Hello Alex,
I'm struggling with the same problem and, not wanting to modify the CDR
backend, I just put in a workaround in the form of a MySQL trigger.
I'll describe what I did in case it helps someone, though I'm very
inexperienced at making compound procedures in MySQL.
In my
Hello,
I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to
providers Cisco 2800 with VWIC-1MFT-E1 card.
the same card runs fine with another E1 provider.
TE420 led's lite green.
Message type: RELEASE COMPLETE (90)
[08 02 80 ac]
Cause (len= 4) [ Ext: 1 Coding: CCITT
Hi,
1) It would be nice to find out the root reason that forces you to
restart the Asterisk. I do run Aheeva with decently high uptimes.
2) Both a and b methods of Jose P. Espinal are functional, but if
I'm having a failure, I up to grab the putty and investigate what's
going on there :) How
Anyone has experience in configuring Redfone to support failover, please share
with me?.
--- On Fri, 3/26/10, Eric Wheeler aster...@ew.ewheeler.org wrote:
From: Eric Wheeler aster...@ew.ewheeler.org
Subject: Re: [asterisk-users] Asterisk load balancing and failover
To: huugiang...@yahoo.com
I'm sorry, Anyone has experience in configuring Redfone to support
load-balancing, please
share with me? I can't find any guide about this feature from RedFone.
--- On Mon, 3/29/10, huu giang huugiang...@yahoo.com wrote:
From: huu giang huugiang...@yahoo.com
Subject: Re: [asterisk-users]
Hi all,
When a user make a call to Asterisk, and when user hang up the call at any
point of the conversation, Asterisk will stop Diaplan intermediately.
At this situation, Are there any way to make Asterisk continue execute the
Diaplan ?, so Asterisk can do something like that delete
Hello list,
how does StopMixMonitor know which 'monitoring channel' to stop when
there are multiple conversations that are being monitored/recorded ??
I want to use StopMixMonitor in a macro, called from within
applicationmap (features.conf).
Jonas.
--
There is the h exten to deal with exactly what you want
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
huu giang wrote:
Hi all,
When a user make a call to Asterisk, and when user hang up the call at
any point of the conversation, Asterisk will stop Diaplan
Hi Ishfaq
When Asterisk continue the dialplan, can it discover that the client has hang
up the call ?.
Is there any way ?.
--- On Mon, 3/29/10, Ishfaq Malik i...@pack-net.co.uk wrote:
From: Ishfaq Malik i...@pack-net.co.uk
Subject: Re: [asterisk-users] Continue a dialplan when the client
Thanks Ishfaq, h extension is the answer for my question :).
--- On Mon, 3/29/10, huu giang huugiang...@yahoo.com wrote:
From: huu giang huugiang...@yahoo.com
Subject: Re: [asterisk-users] Continue a dialplan when the client hang up the
call
To: Asterisk Users Mailing List - Non-Commercial
hi all!
Does anybody know, how to get the status autopaused from queues.
I want to display the status to the agent.
I'm using asterisk-1.4.29.1
thanks
chris
--
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-- Bandwidth and Colocation Provided by
On Fri, Mar 19, 2010 at 2:30 PM, Jonathan Addleman j...@redowl.ca wrote:
If that doesn't work for some reason (In my case, I needed to stream
through a flash applet on a web page, so it needed to be an mp3 stream),
you can use an eagi that pipes through an encoder and then to your
streaming
Actually h extension is for hangup context, g within the dialplan command is
used for this purpose, so you can stay within the same context when the
other party hangs up the call, and execute further commands.
On 2010-03-29 6:18 AM, huu giang huugiang...@yahoo.com wrote:
Thanks Ishfaq, h
If I remember correctly you should do CDR(flavor)=cherry and it should
work. I have added custom fields in my CDR table in the past and didn't need
triggers.
On 2010-03-29 3:40 AM, Robert Price rob...@proxims.com wrote:
Hello Alex,
I'm struggling with the same problem and, not wanting to modify
Adtran also has a 10 year warranty on their products. Doesn't matter if
you bought it off eBay for 30 bucks. Also excellent support. Something
needs fixed, get an RMA you ship it to them, they ship back paid, often
an exchange upgraded.
also there are cheap TE110 cards available on eBay, since
Hello,
I'm wondering if it is possible to ring X number of extensions
simultaneously, and each answered call can be handled with some code.
I can do a huntgroup-esque way of dialling, but I want all the dialled
numbers to be picked up.
I hope this makes sense.. If not please say..
Many thanks!
The built-in Dial command will not satisfy this requirement (first pickup
terminates function). You could do an AGI to do an Asynchronous dialing of
the X extensions simultaneously (Although the realistic limit would probably
be 5-10 extensions at once).
_
From:
Hi Danny,
Thats excellent, thank you. I have limited knowledge on writing AGI, but I
am always up for a challenge!
Thanks
Andy
On 29 March 2010 14:08, Danny Nicholas da...@debsinc.com wrote:
The built-in Dial command will not satisfy this requirement (first pickup
terminates function). You
On Sun, Mar 28, 2010 at 09:16:48PM -0500, Tim Panton wrote:
On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote:
On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
I'm having trouble getting a xorcom set up.
A large part of the problem is that the box is a _long_ way away and
I
It seems that my realtime is not assigning channel variables correctly.
INFO
Asterisk 1.6.0.26
Exten.conf
exten = _X.,1,NoOp()
exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})
exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})
exten = _X.,4,NoOp(DEVICE is ${DEVICE})
exten =
Hi!
I'm wondering if it is possible to ring X number of extensions
simultaneously, and each answered call can be handled with some code.
You might want to explain what you are trying to do.
Dial() can handle this by using something like SIP/peer1SIP/peer2
The first one that answers wins.
Hi!
so after using the sip show channels commands i can see that most of the
the communication are under ulaw format and one or two are under gsm, do you
guys know a way to force the system to use only ulaw, is it a good idea and
is it gonna solve my static noise issue?
actually, the caller and
i have the same model polycom phone configured with another server
(asterisk 1.4), and guess what no noise at all. any guess!
Replace the handset?
--
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Alyed,
I figured it was a freepbx issue but I have had no response on the post i put
on their forum. I am going to look some place else for freepbx support. I am
not impressed at all with the freepbx support. On the other hand i have been
getting answers to most all my post on this mail list
:) all users are having the same issue, even those connected to this server
from abroad!
2010/3/29 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
i have the same model polycom phone configured with another server
(asterisk 1.4), and guess what no noise at all. any guess!
How about...
exten = _X.,1,NoOp()
exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})
exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})
exten = _X.,4,Set(usernamepair=${CUT(NULL,\,,1)})
exten = _X.,5,Set(username=${CUT(usernamepair,=,2)})
exten = _X.,6,NoOp(DEVICE is ${DEVICE})
exten =
You could read the source code, but based on it's name I would say it is a
library responsible for zone specific tone generation. Many parts of the world
have different tone patterns than the U.S. and Asterisk is used worldwide. A
better question is, why are you concerned by it?
I was building
Hi all,
I've cross-posted this to the -users and -biz groups. Hope that's OK.
I have a customer who REALLY needs to be able to send/receive faxes reliably.
I could probably get hylafax configured, but I'm not sure how reliable it is.
If it is considered reliable, would someone let me know?
Khalid-
:) all users are having the same issue, even those connected to this server
from abroad!
Since you have an identical working system, why are you not able to debug this?
First swap the phone... then swap any
cards in the server, then servers, then check carefully software differences
Mike Diehl wrote:
Hi all,
I've cross-posted this to the -users and -biz groups. Hope that's OK.
I have a customer who REALLY needs to be able to send/receive faxes reliably.
I could probably get hylafax configured, but I'm not sure how reliable it is.
If it is considered reliable,
Mike Diehl wrote:
I could probably get hylafax configured, but I'm not sure how reliable it is.
If it is considered reliable, would someone let me know?
It's reliable as long as you're not using FoIP (i.e. as long as you're
faxing with PSTN lines).
Thanks,
Lee.
--
On Monday 29 March 2010 10:15:50 am jon pounder wrote:
Mike Diehl wrote:
Hi all,
I've cross-posted this to the -users and -biz groups. Hope that's OK.
I have a customer who REALLY needs to be able to send/receive faxes
reliably. I could probably get hylafax configured, but I'm not
Is there anyway to get the following scenario to work...
I have 3 IAX trunks that I want to setup to peer with other * boxes. I have 1
physical interface, eth0. I also have 2 sub interfaces, eth0:1 eth0:2. I
want to setup a single IAX trunk on each of the interfaces. All 3 interfaces
are
i posted this on the freepbx site. here is the response
from the trace, everything is working. Check your asterisk log for file
errors playing back the audio, could be your sound files are not
installed or messed up.
so i checked /etc/log/asterisk/full
and in vi full i did /error and
On Mon, Mar 29, 2010 at 2:26 PM, Mike Diehl mdi...@diehlnet.com wrote:
On Monday 29 March 2010 10:15:50 am jon pounder wrote:
Mike Diehl wrote:
Hi all,
I've cross-posted this to the -users and -biz groups. Hope that's OK.
I have a customer who REALLY needs to be able to send/receive
Hi,
I have done it a few times. Just posted a small blog about it with code.
Check it at www.ilovetovoip.com/?p=322. I hope it'll help you.
--
Zeeshan A Zakaria
On 2010-03-29 11:07 AM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
I'm wondering if it is possible
When it comes to FoIP, it is a good idea to stay with analog lines and
regular fax. FoIP is a pain and not recommended where fax is a regular part
of a business.
On 2010-03-29 3:20 PM, David Backeberg dbackeb...@gmail.com wrote:
On Mon, Mar 29, 2010 at 2:26 PM, Mike Diehl mdi...@diehlnet.com
Thanks for the feedback. Btw, I meant SIP / IAX gateway. I'll take a look
at the suggestions.
Best Regards,
Joseph
On Sun, Mar 28, 2010 at 7:28 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sun, 28 Mar 2010, Joseph Begumisa wrote:
Can anyone recommend a 24 fxs port voip gateway
At 23:26 3/28/2010, James Lamanna wrote:
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 28 Mar 2010, Joseph Begumisa wrote:
Can anyone recommend a 24 fxs port voip gateway that has worked well with
asterisk? I have a couple of analog handsets
On Mon, Mar 29, 2010 at 04:03:12PM +, Joseph L. Casale wrote:
You could read the source code, but based on it's name I would say it is a
library responsible for zone specific tone generation. Many parts of the
world have different tone patterns than the U.S. and Asterisk is used
So I realize the error in my question/request. The section I was thinking of
using for the binding IP address, is, itself, the wrong place to do such a
thing. It would need to be in the register statement... something like the
following...
register = user:pass:fro...@targetpeer
I do realize
- trebaum treb...@telepaths.org wrote:
So I realize the error in my question/request. The section I was
thinking of using for the binding IP address, is, itself, the wrong
place to do such a thing. It would need to be in the register
statement... something like the following...
I have been asked by my church to recommend a VoIP system which can do
the following.
They do internet radio shows which are sometimes broadcast on radio.
They are looking for a system which does the following for about 5
agents, exactly as they have described it.
1. Take incoming calls
2. Put
I thought about that, but there are a couple issues with that. Currently the
physical interface has a single gateway (the only way to change this is to add
more physical interfaces), and that gateway is what does the routing to the
specific peers. Being that all of the traffic is going to a
Un-top-posting...
- trebaum treb...@telepaths.org wrote:
So I realize the error in my question/request. The section I was
thinking of using for the binding IP address, is, itself, the wrong
place to do such a thing. It would need to be in the register
statement... something like
I have been asked by my church to recommend a VoIP system which can do
the following.
They do internet radio shows which are sometimes broadcast on radio.
They are looking for a system which does the following for about 5
agents, exactly as they have described it.
1. Take incoming calls
2. Put
Apart from the call back function, appears to me that any asterisk
distribution interfaced with freepbx will do what you need.
I would recommend pbxinaflash (http://www.pbxinaflash.com), they have a very
active forum, and will get up and running very fast.
Gondar
On Mon, Mar 29, 2010 at 1:46 PM,
Just noticed that packman has precompiled versions of amr codec.
Both wideband and narrowband. Can these be used for asterisk?
Heard some nice about AMR (in general)
If so, any one around with experience with either??
hw
--
_
Hi Frank
I have found Freepbx on top of Asterisk a good solution for the church I look
after and the rest of my customers, the callcentre functions you need are built
in it and if they have someone technical then they can expand what they are
doing
It has both queues and ring groups (which
Sounds like the church has strayed from its core competencies and
invited the money-changers into the temple.
Being the official asterisk-biz harbinger of God's wrath, I suggest an
intensely commercial platform, for the meek shall inherit the Earth,
not the 700 Club. Fight the power.
--
I would appreciate any ideas of what I'm doing wrong on this. My
dialplan calls an AGI which records a file. That works, but I'm trying
to find a way to determine whether the caller pressed # to stop a
recording before the maxtime expired, or if the recording ended due to
reaching the max
On Mon, 29 Mar 2010, Jeff Johnson wrote:
I would appreciate any ideas of what I’m doing wrong on this. My
dialplan calls an AGI which records a file. That works, but I’m trying
to find a way to determine whether the caller pressed # to stop a
recording before the maxtime expired, or if the
That worked...The help is very much appreciated.
Jeff
http://www.neturallyspeaking.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, March 29, 2010 8:04 PM
To: Asterisk Users
Hello,
I was planning on using a call file to test my IVR on a regular basis to
ensure it is operational
Channel: local/1...@from-internal
Application: SendDTMF
Data: ww12345678#1w1234#w1ww
But what ever I try so far the IVR does not seem to
Hi,
I'm trying to figure out the cause of a soft lockup I experienced:
Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s!
[asterisk:32029]
Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk
Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[c046e7fe] CPU: 0
Mar 29
They say confession is good for the soul. Perhaps they are offering a
phone in confessional service?
Unfortunately the business of the church often flies in the face of
the business of the Church.
On 03/29/2010 07:48 PM, Alex Balashov wrote:
Sounds like the church has strayed from its core
Hi,
I need a Diameter protocol stack (DDA/Ro) for Asterisk.
I came across Traffix Systems Diameter stack. according to the site they got
Asterisk ready Diameter stack and also Diameter Gateway that can interface to
Asterisk, has anyone tried those out ?
thanks
Maria
--
I am running into an issue with A2Billing. I will explain first of all that
everything else works! the system is 90% complete its just this one small
problem I am running into.
So my problem is that when I place a call,
1. I dial my number that I want and A2Billing gets activated
2. it asks
On 29 Mar 2010, at 08:13, Tzafrir Cohen wrote:
On Sun, Mar 28, 2010 at 09:16:48PM -0500, Tim Panton wrote:
On 28 Mar 2010, at 10:13, Tzafrir Cohen wrote:
On Sat, Mar 27, 2010 at 04:48:41PM -0500, Tim Panton wrote:
I'm having trouble getting a xorcom set up.
A large part of the problem
Hi all,
Is there anyway to catch DTMF keypress while a music file is playing
without stop the music?
Thanks,
Quyps
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New to Asterisk? Join us for a
Dell server by any chance?
I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
several other people having issues with digium cards in dell servers as
well.
I've actually done something similar to what you have done - isolated the
TE220B onto its own IRQ and set
On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson m...@mattgwatson.ca wrote:
Dell server by any chance?
I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
several other people having issues with digium cards in dell servers as
well.
I've actually done something similar to
When you say 'a2billing' won't pass the number, you mean you are calling to an
IVR or something like that.
And when did you dial you destination number twice???
Saludos,
Juan E. Rodríguez
-Original Message-
From: Nathanial Allan nathanial.al...@gmail.com
Date: Tue, 30 Mar 2010
On Tue, 30 Mar 2010, Pham Quy wrote:
Is there anyway to catch DTMF keypress while a music file is playing
without stop the music?
Have you tried externivr(). I've never used it, but it looks interesting.
--
Thanks in advance,
On Mon, Mar 29, 2010 at 9:23 PM, James Lamanna jlama...@gmail.com wrote:
On Mon, Mar 29, 2010 at 8:38 PM, Matt Watson m...@mattgwatson.ca wrote:
Dell server by any chance?
I have a similar problem with a TE220B in a Dell 1950 III server - i've seen
several other people having issues with
I'm not an Amportal expert so all I can say from:
-- Executing [...@from-internal:8] Playback(SIP/117-01f6,
do-not-disturbactivated) in new stack
-- Executing [...@from-internal:9] Macro(SIP/117-01f6,
hangupcall,) in new stack
is that Asterisk is playing the do-not-disturbactivated
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