Kevin's entry in sip.conf does not have caller id properly defined
NOTICE: If received in error, please destroy and notify sender. Sender does
not waive confidentiality or privilege, and use is prohibited.
Peter,
I also made it a point to voice my appreciation and recognize the fact
that Stephen is major contributor here. I also acknowledged his
generous explanations. I have also since replied to his reply and
thanked him again as well.
A consultant so I can get a T1 PRI on my wall and use it
See
http://www.wheely-bin.co.uk/asterisk/ check this link - I've
implemented it and it works, at least in the test environment.
John
On Wed, 5 Jan 2005 16:00:56 +, Mike Dent [EMAIL PROTECTED] wrote:
Hi,
Is there some script which can be called from a * extension to
playback the recent
On Jan 5, 2005, at 11:23 AM, Wiley Siler wrote:
snip
To further explain my siutation, I should give you some more background
on my setup. My current setup has an AdTran 616 on the wall breaking
out my 6 analog lines and delivering my data to the office. I have two
TDM400P cards receiving 6
As well as allowing *xx to be dialed in your device dialplan, do you also
have those codes set up in extensions.conf to do TheRightThing(tm)? (ie set
a database flag that then gets checked by your call an extension macro to
see if DND is activated or not?)
Paul
I think you might have to add the line below to [sip.broadvoice.com]:
insecure=very
I know that it's required for other services, and probably with broadvoice
as well.
/Anders
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip =
On Wed, 5 Jan 2005, Eric Bishop wrote:
I will certainly try that. Please also let me know your progress..
Didn't help for me.
I also tried removing one processor with no benefit.
So I've now given up.
Steve
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Hi
On the www.asterisk.org main page it says Music provided by Freeplay
Music with a link - Where is the music in the *config? I cant find
any supplied music - is there any?
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I have this error compiling ASTCC:
[EMAIL PROTECTED] astcc]# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
chmod 755 ./astcc-admin.cgi
echo | ./astcc.agi /dev/null
Can't locate DBI.pm in @INC (@INC contains:
/var/lib/asterisk/mohmp3/
__
Dana Olson
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Middleton
Sent: Wednesday, January 05, 2005 2:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Music from Freeplay music included in * ??
Hi
Tim,
Thanks for the reply!
Your expanation is correct. The AdTran delivers the FXS on the wall
and is being converted from digital.
I hope you are correct about the swapout and I will chase this up with
ISP again. Originally, they told me that changing my service required
making changes
Should a watchdog be an internal part of the Asterisk core?
The problem is generic. I.e. any real time process may swamp a machine,
and therefor it is not Asterisk specific.
arve5
This is a problem that can be solved in asterisk, though, with a
watchdog, and/or something more elegant. I've
Hi,
Hmmm they aren't there - I did a cvs checkout -r v1-0_stable asterisk
from the digium web server - Whats the CVS command for a 'head'
install ?
Thanks
On Wed, 5 Jan 2005 14:43:23 -0500, Steven Frazier [EMAIL PROTECTED] wrote:
Hi John,
Yes when you do the cvs head install, look in
Julian,
I'm also following this issue, so I guess you're not alone in the universe,
even more I'm not sure why nobody's following this issue usefull as it
seems.
Anyway we'll probably start working on it soon if this happens I'll let you
know.
What I'm not sure is why this didn't make it to the
You are reading the instructions for the STABLE 1.0 version of asterisk and
are using the CVS version.
Goto the wiki and read the instructions for RealTime.
-Matthew
- Original Message -
From: Muhammad Rizwan Khan [EMAIL PROTECTED]
To: Asterisk-Dev@lists.digium.com
Sent: Wednesday,
Hello everyone,
As far as I can tell, if we try to forward
a voicemail (by going into voicemail and saying that we want to forward
it to another extension) it crashes asterisk.
voicemail.conf does not seem to be where
I should be looking. Any ideas?
I did a 'cvs checkout -r v1-0_stable
Hi folks,
Until now I have used only SIP IAX2 with success and understand
them pretty well. The point is that someone has asked me to configure an *
box for them, the problem is that they want to use H.323. I have already
compiled and tested the chan_oh323 with asterisk and works. The
When you do a checkout, you will get 3 mp3 files that all begin with fpm-
These are the 3 freeplay music files.
-Matthew
- Original Message -
From: John Middleton [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Sent: Wednesday, January 05, 2005 1:47 PM
Subject: Re:
On Wed, 5 Jan 2005, Wiley Siler wrote:
Your expanation is correct. The AdTran delivers the FXS on the wall
and is being converted from digital.
I hope you are correct about the swapout and I will chase this up with
ISP again. Originally, they told me that changing my service required
making
My goal is to have only 1 primary phone number that can seamlessly
pool multiple VoicePulse accounts. Let's say I have 3 accounts with
VoicePulse Connect
212-555-1000 (primary)
212-555-1001
212-555-1002
When I receive inbound calls on 212-555-1000, I want to forward or
roll over the connection
Tim,
Just confirmed with ISP that the NIU connects to the AdTran over HDLC.
Thanks!
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Wednesday, January 05, 2005 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial
Matt Riddell wrote:
Corvin wrote:
Hi!
I would like to use lcdproc and asterisk.
Any hints or links? Maybe someone
has experience in such matter. I am working
on such solution. I've heard of SAPBX.
Thanks for any help.
Hi, I was working with someone on this until my BB Forum fell
On Wed, 2005-01-05 at 12:45 -0700, Wiley Siler wrote:
Tim,
Thanks for the reply!
Your expanation is correct. The AdTran delivers the FXS on the wall
and is being converted from digital.
I hope you are correct about the swapout and I will chase this up with
ISP again. Originally,
I usually do it by finding out the smtp address to the cellualr
subscribers sms address, and send the message to that address. To find
out an email address that ends up in ones sms inbox: send an email
from the phone to any other email address using sms (most american
phones allow you to send
You can configure the gain to be lower on the SPA2000
via the web interface - Ido not remember the exact
location, but you will find it under advanced
settings.
--- CClarke [EMAIL PROTECTED] wrote:
Dear All ~
I have * setup running ok (with two Wildcard
X100P's to PSTN). I also have
two
On Wed, 2005-01-05 at 10:23 -0700, Wiley Siler wrote:
LOL - Thanks for not getting mad about my email. I just felt a little
stung for being uneducated about T1s but we have to learn somewhere!
I completely understand your concerns and will try to comply as best as
I can.
Again, thanks
On Tue, 2005-01-04 at 17:05 +1100, Eric Bishop wrote:
And I thought it was just me going crazy. I have the exact same issue
on a HP-Compaq DL360 G4 server (1U rackmount version). I have tried
everything that has been mentioned here and more. Even replaced the
TE410P card (so know it's not the
In article [EMAIL PROTECTED],
John Middleton [EMAIL PROTECTED] wrote:
Hi,
Hmmm they aren't there - I did a cvs checkout -r v1-0_stable asterisk
from the digium web server - Whats the CVS command for a 'head'
install ?
They should be in /usr/src/asterisk/sounds, but they don't appear to
have
Corvin wrote:
Matt Riddell wrote:
Corvin wrote:
Hi!
I would like to use lcdproc and asterisk.
Any hints or links? Maybe someone
has experience in such matter. I am working
on such solution. I've heard of SAPBX.
Thanks for any help.
Hi, I was working with someone on this until my BB Forum fell
Some commerical SMS gateways can provision a # for routing inbound
messages. An example or 2 would be clickatell and ippipi
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That's a known, yet not feasible work-around over accessing an
SMS-center directly. But the question remains how to accept IMCOMING
messages with *.
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 05, 2005 2:14 PM
To: Asterisk Users Mailing List -
On Wed, 2005-01-05 at 23:49, PHP Mechanic wrote:
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID})
M. Tried that, but it didn't deliver
Good points all. Apologies and thanks again.
I guess I am the master at leaving out pertinent information. We are
locate in Phoenix AZ. I currently have a fully functional phone system
built on * that uses Polycom IP 500s over SIP internally. Lines from
the AdtTran are delivered via two
If anyone in the list has a working version of the chan_oh323.so file
for Fedora Core 2 and Redhat, can he email the same to the list as
attachment. This will reduce the pain for many of the users who are
trying to compile the same from the libraries, which never seemed to
work.
Seshu Kanuri
That sounds like it might just be the ticket Roger.
I like the web page idea too.
Would you be willing to share it please?
Thanks
Mike
On Wed, 05 Jan 2005 11:32:08 -0500, Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Wed, 2005-01-05 at 11:00, Mike Dent wrote:
Hi,
Is there some script which
From: Rich Adamson [mailto:[EMAIL PROTECTED]
implementation. Since you mentioned T-Mobile, I'm assuming
you're in the US.
The phrase voip-based US landline should have given that away as well
:) On a related note, T-Mobile or T-Mobil is the European parent of
T-Mobile US (formerly
On Tue Jan 04, 2005 at 01:27:21PM -0500, Philippe Daoust wrote:
Anybody know anything about this F-1000 phone?
100 hours of battery life, not bad at all...
http://www.utstar.com/Solutions/Document_Library/Handsets/docs/WiFi/F1000DataSheet.pdf
This quotes 48-80 hours standby, so you can
Does any body know an IP phone that has at least 2 line appearances,
POE, is around $150 USD, and works nice with *. I've been looking at
the UIP 200 but it's only a single line phone, and I'm looking for
something that has at least 2.
___
Asterisk-Users
Hi,
I have the following scenario.
I have an Asterisk server running on an internal IP address behind a
firewall, and I have a remote user trying to connect to my Asterisk box
behind his firewall, but he can't seem to get a connection.
I have opened up the port (5060) so that he can connect
SSH runs on port 22, so either that's a typo or you've got something else going
on.
Did you forward port 5060, or just open it on the router? You probably need to
forward it to the Asterisk box's IP.
__
Dana Olson
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi guys,
I've just started playing with Asterisk, and I must say that I'm very
impressed.
I'm now looking at hooking this up to a single phone line, and looking for
cheapish hardware to do so. While doing this, I've stumbled across a
Personal Phone Gateway PCI card at:
To all those who answer 50+% of the questions on this list with '*
cannot do that since it is a pbx and what you talk about is the
functionality that a key system provides'... I pose a question.
What would it take - from any point of view you wish to use - to change
that statement to '*
What type of information would you be looking for from the DID providers?
I know the company that I am with works with one DID provider, but might
be interested in expanding beyond that. I would also recommend/request
that the database have info in it letting people know about the outbound
On Fri, 31 Dec 2004 [EMAIL PROTECTED] wrote:
From configs/queues.conf.sample:
[general]
...
; Persistent Members
;Store each dynamic agent in each queue in the astdb so that
;when asterisk is restarted, each agent will be automatically
;readded into their recorded queues.
Not all providers bind the number to a email address.
I havent set it up, but in terms of sms, if asterisk could send out the
message to a URL, or connect using SMPP then it could be done.
Asterisk --- over http ---url--- url parses number in the GET request
and then fires that request by a
Paul wrote:
Having all sorts of nightmares getting IAX working from voiptalk.org
Originally I did too but it was all my fault. We have been using VoIP Talk
for about 3 months and have no complaints.
Getting outbound IAX (from PBX to PSTN via VoIPTalk) is straight forward the
guide on their web
This is more of a VoicePulse thing than an asterisk thing - you'd need them
to roll over to 5551001 after presenting you with 4 calls on 5551000..
although really, this is kinda silly.. It would be better to talk to them
about upping the limit on the number of simultaneous calls you can receive
I got it, but email it to the list is not a good option.
Who 're interested just email me, I'll send it asap.
But AFAIK, you still need the wrapper.
LTenorio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanuri,
Seshu (Company IT)
Sent: Wednesday,
Does any body know an IP phone that has at least 2 line appearances,
POE, is around $150 USD, and works nice with *. I've been looking at
the UIP 200 but it's only a single line phone, and I'm looking for
something that has at least 2.
This information is on the wiki... www.voip-info.org.
On Tue Jan 04, 2005 at 01:27:21PM -0500, Philippe Daoust wrote:
Anybody know anything about this F-1000 phone?
100 hours of battery life, not bad at all...
The peanut gallery chimed in on this yesterday:
http://slashdot.org/article.pl?sid=05/01/04/1816228tid=193tid=215
I saw the post on the wiki a last month stating the meeting was this
Saturday. Is that confirmed? Still on for 1/8?
Roger
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To
I can't get the debug channel command to work. In each case *
responds with No such channel I've tried:
debug channel 1
debug channel Zap/1
debug channel Zap/1-1
debug channel 25
debug channel Zap/25
debug channel Zap/25-1
etc.
The zap show channels command shows all channels to be
Hi,
First of all you have to configure the externip and localnet parameters
at the sip.conf file. You have to write the external ip address of your
internet connection to the extern ip parameter like
exterip=XXX.YYY.ZZZ.WWW and your local address for
ex.localnet=192.168.1.0/255.255.255.0 after
That's the problem.
You need the chan_oh323.so and the oh323wrapper.
You can try it, but, i guess it i'll not work.
A little help from Michael Manousos at this point i'll be great ;)
Tomorrow i'll try to get it working, but, if i can't,
maybe i'll need to do downgrade asterisk chan_oh323
On Wed, 2005-01-05 at 15:52, Mike Dent wrote:
That sounds like it might just be the ticket Roger.
I like the web page idea too.
Would you be willing to share it please?
I've attached the agi script.
My web site is written in Mason which probably doesn't interest many
folks.
The table I use
On Wed, 2005-01-05 at 16:03, richard wrote:
Hi,
I have the following scenario.
I have an Asterisk server running on an internal IP address behind a
firewall, and I have a remote user trying to connect to my Asterisk box
behind his firewall, but he can't seem to get a connection.
I have
-Original Message-
Does any body know an IP phone that has at least 2 line appearances,
POE, is around $150 USD, and works nice with *. I've been looking at the
UIP 200 but it's only a single line phone, and I'm looking for something
that has at least 2.
---
Yes, Polycom
i have tried to connect my asterisk server to vonage like this:
Sip.conf:
register = 1yournumber:secret@atlas-east.vonage.net:5060
[vonage]
type=friend
username=1yournumber
secret=secret
host=atlas-east.vonage.net
port=5060
allow=all
maxexpirey=15
dtmfmode=inband
fromuser=1yournumber
I can't get the debug channel command to work. In each case *
responds with No such channel I've tried:
debug channel 1
debug channel Zap/1
debug channel Zap/1-1
debug channel 25
debug channel Zap/25
debug channel Zap/25-1
I beleive you have to have the channels in use (as per show
After about 4 or 5 minutes, however, I cannot get incoming
calls. It either just rings or goes busy, and never executes
the dialplan in extensions.conf.
Broadvoice has four servers that may send your * server calls. This was
my sip.conf setup until last week (when I cancelled Broadvoice):
On Wed, 5 Jan 2005, Wiley Siler wrote:
A consultant so I can get a T1 PRI on my wall and use it with my
Asterisk box? LMAO. That is the dumbest thing I have ever heard. I
need a consultant so I can get a T1 with PRI? Please. I am just trying
to better understand how the Digium PRI card
Until now I have used only SIP IAX2 with success and understand
them pretty well. The point is that someone has asked me to configure an *
box for them, the problem is that they want to use H.323. I have already
compiled and tested the chan_oh323 with asterisk and works. The problem is
Hi,
Jan 6 01:43:09 WARNING[12209]: pbx.c:796
pbx_find_extension: No such switch 'Realtime'
What does this message mean ?
Something wrong with the switch statement in my
extensions.conf or maybe is the module net correctly installed ?
Thnx.
I was wondering, does anyone know if it is possible to have a stream of
audio coming from a Microsoft compressed audio stream fed to the caller if
they are placed on hold and if so how might this be done?
Dan - [EMAIL PROTECTED]
___
Asterisk-Users
Hi!
I have benn playing a little with quesues tonight and I found out if there
are at least one member-extension free the announcement with p'the place
in the queue wont be played to the person who called in.
Is this possible to change so the announcement will be played even if
there are free
After emerging some updates this morning asterisk 1.0.3 fails to start
I get the following errors:
..Jan 6 00:39:24 WARNING[28998]: chan_zap.c:765 zt_open: Unable to
specify channel 1: No such device or address
Jan 6 00:39:24 ERROR[28998]: chan_zap.c:6197 mkintf: Unable to open
channel 1: No
Well, I wont say my time is cheap or that I know everything about the
T1s. However, I did manage to build my PBX on *, implement Polycom IP
500 phones pulling configs from the network, and script my own
extensions all with an initially minimal understanding of Linux. I have
dealt with problems as
Hi list,
I am having some difficulty implementing a certain dialplan where the
following
happens. If the first Dial() is not answered, I want to play a small
greeting then
ask the caller to either hold the line (try calling again) or press 1
to leave
voicemail.
exten = s,1,Dial(${BLAH},10,Tt)
Serge Schumacher wrote:
Hi,
Jan 6 01:43:09 WARNING[12209]: pbx.c:796 pbx_find_extension: No such
switch 'Realtime'
What does this message mean ?
Something wrong with the switch statement in my extensions.conf or
maybe is the module net correctly installed ?
Perhaps you might
Well it's clear now that this is not an isolated issue. Has anyone
been in touch with Digium about this issue? I have logged a support
issue with them, but thus far have not received a response. Anyone
had better luck with Digium support and the Compaq/HP G4 server
series?
On Wed, 5 Jan 2005
Hi all,
I've struggled for several days trying to get a Digium TDM04B 4-port
wxfco card working on a Dell 1U PowerEdge 750 machine running
Fedora Core 1. I finally got a call back from Digium who indicated that
there is a fundamental conflict between the card and the PowerEdge
having to do with
Hi, again.
I've spent a week trying to get asterisk to work on FreeBSD unix, with some
success. Everything works until I plug the box into the TELCO line and then
the line goes off-hook and stays that way.
So I bit the bullet and decided to install the application on a fresh linux
install.
On Wed, 5 Jan 2005 15:12:08 -0500, Zeno Lee wrote:
My goal is to have only 1 primary phone number that can seamlessly
pool multiple VoicePulse accounts. Let's say I have 3 accounts with
VoicePulse Connect
212-555-1000 (primary)
212-555-1001
212-555-1002
When I receive inbound calls on
You're right it works, but how about receiving calls, how can you register
so the FXO gateways knows where to send the calls? Or I just setup the FXO
gateway with the IP address of the * box?
Humberto Aicardi
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de
Hi,
Did you update the kernel or modutils?
Maybe try to recompile Zaptel modules...
bye,
Samuel T. Cossette
[EMAIL PROTECTED], 1.418.8o2.784o
Well, that's for me to know and you to find out. Jeffrey, Blue Velvet
After emerging some updates this morning asterisk 1.0.3 fails to start
I get
I am having all sorts of probs. It just won't connect. Anyone got any
example configs I could look at?
Thanks
Paul
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Lane wrote:
Hi, again.
I've spent a week trying to get asterisk to work on FreeBSD unix, with some
success. Everything works until I plug the box into the TELCO line and then
the line goes off-hook and stays that way.
So I bit the bullet and decided to install the application on a fresh linux
Hi All -
I've got a load of Polycom phones, and for the most part, I think
they're great, but one thing that is bugging the heck out of me (and my
users) is the on-hold feature. When you're on a call, and another
one comes in, it doesn't ring the second line appearance on the phone,
even
A good alternative would be to try a free rebuild of RedHat Enterprise
Linux, for example www.taolinux.org. Just use the 32 bit version, the
64 bit version (if you would have the cpu) gives me trouble compiling the
kernel modules.
With 32 bit Tao it runs almost out of the box and works like a
I've struggled for several days trying to get a Digium TDM04B 4-port
wxfco card working on a Dell 1U PowerEdge 750 machine running
Fedora Core 1. I finally got a call back from Digium who indicated that
there is a fundamental conflict between the card and the PowerEdge
having to do with PCI
You need some more perl modules. DBI and DBD-mysql I believe.
Darren Wiebe
[EMAIL PROTECTED]
Rafael J. Risco G.V. wrote:
I have this error compiling ASTCC:
[EMAIL PROTECTED] astcc]# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755
I have these very phones and took me a while to figure this out myself.
The phone considers each line registration to be a line with a second
line. So, call line while someone is on a call and another instance
will appear below. That means you only need one registered instance
for the phones to
After getting zaptel from the CVS server, compiling and installing it I
type:
modprobe zaptel
and all is well. Then I type:
modprobe wctdm
and I get this:
modprobe: Can't locate module wctdm
Any idea why?
I did this yesterday but with the CVS head of Asterisk and I got by this
part without
On Thu, Jan 06, 2005 at 07:25:35AM +1100, Howard Lowndes arranged a set of bits
into the following:
On Wed, 2005-01-05 at 23:49, PHP Mechanic wrote:
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten =
On Wed, 2005-01-05 at 18:14 -0600, Lane wrote:
Hi, again.
I've spent a week trying to get asterisk to work on FreeBSD unix, with some
success. Everything works until I plug the box into the TELCO line and then
the line goes off-hook and stays that way.
So I bit the bullet and decided
On Thu, Jan 06, 2005 at 12:14:12PM +1100, Julien Goodwin wrote:
On Thu, Jan 06, 2005 at 07:25:35AM +1100, Howard Lowndes arranged a set of bits into the following:
Sure was. It was me calling myself from my mobile (cell) phone, and
that definitely has CLID enabled. In AU CLID is enabled by
Hello all,
After reading through the Wiki and archives, I decided to take a stab at
installing * on Solaris 9 SPARC. I checked it out via CVS last night as
well as about an hour ago, and have run into a compile problem that I
can't quite figure out.
After running into some unlisted dependencies,
TDM400's use the wcfxs module to drive both FXO and FXS ports on them.
I have an IBM xSeries 305 (1U P4 2.4ghz 1gb of RAM) server and I just
picked up a TDM04B today, and I am getting the exact same problem.
When I make calls to/from the TDM04B card I get this really really
staticky sound.
At 06:53 PM 1/5/2005 -0600, you wrote:
I've struggled for several days trying to get a Digium TDM04B 4-port
wxfco card working on a Dell 1U PowerEdge 750 machine running
Fedora Core 1. I finally got a call back from Digium who indicated that
there is a fundamental conflict between the card and
On Wed, 5 Jan 2005 19:09:00 -0600, Me wrote:
After getting zaptel from the CVS server, compiling and installing it I
type:
modprobe zaptel
and all is well. Then I type:
modprobe wctdm
and I get this:
modprobe: Can't locate module wctdm
Any idea why?
I did this yesterday but with the CVS
CClarke wrote:
Dear All ~
I have * setup running ok (with two Wildcard X100P's to PSTN). I also have
two analog phones connected into same through a SIPURA 2000. These work fine,
except that when I call out through PSTN try to send DTMF tones to (say) a
remote PBX to dial an extension, the gain
On Wed, 5 Jan 2005 16:50:12 -0700 (MST), Dan Adams
[EMAIL PROTECTED] wrote:
I was wondering, does anyone know if it is possible to have a stream of
audio coming from a Microsoft compressed audio stream fed to the caller if
they are placed on hold and if so how might this be done?
I have not
I get the same error with modprobe wcfxs.
It's weird, yesterday I installed CVS Head and the latest Zaptel and did not
have these problems..
I tried updated Zaptel via CVS then ran make clean; make install. When I
tried modprobe wctdm it still flaked and when I tried to start asterisk it
totally
Also, I wonder if there is some sort of issue with the fact that I compiled
and installed Asterisk before Zaptel?
??
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday,
I dug around and found my newest UpdateXpress cd from IBM and ran it on
this box and updated the BIOS and my problem went away. *shrugs*
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Swan
Sent: Wednesday, January 05, 2005 7:35 PM
To:
Thanks for the suggestion!
I have tuned the SIPURA SPA 2000 gain as suggested and also re-adjusted X100P
gain and echo cancel. That helped a lot with audio quality. But still cannot
send DTMF tones out over PSTN. As well I discovered DTMF transmission from IAX
soft phone out to PSTN is not at all
On Wed, 2005-01-05 at 18:53 -0600, Rich Adamson wrote:
The symptoms of the problem were as follows:
1. issue modprobe zaptel which immediately returns with no feedback
Right, there isn't any output from loading this module.
2. issue modprobe wcfxo which returns
init_module: No
I'm not entirely sure this phone supports sip. Have you tried building
the asterisk extra's and configuring it with skinny?
Erik
On Thu, 6 Jan 2005 00:37:32 -, Paul A Brown [EMAIL PROTECTED] wrote:
I am having all sorts of probs. It just won't connect. Anyone got any
example configs I
On Wed, 5 Jan 2005 Remco BarendeB wrote:
After emerging some updates this morning asterisk 1.0.3 fails to start
I get the following errors:
CUT
Check your file permissions. * recently got pretty picky about them lately
--
Andrew McRory - President/CTO
Linux Systems Engineers, Inc. -
Yes, I believe that this is a problem. Everything I've read says you
compile and install zaptel first...then asterisk. On Monday I rebooted
my server again, the just did a CVS update of zaptel. That was all the
was required.
Michael
On Wed, 5 Jan 2005 20:10:27 -0600, Me wrote:
Also, I wonder if
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