Matt Riddell li...@venturevoip.com wrote:
On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote:
Hi. I have a soft phone -- expresstalk-- on a computer in my network
and I use the internal ip address of the asterisk box to register the
phone. But using asterisk-1.8 between revisions 281912
Hi Everyone,
I have two servers as the following that are trunked with each other via
IAX2 trunk:
Server A:
Asterisk 1.4.21.2 (Elastix Flavor)
Server B (IP # 72.72.72.72):
Asterisk 1.6.2.0 (Vanilla)
Server B can place calls to Server A but when trying to place calls from
Server A to Server B
On Thu, Sep 02, 2010 at 02:21:11AM -0400, bruce bruce wrote:
Hi Everyone,
I have two servers as the following that are trunked with each other via
IAX2 trunk:
Server A:
Asterisk 1.4.21.2 (Elastix Flavor)
Any chance you could upgrade that? Elastix has newer versions of
Asterisk, for
I'd rather find the problem than upgrade blindly. Upgrading not always
solves the problem and has the potential to break other things. Thanks for
the offer though.
Bug # 16753 applies.
call token not required was set in the trunk and problem solved.
There is not warning for this in iax2 debug
Maybe dvossel can re-open issue # 16753 and fix the warning to show on iax2
debug as well along with core set debug like all other warnings. That way
it's straight forward. That ticket shouldn't have been closed without a fix.
On Thu, Sep 2, 2010 at 4:11 AM, bruce bruce bruceb...@gmail.com wrote:
Hi all,
I am using asterisk-1.6.2.10. I changed say.conf script for customized
number reading.
In the extension.conf:
--
[number-to-voice]
exten = 8765,1,playback(num:344345,say)
exten = 8765,n,hangup
It executes corresponding say.conf script and produces good results
Hi max,
Please read India-CID.txt file in asterisk documentation directory.
Thanks,
Ashik
On Wed, Aug 11, 2010 at 10:35 AM, Max Alex max.aster...@gmail.com wrote:
Hi,
Thanks for this information, but it is not working for both the issues,
I have tried with the configuration with
Hi,
1) I want to create add *1 call recording and wanted to know whether the
file is created during recording or only after? I want to syncronise the
recorded files with my web server (on a different machine (Windows)) so I need
a way of telling when the recorded call has ended before
1) I want to create add *1 call recording and wanted to know whether the file
is created during recording or only after? I want to syncronise the
recorded files with my web server (on a different machine (Windows)) so I
need a way of telling when the recorded call has ended before copying
1) The file is written in real time. Personally I would add a dialplan
entry into the 'h' extension to move the recording into a different
directory when the call ends. That will make your syncronisation much
easier.
Dan Journo wrote:
Hi,
1) I want to create add *1 call recording
The DTMF mode can cause problems. The main rule is to make sure
everything is using the same method. I normally use SIP-Info as the
method as it allows to rtp stream to be switch directly between the two
end points but asterisk still sees all the dtmf digits.
Dan Journo wrote:
1) I want to
1) I use a bash script I wrote to check if call recordings are being
written to and if not then move them. I move them to a locally mounted
NFS share but this will work with any type of locally mounted share
(Samba for Windows). I run the script every minute with cron. It also
sorts the
On Thu, 2010-09-02 at 07:21 -0400, Dan Journo wrote:
1) I want to create add *1 call recording and wanted to know whether the
file is created during recording or only after? I want to syncronise the
recorded files with my web server (on a different machine (Windows)) so I
need a way of
How do you sort out the issue of having 2 wav files per call?
Also, when I press *1, asterisk thinks that both the caller and the callee have
pressed *1 and therefore it starts recording twice (therefore making 4 wav
files). Any idea what's going on there?
Heres the CLI output:-
-- Called
Hi all
Been looking to find a way to stop the dtmf keys * 0 and # managing call
flow in the dialplan - I just want VM to stop recording on silence or
hangup.
I know I can trap the exit and loop back around but just want to ignore the
keys totally.
Any suggestions
P
--
oops - forgot to say this is voicemail() on Version 1.4.33.1
P
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice
Sent: 02 September 2010 13:32
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voicemail
On Thu, 2010-09-02 at 08:20 -0400, Dan Journo wrote:
How do you sort out the issue of having 2 wav files per call?
Also, when I press *1, asterisk thinks that both the caller and the callee
have pressed *1 and therefore it starts recording twice (therefore making 4
wav files). Any idea
On Thu, Sep 2, 2010 at 8:31 AM, Paddy Grice pa...@wizaner.com wrote:
Any suggestions
You have to modify the source code.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
1.4 will talk to 1.6 but results haven't been as good in my experience as
native 1.4-1.4 or 1.6-1.6 communication.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asteriskguru
asteriskguru
Subject: [asterisk-users] agi playback to execute say.conf settings
Hi all,
I am using asterisk-1.6.2.10. I changed say.conf script for customized
number
On 09/02/2010 01:09 PM, Ishfaq Malik wrote:
On Thu, 2010-09-02 at 07:21 -0400, Dan Journo wrote:
1) I want to create add *1 call recording and wanted to know whether the
file is created during recording or only after? I want to syncronise the
recorded files with my web server (on a
I have our recordings written to a solid state drive rather than straight to
storage disks then moved to long term storage to avoid this problem.
Not an option for me at the moment.
I'm running Asterisk on a cloud to reduce startup costs.
Once I reach around 1,000 extensions, I'll move over
On Thursday 02 September 2010 01:13:35 cov...@ccs.covici.com wrote:
Matt Riddell li...@venturevoip.com wrote:
On 25/08/10 8:54 PM, cov...@ccs.covici.com wrote:
Hi. I have a soft phone -- expresstalk-- on a computer in my network
and I use the internal ip address of the asterisk box to
Hi, I have the same problem too, but i can´t provide more information,
because i can´t find more information, just in the log show me that´s
it´s starting a MOH normally. It´s happen on random way, without nothing
similar on each call.
Using Elastix 1.6, x64 with ATA LinkSys PAP2.
Em
I'd *really* like to be able to have a call ring three different cell
phones; then, if someone answers, they have to somehow acknowledge the
call for it to be directed to them. That way, if one of the phones is
off, or out of range, it doesn't go straight to that phone's voicemail.
Asterisk 1.4
Ken D'Ambrosio wrote:
I'd *really* like to be able to have a call ring three different cell
phones; then, if someone answers, they have to somehow acknowledge the
call for it to be directed to them. That way, if one of the phones is
off, or out of range, it doesn't go straight to that phone's
On Thursday 02 Sep 2010, Ken D'Ambrosio wrote:
I'd *really* like to be able to have a call ring three different cell
phones; then, if someone answers, they have to somehow acknowledge the
call for it to be directed to them. That way, if one of the phones is
off, or out of range, it doesn't go
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Subject: [asterisk-users] Google Voice-like feature.
I'd *really* like to be able to have a call ring three different cell
phones; then, if someone
Ken D'Ambrosio ken at jots.org writes:
I'd *really* like to be able to have a call ring three different cell
phones; then, if someone answers, they have to somehow acknowledge the
call for it to be directed to them. That way, if one of the phones is
off, or out of range, it doesn't go
PROBLEM: result of dahdi_cfg:
DAHDI Tools Version - 2.2.1
How nice. Why don't you use the latest? 2.3.0[.1]? (2.4.0 should be
released shortly, but that's not really something users are expected to
guess).
-- I decided to use the same versions with the tutorial. I did the other
way but I
Now it is done...
asterisk*CLI dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState
pseudodefault
default In Service
1from-pstn en
default In Service
2
Want to thank everyone who mailed; a couple of your ideas got me going
down certain paths, and found the answer here:
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
Again, thanks!
-Ken
original message -
I'd *really* like to be able to
One more thing.
Can anybody point me to a sample configuration for
2 PSTN lines and 2 internal phones. (May be plus a SIP server)
for Asterisk 1.6.x
On 1 September 2010 15:41, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Wed, Sep 01, 2010 at 02:58:59PM +0300, Mehmet Kuzulugil wrote:
I am not interested in open source solutions. I want to know how much the
propriety systems cost in terms of licensing. Specially Avaya now a days per
extension. Exclusive or Inclusive of the hardware for 10 agents as noted.
Thanks
On Thu, Sep 2, 2010 at 8:18 AM, Muhammad Shomail Haider
El 02/09/10 11:32, bruce bruce escribió:
I am not interested in open source solutions.
Then what are your doing here?
I want to know how much the propriety systems cost in terms of
licensing. Specially Avaya now a days per extension. Exclusive or
Inclusive of the hardware for 10 agents as
He is looking for competitive information...what are prospects paying for
Avaya when they could be saving lots of money with Asterisk systems.
Probably a better question for the biz list, but he doesn't deserve the
responses he's getting.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
Hello community,
I need to finish an AGI script when it invokes a macro from dialplan, how
can i do that? it's quite confusing...the macro is making a hangup but the
script continues
Thanks
--
Salu2
--
_
-- Bandwidth and
On Thu, 2 Sep 2010, Danny Dias wrote:
I need to finish an AGI script when it invokes a macro from dialplan,
how can i do that? it's quite confusing...the macro is making a hangup
but the script continues
I don't understand your question, but I'm guessing it has something to do
with:
Hi everybody,
sometimes we have an Asterisk-crash, but no clue why this is happening,
so I'm trying to make a coredump to analyse it.
I compiled Asterisk 1.4.20.1 on CentOS 5.4 i386 with DEBUG_THREADS and
DONT_OPTIMIZE, then I start it with:
# /bin/bash /usr/sbin/safe_asterisk
This should do
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorolf Godawa
Subject: [asterisk-users] How to create a coredump for Asterisk
snip
Just my opinion, but Asterisk probably isn't going to dump when you kill the
process; something internal
Don-
He is looking for competitive information...what are prospects paying for
Avaya when they could be saving lots of money with Asterisk systems.
Probably a better question for the biz list, but he doesn't deserve the
responses he's getting.
Agree. If it weren't for extreme high cost of
He doesn't deserve the responses, but it seems that boundaries are being
pushed in both sides of the response. If he thinks he's on the biz list,
that's one thing, but in the purely open discussion, don't be dissing open
source either.
--
Hi,
I have a problem that the machine running asterisk 1.6.2.11 freezes
unexpectly time to time. Sometimes it runs for 4 weeks without any
problem, sometimes after a free it freezes again in 24 hours. But
usually it runs normally for 1 month or so before it freezes again.
I could not find any
It could be that I'm entirely confused, but I think he asked what people are
paying for Avaya solutions--so he'd know what competitive pricing would be
for the open source solution he's prepared to offer.
When someone replied with open-source suggestions, he pointed out that that
was not the
There´s a way to get the channel signalling in dialplan?
I have changed the code in channels/chan_dahdi.c and includes:
} else if (!strcasecmp(data, signalling)) {
ast_mutex_lock(p-lock);
snprintf(buf, len, %s,
Thanks Don for clarification.
There are lots of people on this list that hastily decide to answer without
even reading a post properly. I am sure they won't even read the follow-ups.
They just talk for the sake of talking. Sickens me!
Please note the subject line in my original post: To compete
Hello Steven...
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
from my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified
for every peer with setvar it should Hangup the call (frobidden,1 in
What should i do to finish the macro if this macro reachs the Hangup?
I tried to say: What should i do to finish the *AGI* if this macro reachs
the Hangup?
2010/9/2 Danny Dias ing.diasda...@gmail.com
Hello Steven...
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Subject: Re: [asterisk-users] How to finish an AGI
snip
This isn't really a task for AGI since it is by nature single-call specific.
As I interpret what I read, you are
No nicolas...that's not what i want...by the way sound very complicated :(
What i need is to FINISH THE AGI when the MACRO reachs the Hangup...dont
worry for the purpose of the macro, if the macro reachs the hangup the Agi
should stop working, but it continues with his job... :(
2010/9/2 Danny
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Subject: Re: [asterisk-users] How to finish an AGI
No nicolas...that's not what i want...by the way sound very complicated :(
What i need is to FINISH THE AGI when the
YES YES...that's what i want ;)
so simple but i was so tired :(
I will try it and let you know ;)
THANKS my friend
2010/9/2 Danny Nicholas da...@debsinc.com
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Dias
On Thu, 2 Sep 2010, Danny Dias wrote:
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro from
my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified for
every peer with setvar it should Hangup
How does asterisk process URI's that get sent to it?
I am having a issue with a Cisco phone, where 99% works except the call
forwarding. The phone issues a X-cisco-serviceuri-cfwdall which can be seen
when running a sip debug on the peer directly. However the system tries to
lookup the request as
Unfortunately, if I kill all asterisk-processes with kill -9 ..., a
coredump never is writen to /tmp, I also looked in other dirs.
Try kill -6 (i.e. SIGABRT). That usually triggers a core dump for me.
Luki
--
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-- Bandwidth
We have a server that has been in operation since December of last year.
Two days ago we started seeing this messages over and over (maybe a couple
thousand in a minute):
[Sep 2 17:46:19] DEBUG[7422] audiohook.c: Write factory 0x2aaad40a0038 was
pretty quick last time, waiting for them.
Has anyone successfully made this scenario work in 1.4. I found info at
http://www.voip-info.org/wiki/view/Asterisk+presence indicating that this
does not work with 1.4 implementations.
--
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-- Bandwidth and Colocation Provided
In asterisk.conf, use these options:-
cache_record_files = yes ; Cache recorded sound files to another directory
during recording
record_cache_dir = /tmp ; Specify cache directory (used in cnjunction with
cache_record_files)
--
Regards,
Prince Singh
Drishti-Soft Solutions Pvt Ltd
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