Hi All,
I seem to have stumbled on a bit of a problem. When trying to send a fax
with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the
current svn version, with FFA 1.2 I get a core dump each time.
Here is an extract form the console:
[May 12 22:47:09] DEBUG[22584]:
Kevin P. Fleming wrote:
Like I said, it's a known problem, and the fix should be out within a
day or two. It was reported to us about a week ago, so if you had
contacted the support department, it's likely they would have been able
to shortcut your hair-pulling experience :-)
Hi Kevin,
Well, I have managed to get my hands on a copy of 1.2.1 rc1 FFA which
seems to have fixed the core dumping issue but does not appear to have
fixed the issue that was causing the core dump.
We are still getting an issue with a particular file which I have tried
multiple different ways to create
Sean Brady wrote:
Have you tried removing the /n option from the local channel? Just a
thought, but it's probably worth a try. You could also try calculating the
billsec in the dialplan and write it to the CDR with the adaptive CDR feature
in 1.6.2.
Not sure if this is helpful but it
Hi All,
I am in the process of migrating from 1.4.20 to 1.6.2.x and have
stumbled across a number of differences between the 2 versions that
are forcing me to use local channels in my dialplan (mainly centered
around the different behavior of CDR fields in the 2 versions) .
Previously, I
Leo Burd wrote:
Hello there,
I'm trying to figure out how to run a PHP script on a remote machine and
still have access to the audio stream associated with the call.
Ideally, I'd love to play/record audio files directly from/to the remote
server without having to copy them back and
Per Jessen wrote:
I have the following dialplan:
; calls prefix by '8' are recorded
exten = _8[01]./_251,1,Set(something=shortened)
exten = _8[01]./_251,n,Set(WAV=filename)
exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
exten =
Why dont you use the MixMonitor application which allows for a system
command to be passed in as an argument that is executed once the
recording is finished??? -
MixMonitor(file.ext[|options[|command]])
command will be executed when the recording is over. Any strings
matching ^{X} will be
I get a somewhat minimal set of standard shell environment variables
(BASH*, HOSTNAME, PWD, TERM, etc) including the same PATH environment
variable I passed to Asterisk when it was started.
That just means that you cant rely on environment variables in the
script that you execute and you
This is usually due to an error with the SIP stack not being loaded due
to an error - make sure that full logging is on and check your log file
and search for ERROR and see if there is any mention to SIP (chan_sip.o
etc), alternatively, start asterisk from the command like with asterisk
Hi All,
I have been running a environment with asterisk 1.4.20.1 for some time
now with no issue but have recently added some extra functionality
(enabled call recording via MixMonitor) and ran into some deadlock
issues which seem to be well documented with earlier 1.4.x releases so
have
Hi All,
I hate to post yet another bloody hangup detection issue on the list, but
I have been pulling my hair out no end of late with a hangup detection issue
on 1 system (have a few others out there with TDM400's and no issue but this
one is causing a real headache)
The scenario is - system
Kevin P. Fleming wrote:
Ben Dinnerville wrote:
The problem occurs when we have external (pstn) calls coming into / out
of the system (via an iax trunk), in which case we have no control over
frame size, as well as occurring with handsets directly connected to the
system.
Please
Hi All,
I have recently installed a TC400B card into a system and am trying to
get it to work. As far as I ca tell from the docco on Digiums website,
there is no config as such unless you want to enable / disable only 1
codec, otherwise by default it runs as 92 channels of either.
I have
Andres wrote:
Try to compare the frame size you are receiving from asterisk and set
your phone to transmit the same frame size. I would guess the card
appears to have problems when the frame size is different. Please try
and report back. I am curious about this.
The problem occurs
Hi All,
I am trying to tune out some echo on a analogue line and have run
ztmonitor to get some info. When i run it, i get a RX reading when the
line is idle - is this normal? eg:
[EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv
Visual Audio Levels.
Use zapata.conf
Nudge?
Ben Dinnerville wrote:
Hi All,
Is it possible to configure asterisk to play a beep at a regular
interval when a conversation is being recorded / monitored?
There are a number of ways indicating to a user that a conversation is
being recorded, one is to play an announcement, another
Hi All,
Is it possible to configure asterisk to play a beep at a regular
interval when a conversation is being recorded / monitored?
There are a number of ways indicating to a user that a conversation is
being recorded, one is to play an announcement, another accepted way is
to play these
Has anyone seen these solid state Drives from gigabyte yet? -
http://www.pcper.com/article.php?aid=224type=expertpid=3
For those who havnt, they are basically a pci card with 4 DDR memory
slots on board, coupled to a SATA interface and with a battery on board
to ensure that you can use
with 1.2.4 on it and the A party was also running 1.2.4 and calls will
now go through without a problem so all is now sorted.
Cheers,
Ben
Ben Dinnerville wrote:
Hi All,
We are experiencing a a problem when running calls over IAX with g.729.
The call flow is as follows:
Sip handset -(SIP
I am running 2 TDM400's in a single machine without a drama. It is a
scenario that will work, but you have to be careful with your PCI slots
and IRQ assignments. I basically disabled everything on board that was
not needed (USB, floppy, IDE2 etc) and had to play around with which
slot the
Hi All,
Is there some way to verify that a channel is using iax trunking?
The reason i ask is that i have a scenario where 1 Asterisk system is
communicating with another over IAX. System A is using static
configuration from the standard files, System B is using realtime with
MySQL config
Hi All,
We are experiencing a a problem when running calls over IAX with g.729.
The call flow is as follows:
Sip handset -(SIP) Asterisk1 -(IAX) Asterisk2 -(SIP) Carrier
The first Asterisk system is running 1.2 and the second is running 1.0.
When using g726 from the handset all the way thru
The carrier does not support speex, only g729, 723 and 711, so to
minimise codec coversions etc, and due to the fact that licensing 723
is so expensive and 711 is a bit fat on bandwidth (asterisk 1 is
connecintg over 128k ISDN) we are kind of stuck with g729 (not that it
has ever proved to be
Nudge
Ben Dinnerville wrote:
Hi All,
I am looking at pacthing up Asterisk to test T.38 passthrough and see
that there is a bug with some pacth / diff / new files that need to be
applied and compiled with Asterisk.
Can anyone provide a quick how-to on applying this patch?
Do all the files
Hi All,
I am looking at pacthing up Asterisk to test T.38 passthrough and see
that there is a bug with some pacth / diff / new files that need to be
applied and compiled with Asterisk.
Can anyone provide a quick how-to on applying this patch?
Do all the files that are applied to the bug
at www.voicelogic.com.au - check back shortly for more details.
Regards,
Ben Dinnerville
Voice Logic Pty Ltd
Dean Collins wrote:
*From:* Dean Collins
*Sent:* Wednesday, November 30, 2005 10:44 AM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* IAX Service providers in Australia
HI All,
Is there some way of telling h323 / oh323 to use an externip in the
rtp streams and 323 data similar to how you can configure sip on
asterisk?
We have a system that we need to be able to send outbound calls via
sip (working fine) and receive calls via h323. The machine is on a
private
HI All,
Have searched a bit for this one, but cant seem to find any indication
of how you might use a dial prefix with the hash / pound sign as part
fo the prefix.
I have a gsm pod set up on asterisk which all works fine, but i want
to disable cli on the outbound calls over this particular
Hi All,
Just wondering if anyone has managed to get line presence working on the
7 indicator lights on a grandstream gxp-2000 with asterisk? If so, what
is the trick? :)
I have simple presence working with my polycom phones but cant seem to
get it working with the gxp-2000 - is it available
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