Sure got it working yesterday
In sip.conf add these 2 lines to each user
callgroup=1
pickupgroup=1
and in extensions.conf add
exten = *8,1,PickUp(1)
In short and as I understand
You put each user in pickup group 1 and Picjup(1) tells to pickup group (1)
Remember restart
Prof. Marcelo Kruk wrote:
What I want the users to do is something like pressing *333# and this will
enable divert
[EMAIL PROTECTED] wrote:
On Tue, 12 Oct 2004, Altus Syman wrote:
Good day all
We have a pbx system running sip and sipphone(Bughtone)
My question is.If a user is not at their desk,how do I
FWIW, I used to be in Covad territory with a $400/month 1.1MB SDSL over
a dry pair. Now I'm connecting via SBC/Ameritech, and basic dialtone
service around here is only about $6/month, including all fees taxes.
Even if it were a buck or two more than a dry pair, it might be worth
having the
On Wed, 13 Oct 2004, Ed DeHart wrote:
When you call my system your call is handled by the auto
attendant. It works fine with one little problem. In addition
to being able to dial any extension during the announcement, you
can dial a telephone number. The system will bridge the Zap
Any chance you could include support for placing a call on hold?
And like BKW said, It'd be awesome if we could talk ya into releasing
the source... ;)
-Brian
Brian West wrote:
Anyway we could talk you into releasing the source? I would love to see
wider codec support. And the ability to launch
On Wed, 13 Oct 2004, Chris Travers wrote:
What happens then is that a dropped packet will not cause jitter but
rather a delay in the audio. This is the problem.
A delay in the audio IS jitter.
Steve
___
Asterisk-Users mailing list
[EMAIL
Hi
Is there a way to limit the number of concurrent connections in
chan_sip? I want to allow only one connection for each SIP peer, but I
need to allow dynamic IP addresses.
Is this possible?
roy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] writes:
Use OpenSwan http://www.openswan.org
I use OpenVPN http://openvpn.sourceforge.net/, and am very happy
with it. It's easy to set up, but extremely powerful and flexible.
Unlike Swan, which is Linux only, OpenVPN runs on Linux, the
Joe,
Who cares if people agree with it or not. Just do it if it bothers you.
I doubt many people agreed with an Open Source PBX (a good many dont now)
- Original Message -
From: Joe Greco [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 11:46 AM
Subject:
Hello !
We have 7912G SIP phones with the 1.02.00 firmware.
*Sometimes* when you call someone who is already on the phone, our PBX receives
immediatly a 302 Moved Temporarily SIP message, so that the 2nd caller is
forwarded to the voicemail instead of waiting 20s (Allow Call Waiting is set to
1,
Hello
Is there a way once you're inside a conference call to invite an
external party to join?
Of course I could tell the party what the extension number and password
is, but unfortunately, often people are unable to dial the password
especially if calling from overseas.
I've looked a lot and
Hi Brian,
Any chance you could include support for placing a call on hold?
I work on this too...
In the mean time G.711 and Speex codec support was added to the iaxclient
library,
so they will be available in DIAX soon...:-)
Best regards,
Dan
___
I like it but it always generates errors and closes on my win2k box.
- Original Message -
From: Brian [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, October 14, 2004 3:15 AM
Subject: Re: [Asterisk-Users] Prerelease of DIAX
Good question.
So far in this situation I have just used a conference room with no
password, call the 3rd parties and transfer them to the conference room.
Then join them.
- Original Message -
From: Jean-Yves Avenard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 14,
Same for me. After a few minutes the program crashes.
Any chance of support for ULAW / ALAW which is mandatory for FWD IAX?
Thanks
Peter
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: 14 October 2004 09:33
To: Asterisk Users Mailing List - Non-Commercial
I have 190's and a 200. Very pleased with the phones (headset 'Chinch
connectors' are a little quiet on the 190's but better on the 200,
cetainly loud enough to be usable. Not tried the RJ connector as don't
have any compatible headsets.)
Never experienced a problem with them losing
Hi Sudhir,
I purchased couple of SNOM 190 phones last week. Connected them to the
Asterisk server, and they seemed to work fine. However, after sometime
they seem to lose registration with Asterisk as I can make calls but
cannot receive calls.
We do not have these problems with the snom 190. It is
Good day all
We have a voicetronix openline4 card
If someone calls from the outside and asterisk answers the phone and
diverts it to the users the ringing sound is very fast?
Is there a way you can change intervals?
Please Help
Thanks
Altus
___
could you tell me how you changed the headset volumes ? does that option
also work on snom 200s ?
Joachim
At 02:03 14/10/2004, you wrote:
Hi Sudhir,
I purchased couple of SNOM 190 phones last week. Connected them to the
Asterisk server, and they seemed to work fine. However, after sometime
they
Hi all,
Proof that you learn something new everyday... I found an app (by
accident) that should be standard on most linux distros called pmap. You
can run it by:
pmap `pidof asterisk`
to see how much memory asterisk is consuming. of course you could do it
via top as well, but pmap will also
Hello Asterisk dudes and dudetes;
i got -what seems to me a simple a fair question- question about
transferring a call during a conversation.
Like in the following example;
CAPI takes call; sends it out to SIP/phone1
the operator of phone1 picks up; talks to the caller; and after a
while he
What is the voice quality like on the 500?
I've tried a number of phones and found that the voice quality on the 7960
to be far better than any other make that I have tried. You always know when
the other party in a call has a cisco phone!
- Original Message -
From: Kevin P. Fleming
That should also work for the snom 200. We have an apache server on the
asterisk box to update and configure all our snom phones at a central
place via the 'Settings URL' :
http://asteriskbox/snom_autoconfig/{mac}.snom
We can also update the firmware on all phones from one central server.
To
On Thu, 2004-10-14 at 08:36 +0200, Altus Syman wrote:
What I want the users to do is something like pressing *333# and this
will enable divert
And 24 hours ago I posted to this thread something that would enable you
to do just that.
--
Dave Cotton [EMAIL PROTECTED]
On Wed, 13 Oct 2004 20:46:48 -0600, Darren Wiebe [EMAIL PROTECTED] wrote:
I have taken the astcc program which is designed for calling cards and
used it to create a very basic post-pay system. This allows your users
to make multiple calls at one and puts the cost against their card.
Their
Thanks, maybe one day I can be a linux guru but until then, posts like
this help me out.
- Original Message -
From: el Flynn [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, October 14, 2004 5:22 AM
Subject: [Asterisk-Users]
On Thu, 2004-10-14 at 20:09, Dido Sevilla wrote:
On Wed, 13 Oct 2004 20:46:48 -0600, Darren Wiebe [EMAIL PROTECTED] wrote:
I have taken the astcc program which is designed for calling cards and
used it to create a very basic post-pay system. This allows your users
to make multiple calls at
negative balances would imply paying after. it doesn't imply a monthly fee,
just a bill.
- Original Message -
From: Dido Sevilla [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, October 14, 2004 6:09 AM
Subject: Re:
I am seting up to log cdr records via the postgresql module
and have suggestion:
Would it not be nice to have an option in the config file that
lets you specify the table name?
Also, here is the table creation that I used to make the table in
postgresql in case it would help anyone else:
Or
I like it but it always generates errors and closes on my win2k box.
Wait a little bit.
Now I work on the DLL and hope to solve all those crashes...
Thank you for your understanding.
Best regards,
Dan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
Same for me. After a few minutes the program crashes.
Se my previuos post.
Any chance of support for ULAW / ALAW which is mandatory for FWD IAX?
Yup. alaw will be available at the end of the week.
Best regards,
Dan
___
Asterisk-Users mailing list
On Thu, 14 Oct 2004 09:30:51 +0200, Tom Ivar Helbekkmo
[EMAIL PROTECTED] wrote:
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] writes:
Use OpenSwan http://www.openswan.org
I use OpenVPN http://openvpn.sourceforge.net/
As far as I know OpenVPN is not IPsec and thereby non-standard.
Philipp von Klitzing wrote:
How about this pseudo code:
[default]
1,Dial(Sip/1Sip/2)
2,SetVar(foo=x)
3,Goto(international,8500,1)
102,SetVar(foo=x)
103,Goto(international,8500,1)
[international]
8500,1,GotoIf(foo=x THEN voicemail ELSE callotherphones)
Many thanks for the reply, but with
Hi,
Thanks Tim, we try this and works fine at first page but when the page
graphic dense or more than one page, we have an error.
Un saludo,
Pedro
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Robinson
Tim-W10277
Enviado el: miércoles, 13 de octubre de
Hello,
I have teh same problem with:
QuadBRI - * - TDM400 - Modem
Thanks in advance for your help.
Regards,
Pedro
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Carl Sempla
Enviado el: jueves, 23 de septiembre de 2004 3:56
Para: [EMAIL PROTECTED]
Asunto:
Paul Crick [EMAIL PROTECTED] wrote:
I implemented DND using *78 and *79 in conjunction with AstDB and the
dialplan (and some tweaks to the Cisco dialplan.xml) for a client of mine.
We added some checking to the dial macro to see if the DB flag was set or
not, dumping the caller to voicemail if
We do the same thing here. The customer has a say a starting balance of
$100.00 and they are allowed to use it up. If it goes into the negative, then
they are still allowed to place calls. When it comes time to invoice them, we
invoice them for the amount and they pay. When they pay, we
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] writes:
As far as I know OpenVPN is not IPsec and thereby non-standard.
Oh, absolutely. I never claimed it adhered to any standard. It just
does one heck of a great job implementing a VPN solution with lots of
useful features, and
G'Day,
Where might I find documentation on setting up diax, Dante's IAX Phone?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Sent: Thursday, October 14, 2004 7:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Thu, 14 Oct 2004 14:10:23 +0200, Tom Ivar Helbekkmo
[EMAIL PROTECTED] wrote:
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] writes:
As far as I know OpenVPN is not IPsec and thereby non-standard.
Oh, absolutely. I never claimed it adhered to any standard.
But you made it sound
Hi,
i have a basic asterisk server running with an incoming IAX number
routed correctly.
now incoming calls are fine, but when i make outbound calls (IAX) the
receiving person cant hear a thing although i can hear them fine.
The server is sitting behind IPTABLES with both SIP and IAX allowed.
Hi,
Where might I find documentation on setting up diax, Dante's IAX Phone?
For the version 0.9.8 the help is available online at:
http://www.laser.com/dante/diax/diaxhlp.htm
or the CHM version in the 0.9.8c package at:
http://www.laser.com/dante/diax/diax098c.zip
The new help (for 0.9.9) will be
I'm using vtun - I think it's just the best choice for secure tunnels :)
http://vtun.sourceforge.net/
It supports both TCP and UDP connection - you decide what to use
Lubo
-
Appradius Project: RADIUS authentication and accounting support for
Asterisk PBX
http://appradius.minitelecom.org/
[EMAIL PROTECTED] wrote:
On Wed, 13 Oct 2004, Chris Travers wrote:
What happens then is that a dropped packet will not cause jitter but
rather a delay in the audio. This is the problem.
A delay in the audio IS jitter.
Actually isn't it rather that a *variation* in delay from sample to
sample
I have v0.9.9a
And have no idea what to do with it or what it does.
Will v0.9.8 help be of any value to me?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Sent: Thursday, October 14, 2004 8:42 AM
To: Asterisk Users Mailing List -
On Thu, 2004-10-14 at 07:59, Kevin Walsh wrote:
Paul Crick [EMAIL PROTECTED] wrote:
On the Cisco 7960, I prefer to use the built-in DND facility.
Switch it on with the Settings-6-Yes-Save-Back sequence, which is
easy to remember once you've done it a few times. To switch it off,
simply
Danny Froberg wrote:
Hi folks,
Working on getting AlarmReceiver to work on newer SIA protocols and
have some thoughts if anyone has used i.e. t38modem to receive the
short bursts of data that an alarm communicator sends inside Asterisk.
This is a lot simpler than i.e. receiving a fax, so maybe
oi geli wrote:
I am using Uniden UIP200 SIP Phone. While I was
talking in one line, another call came in. I tried the
to put the first call on hold. It would not put the
call on hold. But I could switch between the lines
with Flash.
When there is one call, the hold works fine.
Has anybody else
Title: Message
Hi
all,
I am currently
trying to decide what Operating System is best to go for on a customer
site. Server will only be running Asterisk / MySQL / Apache / PHP but
nothing else.
I have only tested
Asterisk on SLES 8.1 however I do have experience with RedHat 9 as
well.
This is done with contexts. The context that incoming calls from the
telco land in is allowed to dial out. See the MANY examples on the
web, including www.fnords.org/~eric/asterisk for examples of asterisk
configurations
On Wed, 13 Oct 2004 21:08:41 -0400, Ed DeHart wrote:
When you call my
Joseph [EMAIL PROTECTED] writes:
exten = 14,104,Hangup ; and then hangup.
How Shakespearian! ;-)
-tih
--
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
Yes, Fedora works fine. Debian too. (I've used both)
...and others have successfully used other flavours... See the Wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk
Regards,
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
I tried to setup DIAX and connect to Asterisk for the last few versions.
It never actally been able to connect to Asterisk or my Other SIP
Proxies like VoiceMaster.
What does DIAX do actually?
Is there anyone in this list who has connected to Asterisk and made call
for real?
Seshu Kanuri
Ah Qiang wrote:
Has anyone able to implement the 3 way calling feature? I have a 3
way calling plan with my telco. what I need to do is to call a number
,flashhook,call a 2nd number,flashhook , then the two party will be
able to talk to each other. I am doing all this on a FXO card and all the
Yeah I have gone through the list of Operating Systems on the Wiki.
But I was looking for more specific info on What makes a good Linux server.
-Original Message-
From: Scott Stingel [mailto:[EMAIL PROTECTED]
Sent: 14 October 2004 14:22
To: 'Asterisk Users Mailing List - Non-Commercial
On Thu, 2004-10-14 at 23:10, Alex Barnes wrote:
Hi all,
I am currently trying to decide what Operating System is best to go
for on a customer site. Server will only be running Asterisk / MySQL
/ Apache / PHP but nothing else.
I have only tested Asterisk on SLES 8.1 however I do have
Hi. I am evaluating the installation of ~ 20 extensions and 4 telco lines. The customer asked me to compare costs and features of doing it all with voip phones or using analog phones.
I think that the analog route would involve a T1 card, a channelbank (probably adit 600) and 20 new phones
Joseph wrote:
On Thu, 2004-10-14 at 07:59, Kevin Walsh wrote:
Paul Crick [EMAIL PROTECTED] wrote:
On the Cisco 7960, I prefer to use the built-in DND facility.
Switch it on with the Settings-6-Yes-Save-Back sequence, which is
easy to remember once you've done it a few times. To switch it off,
What did he mean by RH9 is no longer available? I just downloaded it
yesterday and burned it to CD. You can get on ftp.redhat.com and download
every version RedHat ever made.
Matthew
- Original Message -
From: Adam Goryachev [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Hi
everyone,
Is anyone using
Xten eyeBeam Video softphone with Asterisk? It supports few types of H.263
codecs for video. I have tried to use it with Asterisk with enabled video
support in sip.conf and allowedh263, but in the moment I click to start
sending video I get this error in
On Thu, 14 Oct 2004, Brian Capouch wrote:
[EMAIL PROTECTED] wrote:
On Wed, 13 Oct 2004, Chris Travers wrote:
What happens then is that a dropped packet will not cause jitter but
rather a delay in the audio. This is the problem.
A delay in the audio IS jitter.
Hi,
- Original Message -
From: Ferguson, Michael [EMAIL PROTECTED]
I have v0.9.9a
And have no idea what to do with it or what it does.
Will v0.9.8 help be of any value to me?
Yes for the aprox. 80% of the functionality.
Only the new features in DIAX 0.9.9 are not yet documented, but you
Title: Message
Alex use the one you like best or know best. I use
RH 9 and Fedora Core 1 due to they work and I know them well. Asterisk will run
on most linux distro with the proper setups.
- Original Message -
From:
Alex Barnes
To: [EMAIL PROTECTED]
Sent: Thursday,
On Thu, 2004-10-14 at 09:36, Eric Wieling wrote:
Joseph wrote:
On Thu, 2004-10-14 at 07:59, Kevin Walsh wrote:
Paul Crick [EMAIL PROTECTED] wrote:
On the Cisco 7960, I prefer to use the built-in DND facility.
Switch it on with the Settings-6-Yes-Save-Back sequence, which is
easy to
How to have an authentification for H323 users.
I define user in oh323 but it's doesn't work.
[3602];OpenPhone
type=h323
defaultip=192.168.0.32
username=3602
callerid=3602
context=communication_local
But i can use any username/alias i want .
___
Hi,
- Original Message -
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
I tried to setup DIAX and connect to Asterisk for the last few versions.
It never actally been able to connect to Asterisk or my Other SIP
Proxies like VoiceMaster.
Have you configured IAX2 accounts on Asterisk
I would like to see those configs as
well.
Gary
- Original Message -
From:
Tim Connolly
To: [EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 6:02
PM
Subject: [Asterisk-Users] Using
Lucent/Ascend TNT as a PSTN Gateway?
About a year ago, a
I agree with the DND setting on the phone. This would be my ideal solution.
The only problem is, if the user forgets to set it and we ring X number of
times, I need to set it programmatically. Do you know of anyway I can do
this?
~c
Message: 9
Date: Thu, 14 Oct 2004 08:36:14 -0500
From: Eric
Joseph wrote:
exten = _XX09,1,GoToIf($[X${RDNIS} != X]?${EXTEN},4)
Could you show me the part that dials the extension in this case?
If RDNIS is set (which is the case if DND is on in the Cisco) then the
RDNIS variable will not be empty therefore we can test to see if the
RDNIS variable is
As far as I know OpenVPN is not IPsec and thereby non-standard.
Oh, absolutely. I never claimed it adhered to any standard.
But you made it sound as if it had cross-platform advantages over
OpenSwan which I believe to have been misleading -- not intentionally
of course -- because
Thanks everyone for the replies.
RH 9 Availability:
According to the RedHat site http://www.redhat.com/software/rhelorfedora/
RedHat 9 is no longer available. If you say you can still download it from RH that's
great, can I assume this means it is totally free to use / install on
With analog phones the different features provided by asterisk are
not natevily supported. For example, on my 10 dollar analog phone, I have
to press the flash button and hear a dial tone to put the caller on hold.
Thats sort of confusing. Most users would think of the Hold button when
putting
You would need a TCP version of IAX to use SSH as I don't think it supports
UDP.
Asterisk does work (tunelling IAX) through Zebedee (an SSH-like TCP UDP
tunnel).
Peter
-Original Message-
From: Tom Neville [mailto:[EMAIL PROTECTED]
Sent: 13 October 2004 16:55
To: Asterisk Users Mailing
I've been doing some work on a queue log analyser for a while now,
getting the basics in place, an example of which you can find at the URL
below. However, just wondering what information people think is most
useful in a log analyser?
At present it includes the following features:
# Time periods
Sudhir,
I have SNOM 200's. I agree the physical design is poor and have the
same problem with finding the cradle and keeping the handset in the
cradle.
As for registration, I have my snom's set to register every minute per
the settings on the phone. In this way it tries to
Joseph wrote:
How do you get * to forward the call to the next dialplan
entry? I always get a busy signal when turning on the DND
option of the 7960.
The customer in question didn't like the hassle of
Settings-6-Yes-Save-Back (despite the visual confirmation of DND status
on the screen). We
On Thu, 2004-10-14 at 10:13, Eric Wieling wrote:
Joseph wrote:
exten = _XX09,1,GoToIf($[X${RDNIS} != X]?${EXTEN},4)
Could you show me the part that dials the extension in this case?
If RDNIS is set (which is the case if DND is on in the Cisco) then the
RDNIS variable will not be empty
Nice work, I've been thinking on doing almost the same thing but how many users can it handle at the same time?
On Thu, 2004-10-14 at 04:39, JR Richardson wrote:
I have had an embedded * server for a while, a one-off project I've been
working on in some spare time. I want to write a white
Henry Devito wrote:
A transformer would not work correctly on a phone line due to talk path
being a DC voltage. 1:1 transformers only work with AC voltage.
Yes it would work - otherwise hybrids, and the transformers in my Adtran
interfaces, wouldn't work. The voice and ringing signals are AC
Hi
I know this question has been asked before and i have
also checked the previous archives..but still in vain
Ok i am receving this message (Unable To retrieve DTMF
tone from INFO message) in asterisk console when i
send the following info message
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP
On Thu, 14 Oct 2004 07:13:04 -0700, Geoff Nordli [EMAIL PROTECTED] wrote:
OpenVPN runs on: Linux, Windows 2000/XP and higher, OpenBSD, FreeBSD,
NetBSD, Mac OS X, and Solaris.
And how many routers and firewalls out there do support OpenVPN? Do
Cisco routers support it?
On the other hand, IPsec
TC, please re-read my statement:
T1 plus channels bank is a solution for USA but not ... other countries
and note the T1.
So T1 channels banks (the ones proved an referenced in the wiki, most of the list messages and all the asterisk documentation) are developed to be used in USA so their
Alex Barnes [EMAIL PROTECTED] wrote:
RH 9 Availability:
According to the RedHat site
http://www.redhat.com/software/rhelorfedora/ RedHat 9 is no longer
available. If
you say you can still download it from RH that's great, can I assume this
means it is totally free to use / install
- Original Message - Message: 1
Like in the following example;
CAPI takes call; sends it out to SIP/phone1
the operator of phone1 picks up; talks to the caller; and after a
while he decides to dispatch/transfer/send out the call the a other
operator that is on phone2.
WHAT do i need to do
Kanuri, Seshu (Company IT) wrote:
I tried to setup DIAX and connect to Asterisk for the last few versions.
It never actally been able to connect to Asterisk or my Other SIP
Proxies like VoiceMaster.
What does DIAX do actually?
Is there anyone in this list who has connected to Asterisk and made
Adam Goryachev wrote:
On Thu, 2004-10-14 at 23:10, Alex Barnes wrote:
snip
I assume that I cannot actually legally install this now for
NEW installs
snip
Oh, you can probably 'legally' install RH9 in another 20 years, I doubt
there was any sort of time limit on installation for it... (not that
Hi
I see on this page
(http://www.voip-info.org/tiki-print.php?page=Asterisk+PBX+functions)
that the 3 way calling is normaly implemented by the SIP phone. I have a
GrandStream Budge Tone 100 and i wanted to use this feature but i don't
find any tips for that. So there is my question : Is this
Joe,
Who cares if people agree with it or not. Just do it if it bothers you.
It's a fair amount of putzing around. Digium wants some sort of release
signed, so I have to run it by our IP lawyer, fax it to Digium, probably
after negotiating some variation, because I know she won't
Steve,
Thanks for sharing your configuration. I will experiment with this conf
and try to connect.
Context - I agree with you on that and the DNS Entry. I find no need for
this to be user
entered as there must be a way pickup the default context. Also other
problems I found
were where the text
Wow! That\'s great! Our company could really benefit from this level
of analysis. Previously we were using Nortel Merridian, and everyone is
used to that level of reporting. Your report(s) are the closest I\'ve
seen in their ability to provide the necessary statistics to manage a
call center.
Steven Critchfield wrote:
On Wed, 2004-10-13 at 13:22 -0700, Richard Lyman wrote:
Steven Critchfield wrote:
On Wed, 2004-10-13 at 11:06 -0700, Steve Edwards wrote:
How much does the break out box cost?
They call it a mini patch panel which sounds a bit more descriptive
than a breakout box.
I
A transformer would not work correctly on a phone line due to talk path
being a DC voltage. 1:1 transformers only work with AC voltage.
Yes it would work - otherwise hybrids, and the transformers in my Adtran
interfaces, wouldn't work. The voice and ringing signals are AC voltages
hi,
our asterisk server is currently connected via 4 isdn trunks to our main
pbx using it as a voip gateway for homeworkers.
currently this is the dial command for outgoing calls
exten = _., 1, Dial,CAPI/141:${EXTEN}
what i like to do, is giving each sip-user a different outgoing msn (the
I wonder if anyone can give me any ideas how to go about achieving
the following?
I want to create a CGI or PHP script that, when invoked from the web,
will make a call to Asterisk on a given extension number, and the audio
that is played to that extension gets streamed to the remote client.
I'm
On 14-Oct-2004, Stephen R. Besch wrote:
My understanding is that if you don't ask for or need support, RedHat
(and the other distro's as well) can be downloaded and installed for free.
Am I wrong about this? Isn't that the whole point of the GPL on linux?
No, it is a violation of the EULA on
Tony Mountifield wrote:
I want to create a CGI or PHP script that, when invoked from the web,
will make a call to Asterisk on a given extension number, and the audio
that is played to that extension gets streamed to the remote client.
A EAGI interface with FFMPEG or Helix (using Windows
Hi,
I have a couple of IpDialog SipToneII phones and although I understood that
they had a choice of 5 ringtones, it turns out that it is Distinctive
Ringing. I contacted IpDialog support and sent me this.
---snip--
The phones you have support 5 different ringtones and 4 call
What about using a vers of fedora or debian, where the code is free and
the upgrades to the system aren't needing conformity to any type of
regulation?
Gus Coutin
Network Engineer
Triple Canopy
[EMAIL PROTECTED]
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
Joseph wrote:
I am seting up to log cdr records via the postgresql module
and have suggestion:
Would it not be nice to have an option in the config file that
lets you specify the table name?
Also, here is the table creation that I used to make the table in
postgresql in case it would help anyone
I realize this is slightly off-topic here, but I know quite a few people
on this list use Sipura products. Has anyone else experienced the same
rebooting problem I'am?
I have about 8 SPA-2000's and about half of them just started rebooting
4-8times/day in the last month or so. (they used to be
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