Trying to install asterfax-1.1-freeb2.i386.rpm, I get following error. How can I get rid of it.
Installing jreInstalling libtiffInstalling ghostscriptInstalling XvfbInstalling openoffice.orgInstalling spandspInstalling spandsp0.0.3
Spandsp did not install correctly.error:
I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone
because it works with Skype (from Linux), but can do SIP, too.
Not necessarily asterisk related, but possibly. My networking situation
might require IAX if I'm running Linux and want to use SIP, I'm not
certain (Skype works
Hi
For add a analog line to my asterisk, i want add a Dgium Fxo card.
but i want know a small information:
The quality of the call are good or not with this type of card ?
Thanks for your returns
___
--Bandwidth and Colocation provided by
It''s a USB Sound card / keypad / display, not a phone. It contols a
softphone on the PC it's plugged into - they say it works with XLite -
the SIP setup will be done in Xlite, not the 'phone'.
Peter
On 05/11/06, Thufir [EMAIL PROTECTED] wrote:
I'm looking at the
Tried inspecting packet dumps with an analyser like Wireshark (ex
Ethereal)? They can prove very useful when troubleshooting issues like
these.
On 11/5/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Seems likes I am the only person in Asterisk world with this problem,
everybody else is fine with
Hi all,
I have been trying to get my asterisk (v1.2.10) to lookup voicemail
config data from my mysql database as opposed to voicemail.conf +
sip.conf for my users. Users register with SER and get passed through
to asterisk when they dial out. I followed the instructions as per
Can anyone help with the following problem please?
1) On a receptionist's phone (Snom 360 latest firmware), a call is answered.
2) While on this call a second call comes to the phone but she does not answer it.
3) The receptionist makes an attended transfer placing the first caller on hold
Yes. You can use the TDM400P. It should do the trick. Make sure to look in
to echo cancelation.
- Original Message -
From: Noc Phibee [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, November 05, 2006 11:51 AM
Subject: [asterisk-users] Asterisk and FXO Digium Card
I experimented with my router, and setup DHCP Lease time to expire every minute. After doing this, my phone started to register every hour. But in the above example, on same phone, one account registers every minute and other account every other minute. This is how it is setup in the phone. But
On Sun, 05 Nov 2006 09:53:52 +, Peter Bowyer wrote:
It''s a USB Sound card / keypad / display, not a phone. It contols a
softphone on the PC it's plugged into - they say it works with XLite -
the SIP setup will be done in Xlite, not the 'phone'.
Peter
On 05/11/06, Thufir [EMAIL
Sorry, just a correction. DHCP lease time setup to expire every hour, not every minute.
On 11/5/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I experimented with my router, and setup DHCP Lease time to expire every minute. After doing this, my phone started to register every hour. But in the above
It seems that xlite doesn't support IAX? Too bad.
While xlite does, apparently, run under linux it's not clear to me whether
or not the a-link device will work with the linux version of xlite.
-Thufir
___
--Bandwidth and Colocation provided by
On Sun, 05 Nov 2006 09:53:52 +, Peter Bowyer wrote:
It''s a USB Sound card / keypad / display, not a phone. It contols a
softphone on the PC it's plugged into - they say it works with XLite -
the SIP setup will be done in Xlite, not the 'phone'.
[...]
Heh, I did miss it. Yes, for
I downloaded a softphone called kiax last night. Its working great. I was
real tired then so I dont remember where I got it from. Hope that helps.
(and its open source as well as they give you the source files for it :) )
- Original Message -
From: Thufir [EMAIL PROTECTED]
To:
Sounds like a bad Internet connection messing with the IAX
jitterbuffer. Try running ping plotter from your location to your
host, and see if it goes 'red'/down.
On 11/3/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Hi everybody,
I finally want to get rid of 1-way audio problem. Please help me
Use the set absolute timeout option on all inbound calls, and then
reset that time to something really high when it connects to a sip
phone.
On 11/5/06, Dovid B [EMAIL PROTECTED] wrote:
Is there any way to run a script and or agi that looks on asterisk and looks
for calls that are connected
I am hoping someone could shed some light and point me in
the right direction? Im attempting to get callerid work between an Avaya
Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From
the Definity side Ive searched endlessly and came with an example which
we
I'm in the US and had bad echo problems with the SPA3K and the latest
firmware. I was under the impression that the echo was due to my long
cable run to the CO ~15000'. Changing the impedance (900 ohms) would
help for a while, but after a few days the echo came back. If I rebooted
the SPA3K
Tom Vile[EMAIL PROTECTED] Wrote on: 11/4/2006 8:45 PM:
He is not talking about Trixbox but FreePBX and his assumption is
correct.
Just load Asterisk and then FreePBX later.
Thanks. I see that 2.2.x is spoken about, but 2.1.3 is the latest that
sourceforge offers.
Is 2.2.x out or still in a
Hello,
Im a student of the
school of engineer of Yverdon Switzerland
and Im working for my project of diploma (VoIP-Asterisk)
Im
wondering if it is possible to have multi languages email with the
voicemail.conf. I wish to set the emailbody/emailsubject relatively to the user
language
Dean
Thanks for responding. I have added more info in your reply. Right now we
do not operate our own PBX or voice mail system. All of the service is
provided by the telco. As a start I was wondering if I could simply put in
asterisk to do just voicemail. I am assuming the telco can configure
On 05/11/06, James Harper [EMAIL PROTECTED] wrote:
Even in this configuration, with my impedance settings set to the
Australian standard of 220+820||120nf, and the PSTN and PAP2 echo
cancellers enabled (or not, and all combinations of) I get local echo as
soon as I pick up the handset (I hear my
Hi Brian,
Uhmmm as it appears you are using a centrex service from your telco
(your comment about not having any pabx)
I need to ask this question..are you sure that under your current
commercial arrangements you are actually allowed to continue to use the
telco as your centrex provider but
On 26/10/06, Guillermo Salas M. [EMAIL PROTECTED] wrote:
What about the bandwidth used for both protocols? Is IAX using less or
more bandwidth than SIP?
I'll give you an actual measured result.
A trunked IAX2 link, carrying 30 simultaneous calls using
variable-bit-rate Speex - we saw 7
On Sun, 5 Nov 2006, Dean Collins wrote:
Date: Sun, 5 Nov 2006 15:21:19 -0500
From: Dean Collins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Newbie questions about Voice mail
Hey all,
I recently got a message from my provider about IAX:
We do not recommend the use of IAX. It is a lossy protocol that is
known to cause crackling, loss of audio and other issues. You can
use IAX if you want, but we will not assist with any issues you may
encounter.
Does anyone else
cp wrote:
I am hoping someone could shed some light and point me in the right
direction? I’m attempting to get callerid work between an Avaya
Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover
PRI. From the Definity side I’ve searched endlessly and came with an
example
On Sun, 05 Nov 2006 15:21:24 +0200, Dovid B wrote:
I downloaded a softphone called kiax last night. Its working great. I was
real tired then so I dont remember where I got it from. Hope that helps.
(and its open source as well as they give you the source files for it :) )
[...]
Matt wrote:
Hi,
I am trying to upgrade my system (running 2.4 kernel) from 1.0.9 to
1.2.6, everything upgraded fine, however asterisk is not seeing any
zap/sip/iax2 channels.
I compiled in this order: libpri/zaptel/asterisk. Zaptel comes up
fine... ztcfg -vv shows all of my channels, however
On Sat, 4 Nov 2006 06:36:06 -0500
Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Trixbox
www.trixbox.org
___
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To UNSUBSCRIBE or update options visit:
Steve Totaro wrote:
cp wrote:
I am hoping someone could shed some light and point me in the right
direction? I’m attempting to get callerid work between an Avaya
Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover
PRI. From the Definity side I’ve searched endlessly and came
Thanks for the reply but I got it worked out a few moments after I sent
the email.
BTW, exten = _*NXXNXX*NXXNXX*,8,NoOP(${CALLERID}) works just
fine. My only problem was the double underscores before setting the
callerID.
Thanks,
Steve Totaro
Kevin Bockman wrote:
Response inline.
Hey gang,
I'm hoping someone can help me out here. I've just noticed that on
two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm
getting the following translation cost for g729:
asterisk*CLI show translation
Server 1: g729 -26252525252426
Try forcing asterisk recalculate those costs:
CLI show translation recalc 20
Julian J. M.
On 11/5/06, Avi Miller [EMAIL PROTECTED] wrote:
Hey gang,
I'm hoping someone can help me out here. I've just noticed that on
two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm
getting
Besides ranch networks and borderware, what other SIP aware firewalls
for the SOHO/medium market exists?
--
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel
The next step should be
1a/ You boss decides You or someone in your team skill up in asterisk
Or
Does the asterisk communitty have a presence at any of the IP telephony
conference?
..Brian
You just missed it check out www.astricon.net it was 2 weeks ago in
Dallas.
(but yes Digium
Hi,
After some more searching I decided to try USING unix ODBC for the
connection. I have both the unixODBC and unixODBC-devel packages on my
fedora box:
[EMAIL PROTECTED] /]# rpm -qa | grep -i unixodbc
unixODBC-2.2.11-7.1
unixODBC-devel-2.2.11-7.1
Here are my odbcinsi.ini and odbc.ini files
There is firmware out there that is made for asterisk users that can be
loaded on to some linksys routers. Dont remember the URL. Do a google search
for linksys hacks.
- Original Message -
From: Erick Perez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
You should run the 4ESS protocol ("a" on the Definity command "change DS1 board#) as the Definity may not send Display Name for the NI-2 protocol (settings "b" or "d")
Remember tomake the Asterisk zapata settings consistent with the Definity.
On the Definity trunk group form Page 1, change
Funny you mention this because I've run into some voice degradation
problems with IAX2 myself recently...
When I have an external call come in on a DiD I frequently have to
send it back out to the PSTN (i.e. to a cell phone). When this
happens I don't want my server in the media path, I want to
On Sunday 05 November 2006 13:54, Erick Perez wrote:
Besides ranch networks and borderware, what other SIP aware firewalls
for the SOHO/medium market exists?
Netfilter's SIP connection-tracking module is ready for prime time, and will
be included in 2.6.18 Linux kernels. Early birds can patch
On 06/11/2006, at 8:53 AM, Julian J. M. wrote:
Try forcing asterisk recalculate those costs:
Ok, that fixed it. Thanks! :)
--
National Manager - Special Projects
Sydney / Melbourne / Canberra / Hobart / London /
2/340 Gore StreetT: +61 (0) 3 9235 5400
Fitzroy, VIC
On 05/11/06, James Harper [EMAIL PROTECTED] wrote:
Even in this configuration, with my impedance settings set to the
Australian standard of 220+820||120nf, and the PSTN and PAP2 echo
cancellers enabled (or not, and all combinations of) I get local
echo as
soon as I pick up the handset (I
Thanks for responding.
Yes I am doing pretty much exactly what you showed. When I try to
dial without answering, I get a busy tone on the DiD (the local Telco
offers to let them notify me when it becomes available). Sometimes I
get half a ring on the destination cell phone b4 receiving the
newbie alert
Glad to see this got fixed so quickly, but could someone give a brief
explanation of what this was? What did Asterisk do? Where does the
cost come into play or get calculated?
Jason
The place where you made your stand never mattered,
only that you were there... and still on your
Please post AsterFax enquiries on the AsterFax help forum at
https://sourceforge.net/forum/forum.php?forum_id=510878. In the mean
time please look for /var/log/asterfax_install.log and post any errors
you see in there.
Zeeshan Zakaria wrote:
Trying to install asterfax-1.1-freeb2.i386.rpm,
On Sat November 4 2006 06:43, Steve Murphy [EMAIL PROTECTED] wrote:
I was encouraged to post this notice on both asterisk-users and
asterisk-dev;
sorry if this is overkill, but it **is** applicable to both communities.
Since the report is fairly large, has a pretty graph, and the whole bit,
Ale,
We had a simiarly problem here, not sure if its the same. The Telco here
has 'ISDN suspension' (think thats the correct term) activated on landlines
here by default. When you phone some one, the person who recieves the call
can put down the reciever and goto another room and pick it up
Hello everyone,
I'm trying to set up a system wherein Trixbox handles the calls but
it's astbill that's billing the calls.
Has anyone set up something similar? How would you go about with this
kind of set up?
Best regards,
Matt
--
Stand before it and there is no beginning.
Follow it and there
Hi,
Where can I find that option?
Thanks
Jesus
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto:
I have the value of xfersound = beep in my features.conf
file but when a call is transferred there is no beep noise. Can someone please
assist?
features.conf
xfersound =
beep
; to indicate an attended transfer is complete
___
On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..
Doug
On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:
Hi,
Where can I find that option?
Thanks
Jesus
-Mensaje original-
De: [EMAIL PROTECTED]
Yes I agree, the SPA3000 can be a bear with echo on the PSTN. I did find
that using older fimware helped some and that the levels - there are 4
settings - FXO/FXS in/out can be juggled to help. I also found out after
adding a Budgetone 200 that I had much less echo problem going through it
and the
I'm still waiting on the 2.0 firmware from Voipsupply. No luck. Don't hold your breath, I would have died a couple of weeks ago.On 11/4/06, Eric Bishop
[EMAIL PROTECTED] wrote:I second that request.
On 11/4/06, Kevin Bockman [EMAIL PROTECTED]
wrote:
Hi,Would anyone be kind enough to send
Besides ranch networks and borderware, what other SIP aware firewalls
for the SOHO/medium market exists?
Anything Cisco
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
FreePBX allows you to specify an extension range
per login so that only extensions within the range are visible to that user.
Cameron
- Original Message -
From:
Curt Shaffer
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Wednesday, November
Googling for a while has turned up evidence that this can be corrected
by a carefully-crafted dialplan for the Sipuras, at least, but the
avaialable documentation is, let's say, a little convoluted.
Try this on Sipura
(*x.*x.)
Seemed to work for me.
Cameron
Hi,
Hi All,
I've just delved into the world of asterisk and I'm having a few dtmf issues.
Internally, amongst sip phones, dtmf is fine.
Externally, if you ring from a GSM mobile, DTMF is fine, however if
you ring from a standard phone, DTMF fails to register.
I am attempting to use a quad
Intertex
Not cheap, licensed per number of users
But seem to work great and have some nifty tools
very confusing picking models though
On Nov 5, 2006, at 3:54 PM, Erick Perez wrote:
Besides ranch networks and borderware, what other SIP aware firewalls
for the SOHO/medium market exists?
--
Carla Schroder wrote:
On Sunday 05 November 2006 13:54, Erick Perez wrote:
Besides ranch networks and borderware, what other SIP aware firewalls
for the SOHO/medium market exists?
Netfilter's SIP connection-tracking module is ready for prime time, and will
be included in 2.6.18 Linux
Matt Florell wrote:
Hello,
We will be having another Tampa Bay Area Asterisk Users Meetup on
Monday, November 6th at 7:30 PM.
Asterisk users from gurus to new users are welcome.
Along with user discussions, we will be talking about Astricon and
Asterisk 1.4 at this meeting.
We will also have
Erick Perez wrote:
Besides ranch networks and borderware, what other SIP aware firewalls
for the SOHO/medium market exists?
Alcatel 610x (discontinued?)/620x. These routers can act as standalone
SIP PBX or outbound proxy to allow phones to register to central registry.
Leo
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