On 04/13/2011 09:18 PM, Rob Coward wrote:
Rather than add extra overhead to your dialplan and the asterisk
server, why not make use of the AMI and have a background process
listening for the various events and updating your database accordingly ?
See
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk itself has been
broken.
the last entry I have is:
/var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c:
Registration from 'sip:22942@10.0.0.3'
On 4/15/2011 3:39 AM, Jeremy Kister wrote:
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk itself has been
broken.
forgot to mention that I can replicate this behavior on 1.8.2.3 and 1.8.3.2
--
Jeremy Kister
On 110414 1346, satish patel wrote:
We are running asterisk 1.8.3.2 and if in future i need to upgrade with new
release then how should i upgrade it. The reason i am asking is i did lots of
customization in make menuselect so do i need to keep all option remember or is
there anyway i just
We are hiring a VoIP developer. Would it be within the list guidelines
to post a position description? Not sure if that's in the scope of this
list, but does not match asterisk-biz either.
-kkm
--
_
-- Bandwidth and
Registry type Event will give you information about your peer.
Adolphe Cher-aime
From my Iphone
On Apr 15, 2011, at 1:15 AM, Jonas Kellens jonas.kell...@telenet.be
wrote:
On 04/13/2011 09:18 PM, Rob Coward wrote:
Rather than add extra overhead to your dialplan and the asterisk
server,
On 04/15/2011 11:53 AM, Adolphe Cher-aime wrote:
Registry type Event will give you information about your peer.
Adolphe Cher-aime
From my Iphone
I don't find information on how this event tells me whether the SIP peer
is occupied with a call or not.
How can I capture the notify messages
Hi ,
How I will get my park call back?
*1 ) features.conf *
[general]
parkext = 900; What extension to dial to park
parkpos = 901-920 ; What extensions to park calls on.
context = bhati-park
*2) Extensions.conf *
[incoming]
include = bhati-park
exten = XXX,1,Answer
exten =
Hi, I'd like to replace my DECT + fxs phones with some wireless phones.
Main problem is: what about the roaming from one ap to another? Do
someone adopted the 802.11r standard? What APs and what phones do you
suggest me? I'm open to suggestions but I'd like to avoid proprietary
solutions, I don't
Hi all,
You're welcome as always to join the talk on the VoIP Users
Conference, VUC for short. VUC began as the Asterisk Users
Conference but for obvious reasons, we changed the name in the first
year, although Digium was our sponsor for three years. We still have
plenty of you who are asterisk
Hi
I want to join this conference but please tell me the topic of conference
and the process of joining step by step.
Please tell me the time in IST too
On Fri, Apr 15, 2011 at 5:09 PM, randulo rand...@randulo.com wrote:
Hi all,
You're welcome as always to join the talk on the VoIP Users
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause
code) commands,
On Fri, Apr 15, 2011 at 1:56 PM, virendra bhati virbh...@gmail.com wrote:
I want to join this conference but please tell me the topic of conference
and the process of joining step by step.
You're welcome to join us! All this information is at the top of the
main site: http://vuc.me
Please
Hey guys,
I have stdexten macro dialplan and I have to handle those who doesn't
have voicemail box setup. Right now if someone call and if person
unavailable the it's just hangup that call. I want it say person
doest have vm setup yet. smthing like that. How should I handle this
in my
Is this online conf? Or are there archived files we can review?
--
Sent from my iPhone
On Apr 15, 2011, at 8:06 AM, randulo rand...@randulo.com wrote:
On Fri, Apr 15, 2011 at 1:56 PM, virendra bhati virbh...@gmail.com
wrote:
I want to join this conference but please tell me the topic of
On Fri, Apr 15, 2011 at 2:16 PM, Satish Patel satish...@hotmail.com wrote:
Is this online conf? Or are there archived files we can review?
There are over 300 recordings here:
http://vuc.me
:r
--
_
-- Bandwidth and Colocation
Satish Patel wrote:
want it say person doest have vm setup yet. smthing like that. How
should I handle this in my dialplan ?
Here is a snippet of what I do:
[s-NOANSWER]
exten = s,1,Gosub(mailbox_exist,s,1)
[mailbox_exist]
exten = s,1,Set(_direct_vm=${ARG1})
exten =
Security only fixes: 2011-04-21 So in six days, no more bugfix patches will
committed into 1.4-branch :(
Is a prolongation possible? Because 1.4 is so reliable now. It would
be a great loss.
And no, 1.8 is not (yet) a replacement.
Kristijan
--
On 16/04/11 12:33 AM, Kristijan Vrban wrote:
Security only fixes: 2011-04-21 So in six days, no more bugfix patches will
committed into 1.4-branch :(
Is a prolongation possible? Because 1.4 is so reliable now. It would
be a great loss.
And no, 1.8 is not (yet) a replacement.
It actually
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 5:02 AM, Vlasis Hatzistavrou wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten =
You know we don't have choise. I had remembered when we shifted 1.2 to
first release of 1.4 and we had many issue. Same thing right now I'm
dealing with 1.8 things take time to stabilized.
Good luck!!
--
Sent from my iPhone
On Apr 15, 2011, at 8:33 AM, Kristijan Vrban
ViciDial doesn't work that way, you have to use the agent web interface or
the API to disposition a call.
MATT---
On Mon, Apr 11, 2011 at 10:04 AM, mahesh katta maheshka...@flexydial.comwrote:
Hi ,
In vicidial dialer
I need small Dialplan require. when i call from hardphone , in that has
Hello Jim,
Thank you for the reply.
The problem is not reading the ${HANGUPCAUSE} or the ${DIALSTATUS}. It
is that the Hangup(cause) command seems to ignore its argument and
just sends a 503 cause to the caller for all unanswered calls no matter
what...
Hangup(cause) was working as
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 6:46 AM, Vlasis Hatzistavrou wrote:
Hello Jim,
Thank you for the
On 15 April 2011 13:02, Vlasis Hatzistavrou vh...@kinetix.gr wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have
The h extension is executed after the remote end peer01 rejects the call
with a 408. I verified it by altering the dialplan as:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,NoOp(Hangup cause is: ${HANGUPCAUSE})
Hello Steve,
On 15/4/2011 5:07 μμ, Steve Davies wrote:
Strictly speaking you can only Hangup (BYE) an answered and fully
established call. In SIP terms, a hangup that occurs before an answer
is a CANCEL, and I believe a CANCEL is always represented by a 503
code in chan_sip.
Regards,
Steve
I
Perfect! Thanks
Date: Fri, 15 Apr 2011 08:21:48 -0400
From: supp...@drdos.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] If voice mail not found dialplan
Satish Patel wrote:
want it say person doest have vm setup yet. smthing like that. How
should I handle
Hey Guys!
We had echo issue before so we replaced old PRI card with Sangoma A102D HWEC.
Now my question is i set rx/txgain level 0.0 default do i need to touch this
value or default is best. I have read on google and people say it should around
14844 on ztmonitor for rx/tx level same.
I
Hi,
I am working on call packing feature of asterisk. Call packing is working
fine but I want to make this feature as multi tenant.
exp:-
*for A client*
packing extension are
parkext = 700
parkpos = 701-720
context = parkedcalls_A
parkingtime = 45
*for B client
*packing extension are
Dear, we have the following:
- Asterisk A with E1 to PSTN connection.
- Asterisk B with IAX trunk to Asterisk A
- Outgoing routes between Asterisk A and B
- Asterisk A with an outgoing route to PSTN with 9|. dial rule
How can I reach the PSTN from Asterisk B through Asterisk A ???
Thanks a lot
yes, ami is your unique answer.
what is msisdns ?
On Wed, Apr 13, 2011 at 3:18 PM, Albert alber...@wp.pl wrote:
Hi,
I am working on integration of 2 systems: asterisk and messaging platform.
What I need is to access somehow information about current calls. Should I
do it over AMI ?
I
On server B use
IAX2/iax-trunk-name/whatever-needs-to-be-dialed-on-A-to-make-call in the
Dial command.
like Dial(IAX2/sfserver1/9411212)
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 12:29 PM, Alejandro Cabrera Obed wrote:
Dear, we have the
1.4 svn has a nasty bug in it at the moment. Would love to see that fixed ;)
https://issues.asterisk.org/view.php?id=18951
Julian
On 15 April 2011 14:22, Satish Patel satish...@hotmail.com wrote:
You know we don't have choise. I had remembered when we shifted 1.2 to first
release of 1.4 and
On a test fax:
-- Executing [s@incoming-fax:1] Set(DAHDI/4-1,
FAXFILE=/var/spool/asterisk/fax/20110415_1825) in new stack
-- Executing [s@incoming-fax:2] Answer(DAHDI/4-1, ) in new stack
-- Executing [s@incoming-fax:3] ReceiveFAX(DAHDI/4-1,
On Fri, Apr 15, 2011 at 07:00:27PM -0400, sean darcy wrote:
...
Do the log notes of the CED tone or the T.30 ECM warning have anything
to do with this?
I can't help with this.
faxbuffers=6,full ; who knows what this does
But I do know that this faxbuffers policy basically is telling the
On Fri, Apr 15, 2011 at 7:00 PM, sean darcy seandar...@gmail.com wrote:
Using spandsp-0.0.6-pre18, the Jan 22 release.
You might try using spandsp-0.0.6-pre17. That version works great for
me with 1.8.4-rc2. When I tried pre18 it failed to receive any faxes.
Ryan
--
On Apr 15, 2011, at 12:50 PM, satish patel wrote:
We had echo issue before so we replaced old PRI card with Sangoma
A102D HWEC. Now my question is i set rx/txgain level 0.0 default do
i need to touch this value or default is best. I have read on google
and people say it should around 14844
Hi list,
I use AMI to originate calls for outbound campaigns via local
channel. I use CDR records to have status of the calls such as :
disposition, if call was answered by live human or answering machine, reason
of call failure and make accounting based on total calls that have
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