tried out asterisk on the LAN ,
now I want to call outside using sipdiscount or using
http://exgn.net
my asterisk box is behing a Firewall and the
Internet usage is through a proxy server located at
192.168.20.20:8080
Now I want to configure asterisk
no, this doesn't make a difference
On Mon, 20 Mar 2006 13:01:00 +0100
René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote:
Tried:
$DIALSTRING???
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Lenz
Gesendet: Montag, 20. März
Melcon Moraes wrote:
On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote:
Hello all,
I have an asterisk @ home system running 1.2.4. Call pickup seems to
be a bit of a problem. I’ve looked at a lot of posts and the wiki,
which states that you need to define the pickup extension in
Hello All !!!I have 4 PSTNlines in the PBX server 1,2,3,4. Firstline will be usedby only one extension (i.e. for the boss) for incom
ing and
outgoing. This line is dedicated for him only.(The remaining lines will be shared bythe employees 1) Group Ahave access to lines 2 , 3 4.
Hi guys,
maybe youìve got the answer...!
When a caller(not internal, but from PSTN) call *, I need to let him hear a message, before * answer and the bill start running.
If is not clear, just let me know.
caller-telco(telco bill to the caller as soon as * answer)-asterisk
Thanks in advance.
--
One thing that may help:
I use outlook rule to move all the messages into a folder.
Then outlook has a feature, instead of sorting by date, or subject, you can
sort by conservation.
It then groups the messages by thread in date order, so you can sort through
the emails very quickly and allows
And don't forget to set callgroup/pickupgroup to
each one in your sip.conf
Call pickup works among IAX phones?
Mimmus
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I also may be able to contribute as well.
Thanks
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: Sunday, 19 March 2006 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How
Dear Francois,
Thanks for your advise,, I'll buy the echocan module
Best Regards,
Fernando
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Saturday, March 18, 2006 6:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial
James Sturges wrote:
Hi,
Has anyone had experience installing AMP/FreePBX on Asterisk Business
Edition?
The main issue we have come across is FreePBX requires a dependency
PHP-PEAR PHP-GD which is not available on RedHat RHEL3 (ES)
Thanks
James
Hello
I've been trying to compile zaptel 1.2.4 on Mandriva 10.2 , kernel
2.6.11-6mdk and i keep getting these errors:
#make linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-Drw_lock_t=rwlock_t
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o
gendigits.o gendigits.c
cc -o
There are about 10k subscribers to this list, there is a good number
to start with.
On 3/18/06, Rob Gillan [EMAIL PROTECTED] wrote:
Does anyone have a guesstimate of how many active Asterisk
installations there are? Sorry this is off topic, need it for a
customer proposal and they need
Also note the use of FXO for Overhead Paging needs. Most all systems
from Valcom, Bogen and some others are C.O. Line only and the line
converter can cause huge delays in broadcast..
On 3/19/06, Rich Adamson [EMAIL PROTECTED] wrote:
In the interest of Symmetry, does anyone else in the world
Google is a good friend, unfortunately the system admin who represent the
company we are installing is not so.
They a requiring an audited stable platform, aka Asterisk Business Edition.
So when we say we need to install non-certified package onto their
Enterprise Server, they say na!
Thanks
You should try '$res = $AGI-exec(DIAL, $dialstring);'
Christian B wrote:
Hello!
I'm trying to setup a perl-deadagi, but my perl skills lack. can
someone tell me why the following code doesn't work:
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
$dialstring =
On 20/03/06, James Sturges [EMAIL PROTECTED] wrote:
Google is a good friend, unfortunately the system admin who represent the
company we are installing is not so.
They a requiring an audited stable platform, aka Asterisk Business Edition.
So when we say we need to install non-certified
Hello,I can pickup a call from a specific number:exten = _8XXX, 1, Pickup(${EXTEN:1})But i couldnt pickup calls coming from PSTN to local extensions.Another question is it possible to pickup the last calling number without any exten.Can you help me?erkaN
Yahoo! Mail
Use Photomail to share
http://www.asterisk.org/node/36
Boasting close to a quarter-million users in over 200 countries...
MATT---
On 3/20/06, Andrew Latham [EMAIL PROTECTED] wrote:
There are about 10k subscribers to this list, there is a good number
to start with.
On 3/18/06, Rob Gillan [EMAIL PROTECTED] wrote:
On Mon, Mar 20, 2006 at 03:38:21PM +0200, Assaf Flatto wrote:
Hello
I've been trying to compile zaptel 1.2.4 on Mandriva 10.2 , kernel
2.6.11-6mdk and i keep getting these errors:
#make linux26
[ snip ]
/lib/modules/2.6.11-6mdk/build
make -C /lib/modules/2.6.11-6mdk/build
On 3/19/06, James Harper [EMAIL PROTECTED] wrote:
That being said, a mailing list with a forum interface (or a forum witha mailing list option) might be a reasonable compromise as it should
meet the needs of both mailing list lovers and forum lovers (assuming itis implemented properly!)
All- if
On Sat, Mar 18, 2006 at 08:23:03PM -0600, Rich Adamson wrote:
This same issue has been discussed many times over the last two years.
Not likely its going to change now.
I just love this attitude.
Could someone managing these lists outline the
requirements to change the lists?
Do we need a vote
What about the Monitor command from the manage api. It allows for
monitoring but not coaching.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Totaro
Sent: Saturday, March 18, 2006 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial
Kristian Larsson wrote:
On Sat, Mar 18, 2006 at 08:23:03PM -0600, Rich Adamson wrote:
This same issue has been discussed many times over the last two years.
Not likely its going to change now.
I just love this attitude.
Guess its not an attitude as much as having been around this list for
Hi,
I just download the SIP image (cmterm-7970_7971-sip.8-0-2-0.cop) from
Cisco, copy all files on my tftpboot, create a SEP{mac}.cnf.xml file
(take the one posted by Greg Oliver) with some modification.
If the secret= is empty on the server, I receive now request on the
Asterisk server but
On Monday 20 March 2006 09:39, Kristian Larsson wrote:
I just love this attitude.
A modicum of thought for others may save you from yourself. This has been
discussed many, many times. The problem is a complex one, and one that has
been thought through many times by people much smarter than
erkan kolemen wrote:
Hello,
I can pickup a call from a specific number:
exten = _8XXX, 1, Pickup(${EXTEN:1})
But i couldnt pickup calls coming from PSTN to local extensions.
I'm using a dialplan entry like yours:
exten = _*9,1,Pickup(${EXTEN:2})
and just tested it. Working fine using
On Monday 20 March 2006 09:56, Rich Adamson wrote:
FWIW, I'd vote to keep it the way it is now and I'll just make use of
the delete key to handle uninteresting noise.
Amen. I currently have 12619 messages in -users, and that's with kmail
expiring old messages. I've been on these lists as
Guys,
Thanks again for all your help. I've updated /etc/sysconfig/network and /etc/hosts as per your suggestions:
/etc/sysconfig/network:
NETWORKING=yes
HOSTNAME=localhost.localdomain
127.0.0.1 my external, static IP address asterisk localhost
/etc/hosts:
# Do not remove the following line,
no. the result is sligthly different(no quotes), but the variable still
can not be written:
GET VARIABLE DIALSTRING
AGI Tx 200 result=1 (Zap/G1/0892343242343)
AGI Rx EXEC DIAL
-- AGI Script Executing Application: (DIAL) Options: ((null))
Mar 20 16:12:10 WARNING[4478]: app_dial.c:773
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Monday, March 20, 2006 4:06 PM
there is also a more generic call pickup
using 'callgroup=2' and 'pickupgroup=2' in your sip
definitions. That approach uses *8 or *8# to pickup
Maybe I am misunderstanding what you did here, but I just want to make sure...
First, in the network' file, the goal was to change the hostname from
localhost.localdomain reference to a real hostname that would be accepted, so
that the file would look more like:
NETWORKING=yes
Mimmus wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Monday, March 20, 2006 4:06 PM
there is also a more generic call pickup
using 'callgroup=2' and 'pickupgroup=2' in your sip
definitions. That approach uses *8 or *8#
On 20 Mar 2006, at 15:39, Rich Adamson wrote:
Mimmus wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, March 20, 2006 4:06 PM
there is also a more generic call pickup using 'callgroup=2' and
Hi,
How often do you all have your ATAs and phone register with the
asterisk server. I am doing it once an hour, but now I am wondering
if maybe that is too long in between registrations?
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Chris Mason (Lists) wrote:
FYI:
I am trying to build zaptel-1.2.4 against the recently updated kernel
version 2.6.9-34.EL on Centos 4.2. but I am getting errors and it will
not build. This is apparently due to a typo in a kernel header
spinlock.h although I have not successfully modified
You'll want to learn all about Channels and groups. You can try here: http://www.voip-info.org/wiki/view/Channels+and+Groups
.
I've assumed that you have 4 FXO modules (to support 4 external phone lines) and 4 FXS modules (to support 4 local extensions).
Essentially you'll need
to define
Woops, noticed that the channels in my example are channel =
1. You'll need to change that so it jibes with your ZAPTEL.CONF
file...
H
On 3/20/06, hugolivude [EMAIL PROTECTED] wrote:
You'll want to learn all about Channels and groups. You can try here:
Hi,
Being SER user I use 5 minutes (300 seconds).
But you have to balance between load on your registrar server (like * in
this case) and keeping your database up to date. Too short
re-registration in huge system means literally tens of registration per
second. To long registration means:
hello, sometimes per day, below messages appears in my asterisk/messages
log...
any suggestion, what this mean? thx
PJ
Mar 20 07:57:53 WARNING[4672] channel.c: Prodding channel
'H323/ip$172.20.1.11:53473/331' failed
Mar 20 07:57:53 NOTICE[14280] chan_h323.c: Avoiding H.323 destory
deadlock
Ok,
If a user drops off (power failure, etc). I detect them in asterisk
as going offline within about 2 minutes. However, registration is
only happening once an hour. I have qualify=yes set in asterisk.
On 3/20/06, Arek Bekiersz [EMAIL PROTECTED] wrote:
Hi,
Being SER user I use 5 minutes
Hi,
I am planning to deploy an asterisk installation but I need to convince
a few managers that its a good idea.
Theres something I don't quite understand though,
I plan deploy a box on the end of 4 channel BRI ISDN and provide it an
ADSL internet connection.
Should a phone behind the asterisk
My thought was.. I wondered if having it register more often (Every 5
minutes) might help some users who experience intermitent 'no dial
tone' and have to 'reboot their device'
On 3/20/06, Arek Bekiersz [EMAIL PROTECTED] wrote:
Hi,
Being SER user I use 5 minutes (300 seconds).
But you have
Hello,
We are faced with a problem concerning queues.
When we have several calls in different queues, is there some sort of
way to open a channel between a (sip-)phone and a SPECIFIC call in a
queue using the Asterisk manager api?
We would like to do this even when we are not a member of
Its all about how you configure your dialplan. Asterisk doesn't know what a
PSTN or VOIP phone number is. If you want all 08444 numbers to go through a
certain trunk, then you set your dialplan up accordingly.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
The project is now soundly set at assman.sf.net with a few more updates committed to SVN. I have not released an official release yet since the package is still considered beta quality, but it's quite easy to check out the SVN.
-- Sig Langehttp://www.signuts.net/
What I would like to do is
exten = 1000,1,Dial(sip/1000)(zap/g1,97837560)
exten= 1000,2,Voicemail(u1000)
Basically a follow me app that rings numerous interfaces and
allows me to answer or it to time out and go to vmail. I didnt include the
time out here as I am hoping someone can
FaberK wrote:
Hi guys,
maybe youìve got the answer...!
When a caller(not internal, but from PSTN) call *, I need to let him
hear a message, before * answer and the bill start running.
If is not clear, just let me know.
caller-telco(telco bill to the caller as soon as * answer)-asterisk
The reason for it being 0 is because as long as you sit on the h
extension the call is not yet done, therefore asterisk has no clue
what those valuse are. If you use the h extension then you are messing
up the CDR.
On 3/20/06, Mark Ackroyd [EMAIL PROTECTED] wrote:
Hello,
I am wondering if
OK!! That's not what I did I've gone back and changed
things according to what you indicated, thanks for making it so simple
to folow...
The Asterisk box is on an internal network so instead of asterisk.mydomain.com
I tried using our external fixed IP address. The error messages
have
Hi. I tried to record a meetme conference using the r option -- using
asterisk 1.2.4 and the volume is so loud it clips.
Any way to fix this -- using the monitor the volume is generally fine.
Thanks.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend
On Mon, 2006-03-20 at 11:59 -0500, hugolivude wrote:
OK!! That's not what I did I've gone back and changed things
according to what you indicated, thanks for making it so simple to
folow...
The Asterisk box is on an internal network so instead of
asterisk.mydomain.com I tried using
On Monday 20 March 2006 11:46, John Daragon wrote:
Alas, most (if not all) telcos object to you transmitting voice over
their circuits before they've started to charge you for the call.
Incorrect. I do this all the time with a PRI. You can't do this with POTS.
Simply don't Answer() until
Thanks a lot!!!
Is exactly what I need to do.
Send a message, before answer.
Thanks to all!
F.2006/3/20, Andrew Kohlsmith [EMAIL PROTECTED]:
On Monday 20 March 2006 11:46, John Daragon wrote: Alas, most (if not all) telcos object to you transmitting voice over their circuits before they've
What I would like to do is…
exten = 1000,1,Dial(sip/1000)(zap/g1,97837560)
exten= 1000,2,Voicemail(u1000)
Basically a follow me app that rings numerous interfaces and allows me
to answer or it to time out and go to vmail. I didn’t include the time
out here as I am hoping someone can tell
Anyone got this working yet?Nope :(
Any update to this status?
JR
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Hi
iam working with asterisk with mysql Realtime
when i have confgured and run the asterisk
iam getting the following error
i dig all the places for help could not find the results
could some one help me what is wrong
iam using 1.2.5 on FC4
Mar 20 23:04:52 NOTICE[2054] cdr.c: CDR simple
On Mon, 2006-03-20 at 23:14 +0530, ram wrote:
Hi
iam working with asterisk with mysql Realtime
when i have confgured and run the asterisk
iam getting the following error
i dig all the places for help could not find the results
could some one help me what is wrong
iam using
Hi,
Is it possible to turn off the request for a security code when
transferring in FOP (Flash Operator Panel)? If not can the security code
be set to use the SIP or voicemail passwords? I know there is a forum
for FOP but no one seems to be answering there... so I thought I would
see if anyone
PickUp2:
http://linux.thorsten-knabe.de/asterisk/pickup.jsp
works very well.
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Panton
Sent: Monday, March 20, 2006 4:50 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Hi everybody. Yesterday I fix typo in spinlock.h and compiled zaptel. But today I have problems with soft phones. I tried to recompile zaptel and it showed errors again. So I don't understand what now it needs.
Brings words and photos together (easily) with
PhotoMail - it's free and works
hi, the command sip show inuse is giving me wrong results , the outgoing
column is not working, look at this, (i have an outgoing call on
22662848 and it appears free)
asterisk*CLI sip show inuse
UsernameincomingLimit outgoingLimit
226628490
That, and make sure you've got extconfig set to use mysql for it's
sippusers and sippeers and not odbc.
Aaron
Patrick wrote:
On Mon, 2006-03-20 at 23:14 +0530, ram wrote:
Hi
iam working with asterisk with mysql Realtime
when i have confgured and run the asterisk
iam getting the
Andrew Kohlsmith wrote:
On Monday 20 March 2006 11:46, John Daragon wrote:
Alas, most (if not all) telcos object to you transmitting voice over
their circuits before they've started to charge you for the call.
Incorrect. I do this all the time with a PRI. You can't do this with POTS.
On Monday 20 March 2006 13:49, John Daragon wrote:
Hell, you learn something new every short period of time. I have to
go try this out...
:-) It's called early audio in PRI parlance, some carriers do not offer it
but almost all do.
-A.
___
Hi
thanks for the reply
this what my extconfig
sipusers = odbc,asterisk,2_Sipsippeers = odbc,asterisk,2_Sipextensions = odbc,asterisk,2_Extensionsvoicemail = odbc,asterisk,2_VMUsersvoicemail_messages = odbc,asterisk,2_VM
waht is wrong with this ?
ram
On 3/20/06, Aaron Daniel [EMAIL
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15 users on the remote
server, all connecting via SIP softphones.
For some reason, there is an increasing number of calls where the callee
does not get any audio although the
I am having this problem also. I have 2 systems running 1.2.5. I had the
problem and one system was running 1.2.4 and the other was running a CVS
HEAD from October so I upgraded them both to 1.2.5 with no success.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
Hi all,
Can someone share with me his experience in
making asterisk-oh323
talk to quintum gateway without gatekeeper.
My set up is QUINTUM GATEWAY --IP ASTERISK (OH323)
Both are gateways..
but I dont know what authentication I will set
up in oh323.conf and how to set it
Barry Flanagan wrote:
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15 users on the remote
server, all connecting via SIP softphones.
For some reason, there is an increasing number of calls where the callee
does not get
Since you say you're using mysql as the backend, you need to change
anything that says odbc to mysql so that the server knows where to
find the db at. Also, you need to make sure the DB info is in
res_mysql.conf.
Aaron
ram wrote:
Hi
thanks for the reply
this what my extconfig
Hi,
I took a piece of Octtel SP200SO (SIP, FXS, FXO, LAN, WAN, QoS ...) from
local distributor for testing.
First surprise came, when I opened it ... no documentation was included,
the second, when I learned, there is no relevant doc on company web too!
Nevertheless in the rest I configured it
Hello, I got a Problem with my HFC Card,
I start my asterisk -c
The console comes up:
---snip--
Asterisk Ready.
*CLI
---snip--
Setting up debuglevel for span 1
---snip--
*CLI pri intense debug span 1
Enabled EXTENSIVE
The reason for it being 0 is because as long as you sit on the h
extension the call is not yet done, therefore asterisk has no clue
what those valuse are. If you use the h extension then you are messing
up the CDR.
So how can I tell it the call is complete and give the CDR values? Is it
just not
The only thing is I want to be sure I understand the statement above because
the only time I can see Asterisk needing to do an SRV lookup is if it is
handing a call to a carrier for termination.
Gabe, that is what I was talking about. Asterisk really needs the
ability to make use of the
Doug Lytle wrote:
Barry Flanagan wrote:
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15 users on the remote
server, all connecting via SIP softphones.
For some reason, there is an increasing number of calls where
Hi,
I am currently integrating our company's Toshiba PBX with the Asterisk version 1.2.1.
I bought Quad T1 card, and making the port 1 to connect to PSTN PRI
(use pri_cpe in zaptel.conf) and making the port 3 to connect to
Toshiba PBX (using pri_net in zaptel.conf).
The first stage goal is to
Is it so difficult to add a line in the dialplan directly under the one
that fails to failover to?
Aaron
David Thomas wrote:
The only thing is I want to be sure I understand the statement above because
the only time I can see Asterisk needing to do an SRV lookup is if it is
handing a call to
Hello, I have started having a strange problem.
Asterisk is connected via 4 analog lines to PSTN and
we have SIP phones internally. All was working fine
but recently each time a user calls from PSTN and when
he is leaving a voicemail for someone, the caller gets
disconnected after 30 secs. We
I am sure that must lead to potential trouble having multiple dial() one
after the other as a form of HA. Besides, that wouldn't work for my
LCDial().
- Gabe
- Original Message -
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
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