Replacing the sip terminal for Vonage isn't possible. The terminal is locked
and will not allow access by the user to get the user/password info, and the
user/password handshaking is encrypted which prevents it from being spied
upon.
The only way this will work is if you plug the analog
I raised this issue on the Viop form on dslreports a few weeks ago. It seems
that the providers do not want to sell unlimited minute plans with a direct
SIP connection (sans adapter) to Asterisk users for fraud reasons. They are
concerned that a bunch of Asterisk users will share the same
=fxo_ls
context=internallines
;immediate=yes
mailbox=21
channel = 2
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Rodgers
Sent: Saturday, January 31, 2004 12:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Internal Lines Dialing Out
exten = _NXX,103,Hangup
exten = _1NXXNXX,1,Dial(Zap/1/$EXTEN)
exten = _1NXXNXX,2,Goto(102)
exten = _1NXXNXX,102,Congestion
exten = _1NXXNXX,103,Hangup
On Friday 30 January 2004 21:51, Steve Rodgers wrote:
Try replacing these lines:
[always-out-pots]
;as generic as possible
Hi,
I occasionally experience a one way audio problem when using the IAX1 protocol
with a service provider. If I call a DID number in by group provided by the
supplier which is redirected to my box, I only get the audio path back from
my box to the phone I used to dial the DID number.
If I
Looking at the code in chan_zap.c, I only see options for feature group B and
feature group D MF. The 2 stage MF signalling you are asking for isn't
implmented in the latest asterisk source code.
I would suggest you post a feature request detailing your 2 stage dialing
requirement to
Sip phones generate their own dialtone. The ignore pat option is meaningless
with regard to SIP phones. I would check the Qrandstream's dialplan and see if
you can program it to ignore the dialtone after a '9' is pressed. I had to do
something similar for my Sipura SPA-2000.
Steve.
On
I'd like to add a test extension to implement ringback so that I can test a
phone's ringer without having to use another channel in another room. The way
I'd like to implement this is to dial a test extension, get a tone, hang up,
then one second later, have the system call me back at that
Anyone else having timeout problems with IAXtel? Here's the logfile output,
user names, passwords, and destination phone numbers have been changed to
protect the guilty
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1,
IAX/someuser:[EMAIL PROTECTED]/[EMAIL
Strange,
It's back up for me as well...
On Friday 28 November 2003 18:49, Joel Maslak wrote:
On Fri, 28 Nov 2003, Steve Rodgers wrote:
Anyone else having timeout problems with IAXtel? Here's the logfile
output, user names, passwords, and destination phone numbers have been
changed
exten = _0119X,1,Congestion
exten = _011[0-8]X,1,Dial(Somechannel,${EXTEN})
See page 27 of the Asterisk Handbook, version 2 for further details.
Steve.
On Friday 28 November 2003 01:53, Isamar Maia wrote:
Hi Folks,
I already know how to make a simple dialplan to
For some reason, I can't get call pickup to work between Sip phones or between
Sip and Zap phones. All phones are in the same call group and pickup group
(1). The source code was downloaded and built as of today 11/15/03.
Here's what's in sip.conf:
[general]
port=5060
bindaddr=192.168.17.2
192.168.17.2
to 192.168.17.6. If anyone could shed some light on what is going on here
it would be sincerely appreciated.
Steve Rodgers
San Diego, CA
The symptoms are caused by your qualify= lines. Every 60 seconds,
an OPTIONS request is sent from Asterisk to the destination. I
don't
sharing the
same IP address? I don't seem to be seeing any traffic being logged from the
SPA2000 to Asterisk; it all seems to be going from 192.168.17.2 to
192.168.17.6. If anyone could shed some light on what is going on here it
would be sincerely appreciated.
Steve Rodgers
San Diego, CA
I figured out what was going on with the lack of/stuck on stuttered dial
tone. Apparently, there are two voicemail directories being referenced:
/var/spool/asterisk/voicemail/default, and
/var/spool/asterisk/voicemail/local. The sip phones were using
/var/spool/asterisk/voicemail/local to
;
101 = ,Steve Rodgers,[EMAIL PROTECTED]
102 = ,Karen Rodgers,[EMAIL PROTECTED]
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: NO
Use OB Proxy in Dialog: YES
Make Call Without Reg: NO
Ans Call without Reg: NO
Display Name: SIP LINE 1
User ID: 101
Use Auth ID: NO
Supplementary Service Subscriptions all set to YES.
Audio Config:
N/A?
Dial Plan:
(6x.|1xx|9,911|9,[2-9]xx|9,1[2-9]xx[2-9]xx)
Thanks,
Steve Rodgers
You have me convinced. It's a forwarding issue not a threeway calling issue.
Also, if the outgoing lines are configured for kewlstart, as long as the
called parties hang up at the same time, the conference will be torn down.
Steve.
___
, then this
case should not not apply.
Steve Rodgers
San Diego CA
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1, Zap/g1/9www8531212) in new stack
-- Called g1/9www8531212
-- Zap/1-1 answered Zap/3-1
-- Attempting native bridge of Zap/3-1 and Zap/1-1
-- Starting
version checked out at
8:00pst October 30 2003.
Thanks
Steve Rodgers
San Diego, CA
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