What is SIP jitter buffer how can i test it ???
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Satish Patel
mobile:- +91-9818875535
http://www.linuxbug.org
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[EMAIL PROTECTED] wrote:
Hi.
Using Asterisk 1.4.13 running on Ubuntu 7.04 with Intel CPU:
1) Not being able to build mISDN on Ubuntu using make b410p I have
used mISDN-1_1_7 which seems to work ok. QUESTION: Should I expect
this version of mISDN to work ok with these cards? Or is there a
On Nov 18, 2007 12:46 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Nov 18, 2007 at 10:46:57AM +0800, Mark Quitoriano wrote:
Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5
im having an error message when in running asterisk with the tdm card
in.
here's the
Can someone explain what the facilityenable setting does in zapata.conf
I've read the wiki archive, but it's not even clear what an ISDN
facility is.
Thanks,
MD
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asterisk-users
Hello
Since SIP is a bit of a pain to use with NAT firewalls in the
way between clients and *, I'm considering IAX for soft/hardphones.
One thing though: Does the client have to also use UDP4569 as its
source port when connecting to * on UDP4569, or can the client use any
UDP port
On Sun, 2007-11-18 at 00:44 -0800, satish patel wrote:
What is SIP jitter buffer how can i test it ???
Why don't you just ask Google? Gives tons of answers:
http://www.google.com/search?q=what+is+a+jitter+buffer
The jitter buffer config can be found in sip.conf
Regards,
Patrick
On Sun, Nov 18, 2007 at 06:46:18AM +0200, Tzafrir Cohen wrote:
On Sun, Nov 18, 2007 at 10:46:57AM +0800, Mark Quitoriano wrote:
Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5
im having an error message when in running asterisk with the tdm card
in.
here's the
Callerid(number) ? or callerid(num) ?
- Original Message -
From: Benjamin Jacob [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, November 16, 2007 4:39 AM
Subject: Re: [asterisk-users] Problem with AGI Script
I have created a conference call solution for a client and works fine. The
next challenge is to let the conference dial out the participant instead.
Has anyone done this before or know the function to achieve this? Thanks.
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On Sunday 18 November 2007 09:58:41 didier wrote:
Callerid(number) ? or callerid(num) ?
Neither. It is CALLERID(num) or CALLERID(number). The function name is
cAsE-sEnSiTiVe, though either argument will work. In fact, you can even do
CALLERD(numthispartmakesnosense) and it will still work.
On Nov 18, 2007 10:11 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Nov 18, 2007 at 06:46:18AM +0200, Tzafrir Cohen wrote:
On Sun, Nov 18, 2007 at 10:46:57AM +0800, Mark Quitoriano wrote:
Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5
im having an error message
didier wrote:
Callerid(number) ? or callerid(num) ?
Grasshopper, you will find many answers you seek by looking in
/path/to/src/asterisk-1.4/doc/channelvariables.txt
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Vincent wrote:
Hello
Since SIP is a bit of a pain to use with NAT firewalls in the
way between clients and *, I'm considering IAX for soft/hardphones.
One thing though: Does the client have to also use UDP4569 as its
source port when connecting to * on UDP4569, or can the client use
On Sunday 18 November 2007 10:20:18 broadband Voice wrote:
I have created a conference call solution for a client and works fine. The
next challenge is to let the conference dial out the participant instead.
Has anyone done this before or know the function to achieve this? Thanks.
Please see
On Mon, Nov 19, 2007 at 12:30:01AM +0800, Mark Quitoriano wrote:
On Nov 18, 2007 10:11 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Nov 18, 2007 at 06:46:18AM +0200, Tzafrir Cohen wrote:
On Sun, Nov 18, 2007 at 10:46:57AM +0800, Mark Quitoriano wrote:
Hi i have a tdm2400p and
I looked through /etc/asterisk and could not find the folder sampl.call.
On 11/18/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Sunday 18 November 2007 10:20:18 broadband Voice wrote:
I have created a conference call solution for a client and works fine.
The
next challenge is to let the
You can find it enclosed
sample.call
Description: Binary data
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On Sun, 18 Nov 2007, broadband Voice wrote:
I looked through /etc/asterisk and could not find the folder sampl.call.
On 11/18/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Sunday 18 November 2007 10:20:18 broadband Voice wrote:
I have created a conference call solution for a client and
broadband Voice wrote:
I looked through /etc/asterisk and could not find the folder sampl.call.
That is the Asterisk configuration directory. You are looking for the
Asterisk SOURCE CODE directory. If you installed from a package (.deb,
.rpm, etc) then you will have to contact the packager
On Sun, Nov 18, 2007 at 01:37:00PM -0600, Eric ManxPower Wieling wrote:
broadband Voice wrote:
I looked through /etc/asterisk and could not find the folder sampl.call.
That is the Asterisk configuration directory. You are looking for the
Asterisk SOURCE CODE directory. If you installed
On Sun, 18 Nov 2007 10:49:02 -0600, Eric \ManxPower\ Wieling
[EMAIL PROTECTED] wrote:
The source port should not matter.
Good to know. I'll give ZoIPer/Idefisk a shot then. Thanks.
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Hi all,
I have successfully compiled and installed Asterisk on an Alix board
(AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian
variant).
I'm using it at home for a month.
I wondered how much it could be loaded, so I tested it with pbx-test:
I could place up to 15 simultaneous SIP
I tried re-sending my previous messages, but they are not coming
through. There is definitely some kind of filtering going on with this
list.
I like the Report website-related issues to the webmaster. link here,
which goes nowhere;
http://www.asterisk.org/support/contact
This is also
hmmm... everythings is in there too...
On 11/19/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Nov 19, 2007 at 12:30:01AM +0800, Mark Quitoriano wrote:
On Nov 18, 2007 10:11 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Nov 18, 2007 at 06:46:18AM +0200, Tzafrir Cohen wrote:
On
Yeah, I posted several hours ago and I haven't seen mine either.
-Philip
Jesse Molina wrote:
I tried re-sending my previous messages, but they are not coming
through. There is definitely some kind of filtering going on with this
list.
I like the Report website-related issues to the
I'm using a bunch of SPA942's, and I'm trying to provision them mostly
by DHCP (and what I can't set that way, I try to provision via HTTP
interface into the phone).
I changed the domain in my AstLinux config from astlinux to
redfish-solutions.com, and set
that in my sip.conf file as well:
Mark Quitoriano wrote:
that's the same question i got(regarding question 1). Is it possible
for PCI compatibility issue? i need to check for the motherboard specs
to post later :)
Hopefully someone will have someone smarter to say. This specific ioctl,
if it actually gets to zaptel, should
http://lists.digium.com/mailman/listinfo/
[...]
If you are having trouble using the lists, please contact
[EMAIL PROTECTED]
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On Nov 18, 2007 10:43 PM, Philip Prindeville wrote:
Yeah, I posted several hours ago and I haven't seen mine either.
-Philip
(Footnote: do I need a default context? I'd rather not having one... I'd
rather specify where
my calls go explicitly...)
I just set the context in [general] to be context=INVALID and not have a
context named INVALID.
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On Nov 18, 2007 11:13 PM, Baji Panchumarti wrote:
http://lists.digium.com/mailman/listinfo/
If you are having trouble using the lists, please contact
[EMAIL PROTECTED]
Clicking on the link for asterisk-users takes you to :
On Nov 19, 2007 12:10 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Mark Quitoriano wrote:
that's the same question i got(regarding question 1). Is it possible
for PCI compatibility issue? i need to check for the motherboard specs
to post later :)
Hopefully someone will have someone
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