Thanks for the replies. I am already ignoring the signal but it doesn't
seem to be making much difference on this system with this script. It's
an old legacy script I should hopefully be dropping and writing properly
within the dial plan.
I will keep trying!
Thanks
Lee
-Original
Hi All,
Is there a way to use the dynamic feature of the meetme application (D)
and to set an option to configure the minimum length of the numbers for
the conference and the associated pin.
In my case, I'd like them to be at least four digits.
Thanks in advance !
--
Xavier wrote:
Hi All,
Is there a way to use the dynamic feature of the meetme application
(D) and to set an option to configure the minimum length of the
numbers for the conference and the associated pin.
You can use the read application to get the password and then check the
length,
Yes but what about the conference number ?
On 08/27/2010 11:58 AM, Doug Lytle wrote:
Xavier wrote:
Hi All,
Is there a way to use the dynamic feature of the meetme application
(D) and to set an option to configure the minimum length of the
numbers for the conference and the associated pin.
Hello,
Is it possible to avoid playing music on hold during a blind transfer ?
Thanks
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Hello,
When an agent does a blind transfer the call hangups for him but shows as
In use in queue in my CRM (used for auto dialing). As a result the agent
have to wait until the transfered call completes. Is there any way to change
this behaviour ?
--
Hi,
I'm currently programming an interface for my Asterisk service.
I've noticed an issue if someone sets up call forwarding on their sip phone.
Asterisk receives a 302 Moved Temporarily message, and forwards the call
successfully.
However, the CDR is not correct.
If I set up call forwarding
On Fri, Aug 27, 2010 at 7:39 AM, Tino t...@sparksupport.com wrote:
Is it possible to avoid playing music on hold during a blind transfer ?
Please do not cross-post the same message to multiple lists.
Yes, configure an empty MoH class or not loading MoH are some options, also:
*CLI core show
Hi all,
with an (attended) transfer i see the following happening:
1) A calls B1
2) B2 calls C
3) B2 transfers call to A
4) A talks to C
At step 3, the channel A is connected to channel C and B1 and B2 are hung up.
In the h extension for channel B2, the channel is renamed to B2ZOMBIE and i
Dan Journo schrieb:
Since it isn't behaving like I want, is there any way to disable the
feature that allows a SIP phone to perform call forwarding?
Thanks
Dan
Hello,
in asterisk 1.6.x there is a Dial option i which suppress a 302 redirect
which is very nice when dialing
Did you find the solution?
On Thu, Aug 26, 2010 at 7:25 PM, Carlos Chavez cur...@telecomabmex.comwrote:
I have searched for some time but I have not found an asnwer on how
to
fix the CDR when a call is transferred. The problem is that if someone
dials a cell phone and then transfers
/asterisk-users/attachments/20100827/6c7202ae/attachment-0001.htm
--
Message: 14
Date: Fri, 27 Aug 2010 05:58:57 -0400
From: Doug Lytle supp...@drdos.info
Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
To: Asterisk Users Mailing List - Non-Commercial Discussion
/asterisk-users/attachments/20100827/6c7202ae/attachment-0001.htm
--
Message: 14
Date: Fri, 27 Aug 2010 05:58:57 -0400
From: Doug Lytle supp...@drdos.info
Subject: Re: [asterisk-users] dynamic MeetMe, min. digits
To: Asterisk Users Mailing List - Non-Commercial Discussion
in asterisk 1.6.x there is a Dial option
Sorry, any solutions for Asterisk 1.4?
Thanks
Dan
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Hi all
I have posted a question on the asterisk dev board about this issue but I
want to see if any users have run up against this.
This issue is that when calls are run through Broadvox and Level 3 the
in-call rfc2833 dtmf is not reliable. This occured for me on asterisk
version 1.6.1.18,
Hey all
We are seeing intrusion attempts coming from address 201.47.236.122 today
They were hitting our switches trying to get in. So we blocked them at our
firewall.
Just wanted to put the word out so you all can protect your self.
Bryant
--
How can we set the CDR Master file to rollover at say 30 Meg and create a new
one
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I am having some difficulties getting my Asterisk box to find the d-channel
from a TELUS PRI and am waiting to hear back from one of their techs. In the
meantime I thought I would check with the brilliant people of the mailing list.
As I understand it is a T1 connection, not an E1 and I am
- Ujjval Karihaloo ujj...@simplesignal.com wrote:
How can we set the CDR Master file to rollover at say 30 Meg and create a new
one
Use 'logrotate'.
--Tim --
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Subject: Re: [asterisk-users] ASterisk CDR file Master.csv
- Ujjval Karihaloo ujj...@simplesignal.com wrote:
How can we set the CDR Master file to rollover at say 30
On 8/27/2010 11:55 AM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tim Nelson
*Subject:* Re: [asterisk-users] ASterisk CDR file Master.csv
- Ujjval Karihaloo ujj...@simplesignal.com wrote:
How can
On Fri, Aug 27, 2010 at 12:46 PM, jeremy.hellst...@synovate.com wrote:
moving the dchannel around, 12 through 24. Does anyone see anything
blatantly wrong?
What alarms are you getting?
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger
I see ...
Chan_dahdi.c 2796 pri_find_dchan No D-channel available using Primary channel
X as D-channel anyway.
With X being whichever number I assigned to the D-channel in chan_dahdi and
system.conf.
Then when dialling I get an error 0 - unknown, which occurs when Asterisk tries
to open a
Thx Dean. I will be interested in testing that as well.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Hoover
Sent: Friday, August 27, 2010 11:03 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi all,
I'm working on a system with 4 Grandstream GP-200 Phones and the base
Asterisk install.
I have added a 5 phone which is remote to the client and located in my
office.
I can't get the phone to transfer a call or put a call on hold. This
applies to all the phones at the location.
Asterisk crashes from time to time and dumps core. So... what do I do with
it?
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What do the logs in /var/log/asterisk/* tell you?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Jayson Baker jay...@spectrasurf.com wrote:
Asterisk crashes from time to time and dumps core. So... what do I do with
it?
--
On Fri, 27 Aug 2010, Jayson Baker wrote:
Asterisk crashes from time to time and dumps core. So... what do I do
with it?
Depending on the version, start reading
asterisk-source-directory/doc/README.backtrace or
asterisk-source-directory/doc/backtrace.txt.
--
Thanks in advance,
Carlos Chavez cur...@telecomabmex.com writes:
I have searched for some time but I have not found an asnwer on how to
fix the CDR when a call is transferred. The problem is that if someone
dials a cell phone and then transfers the call to another extensión the
CDR for the cell call
to get accurate cdr's i just use a border server to send every call
through that logs cdr... doesn't matter how many times it gets transferred
internally the border server still gets accurate records of the whole
call.
On 27 August 2010 21:07, Benny Amorsen
Please don't top-post.
Geraint Lee gera...@gmail.com writes:
to get accurate cdr's i just use a border server to send every call
through that logs cdr... doesn't matter how many times it gets transferred
internally the border server still gets accurate records of the whole
call.
That is
There is nothing in /var/log/asterisk... hmm, which log should I turn on?
Debug?
On Fri, Aug 27, 2010 at 1:25 PM, Tim Nelson tnel...@rockbochs.com wrote:
What do the logs in /var/log/asterisk/* tell you?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Jayson
much static testing of my realtime configuration and applications I'm
almost ready to pull the trigger.
The one thing I've been able to determine is what I need to do to
migrate my g729 licenses.
Has anyone got any advice for me on this? The Digium site is...
difficult to navigate
TIA
Bruce
I installed AsteriskNow1.7 and am trying to load the res_snmp module to
monitor the system. Am I correct in saying that compiling asterisk from
source and including the module is the only way to accomplish this? I’m a
little worried about simply downloading the same source version as my
current
From: Bruce Ferrell bferr...@baywinds.org much static testing of my
realtime configuration and applications I'm
almost ready to pull the trigger.
The one thing I've been able to determine is what I need to do to
migrate my g729 licenses.
Has anyone got any advice for me on this? The Digium
My IAX2 trunk provider, Teliax, seems to be forcing early media. Early
media is cool and all, but my Asterisk install doesn't seem to be
fully supporting it. My initial setting was using Dial() to call all
of my dahdi (TDM400P) extensions. The results were that incoming calls
would not hear any
Thought a different succinct subject line must drum up an answer or two...
Also, this has been tested from two different carriers: We're getting
an average of 2/10 call success rate.
-- Forwarded message --
From: Joe Wood sch...@gmail.com
Date: Thu, Aug 26, 2010 at 6:58 PM
I'm sorry, I tried this but the SVN version does not seem to work on
my machine. I get no DAHDI support, I can't even select it in
menuselect so I've no idea what to do.
Ira
At 11:28 AM 8/23/2010, you wrote:
On Monday 23 August 2010 12:19:38 Ira wrote:
At 09:26 AM 8/23/2010, you wrote:
I don't see why it does not work. Setting RDNIS and calling most GSM
mobile phones produces a forwarded call annoucement, so why would
the do it any different? We get RDNIS in a SIP field and use it to
keep the same voicemail for a desk phone and cell phone, also can
forward ILEC and most CLEC
Hi!
My question is this. Is it possible to tell Asterisk to execute part
of a macro as a block without allowing any other commands to be
processed during that time?
What would be a correct way to do this in 1.4.x?
*CLI show application MacroExclusive
Philipp
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