Hi all,
I'm not sure wether it is a bug or not, so I'm asking for your opinion
before submitting it to the bugtracker.
The problem:
I use asterisk with in sip.conf a non standard bind port of 5070 set.
Now when asterisk sends out an Invite message to my sip proxy, the
contact header in de
Hello,
1) How are you setting the nonstandard bind port? Just with a bindport
on the specific peer?
2) The Record-Route header is only for proxies, so it is not relevant
here; a B2BUA cannot set one.
3) What happens if you have the proxy append a 'received' parameter to
the Contact URI
Hi all,
I'm not sure wether it is a bug or not, so I'm asking for your opinion
before submitting it to the bugtracker.
The problem:
I use asterisk with in sip.conf a non standard bind port of 5070 set.
Now when asterisk sends out an Invite message to my sip proxy, the
contact header in de
Abel Monzon wrote:
and then in my softphone I call to 1 the asterisk log say this:
-- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php
== a2billing.php: Failed to execute
'/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory
--
Hi
First off, you replied a previous mail to the list, and hence your
message appears as part of a previous thread. To post a new message
start a new message.
Also,
On Sun, Oct 26, 2008 at 01:47:03AM -0400, Abel Monzon wrote:
Hello is my idea or this is a bug? The thing is that I have in my
Also check the file permissions and if you are using a RedHat like OS, check
the SELinux.
And about using a2billing,I recommend you to use version 1.4.21 or less.
On Sun, Oct 26, 2008 at 3:30 AM, Tzafrir Cohen [EMAIL PROTECTED]wrote:
Hi
First off, you replied a previous mail to the list, and
Hello is my idea or this is a bug? The thing is that I have in my
asterisk.conf this:
[directories]
astetcdir = /usr/local/etc/asterisk
astmoddir = /usr/local/lib/asterisk/modules
astvarlibdir = /usr/local/share/asterisk
astdatadir = /usr/local/share/asterisk
astagidir =
( ! ) Fatal error: Maximum execution time of 30 seconds exceeded in
/var/www/insects.digium.com/core/config_api.php on line 32
Call Stack
# Time MemoryFunctionLocation
1 0.0006 98216{main}( )../bugnote_add.php:0
2 0.242013136160
Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in
the voicemail messaging emailing operation. I had set serveremail option
to:
[EMAIL PROTECTED]
and under ast 1.2.x messages arrived at user mailboxes from
[EMAIL PROTECTED] . However, since upgrading emails
PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:41 AM
To: Asterisk Users List
Subject: [asterisk-users] Bug in voicemail's serveremail setting in 1.4.18
Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in
the voicemail messaging emailing operation. I
to tell sendmail to trust the asterisk account or
voicemail from address
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical
Support
Sent: March-19-08 9:57 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Bug in voicemail's serveremail setting in
1.4.18
List
Subject: Re: [asterisk-users] Bug in voicemail's serveremail setting in
1.4.18
For more info, I grab the relevant portion of the maillog. It looks like
asterisk is trying to send using the right from email, but it's getting
changed. This would suggest a sendmail problem, EXCEPT, it works
Hi guys, I'm not sure here is the best place to ask, but, anyone has
some news regarding to this bug? I'm having problems with this in one
customer.
Thanks
Carlos Barros
___
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I thought this would interest a few people on the list - asterisk
enabled home security video recording dvr anyone?
http://deancollinsblog.blogspot.com/2007/09/bug-labs-opensource-hardware
.html
I had a really interesting conference call today about a new startup
called
Subject: [asterisk-users] bug in 1.2.24
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten = 7141,5,MixMonitor
On 13:33, Fri 14 Sep 07, Isaac Xiao wrote:
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten =
Here is our dial plan. You need to avoid double recording as well when
you transfer the call to other extension.
exten = 7141,3,Set(CALLFILENAME=q${EXTEN}-${TIMESTAMP}-${UNIQUEID})
exten = 7141,4,Set(__FROM-EXT-QUEUES=ext-queues)
exten = 7141,5,MixMonitor(${CALLFILENAME}.gsm|b)
exten =
2007 06:24 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bug in 1.2.24
It is not a bug. attended Transfer is using Local channel, if you have a
look the debug log from CLI, you will see why it fails. To solve this
problem, enable recording before the calls go into the queue
GUys.. I dont know if this is a known bug or not but I just tested and
replicated this one over and over again.
It involves call transfer from calls that entered the pbx via a queue.. say
a call comes in and its thrown in a queue, somebody answers the call but
then wants to transfer the call to
It is not a bug. attended Transfer is using Local channel, if you have a
look the debug log from CLI, you will see why it fails. To solve this
problem, enable recording before the calls go into the queue.
Exten = ,1,MixMonitor(...)
Exten = ,2,Goto(ext-queue, , 1)
This will
I have this in my dialplan...
[general]
static=yes
writeprotect=no
clearglobalvars=no
[start]
exten = 5000,1,Answer
exten = 5000,n,Wait(1)
exten = 5000,n,NoOp(${CALLERID(num)})
exten = 5000,n,Playback(tt-monkeys)
which, when I dial 5000, executes this...
== Parsing
We recently installed Asterisk 1.4.2
Tried to make calls using the Originate command (Asterisk Manager Interface)
All of the calls have zero billsec in the CDR.
Stumbled upon this:
http://bugs.digium.com/view.php?id=8680
so I guess the fix is not yet in 1.4.2.
Is this fixed in 1.4.3/1.4.4?
Dear users,
I think I may found a bug in the voicemail module of Asterisk 1.4.2!
Outgoing email notifications should use a real existing domain (let's
call it domain.real) instead of the local domain (domain.local) so that
some mail servers won't reject the mails. That's why I've set the
Sven Jacobs wrote:
Dear users,
I think I may found a bug in the voicemail module of Asterisk 1.4.2!
Outgoing email notifications should use a real existing domain (let's
call it domain.real) instead of the local domain (domain.local) so
that some mail servers won't reject the mails.
Per Jessen wrote:
Outgoing email notifications should use a real existing domain (let's
call it domain.real) instead of the local domain (domain.local) so
that some mail servers won't reject the mails. That's why I've set the
serveremail option in voicemail.conf to [EMAIL PROTECTED]
Sven Jacobs wrote:
You fix that in your mail-server with aliasing and/or canonicalising.
I think the Asterisk behaviour is correct. It is similar to
receiving an email from cron or some other daemon. That is sent
from [EMAIL PROTECTED], which is fine for your internal purposes, but
if you
As far as I can tell (but I'm on 1.4.1), the serveremail option only
sets the From-address, not the envelope-address. The envelope will
probably always be asterisk-user@hostname
The From-address ist set by the fromstring option - which works btw - so
you are wrong :) Unfortunately setting the
Sven Jacobs wrote:
As far as I can tell (but I'm on 1.4.1), the serveremail option only
sets the From-address, not the envelope-address. The envelope will
probably always be asterisk-user@hostname
The From-address ist set by the fromstring option - which works btw -
so you are wrong :)
Maybe I'm misinterpreting things, but this is what I se:
fromstring = the From:-text, not the From:-address.
I'm just using the default fromstring, but I've set
serveremail = asterisk@realdomain
With this I get
From: Asterisk PBX [EMAIL PROTECTED]
Still, the envelope is [EMAIL
Sven Jacobs wrote:
Maybe I'm misinterpreting things, but this is what I se:
fromstring = the From:-text, not the From:-address.
I'm just using the default fromstring, but I've set
serveremail = asterisk@realdomain
With this I get
From: Asterisk PBX [EMAIL PROTECTED]
Still, the envelope is
Joshua Colp wrote:
The voicemail email gets handed off to sendmail for actual sending.
It's adding on the envelope above.
Yes, but asterisk is writing the From: header.
/Per Jessen, Zürich
--
ENIDAN Technologies GmbH - managed email security.
Starting at SFr1/month/user -
Hi All,
I have tried everything to get callerid to work reliably but to no avail.
I have configured zapata.conf as per documentation but still only get 50%
of callerid's through. As a test I called our system with my mobile a
number of times and only 50% get through. I do get warnings about
5 jul 2006 kl. 13.46 skrev Roger Schreiter:
Hi,
I did not yet study the newest chan_sip.c versions, but
it seems, that chan_sip treats mysql-peers different from
other peers, concerning the variable canreinvite.
If this variable is not explicitely set for a peer or user in
sip.conf, the
Hi,
I did not yet study the newest chan_sip.c versions, but
it seems, that chan_sip treats mysql-peers different from
other peers, concerning the variable canreinvite.
If this variable is not explicitely set for a peer or user in
sip.conf, the global value for canreinvite in sip.conf is
taken
Hi folks,
I used the ast2sql.pl script (found on www.voip-info.org) to put into
the database a simple sip.conf. Among other entries, you could find:
[general]
context=sip-in ;incoming sip calls
Well, the script put the comment into the database entry, and asterisk
started complaining about a
20 jun 2006 kl. 13.33 skrev Andrea Spadaccini:
Hi folks,
I used the ast2sql.pl script (found on www.voip-info.org) to put into
the database a simple sip.conf. Among other entries, you could find:
[general]
context=sip-in ;incoming sip calls
Well, the script put the comment into the database
Ciao Olle,
IMHO the comments should be stripped off by asterisk itself!!
It should be easy to modify the script, but the problem would
remain.
Should it be filed as an Asterisk bug?
A semicolon in realtime separates multiple values, it is *not* used
as a comment. So you should fix
Hello,
I have setup an Asterisk 1.2.7.1 system, with a working voicemail box:
/etc/asterisk/extensions.conf
exten = 83086921,1,Answer
exten = 83086921,2,Dial(SIP/stefan,5,r)
exten = 83086921,3,VoiceMail,u111
exten = 83086921,4,Hangup
exten =
Hi,
I'm still a newbie, but try to help you,
my voicemail works ok, I can also record messages ok.
My extension part is:
exten = s,1,Background(welcome-cisl)
exten = 1,1,Dial(Sip/vmoreno,10)
exten = 1,2,Voicemail(victor)
exten = 2,1,Dial(Sip/juliansip,10)
exten = 2,2,Voicemail(aajulian)
exten
Hello Victor,
Hi,
I'm still a newbie, but try to help you,
THX ;-))
And voicemail.conf part is:
[general]
format=wav49
maxmessage=180
minmessage=2
maxsilence=2
silencethreshold=150
maxlogins=3
[EMAIL PROTECTED]
skipms=3000
[victor]
victor = 1234, Victor Moreno, [EMAIL PROTECTED]
Before going to MANTIS, I want to ask if it's ok that the output of:
asterisk -rx show dialplan default
does not include in the output the dialplans available via the INCLUDE feature:
[default]
include = sipmanualoutbound; allow to make manual calls
include = sipoutbound ; _91X
- Erick Perez [EMAIL PROTECTED] wrote:
does not include in the output the dialplans available via the INCLUDE
feature:
[default]
include = sipmanualoutbound; allow to make manual calls
include = sipoutbound ; _91X calls to US48
include = sipcalluk ;
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
I just realised my problem seems to be related to bug 0003710 - 0003710:
[patch] Consultative transfers between asterisk servers. It's unclear from the
bug info if this problem has been resolved yet. Anyone know?
Doug.
-Original Message-
From: Douglas Garstang
Sent: Monday, March
Version - Asterisk SVN-trunk-r12793M (1.2.4)
I have 4 Grandstream GXP 2000 phones configured. However at the moment,
I have had to disable BLF, Hints, and Call Limiting due to an extremely
annoying bug which seems to make the phones channels lock in busy
after a call has been hungup.
If I do a
PROTECTED] Behalf Of Corporate
IT Solutions - Michael Dunne
Sent: Tuesday, March 14, 2006 8:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Bug Help or Suggestion - Grandstream
GXP2000(firmware 1.0.2.8) - BLF, Hints, call-limit
Version - Asterisk SVN-trunk-r12793M (1.2.4)
I
If you define a sip peer, wheather or not you put an entry in the field
OUTBOUND CID, if you
dial an external extension (let's say an extension on another asterisk
server, connected via IAX2 connection) the callerid
received by the foreign asterisk is device YOURNUMBER: i.e device 567
If you take
[Asterisk-Users] Bug in AMP
13/02/2006 17.14 1.10.010 in sip outbound callerid
On Mon, 2006-02-06 at 22:58 +, Conrad Wood wrote:
Unqiueid: asterisk-1713-1139266402.909
^
Please note the spelling of uniqueid. I find the spelling in
res_features.c - but only once I patched it with bristuff patches.
Does anyone know whether that is a known problem with
On Wed, 2006-02-08 at 08:35 +0100, stoffell wrote:
On 2/6/06, Conrad Wood [EMAIL PROTECTED] wrote:
Please note the spelling of uniqueid. I find the spelling in
res_features.c - but only once I patched it with bristuff patches.
Does anyone know whether that is a known problem with bristuff?
On 2/6/06, Conrad Wood [EMAIL PROTECTED] wrote:
Please note the spelling of uniqueid. I find the spelling in
res_features.c - but only once I patched it with bristuff patches.
Does anyone know whether that is a known problem with bristuff? If so is
it fixed in a later version?
What version of
Hi everyone,
I get these events sent like this:
Event: ParkedCall
Privilege: call,all
Exten: 701
Channel: Zap/4-1
From: IAX2/cnw-4
Timeout: 120
CallerID: X
CallerIDName: Conrad Wood
Unqiueid: asterisk-1713-1139266402.909
^
Please note the spelling of uniqueid. I find the
Its not so hard to look into the source code and make small changes.
Im not sure how hard it would be to implement what you want, but i
have tested it, and yes, you are right, the call get disconnected, and
i agree that souldnt be that way. You may want to open a feature
request in bugs.digium.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Moises Silva
Sent: 25 January 2006 16:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bug in attended transfer or as expected?
Its not so
Sorry for bumping my own thread but just hoping that someone out there
can help, I don't want to raise a bug on * if this is a strange issue
with my dial plan
Just to clarify this is attended transfer using asterisk and not a phone
feature (not joining two held calls etc)
Could someone with
The problem is when reception is busy she doesn't always wait for
someone to answer the call, however hanging up a ringing transfer on
attended also hangs up the caller.
If you have enabled Disconnect Call feature, then you can hangup
with *0 for example, that will hangup only the current
I can not get this to work either.
Here is an except from my extensions.conf
exten = 123,1,Answer
exten = 123,2,Authenticate(1|j)
exten = 123,3,SayDigits(3)
exten = 123,4,Hangup
exten = 123,102,SayDigits(102)
exten = 123,103,SayDigits(103)
exten = 123,104,SayDigits(104)
After dialing 123 and
The problem is when reception is busy she doesn't always wait for
someone to answer the call, however hanging up a ringing transfer on
attended also hangs up the caller.
Its the phone that is responsible for hanging up both calls, not Asterisk.
On the SNOM phones you can disable
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Moises Silva
Sent: 23 January 2006 15:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bug in attended transfer or as expected
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: 24 January 2006 05:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bug in attended transfer or as expected
Hi all,
I have had quite a few customer complaints about attended transfer
cutting off callers.
The problem is when reception is busy she doesn't always wait for
someone to answer the call, however hanging up a ringing transfer on
attended also hangs up the caller.
I have checked the scripts I
I'm Japanese. Sorry,English is not so understood,Please let me question by items.
In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's'j' option.
exten = 123,1,Answer()
exten = 123,2,Authenticate(789,j)exten = 123,3,Playback(pin-number-accepted)exten =
I just have upgraded from Asterisk 1.0.7 to 1.2.1 and im having problems with my AGI script that takes care about
routing the calls. It worked perfectly for the last year with 1.0.7,
now is getting stuck when the is launched. I have agi debug enabled and
this is the output:
-- Launched
Hi, i found 2 bugs in asterisk 1.2.0rc1.
I using debian stable.
I start asterisk with:
/usr/sbin/asterisk -U thomas
or an different user,
Asterisk is starting.
Autodialing are Ignored.
(/var/spool/asterisk/outgoing).
Asterisk ignore to dial a Number / Extension, automaticlly.
When i start
Guys.
I just discovered a bug in rc1, whenever We try to do an addqueuemember,
asterisk core dumps.
Here is the dialplan:
exten = 766,1,AddQueueMember(Ventas)
exten = 766,2,AddQueueMember(Soporte-Tecnico)
exten = 766,3,AddQueueMember(Soporte-Contrato)
exten = 766,4,UserEvent(Agentlogin|Agent:
Anton -
Thanks for the report. I've just posted a bug for you on the bug tracker at
http://bugs.digium.com/view.php?id=5705
Please refer to that URL for further information/resolution.
On 11/10/05, Anton Krall [EMAIL PROTECTED] wrote:
Guys.
I just discovered a bug in rc1, whenever We try
: [Asterisk-Users] Bug in 1.2rc1
|
| Anton -
|
| Thanks for the report. I've just posted a bug for you on the
|bug tracker at
|
| http://bugs.digium.com/view.php?id=5705
|
| Please refer to that URL for further information/resolution.
|
|On 11/10/05, Anton Krall [EMAIL PROTECTED] wrote:
| Guys.
| I just
List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Bug in 1.2rc1
|
| Anton -
|
| Thanks for the report. I've just posted a bug for you on the
|bug tracker at
|
| http://bugs.digium.com/view.php?id=5705
|
| Please refer to that URL for further information/resolution
I had the same problem when I upgraded and fixed it by using the
following syntax:
AddQueueMember({queue_name}|{channel})
So, before when I had:
AddQueueMember(Ventas),
Now, I need to have:
AddQueueMember(Ventas|SIP/1234).
Because I don't know of a function that will just give me the
Any way I can help BJ...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|BJ Weschke
|Sent: Thursday, November 10, 2005 12:22 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Bug in 1.2rc1
|
|On 11/10/05
|Subject: Re: [Asterisk-Users] Bug in 1.2rc1
|
|I had the same problem when I upgraded and fixed it by using
|the following syntax:
|
|AddQueueMember({queue_name}|{channel})
|
|So, before when I had:
|
|AddQueueMember(Ventas),
|
|Now, I need to have:
|
|AddQueueMember(Ventas|SIP/1234).
|
|Because I don't
Perfect timing, 1.2beta1 is released and the bug tracker is
broken!
Try to submit a bug and get error 1303, invalid field value.
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Asterisk-Users mailing list
Damon Estep wrote:
ast_expr2f.c:1784: warning: no previous prototype for âast_yyget_columnâ
ast_expr2f.c:1860: warning: no previous prototype for âast_yyset_columnâ
ast_expr2f.c:1259: warning: âyyunputâ defined but not used
These are not errors, that's why they are called 'warnings'.
Damon Estep wrote:
Try to submit a bug and get error 1303, invalid field value.
I just entered a test bug and it worked fine.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] bug tracker down?
Damon Estep wrote:
Try to submit a bug and get error 1303, invalid field value.
I just entered a test bug and it worked fine.
___
--Bandwidth and Colocation
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Saturday, August 27, 2005 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] bug tracker bug?
Damon Estep wrote
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] bug tracker bug?
Damon Estep wrote:
ast_expr2f.c:1784: warning: no previous prototype for âast_yyget_columnâ
ast_expr2f.c:1860: warning: no previous prototype for âast_yyset_columnâ
ast_expr2f.c:1259
-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Saturday, August 27, 2005 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] bug tracker bug?
Damon Estep wrote:
ast_expr2f.c:1784: warning: no previous prototype for
âast_yyget_columnâ
Cant submit bugs error 1303, invalid value for field
when submitting a new issue.
Bug info
Failure on build with 1.2beta1 on fresh FC4 install
ast_expr2f.c:1784: warning: no previous prototype for
âast_yyget_columnâ
ast_expr2f.c:1860: warning: no previous prototype for
Bug in Mailman version 2.1.5
We're sorry, we hit a bug!
If you would like to help us identify the problem,
please email a copy of this page to the webmaster for
this site with a description of what happened. Thanks!
Traceback:
Traceback (most recent call last):
File
Seems as though the dialpad in SJPhone cannot me used to signal *.
*2 doesn't do anything except play a DTMF in your ear. If you use your
keyboard to send shift-8, 2, all works as expected. Bug report submitted
already.
Cheers
Tim
___
Hello, I have found a possible bug in Asterisk. The reason I say this
is that it does not coredump when I call locally over SIP, only when it
goes over the net via IAX.
I'm running an AGI (in PHP) which does not apparantly complete. It
shows in the console that it runs, but never completes.
Darn, I forgot to say that I'm running:
Asterisk CVS-HEAD-04/21/05-16:53:20
FreeBSD 5.3
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Kevin Bockman wrote:
Hello, I have found a possible bug in Asterisk. The reason I say this
is that it does not coredump when I call locally over SIP, only when it
goes over the net via IAX.
I'm running an AGI (in PHP) which does not apparantly complete. It
shows in the console that it runs, but
From all what I can see here is that there might be a bug in the php
program.
Have you already checked the line 47 and 92 in the php program if you
have missed the semicolon at the end?
If that is not the case, can you dump the kernel to the list along with
your bank account info, so
Kevin Bockman wrote:
From all what I can see here is that there might be a bug in the php
program.
Have you already checked the line 47 and 92 in the php program if you
have missed the semicolon at the end?
If that is not the case, can you dump the kernel to the list along with
your bank
Hi all,
I have just released IPSwitchBoard version 0.70. There are no major changes,
but a few important bug fixes.
You can download IPS from the new website I have created for IPSwitchBoard:
http://ipswitchboard.thorben.dk
Regards
Thorben
___
While trying to
deploy a bunch of Polycom IP 500 phones, I ran in to the following. I
limited the RTP ports from 8000-8050 to limit holes in firewall. Pretty
soon Asterisk ran out of RTP ports. Traced the problem back to how * is
handling SUBSCRIBE. A sip structure is allocated as soon as
Sarat Vemuri wrote:
While trying to deploy a bunch of Polycom IP 500 phones, I ran in to the
following. I limited the RTP ports from 8000-8050 to limit holes in
firewall. Pretty soon Asterisk ran out of RTP ports. Traced the
problem back to how * is handling SUBSCRIBE. A sip structure is
Hi,
Using latest cvs.
A comment-line begins with semicolon ;
However - if the line contains
;--
or like this
; -- blabla bla --
You get this error and * stops reading that file:
Feb 21 13:47:12 WARNING[17393]: config.c:664 config_text_file_load:
Unterminated comment detected beginning
I finally figured out my extension D issue. The extension works fine as
long as Background() has finished playing. But during playback, the D
tone is not recognized. Is there any way to configure this? Is this a bug?
___
Asterisk-Users mailing list
-Original Message-
From: David Brodbeck [mailto:[EMAIL PROTECTED]
I finally figured out my extension D issue. The extension
works fine as
long as Background() has finished playing. But during
playback, the D
tone is not recognized. Is there any way to configure this?
Is
Steve Murphy wrote:
--On Tuesday, December 21, 2004 9:37 AM -0700 Steve Murphy
[EMAIL PROTECTED] wrote:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and
IAXPHONE on
a windows (XP) machine on my network, and I'm
Howdy--
I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on
a windows (XP) machine on my network, and I'm running asterisk on a
newly loaded Fedora Core 3 machine.
I set up a separate IAX account for each phone. I was EXPECTING them
to each register seperately with
--On Tuesday, December 21, 2004 9:37 AM -0700 Steve Murphy
[EMAIL PROTECTED] wrote:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on
a windows (XP) machine on my network, and I'm running asterisk on a
newly loaded Fedora Core 3 machine.
I set up a separate IAX
At 11:37 AM 12/21/04, you wrote:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on
a windows (XP) machine on my network, and I'm running asterisk on a
newly loaded Fedora Core 3 machine.
I set up a separate IAX account for each phone. I was EXPECTING them
to each
--On Tuesday, December 21, 2004 9:37 AM -0700 Steve Murphy
[EMAIL PROTECTED] wrote:
Howdy--
I'm playing with different IAX softphones. I've got DIAX and
IAXPHONE on
a windows (XP) machine on my network, and I'm running asterisk on a
newly loaded Fedora Core 3 machine.
I
For any of you that are running CVS (recent enough to have the wildcard
config file include change), I have posted another enhancement that
makes it even more useful :-)
Bug 3020 adds the ability for the same context to appear multiple times,
in multiple files, and have them add together. This
Hi all,
I wrote an AGI script in perli asked the
script to dial a number
$AGI-set_callerid($calleridnum);
if($left
0) {
$AGI-exec('Dial',"H323/$phonenumber");
}else{ $res =
mystreamfile("vm-goodbye");
$AGI-hangup(); }
The chan_h323 registers well to my gnugk. I call
Bit of transcript below, relevant portion is this though::
-- Executing ChanIsAvail(SIP/4111-6358, Agent/@1) in new stack
Oct 14 16:58:43 NOTICE[101393]: chan_agent.c:549 agent_hangup: Agent
'Michael Loftis' didn't answer/confirm within 20 seconds (waited 118)
Agent was logged in via
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