Hello everyone,
I can not get the name of the recoding file of agents calls. I set
agents.conf as following:
; Enable recording calls addressed to agents. It's turned off by default.
recordagentcalls=yes
;
; The format to be used to record the calls (wav, gsm, wav49)
; By default its "wav".
;recor
Hi, I was looking round on the Internet and saw there was no definitive
list of free applications available for use with Asterisk, so I thought
I'd compile a list for you all. If there's anything that you know of
that is actively maintained but not in the list below, let me know (bear
in mind I
2009/8/8 Dan Pilcheck
> Hello all,
>
> This is a VICIDial server and I am looking to send calls to VM box
> 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI).
> This would be inserted between exten => s,8,Background(open) and exten
> => s,9,AGI.
> From what voip-info has [not] to
David Backeberg wrote:
On Fri, Aug 7, 2009 at 1:48 PM, Terry Nathan wrote:
I'm having a weird problem with CallerIDs and I can't tell if it is a
problem with Asterisk, the telco, or the VOIP provider I'm using.
Basically, I am using Asterisk as a proxy for my cell phone. People call
in and t
On Fri, Aug 7, 2009 at 1:48 PM, Terry Nathan wrote:
> I'm having a weird problem with CallerIDs and I can't tell if it is a
> problem with Asterisk, the telco, or the VOIP provider I'm using.
>
> Basically, I am using Asterisk as a proxy for my cell phone. People call
> in and the call gets forward
Thanks for a quick reply... This link just shows how to set MOH feature if
the phone has "hold" feature. I want to place a call on hold irrespective of
SIP phones used... If I create an MOH extension as shown & transfer the
calls to that extension and then if one party disconnects the call, the
oth
If you want to hang more results on this subject, please see the thread here:
http://www.voipusersconference.org/2009/08/sip-for-apple-iphone/
I'm very interested in anyone who is doing development in this space
so keep in touch. Basically, even though I've always preferred
DECT/SIP phones to wif
Does it this link help?
http://www.voip-info.org/wiki/view/Asterisk+cmd+MusicOnHold
On Fri, Aug 7, 2009 at 10:07 PM, Venkateshwarlu
Kakkireni wrote:
> I want to a place a call (SIP) on hold in asterisk? Is there any way to do
> it? If yes, please give me an example. We are using Asterisk 1.4.24.1.
I want to a place a call (SIP) on hold in asterisk? Is there any way to do
it? If yes, please give me an example. We are using Asterisk 1.4.24.1. Any
help would be appreciated...
Thanks & Regards,
Venkat
___
-- Bandwidth and Colocation Provided b
Danny Nicholas wrote:
> Editing my original comment, "linux uname" should have been "linux
> hostname". Tilghman, can you elaborate a bit more?
>
It's definitely not based on that either since changing your hostname
doesn't change your Host-ID.
In case anyone was wondering, I changed the adapte
On Fri, 7 Aug 2009, Jeff LaCoursiere wrote:
>
> Howdy,
>
> My first forray into using res_mysql.conf for realtime access of sip users
> and extensions.
>
> I have the following relevant section of extensions.conf:
>
> ---
>
> [trunklocal]
> exten => _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLO
I was able to get a VMWare Fusion CentOS 5.3 with Asterisk 1.6.0.9
talking to a Xorcom Astribank on my MacBook. I could connect a POTS
line to an FXO port and a phone to an FXS port and make calls.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 7, 2009, at 9:25
Meant to add that this is 1.4.26... :)
On Fri, 7 Aug 2009, Jeff LaCoursiere wrote:
>
> Howdy,
>
> My first forray into using res_mysql.conf for realtime access of sip users
> and extensions.
>
> I have the following relevant section of extensions.conf:
>
> ---
>
> [trunklocal]
> exten => _NXXX
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten => _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include => trunklocal
include => trunkt
So far, the best iPhone platform app I've found is a $10 one called
iPico. It is a one account SIP client, better designed than the others
and it actually works and can dial SIP URI.
I learned about it directly from Ruben Olsen mentioning it on the VUC
call an hour ago. I will be posting the edite
Hi Cary,
Thanks for the quick reply :D I get what you're saying. I have a
suspicion that it is the telco's fault since every other number that
receives a call from my Asterisk box displays the correct number. I'll
give setting the caller id another go and play with that.
I guess what I am look
On Fri, 7 Aug 2009, Dan Pilcheck wrote:
> This is a VICIDial server and I am looking to send calls to VM box
> 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI).
> This would be inserted between exten => s,8,Background(open) and exten
> => s,9,AGI.
>> From what voip-info has [not
Yes, the issue(s) is/are:
1. The VOIP provider may be masking the callerID for their own cost
allocation reasons. That is some of the issue.
2. Your Asterisk box may forward some of the regular phone line calls with
their caller ID.
3. Somehow, the number you want to use may leak through someti
On Fri, 2009-08-07 at 17:18 +0200, harry R wrote:
> Anyone know how to use regcontext et regexten parameter from sip.conf
> and can give an example ?
Sure... let's say I have a phone with the following configuration in
sip.conf:
[myphone]
type=friend
context=inside
host=dynamic ; phone will regis
Could you use AMI from within the AGI to poll the call status and act
accordingly?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Friday, August 07, 2009 12:39 PM
To: Asterisk Users Mailin
I'm having a weird problem with CallerIDs and I can't tell if it is a
problem with Asterisk, the telco, or the VOIP provider I'm using.
Basically, I am using Asterisk as a proxy for my cell phone. People call
in and the call gets forwarded to my personal number. The feature on my
phone allows f
On Friday 07 August 2009 11:04:14 Dan Pilcheck wrote:
> This is a VICIDial server and I am looking to send calls to VM box
> 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI).
> This would be inserted between exten => s,8,Background(open) and exten
> => s,9,AGI.
>
> From what voip-
>On Fri, Aug 7, 2009 at 11:47 AM, James Lamanna wrote:
>> Hi,
>> I'm coming up with ideas about building a cluster of asterisk servers,
>> and am exploring the virtualization option.
>> I'm curious to know some real-world data about how many extensions a
>> VMWare install on good hardware could sup
¡Tienes un mensaje nuevo en Badoo!
Xavier Cardil Coll te dejó un mensaje.
Sigue el link para abrirlo:
http://eu1.badoo.com/0153696585/in/ptA2fUgm9zo/?lang_id=7
Además, alguien ha estado preguntando por ti:
Omar Lloper (Valencia, España)Laura Martin (Barcelona, España)Alberto
Weingartshofer (Asu
On Fri, Aug 7, 2009 at 11:47 AM, James Lamanna wrote:
> Hi,
> I'm coming up with ideas about building a cluster of asterisk servers,
> and am exploring the virtualization option.
> I'm curious to know some real-world data about how many extensions a
> VMWare install on good hardware could support.
Editing my original comment, "linux uname" should have been "linux
hostname". Tilghman, can you elaborate a bit more?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Friday, August 07, 200
Talk to damin AT nacs.net (he's on this mailinglist)
Zoaaa
James Lamanna wrote:
> Hi,
> I'm coming up with ideas about building a cluster of asterisk servers,
> and am exploring the virtualization option.
> I'm curious to know some real-world data about how many extensions a
> VMWare install on g
On Friday 07 August 2009 10:11:23 Thomas Kenyon wrote:
> Danny Nicholas wrote:
> > AFAIK, host-id is tied to ip address and linux uname, so that's all that
> > should matter.
>
> It's definately not tied to uname, otherwise it'd change every time I
> built a new kernel. Basing it on IP address woul
Where you able to compile DAHDI in a virtual environment? How about skype
for asterisk? Has anyone tried that in a virtual environment? Seems like
to register the license, digium tool is looking for a connection on eth0,
and in a virtual environment I see the name as vnet0 or vnet1. At least
th
been testing with Sun VirtualBox and i managed more than 30 extensions on a
2GHz Dual core machine with 1 GB ram for the VBOX.. just not running recodring
or encoding .. things went well
--
AHD Tarek Sawah
> Date: Fri, 7 Aug 2009 08:47:03 -0700
> From:
It depends on processor capability, disk access time and bandwidth. You
will need to dedicate slices of disk and bandwidth for each machine. A
"realworld" scenario of worst case would be this:
You get sucky throughput on VM2 because 3 or 4 folks are monitoring calls or
using voicemail on VM1.
---
Hello all,
This is a VICIDial server and I am looking to send calls to VM box
2100 after 3 minutes of sitting in the queue(via the VICIDial AGI).
This would be inserted between exten => s,8,Background(open) and exten
=> s,9,AGI.
>From what voip-info has [not] told me, the AGI doesn't allow for a
t
Hi,
I'm coming up with ideas about building a cluster of asterisk servers,
and am exploring the virtualization option.
I'm curious to know some real-world data about how many extensions a
VMWare install on good hardware could support.
I've seen stories about how the hypervisor timeslicing can wreak
-users
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de virus 4315 (20090807) __
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http://www.eset.com
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de virus 4315 (20
Hi
the asterisk version is 1.4.21.2
Here is the CLI
-- Executing [...@incomming:1] Set("Zap/4-1",
"DB(lastcaller/zap4)=01942876818") in new stack
-- Executing [...@incomming:2] GotoIf("Zap/4-1", "0?s-spoof|1:") in new
stack
-- Executing [...@incomming:3] Ringing("Zap/4-1", "") in new
Hi
Anyone know how to use regcontext et regexten parameter from sip.conf and
can give an example ?
thx
regards
Harry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register N
randulo a écrit :
> Hi,
>
Hello
> I've tried two SIP clients so far and both have unusable outgoing
> audio quality.
>
[...]
> Anyone have any recommendations?
>
I made few test with various client, Sip and IAX, on iPhone first
generation:
. frings: good quality but to much delay. Also I
Danny Nicholas wrote:
> AFAIK, host-id is tied to ip address and linux uname, so that's all that
> should matter.
>
It's definately not tied to uname, otherwise it'd change every time I
built a new kernel. Basing it on IP address would be extremely foolish,
since most people use one of 3 ranges
Hello all.
I'm rather new. I'm lost and I would really appreciate if someone can
point me in the right direction. I don't know if it something we did
wrong or if it's an issue with our SIP TSP.
We have a new Asterisk 1.4.26 server that has been running without a
hitch for several days.
I have
Ok, so now let me ask the question more directly:
I am looking for the best SIP application for the iPod Touch (Wifi
only). I don't care about 3g, Gsm or anything phone-related.
The app has to be able to register with an arbitrary SIP service
and/or dial arbitrary SIP URI. If it could dial one li
I'm using it rather successfully. Not perfect, but it works.
It is limited to WiFi connectivity... at least here in Spain I cant get
either client to work over 3G.
I'm using Fring and Truphone. Although I have only configured a SIP to my
Asterisk with Fring.
Skype works fine.
We tested with seve
AFAIK, host-id is tied to ip address and linux uname, so that's all that
should matter.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon
Sent: Friday, August 07, 2009 3:44 AM
To: Asterisk Users Mail
Show us your CLI output. I suspect that you're not getting a bridge and/or
you're timing out. Also sip.conf and user.conf would be helpful as well as
Asterisk release.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of rober
Nearly got an SPA922 phone working behind a NAT,
the phone registers, and I can dial out and have two way speech,
on an incoming call the SPA922 rings
I answer and the SPA922 shows "Anwsering" but never does and the far end
continues ringing until the voicemail answers,
this then show as a disc
Guillermo Garron
Alke Technology
T. +591 33 141000
e. ggar...@alketech.com
On 07/08/2009, at 06:41, hh174 wrote:
Fring works perfectly for me.
Tarek Sawah a écrit :
Have you tried installing fring? i still like that app .. supports
GREAT quality voice over EDGE and GPRS .. plus WIFI and
Hi,
I got ast. 1.6.0.10 working for a few weeks without a problem.
A few mins ago..I got the following msgs on ast-cli and asterisk service
crashed.
I coudlnt find anything that might cause this problem.
Any ideas??
[Aug 7 13:54:34] WARNING[10102]: codec_gsm.c:133 gsmtolin_framein:
Invalid GSM d
Fring works perfectly for me.
Tarek Sawah a écrit :
Have you tried installing fring? i still like that app .. supports GREAT quality voice over EDGE and GPRS .. plus WIFI and 3G if available..
i tried it with Skype and it's great..
Asterisk and its great
Callcentric VoIP provider and it w
Have you tried installing fring? i still like that app .. supports GREAT
quality voice over EDGE and GPRS .. plus WIFI and 3G if available..
i tried it with Skype and it's great..
Asterisk and its great
Callcentric VoIP provider and it was great..
one thing though i noticed that at some times
Jonathan Moore writes:
> The idea of RTP being to blame would make sense though. I can
> still transfer and such, and watching the console, I see when I press
> various keys on the phone, so it seems that the SIP traffic is working
> out fine. (I do understand that right? SIP == control RTP ==
Hello everyone,
I have a problem with getting name of the recorded file of agent calls. As
I've googled I found that the name of the recording file should be inserted
in userfield of CDR table. To do this I set
createlink=yes in agents.conf
but still userfield of cdr is empty but the recrding fil
If you are completely new to Asterisk and want to run a professional
call-center, my suggestion is to stick to a hand-made, lean, minimal
configuration.
l.
2009/7/13 ashish chauhan
> Dear all,
> I am new to asterisk.i like to configure call center using
> asterisk.please can anyone t
How do you do the log-on?
l.
2009/8/6 Joao Gomes Pereira
> Hello to all
> I have a queue where often my agents get stuck and cannot logoff.
> This is very bad, because agents cannot login again, and in Queuemetrics
> reports the agents appear to be online.
> How can I create a timeout to my agen
Hi
The only time I've had issues that seem a bit like yours it was down to
the order of codecs in the handset settings. Make sure they match the
order dictated on the server.
Ish
Jonathan Moore wrote:
> On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien
> Cramatte wrote:
>
>> Hi,
>>
>> This sounds
I'm about to change the motherboard in my server machine, (Different
chipset). The most notable thing that will change, is the onboard
network card (eth2) will be an atheros one instead of realtek.
If I change the mac address of eth2 to read the same as the old one,
will my host-id stay the sam
Appearances suggest that some part(s) of the AGI protocol changed
between 1.4 and 1.6.
hh174 wrote:
> Hello,
>
> I have a problem with fastagi.
> In fact I have a fastagi written in Java.
> Communcation between asterisk 1.6 and the server works correctly, except
> when a 'HANGUP' is sent by ast
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