Hi Chris,
FWIW I have a very crude version of what you are looking for. I've been
using it since last July for testing purposes.
It plays a beep, you enter the sip extension number aaa, then it plays
another beep, you enter the ip address bbbcccdddeee, then it plays another
beep, you enter the
I apologise for sending this to the list, it is not my intention to spam
anyone. Please ignore if you cannot help.
If the asterisk-users list administrator is reading this, could you please
fix the mailman-owner account, or supply me with an alternative contact
address. Here is the error, it has
Message-
From: Pavel Litvinenko [mailto:[EMAIL PROTECTED]
Sent: 04 December 2003 10:01
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] oh323 calling party number
Skuse, Phil wrote:
How do I get asterisk to populate the Calling Party Number field in an
H.323 call?
chan_h323 does
.
-Original Message-
From: Skuse, Phil [mailto:[EMAIL PROTECTED]
Sent: 01 December 2003 17:23
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] How do I get caller's number in oh323 ?
We have an h.323 based IVR platform. When we make a call to it using an
h.323 phone, it can see the callers number (ANI
I have a 2621 working with asterisk. See below:
sip.conf
==
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[cisco] ; Cisco 2621 Router
type=friend
canreinvite=no
We have an h.323 based IVR platform. When we make a call to it using an
h.323 phone, it can see the callers number (ANI), but when we make a call to
it via asterisk, the call goes through OK, but we don't get the number. How
can I make this work?
h323.conf
===
[general]
port = 1720
bindaddr
I have a cisco router connected to the Meridian with an E1 QSIG line. The
router converts the meridian calls to SIP and forwards them to asterisk (and
vice-versa). It works really well, but there are probably cheaper ways for a
small setup - Perhaps put an analogue card in the Asterisk server and
Could you not just add it to your /etc/inittab?
-Original Message-
From: WipeOut . [mailto:[EMAIL PROTECTED]
Sent: 24 September 2003 16:02
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Check and restart script..
Has anyone written a script that can be used as a cron job or similar
What's the secret to getting sound through Xlite? The SIP messages all look
OK to me, but the sound isn't coming through.
It was trying to use GSM, so I searched the archive and tried:
disallow=gsm
allow=ulaw
Now it says that it's using ULAW but I still get no sound in either
direction.
Phil
Yes. They are on the same subnet.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: 09 September 2003 11:12
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Xlite = no sound
On Tue, Sep 09, 2003 at 09:15:32AM +0100, Skuse, Phil wrote:
What's the secret
I have a similar setup to you and get the same message regularly. I don't
think it's the cause of your problem. I did some research on it a while ago:
IIRC the cisco uses codec 13 for silence suppression whereas asterisk
(correctly) uses codec 19. The router can be configured to use 19 also, but
We bought two 100's for $75 each, and IIRC they charged an extra $100 or so
for shipping to the UK (which seemed a little excessive to me - I asked our
finance people to look into it).
-Original Message-
From: Reed Wade [mailto:[EMAIL PROTECTED]
Sent: 31 July 2003 06:08
To: [EMAIL
We have some Pingtel phones in our lab. They work well and are nice to use.
-Original Message-
From: Andy Hester [mailto:[EMAIL PROTECTED]
Sent: 30 July 2003 01:51
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE Pingtel Phones
Hello,
Is anybody else out there using pingtel
Some more unusual ones:
http://www.muppetlabs.com/~breadbox/intercal-man/tonsila.html
-Original Message-
From: Gary Gapinski [mailto:[EMAIL PROTECTED]
Sent: 24 July 2003 14:37
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] the 'pound' and '#' are the same? (OT
Rambling)
On
There's something that I want to set up in our lab for testing purposes, but
I'm not sure how to do it.
I would like to be able to call an asterisk extension, and then enter a SIP
address using DTMF, and then have asterisk make a SIP transfer to that
address.
For example:
If I dial extn
-Original Message-
From: Stephen R. Besch [mailto:[EMAIL PROTECTED]
Sent: 04 June 2003 20:09
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Budgettone 100 phone Configuration
snip
mailbox=100 ;Set to use MWI on phone
Stephen, does your MWI work? It doesn't seem to work for
I have it working.
This is what my extensions.conf looks like
exten = 301,1,Dial,sip/301
That 301 corresponds with the [301] section in sip.conf
-Original Message-
From: Benjamin J. Bawkon [mailto:[EMAIL PROTECTED]
Sent: 26 March 2003 16:41
To: [EMAIL PROTECTED]
Subject:
]
[mailto:[EMAIL PROTECTED] On Behalf Of Skuse, Phil
Sent: Monday, March 24, 2003 4:36 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Other IP Phone Possabilities
Yes, I've been using them. They are compatible with asterisk - although
I am
only playing around in our lab, I haven't tried
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