-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Docs
Yes, what is it you attempting? I use DUNDi extensively,
though you
We are running the latest stable version of asterisk
Is there a way yet to load the realtime peers automatically from
database like the reload command does .
Everytime we make a change to the peer we have to manually load the peer
using
sip show peer abc load
Any suggestions appreciated,
Thank
I think [EMAIL PROTECTED] allows a user to search a directory by either first
OR last
name, right? I don't know for sure since I don't run [EMAIL PROTECTED]
I would like to offer that functionality in my system - and I'd have done it
by now if there was a prompt where Allison asks press 1 to
Further to my post, I was going to revert SVN backwards until I got to a
point where AddQueueMember works with the local channel
To this end, if anyone is using this functionality would they send me
the version number and prefably svn number so that I can track things down.
It may be a
HI all,
I was on this list back in Dec-Jan but the asterisk server got pushed
back in the project queue and it seems to have finally risen to the top.
I am looking to deploy * running on Centos 4 on a Dell 2850. I need a
single T-1 (PRI) card with HW echo cancellation. I had been told that
the
It has also just become glaringly apparent to me that a 'reload' does not
always reload the DUNDi configuation.
How can I reload DUNDi without stopping/starting Asterisk?
-Original Message-
From: Douglas Garstang
Sent: Wednesday, June 14, 2006 11:00 AM
To: Asterisk Users Mailing
Does anyone here have any experience using Digium cards with MBX
hardware/servers?
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Asterisk-Users mailing list
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Here is a copy of indications.conf I have not included other country codes. I
noticed that there are no spaces in country=us does that matter?
Thanks
[general]
country=us
[us]
description = United States / North America
ringcadance = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring =
Thomas Kenyon wrote:
I need to be able to connect an old PA system to an asterisk box, which
basically works as a couple of amplifiers taking an analogue phone
signal and playing whatever it produces out of some speakers. There is
no on-hook state in the whole setup.
Obviously If I just
Brent Torrenga wrote:
I would like to offer that functionality in my system - and I'd have done it
by now if there was a prompt where Allison asks press 1 to search by first
name, press 2 to search by last name. But I don't think that prompt exists.
Can someone explain how (or just tell me that
On 11:51, Wed 14 Jun 06, Mike wrote:
Hi,
I'm stuck writing a Web GUI because nothing out there is exactly what I
need. I'm not writing something as extensive as what _is_ out there, but
just something that allows users to change where their calls are forwarded
and other small things like
Following up to my earlier post.
I'm seeing no inbound SIP traffic locally despite, apparently, being
sucessfuly reigstered with my sip provider.
sip show peers give me
Name/username HostDyn Nat ACL Port Status
2201/2201 192.168.1.100D
Brent Torrenga wrote:
I think [EMAIL PROTECTED] allows a user to search a directory by either first
OR last
name, right? I don't know for sure since I don't run [EMAIL PROTECTED]
I would like to offer that functionality in my system - and I'd have done it
by now if there was a prompt where
It should not matter.
Tim Sharp wrote:
Here is a copy of indications.conf I have not included other country codes. I noticed that there are no spaces in country=us does that matter?
Thanks
[general]
country=us
[us]
description = United States / North America
ringcadance = 2000,4000
dial
Has anyone on here used kiax before? I am asking because I have it installed
on several computers and have been able to get it to connect and register
to my Asterisk box. I can even call between them and my SIP softphones.
The problem I am having is this: when I use kiax to call someone else,
Tim Sharp wrote:
I am running on 1.2.7.1 and have an intermittent problem when making outgoing calls. Sometimes the calling party does not hear the ring tone in their handset, but the call goes through. From my extension I have only had 3 calls like this in the last couple of weeks, other
Don't know about the single T1 but the a104d works flawlessly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Warren
Sent: Wednesday, June 14, 2006 1:25 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Single T1 card with Echo
That may not be such a bad idea. I've read people trying to put
Asterisk on a WRTG54 or something like that. Would that be good? I
guess I could do SIP in the office and trunk via IAX2 and save on
bandwidth plus internal calls would be local.
I tried to upgrade them to 512K but because
We do both first and last name searching.
In [EMAIL PROTECTED] just select first or last name.
In asterisk do:
-- Executing AGI(Zap/2-1, directory|general|ext-local|bo) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/directory
-- Playing 'dir-intro-fnln-oper' (language
If you do a reload pbx_dundi.so, it'll reload the dundi configuration. If
you're talking about the strings it returns, if you want to get an
immediate result and not use the cache, use something like dundi lookup
num bypass.
Also, if you have separate entry points for each section of the
We wrote and submitted a patch to do this. Just modify app_directory.c
and recompile. It adds a new flag b to the directory( ) app where you
can have it use both first and last name.
-= Info about application 'Directory' =-
[Synopsis]
Provide directory of voicemail extensions
[Description]
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Users
If you do a reload pbx_dundi.so, it'll reload the dundi
configuration.
If by MBX you mean Motherboards Express (www.mbx.com), these are, for the
most part, OEM'd by SuperMicro, and most of them should be perfectly fine
with a variety of Linux Distros and running Asterisk.
Good idea to get the model # of the mobo, and cross-reference it on the
SuperMicro website,
We have been using MBX for some time and they are great to work with. We supplied them our requirements including the links to the digium site for the various cards we wanted to use and they came up with configuration that works great. Jon Frank is our sales rep and he is VERY attentive and quick
Matt,
Ahh, they use an AGI script. That's what I was looking for. I think I'll go
dig through the [EMAIL PROTECTED] packages now... Thanks!
We do both first and last name searching.
In [EMAIL PROTECTED] just select first or last name.
In asterisk do:
-- Executing AGI(Zap/2-1,
Yeah sorry.. I had two thoughts running togethor there interupted by a
phone call :)
[EMAIL PROTECTED] uses a script to do both.
On 6/14/06, Matt [EMAIL PROTECTED] wrote:
We do both first and last name searching.
In [EMAIL PROTECTED] just select first or last name.
In asterisk do:
--
Very good.. thanks for the reassurance makes branding a PBX easy
:) Just out of curiousity.. what O/S are you all running that seems
to work well with SATA? I for one, hate SATA.
On 6/14/06, tracinet [EMAIL PROTECTED] wrote:
We have been using MBX for some time and they are great to work
voip-info/wiki and write some basics abt the ser.cfg or
somethjing .. then it would be great.
I haven't read the tutorials, so I could be wrong, but I doubt they'd be very
much use. They probably don't do more than give a basic overview, and I'm sure
they don't touch things like
On Wed, Jun 14, 2006 at 06:31:28PM -, [EMAIL PROTECTED] wrote:
Has anyone on here used kiax before? I am asking because I have it installed
on several computers and have been able to get it to connect and register
to my Asterisk box. I can even call between them and my SIP softphones.
I also heard that Sangoma was planning to release a Single T1 card with HW
echo can but I don't know when. My source was a VERY reliable one.
-Original Message-
From: Shane Burrell [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 11:46 AM
To: 'Asterisk Users Mailing List -
I'll second that. I use Ubuntu, actually installed asterisk through
apt-get. Too easy.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Fedyk
Sent: Tuesday, June 13, 2006 10:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing
even after they've been picked up... Here's one users summary:
When I pick up the phone, I hear a dial tone and I am able to dial out.
But for some odd reason, the receiving line picks up while the outgoing line
is still
FreePBX or AAH(aka trixbox) requires 256MB of RAM minimum to run properly.
Just sitting there doing nothing on my test system it is using 170MB.
-Original Message-
From: Lachek Butalek [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 7:27 PM
To: Asterisk Users Mailing List -
Hm.that's interesting to know. I'll bet they boot from CF but I
could be wrong. Any chance you can get some photo's of the inside of that
thing?
Those Mitels have a built in PoE switch do they not? Anything else special
about them that cannot be duplicated in an Asterisk Server? Not
EP == Erick Perez [EMAIL PROTECTED] writes:
EP However if we take into account the lowest performing component on
EP this system (the sata disks) we go down to 1.5gbits/s which still
EP seems to be enough.
Unless you do call recording, disk is a non-issue.
/Benny
On Wed, Jun 14, 2006 at 01:15:03PM -0700, shadowym wrote:
FreePBX or AAH(aka trixbox) requires 256MB of RAM minimum to run properly.
Just sitting there doing nothing on my test system it is using 170MB.
How exactly do you meassure memory usage?
E.g: on my laptop:
$ free
total
They have a 4 port switch, but not PoE. It's decommissioned but we haven't
taken it out of the rack yet, if I'll remember when we derack it I'll snap a
pic.
AFAIC, there is nothing in them that cannot be duplicated in a decent
Asterisk setup, and in fact the featureset that we have fleshed out
This is *not* bizarre behaviour. Check your sip.conf file and add
this line if it doesn't exist:
progressinband=no
Then restart asterisk.
Sounds iike your provider is not correctly sending indications to you.
On 6/14/06, Dan Elder [EMAIL PROTECTED] wrote:
Hi all, using * 1.2.9.1 and this
All,
Is there a way I can perform a lookup to see if a given extension exists within
a given context, on the local system? I could call Dial() and check the result
of $DIALRESULT, but I'm thinking there should be a better way.
Note, that I don't want to use ChanIsAvail(). That's only for
Greetings,
I am testing to connect two Asterisk servers. The IAX2 connection is
great to go. I used two different ways to make the call from one
server
to another, I listed both as following:
1. switch = IAX2/username:[EMAIL PROTECTED]/mycontext
2. exten = _, 1, dial(IAX2/username:[EMAIL
- Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
It may be a config problem on my end - that's why I need to know a
working version so that I can check my config on that version.
No, it's a bug, and we'll get it addressed today.
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
- Russell Horn [EMAIL PROTECTED] wrote:
bindaddr=0.0.0.0; address to bind to (0.0.0.0 binds to
all)
externalip=yyy.yyy.yyy.yy
nat=yes ; NAT settings
allow=all
canreinvite=no
Check your sip.conf.sample again... it's not 'externalip'.
--
Kevin P.
FWIW, the bug may have been introduced between r24293 (which works) and
r24566 (which doesn't).
Julian.
Kevin P. Fleming wrote:
- Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
It may be a config problem on my end - that's why I need to know a
working version so that I can check my config
Sangoma is NOT releasing a single T1 with echo cancellation.
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice - 800.398.VoIP X3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
aim -
Hello there
I have a small php script that Originates a call, it looks like this:
Action: login
Events: off
Username: strUser
Secret: strSecret
Action: originate;
Channel: SIP/SIP04DK/59119994;
WaitTime: 30;
CallerId: ANYthing;
Exten: 27289955;
Context: Mycontext;
Priority: 1;
Action: Logoff;
John Joseph wrote:
--- Markus Schuster [EMAIL PROTECTED] wrote:
Could you please post some details (or even better:
write them in some sort
of Wiki) on the configuration you did on the Nokia?
I tried to put some details on Voip-info.org ,
please check the link
I am in the same situation, I have heard the hw echo can is much better, easier to configure, etc. But it seems like an overkill to use a quad span card when we will only be using 1. Anyone know if digium or sangoma will release a dual span card with hw echo can?
On 6/14/06, Cory Andrews [EMAIL
Worked it out...
ChanIsAvail(Local/[EMAIL PROTECTED])
-Original Message-
From: Douglas Garstang
Sent: Wednesday, June 14, 2006 2:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Determining if extension exists
All,
Is there a way I can perform a
I have to purchase an E1 card to connect an asterisk to the phone
company. Should I go with echocan in hardware (sangoma does not seem
to have is in its a102/101 models) of should I use software
cancellers?
My phones will be SIP-only and a few fax machines connected to
grandstream FSX-to-SIP
http://www.voip-forum.com/news.php?p=187
I am battling this same problem and cannot for the life of me figure out
how to work around it. The above link looks promising but leads to a
server error page.
Basically here is what I need to do. I have three asterisk boxes. One
takes inbound
This has been fixed for me in r34162
Many thanks.
Julian.
Julian Lyndon-Smith wrote:
FWIW, the bug may have been introduced between r24293 (which works) and
r24566 (which doesn't).
Julian.
Kevin P. Fleming wrote:
- Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
It may be a config
I have been getting occasional reports of dropped calls from the users
of our asterisk system. Is there anything I can monitor in my logs or
in the console to see when a call is dropped? I'd like to see if
these drops coincide with network traffic problems.
Thanks,
Carl
I am showing 3 channels stuck 2 in INVITE and 1 in BYE.
I have tried reloading sip, reloading extensions, hangup, rereading configs,
even changing usernames in sip.conf, rebooting phones and so forth but none
are escaping this. Configs show for sip are as follows:
[232]
username=232
type=friend
Carl Youngblood wrote:
I have been getting occasional reports of dropped calls from the users
of our asterisk system. Is there anything I can monitor in my logs or
in the console to see when a call is dropped? I'd like to see if
these drops coincide with network traffic problems.
Thanks,
Carl
This is driving me nuts.
Why doesn't the DUNDILOOKUP function return the weight of a path to a number?
The CLI 'dundi lookup' command does. What about the mac address and expiry
period? The CLI command returns those, but the DUNDILOOKUP function does not.
Why?
We absolutely need this in order
Thanks!MarkVoice
International714-279-0204 Ext 102www.voiceinternational.com
www.dialogicdealer.comwww.digiumboards.comwww.nmsboards.comwww.quintumhardware.comwww.quintumdirect.com
___
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Can anyone explain to me what this means:
Jun 14 19:46:10 NOTICE[7391]: rtp.c:331 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389). Please turn off on
client if possible. Client IP: 66.175.1.1
When I try to make a call from certain IP phones, I see that message
on
Tzafrir Cohen wrote:
On Wed, Jun 14, 2006 at 01:15:03PM -0700, shadowym wrote:
FreePBX or AAH(aka trixbox) requires 256MB of RAM minimum to run properly.
Just sitting there doing nothing on my test system it is using 170MB.
How exactly do you meassure memory usage?
E.g: on my laptop:
Of course I'm trying to deal with the network problems, but it's nice
to have another method of verifying that everything is working.
Frequently there are people who don't complain, so we don't realize
that their call quality is sub-par.
We are using iax.
It seems like there should be a record
Comfort noise is the sound you hear from the phone to assure the user
that there is still a connection to the other end. It is there to keep
you from hearing no sound through the speaker and thinking you have been
disconnected.
Check your phone's config for comfort noise or silence
Mainly GXP-2000 (with silence suppression off) and Eyebeam (with
Enable microphone noise reduction off)
Thanks,
Daniel
On Jun 14, 2006, at 7:55 PM, Mike Fedyk wrote:
Comfort noise is the sound you hear from the phone to assure the
user that there is still a connection to the other end. It
pbx/pbx_dundi.c in dundifunc_read(). shouldn't be too hard to have it set some variables (i.e. DUNDI_RESULT_n) and add the the weight in a CUTable string. Can't return multiple results in a nice manner with the result from a custom dialplan function... I'm working on some other projects right now,
On Wed, 2006-06-14 at 19:58 -0400, Daniel Salama wrote:
Mainly GXP-2000 (with silence suppression off) and Eyebeam (with
Enable microphone noise reduction off)
its safe to ignore that too, it just means that asterisk doesnt support
a sip feature that your phone does and its telling you hey I
Hi,I found your post onhttp://threebit.net/mail-archive/asterisk-users/msg04580.html
I am having the exact same issue with the Polycom IP601 (SIP version1.6.6.0036) with Asterisk
1.2.7.1.I was wondering if you found any solution to it. I would really appreciateif you could share your
-Original Message-
From: Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 6/10/06 8:47 AM
Subject: [Asterisk-Users] Voicemail records nonsense, but record() works
(??)
Hello,
I have setup an Asterisk 1.2.7.1 system, with a working
You could also look into the official distribution from Digium called Pound Key.
http://www.rpath.org/rbuilder/project/asterisk/
On 6/14/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
On Wed, Jun 14, 2006 at 01:15:03PM -0700, shadowym wrote:
FreePBX or AAH(aka trixbox)
On 6/15/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Wed, 2006-06-14 at 19:58 -0400, Daniel Salama wrote:
Mainly GXP-2000 (with silence suppression off) and Eyebeam (with
Enable microphone noise reduction off)
its safe to ignore that too, it just means that asterisk doesnt support
On Wed, 14 Jun 2006, Douglas Garstang wrote:
Why doesn't the DUNDILOOKUP function return the weight of a path to a number?
The CLI 'dundi lookup' command does. What about the mac address and expiry
period? The CLI command returns those, but the DUNDILOOKUP function does not.
Why?
Correct me
Hello,
We are planning to biuld a 100 lines
PBX based on asterisk.
How do you decide on the system config,
e.i motherboard, cpu , how much ram , etc ?
We will have all 100 phone plugged in. But
we expect about 20 calls at any given moment.
Thanks
Varun
Hi,
is it possible to have one central phonebook and install it on the
phone or using ldap?
Best regards,
Matthias
--
Programming today is a race between software engineers striving to
build bigger and better idiot-proof programs, and the universe trying to
produce bigger and better idiots.
Has anybody else experienced bad echo issues with this SPA941 phone when
calling SIP-SIP to another SPA ATA? When I call remote office phones
that are attached to SPA ATAs, I get very annoying echo. One can sure
blame it on the reflected signal from the phone on the remote end, but
how can
Hi,
There seems to be no solution for call progress on analog lines
and using outgoing spool call files . My wave file starts playing before
the person has answered the phone so the first part of the message is
missed.
Can the backgrounddetect app be used for this. I have tried but
the
Hello Varun,
Every system is different and simply suggesting a botherboard or cpu
just isn't enough...
You have two good options, your first is to do a lot of reading and
research to determine your needs, and the best place to start is here:
Try reducing the gain on the microphone. These phones pick up room
sounds *very* well.
Andres wrote:
Has anybody else experienced bad echo issues with this SPA941 phone
when calling SIP-SIP to another SPA ATA? When I call remote office
phones that are attached to SPA ATAs, I get very
Little or no complaints means everything is working. Are your
extensions IAX softphones or do you use IAX for PSTN connectivity?
What network problems are we talking about? How about taking
initiative and creating a user survey and sending it to everyone? You
can see IAX stats on the CLI,
Jerry Geis wrote:
Hi,
There seems to be no solution for call progress on analog lines
and using outgoing spool call files . My wave file starts playing before
the person has answered the phone so the first part of the message is
missed.
Can the backgrounddetect app be used for this. I have
Does anyone know how many simultaneous calls can a WRTG54GS handle?
Assuming SIP phones are connected locally using G711.u codec and the
WRTG54GS connects to a remote Asterisk server using IAX2 trunking
using GSM codec.
Thanks,
Daniel
___
I really like IBM X305 or X306 for medium load systems.
[EMAIL PROTECTED] wrote:
Hello Varun,
Every system is different and simply suggesting a botherboard or cpu
just isn't enough...
You have two good options, your first is to do a lot of reading and
research to determine your needs, and
Telnet uses TCP, Asterisk's SIP is UDP
Wojtek
- Original Message -
From: John Klimek [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, June 12, 2006 10:42 PM
Subject: [Asterisk-Users] Unable to connect to Asterisk? (simple[?]
question)
I'm trying to setup
Mike Fedyk wrote:
Try reducing the gain on the microphone. These phones pick up room
sounds *very* well.
WellI'm not using the speakerphone. Plus there is no gain setting
at all that I am aware off. Just Handset Volume or Speaker Volume.
Thanks.
--
Andres
What do you mean by smartphone? It is the hardphone, connected via SIP,
IAX2, Analog? What type of trunks, ie. Analog, Digital gateway, TDM card,
etc. You need to provide more information so that we can assist you.
Noc Phibee [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hi,
Who said I was a C programmer?
-Original Message-
From: Terry Wilson [mailto:[EMAIL PROTECTED]
Sent: Wed 6/14/2006 6:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [Asterisk-Users] DUNDi Not Able to
Mimmus wrote:
Hi,
using Sangoma drivers:
- doing 'lsmod', I see:
zaptel ... wanpipe,wctdm24xxp,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2
I'd like to avoid loading all these modules. What have I to do?
- do I need to have 'zaptel' startup script under /etc/init.d ?
Thanks
AFAIK, there is
I really like IBM X305 or X306 for medium load systems.
Second dat. IBM + direct Linux support = Tier 1 goodness
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Michiel van Baak wrote:
On 11:51, Wed 14 Jun 06, Mike wrote:
Hi,
I'm stuck writing a Web GUI because nothing out there is exactly what I
need. I'm not writing something as extensive as what _is_ out there, but
just something that allows users to change where their calls are forwarded
and
I doubt centos requires much more memory than debian.
I run AAH on a Deskpro EN Celery 500 w/ 128 mb RAM for my home PBX, 8
extensions, TDM400, IAX connectivity to PSTN. Had to shut off the fluff,
like mysql etc but it runs fine. No swapping, 12mb free, with httpd +
sendmail running.
Andres wrote:
Mike Fedyk wrote:
Try reducing the gain on the microphone. These phones pick up room
sounds *very* well.
WellI'm not using the speakerphone. Plus there is no gain setting
at all that I am aware off. Just Handset Volume or Speaker Volume.
I'm not talking about the
are there any open source sip softphone (Window OS version )?
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yes. you can use xten http://www.xten.net/index.php?menu=download. free to download.On 6/15/06, Asterisk guy
[EMAIL PROTECTED] wrote:
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Hi!
I have used KIAX and have made calls from KIAX to KIAX and KIAX to DIAX soft phone which u can find on:
http://www.laser.com/dante/diax/diax.html
I didn`t get any MOH.
Can you send me your MOH settings?musiconhold.conf
also can you post me the sip.conf file
which sip phone are you using?
What
I am still getting delay.
I have tried the q option.Did decrease the delay but not that much.
Anyone having any idea why
Regards,
Amna Saleem
On 6/14/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
The problem was fixed in 1.2.0amna saleem wrote: No , actually I am using Asterisk-1.2.9.1
I
Here is the config for one of several boxes we run in similar
environments;
A dell SC1425 1u rackmount with dual Xeon CPUs, 1GB ram, dual 80gb sata
drives (software raid 1), fedora core 4, and a sangoma a104 4 port T1
interface card.
A good choice for business quality SIP phones is the Polycom
Asterisk guy wrote:
are there any open source sip softphone (Window OS version )?
Quite a number, some links:
a. http://www.openwengo.org/
b. http://www.sipfoundry.org/sipXphone/ (The Rolls-Royce, supports Java
phonelet)
c. http://www.sflphone.org/
Leo
Get some hardware, a TDM410b is only $125. Or upgrade to 2.6.13 or
later. Don't compile the kernel unless you know what you are doing.
You might try, Ubuntu 6.06, FC4 with updates or FC5 to see if that makes
a difference.
Also there are patches on mantis for delays in meetme conferences
Mike Fedyk wrote:
Andres wrote:
Mike Fedyk wrote:
Try reducing the gain on the microphone. These phones pick up room
sounds *very* well.
WellI'm not using the speakerphone. Plus there is no gain
setting at all that I am aware off. Just Handset Volume or Speaker
Volume.
I'm not
Hi,Thank you for your response. I have a doubt. May I know what is meant by simultaneous calls? Looking forward for your response.ThanksRegards,Chandra.Zoa [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Friends, I am implementing Asterisk PBX in our office with 180 extensions. In our office, we
Agreed - and I have found the Polycom speakerphone to be quite good as well.
PaulH
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Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
bus: 03 8320 8100
mob: 0434 673 529
Rod Bacon wrote:
I just got my 1st batch of GXV3000's. I can attest that the speakerphone is
every bit as
Does anyone know how many simultaneous calls can a WRTG54GS handle?
Assuming SIP phones are connected locally using G711.u codec and the
WRTG54GS connects to a remote Asterisk server using IAX2 trunking
using GSM codec.
Thanks,
Daniel
You'll have to do a little experimenting, although I
Asterisk guy wrote:
are there any open source sip softphone (Window OS version )?
http://www.voip-info.org/wiki-Open+Source+VOIP+Software
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Daniel,
Does anyone know how many simultaneous calls can a WRTG54GS handle?
Assuming SIP phones are connected locally using G711.u codec and the
WRTG54GS connects to a remote Asterisk server using IAX2 trunking
using GSM codec.
Here are some of my experiences with Asterisk (I think 1.0.7)
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