..
Do you have a specific URL, the only thing I can find is
http://www.gnugk.org/interoperability.html, which doesn't sound
exactly like what you're talking about.
Thanks,
Barney.
--
Barney Sowood [EMAIL PROTECTED]
Tel: +44 (0)845 226 5841
Sowood Co Ltd, 22 Manor Place, Edinburgh, EH3 7DS
Nobody ? :-(
-b
- Original Message -
From:
barney
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, June 08, 2005 11:39
AM
Subject: [Asterisk-Users] performance of
* in several scenarios
Hi,
Is here someone who could
Hi,
Is here someone who could provide meany information from
practical using of * ?
I need to know more about performance. The
main question is:
"How many extensions should i have configuredin and
provided with my * box in several cases":
1. * is usedonly for SIP signalling, no rtp
What are you talking about ? I`m using [EMAIL PROTECTED] with 4 digit extensions over two
months with absolutely no problem...
Just put 4 digits instead of 3 digits in Add an Extension in AMP -
SETUP - EXTENSIONS
-b
- Original Message -
From: Alan Bunch [EMAIL PROTECTED]
To:
Hi,
I`m looking for SIP SoftPhone for debuging some situations in
my SIP VoIP network.
Requirements:
- it mustn`t be registered with any
registrar/proxy/anything
- it must be able to send INVITE msgs without registration, so
it must accept characters @ and .
- it must be able to receive
pridialplan: - called party
prilocaldialplan: - calling party
try:
prilocaldialplan= international
-b
- Original Message -
From: Leon de Rooij [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, May 25, 2005 11:41 AM
Subject: [Asterisk-Users] Possible to send
I have that same problem just now. I`m trying to find some
solution with serveral tests, using IOS v.12.3(8r)T7 on the C2821 box with two
PRI ports.
When i find something, it`ll be posted here, and i`m
awaitingto do it also from your side.
-b
- Original Message -
From:
I tried that, but it is not working for me with [EMAIL PROTECTED] v1.0 :-(
-b
Mark,
Try writing the sip.conf stanza as:
[192.168.44.23]
context=from-pstn
host=192.168.44.23
type=friend
insecure=very
The 'insecure=very' allows any calls from this IP address to match.
Alistair Cunningham,
I think, that In C3600 platform, there is no 0:D port, but D channel is
named as 0:15.
So try port 0/0:15, if you want to use first E1 port in first slot of
router.
Anyway, try to use ? character. Look:
SIP-3640(config-dial-peer)#port ?
2-3 Voice interface slot #
delta(config-dial-peer)#port
Julian
barney wrote:
I think, that In C3600 platform, there is no 0:D port, but D channel is
named as 0:15.
So try port 0/0:15, if you want to use first E1 port in first slot of
router.
Anyway, try to use ? character. Look:
SIP-3640(config-dial-peer)#port ?
2-3
cancel 2
sip-server ipv4:192.168.1.20
!
!
telephony-service
!
!
line con 0
exec-timeout 0 0
password 7 ###
login
line aux 0
line vty 0 4
password 7
login
!
!
end
delta#
Julian.
barney wrote:
Wow :-) . Ok, try to send whole running-config from your router and also
send
Hi.
This is working and OK, but I have other problem now.
When I add prefix before my local extension number ( is my extension
number, and I`m adding 02) i see whole number (02) in CDR
entries. Is there any way to put into CDR only pure extension number
(without prefix) ?
-b
I
Your configuration is OK. Cisco is counting from 0, so Serial 0:15 is 16th
channel (D-channel) of first E1 (if you don`t have serial interfaces
also...).
zaptel/asterisk is counting from 1, so 1-16 is D-channel of first E1
interface.
See archive for thread named Asterisk and Cisco AS5300 or
, you could understand it :-)-b- Original
Message - From: "Anton Krall" [EMAIL PROTECTED]To:
[EMAIL PROTECTED]Sent: Wednesday, May 11, 2005 7:08 PMSubject: RE:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600 Hey
Barney What are the steps necessary to make that work on
Hi,
Don`t you know where can I download some older firmware ? I`m interested for
latest stable one.
Can anybody send me a 1.0.5.23 or anything else, which is good working ?
Currently, I have version 1.0.5.18 and 1.0.5.11.
Thanks,
-b
- Original Message -
From: Michael D Schelin [EMAIL
Just in case you don't know, AS5350 supports SIP *and* H323 after IOS
version
12.3 (maybe a little earlier).
It allows you to use both at the same time, without needing to set it up
for
one system specifically.
Haven't tried it with Asterisk yet though.
I have tried it. I have SIP trunk between
It was also my problem...
Beware of generating ringtone (r, or rt string at the end of the call
command).
-b
- Original Message -
From: Torbjørn Lium [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 10, 2005 12:58 PM
Subject: [Asterisk-Users] BYE from Cisco
Look at [EMAIL PROTECTED]
project. Its better to starting with it...
-b
- Original Message -
From:
Marc
Khayat
To: asterisk-users@lists.digium.com
Sent: Friday, May 06, 2005 3:46 PM
Subject: [Asterisk-Users] Web GUI
Hello
all,
I just installed
Try to use g711 codec in communication between you IP Phone
and *.
-b
- Original Message -
From:
scott
kerschner
To: asterisk-users@lists.digium.com
Sent: Thursday, May 05, 2005 9:23
AM
Subject: [Asterisk-Users] Fritz Card
sound quality
Hi boys and
Hi,
Look at http://www.voip-info.org/wiki-Asterisk+billing
duration: Total time in system, in seconds (integer), from dial to hangup
What are you looking for (from my point of view) is
billsec: Total time call is up, in seconds (integer), from answer to hangup
-b
- Original Message -
Try to use qualify=yes in sip.conf . I`m actually not sure, if
qualify is only testingof IP addressavailability, or it does
something more, but you can try it :-).
-b
- Original Message -
From:
Luca Maccarini
To: asterisk-users@lists.digium.com
Sent: Thursday, May
]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 28, 2005 8:23 PM
Subject: Re: [Asterisk-Users] Prefix to CALLING Number ?
On Thursday 28 April 2005 11:07 am, barney wrote:
Hi there,
I`m trying to add some prefix before my local
Thanks Tony, that is exactly what i was looking for :)
-b
- Original Message -
From: Tony Mountifield [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005 9:41 AM
Subject: [Asterisk-Users] Re: Prefix to CALLING Number ?
In article [EMAIL PROTECTED],
barney
(via prefix application) to the called number (but not calling).
Thanks,
barney
PS: sorry for my poor english
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- Original Message -
From: Andres Paglayan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 28, 2005 6:39 PM
Subject: [Asterisk-Users] BIND VoIP anyone?
Hi List,
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