On Fri, 2004-07-23 at 22:17 +0200, wrote:
Yeah! Like having your dialplan listening in on the bridged call. Add
some helper apps and we can program consultative transfers and much much
more in a channel/device independent way!
Channel/device independent consultative transfers. For me that
Any more information than that? I have a copy here as well but haven't
had time to read through it.
P.S. Yes I know my name is mentioned in the book. No need to flame me
on that fact. I am a regular consumer like anyone. Author felt
inclined to put it in there.
- Original Message -
From:
please do tell what the problem is with the book, all the content with in
the book works and has been tested, by myself and other no the book is not
designed for the more advance users of asterisk but the beginners, which
one could be figure out by the title Asterisk For Small Office Setup
I think this will be coming in kapejod's bri-stuff in the next few days.
Rgds
Tim
Dave Cotton wrote:
On Fri, 2004-07-23 at 22:17 +0200, wrote:
Yeah! Like having your dialplan listening in on the bridged call. Add
some helper apps and we can program consultative transfers and much much
more
Does anyone do any large scale SIP to H323
conversion? How many simultaneous calls can your server handle and on what
hardware? I think I read on the wiki that twenty five would max out most
servers.
i usually demo a system to show that it is real
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 23, 2004 3:35 PM
Subject: [Asterisk-Users] 'Asterisk for Small Office Setup'
please do tell what the problem is with the book, all the content with in
- Original Message -
From:
Steve Totaro
To: [EMAIL PROTECTED]
Sent: Saturday, July 24, 2004 6:31
AM
Subject: [Asterisk-Users] h323 to SIP
Server Load
Does anyone do any large scale SIP to H323
conversion? How many simultaneous calls can your server
I am using ISDN 64k + VOIP (iLBC, G729, GSM codecs only!) + bulk traffic
(FTP, P2P, E-Mail, etc.).
I am using tc, HTB 3.6 + finer tuned wshaper script.
It works pretty well for me.
The callee never misses any VOIP packet from my side.
So I guess HTB + QOS works pretty well, even for VOIP.
I use
Are you sure you have a mailbox for this number ?
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew
Simpson
Sent: 23 July 2004 16:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] hang up when going to voicemail
I have a little menu set up
I tried that but I still get:
-- Executing Answer(SIP/2000-00b8, ) in new stack
-- Executing Echo(SIP/2000-00b8, ) in new stack
== Spawn extension (from-sip, 700, 2) exited non-zero on 'SIP/2000-00b8'
dev*CLI
Just tried from a Cisco 7960 with sip and it works fine:
Hello all,
Anyone know where can I get a complete source that describe all
options available in the configuration files?
I like to know all available options in configuration files with a
description and a correct syntax.
Another think I would like to understand is what´s
I do that. But when I play MusicOnHold the music is played slowly! I don´t
know why... but is how the bitrate is playing with a different number.
-Mensagem original-
De: Chris Foster [mailto:[EMAIL PROTECTED]
Enviada em: sábado, 24 de julho de 2004 01:37
Para: [EMAIL PROTECTED]
Assunto:
On Jul 23, 2004, at 1:22 AM, Andres Junge wrote:
What is a RMA?
Return Merchandise/Materials(something like that) Authorization.
It's a number from the mfr, that when the product arrives with it on
the box, tells them to expect some dead hardware.
rt
Hello all,
Me again! How to use [Br] on indications.conf file?
When I set loadzone = br; defaultzone = br; on /etc/zaptel.conf I
receive an error... Maybe I need to setup it from other file... Anyone can
help?
___
Asterisk-Users
hi there
was wondering if anybody knows this..
have successfully installed shady dial and the agent is now logging in successfully
i've enabled postgres debugging and i found out that no request had been made by the shady dialer to query the database for the numbers...
also i am able to login to
hi there
was wondering if anybody knows this..
have successfully installed shady dial and the agent is now logging in successfully
i've enabled postgres debugging and i found out that no request had been made by the shady dialer to query the database for the numbers...
also i am able to login to
HI all
I have a little question, and since there is a alot of Cisco Gurus
somebody might be able to help me.
I think It is an easy problem.
My PSTN proviver strips the first digit in the callerid on all incoming
calls.
So when the call reaches my Asterisk I am missing a 0 in the CLID
I
I want to clarify how this works now on *.
First, on a legacy PBX, like a merlin legend, I have phones that have
shared call appearances so that my assistant can answer my calls or see
that I am on the phone, or so that I can have one phone on my desk and
one at my conference table.
This means
Hi everybody,,
I need to block incoming collect calls to my Asterisk box but I could not
find out where to do that.
Went to zaptel.h but I did not see any timing which can be applied to
collect calls. Does anybody knows if I can set this up in Asterisk?
I'm using an E100P connected to the PSTN
I'm willing to take it on faith that everything in the book works. However,
that is not the only (or even the most important) criteria for a technical
book. A short list of deficiencies includes:
1. Poor spelling, punctuation, grammer. A flat, hard-to-read table of
contents. These
At the CLI, should the command show channels list the configured devices
that may be used for calls? Or, is that command simply a way of viewing the
channels on a T1/E1 if you have that type of interface? If the latter, is
there an equiv. command to list the single FXO interfaces?
If the answer
Hi All,
I need to replace old analog PBX with Asteriskl and X-Lise SIP
SoftPhones as client phones.
First: I have problems with implementation of PBX functions. I need and
unsuccesfully tried theese functions (took info at
http://voip-info.org/wiki-Asterisk+PBX+functions)
Call Pickup: Supported
Anyone have a user configured auto attendant setup? Something that can
be used without the * admin helping to make changes.
Something where the operator can record the message like 'press 1 for
john, 2 for bill, 3 for jean' and then the operator can enter the
extension that gets dialed when the
The hack below is for OpenH323, not Asterisk. This is not an Asterisk
problem AFAICT. I am posting it here so that any other Asterisk user with a
similar problem might benefit from it. I may or may not post it to an
OpenH323 list, but since both variants of the H.323 channel in Asterisk
use
Steve Totaro wrote:
Does anyone do any large scale SIP to H323 conversion? How many
simultaneous calls can your server handle and on what hardware? I think
I read on the wiki that twenty five would max out most servers.
The wiki is very wrong then.
Jeremy McNamara
Chris Luke wrote:
The chan_h323 in CVS and chan_oh323 both exhibited the same behaviour.
I have setup chan_h323 to talk to CCM without any trouble, after someone
informed me we had to override the External RTP object, which is part of
cvs -head now. I highly doubt the obsolete -stable has it.
Steve Totaro wrote:
Does anyone do any large scale SIP to H323 conversion? How many
simultaneous calls can your server handle and on what hardware? I think
I read on the wiki that twenty five would max out most servers.
The wiki is very wrong then.
Jeremy McNamara
That is what
I dont know about blocking in * but you should be able give the telco a call
and tell them no collect calls.
- Original Message -
From: Osvaldo Mundim Junior [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 24, 2004 10:06 AM
Subject: [Asterisk-Users] Need to block incoming
The recording part is easy.
; exten for recording greetings/menus
exten = 12,1,Wait(2)
exten = 12,2,Record(/var/lib/asterisk/sounds/maingreeting:gsm)
exten = 12,3,Wait(2)
exten = 12,4,Playback(/var/lib/asterisk/sounds/swelcome)
exten = 12,5,Wait(2)
exten = 12,6,Hangup
add authenticate to prevent
All right Steve. I'll ask them..
But if anybody knows that, please post an answer to the list. This is a very
important Asterisk security configuration to avoid people call you without
having to pay the call..
thank you
Oz
From: Steve Totaro [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To:
Jeremy McNamara wrote (on Jul 24):
Chris Luke wrote:
The chan_h323 in CVS and chan_oh323 both exhibited the same behaviour.
I have setup chan_h323 to talk to CCM without any trouble, after someone
informed me we had to override the External RTP object, which is part of
cvs -head now. I
MP3s have to use constant bitrate not variable bit rate. Look in the documentation
for mpg123.
-Original Message-
From: Jozeph Brasil [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 24, 2004 5:30 AM
To: [EMAIL PROTECTED]
Subject: RES: [Asterisk-Users] Play CD!
I do that. But when
Sorry about this, I have been struggling with the basics of my
asterisk config.
I set up two sip peers and two phones. And I set up lots of dial
masks for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get
Title: Do not buy Asterisk for Small Office Setup!
See my previous email but the book is worse than I thought. In addition, to the things I mentioned earlier.
1. The table of contents is not a table of contents. The only chapter heading is Chapter 1. It is impossible to tell what is in
Looks like you missed 's' extension for incoming calls
You need something like this in extensions.conf
exten = s,1,Answer
exten = s,2,Dial(SIP/1001,20,t)
See sample of extensions.conf in asterisk distribution (make samples if
you didn't install samples)
--
Sincerely,
Elman Efendiyev
[EMAIL
On Sat, 24 Jul 2004 20:41:00 +0200 (CEST), Remco Barende
[EMAIL PROTECTED] wrote:
But isn't this due to the 'intended audience' for the book?
I just started setting up asterisk, I can dial the test message now (yea! :)
) but I have still to find out how to write an extensions.conf file.
This
HI ALL;
I have a sip ua (ATA) registered in my asterisk
box.I want my astersik box to route all incoming calls from sip ua(A-Z
prefix)to h323 GW.
what syntax should I use in extension.conf for
routing all prefixes to h323 GW.
Regards
mohammad
You bet it's for sale! Make me an offer!
BTW I have the other Asterisk book on order too - I'm keeping my fingers
crossed it will be a good one. I'd really like to hand out a good book to
each of my customers. Makes my life easier.
And good luck with Asterisk. It's a lot of fun.
-Original
I had already tried that, however my register statement already specifies to
ring out on ext 1001 so the call isn't an unqualified one so should not look
up the s extension
register = 2:[EMAIL PROTECTED]/1001
However I put the s extension statements in and the results were the same
with or
Has anyone had any luck with dynamic agents in queues (addqueuemember) and autologoff? I
have searched, but so far come up empty for a Sip related example.
Sure, you can assign different contexts to different zap channels, but how
does this help? Normally, the telco will send each call on the first
available channel in a given trunk group, sometimes, they will come in on
random channels.
When a call rings in on Zap/1-1, the only way to know what
Troy Settle wrote:
Sure, you can assign different contexts to different zap channels, but how
does this help? Normally, the telco will send each call on the first
available channel in a given trunk group, sometimes, they will come in on
random channels.
When a call rings in on Zap/1-1, the only
mohammad mirzaee wrote:
HI ALL;
I have a sip ua (ATA) registered in my asterisk box.I want my astersik
box to route all incoming calls from sip ua (A-Z prefix) to h323 GW.
what syntax should I use in extension.conf for routing all prefixes to
h323 GW.
Set the context of the sip user to a
I am trying to hook up my Cisco telephones to Asterisk using
a Layer 3 switch and am having difficulties it getting it to work. I realize
this may not be the proper forum for a discussion on VLAN architecture and
configuration so I wont post the question here. I though I had read
all the
Hello all,
Downloaded, compiled and installed Asterisk CVS-04/15/04-17:54:5. Everything
looks fine except I see a pseudo channel in the 'zap show channels'.
*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudodefault
1default
Shahid wrote (on Jun 07):
The result is, I cant use Zap/g1 in extensions, eg this doesnt work anymore:
Where in your channels section do you describe what group the channel is
in?
If it worked before then it probably shouldn't have, since there's no
group=X (X should be 1 in your example) in
Using the latest code from CVS.
Has anyone figured out a way to setup any kind of Group or Broadcasting
of Voicemail messages?
___
Asterisk-Users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
I'm ordering a PRI T1 for use with Asterisk and a Digium Wildcard TE405P. The provider is asking me a number of questions about how I want to configure the line.
Here goes:
1. Dial Tone - No, Yes - Precise, Yes - SCC
2. Framing - SF, ESF
3. Line Coding - AMI, B8ZS
4. Signaling Type - Ground
[EMAIL PROTECTED] wrote:
1. Dial Tone - No, Yes - Precise, Yes - SCC
Not relevant on a PRI.
2. Framing - SF, ESF
ESF
3. Line Coding - AMI, B8ZS
B8ZS
4. Signaling Type - Ground Start, EM, Loop Start w/Ring, Loop Start w/o Ring
The signalling type is PRI, none of the rest of these are relevant.
5.
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