On Thu, Aug 6, 2009 at 10:15 PM, Alex Balashovabalas...@evaristesys.com wrote:
Which generation of the handset are you using? They differ in their
processing power and that may account for at least some of it.
Alex, this is just an iPod Touch, not even a handset. It doesn't have
a mic at all,
Hello !
I 'm having a machine running asterisk 1.6.0.10 with IAX and dahdi.
The calls are going in and out from IAX2 to dahdi (chan_dahdi + libpri)
and vice versa.
After a period of time, I got the following scenario:
NOTICE[860] chan_iax2.c: I should never be called!
WARNING[752] channel.c:
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 asterisk...
In this case, the java server doesn't read the message.
I have tride with PHP, same result.
A ngrep
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
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
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
Cramattescrama...@zensoluciones.com wrote:
How do you do the log-on?
l.
2009/8/6 Joao Gomes Pereira gomespere...@startel.pt
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
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 ashishchauhan07...@gmail.com
Dear all,
I am new to asterisk.i like to configure call center using
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
Jonathan Moore supermegat...@gmail.com 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?
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
Fring works perfectly for me.
Tarek Sawah a crit:
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
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
Guillermo Garron
Alke Technology
T. +591 33 141000
e. ggar...@alketech.com
On 07/08/2009, at 06:41, hh174 oliv...@hh174.be 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
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
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
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
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
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
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
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 don't
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
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 stack
de ESET NOD32 Antivirus, versión de la base de firmas
de virus 4315 (20090807) __
ESET NOD32 Antivirus ha comprobado este mensaje.
http://www.eset.com
__ Información de ESET NOD32 Antivirus, versión de la base de firmas
de virus 4315 (20090807) __
ESET NOD32
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
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
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
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.
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:
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
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 would be
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 good
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, 2009
On Fri, Aug 7, 2009 at 11:47 AM, James Lamannajlama...@gmail.com 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
¡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
On Fri, Aug 7, 2009 at 11:47 AM, James Lamannajlama...@gmail.com 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
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-info
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
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
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
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
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] told
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
So far, the best iPhone platform app I've found is a $10 one called
iPico. It is a one account SIP client, better designed than the others
and it actually works and can dial SIP URI.
I learned about it directly from Ruben Olsen mentioning it on the VUC
call an hour ago. I will be posting the
Howdy,
My first forray into using res_mysql.conf for realtime access of sip users
and extensions.
I have the following relevant section of extensions.conf:
---
[trunklocal]
exten = _NXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[local]
include = trunklocal
include =
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 =
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
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 =
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 adapter
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
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
Kakkirenivenka...@iconsultech.com 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
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
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
other
On Fri, Aug 7, 2009 at 1:48 PM, Terry Nathantnat...@aiinc.ca 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
David Backeberg wrote:
On Fri, Aug 7, 2009 at 1:48 PM, Terry Nathantnat...@aiinc.ca 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.
2009/8/8 Dan Pilcheck pilch...@gmail.com
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
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
57 matches
Mail list logo
