Displaying 20 results from an estimated 2000 matches similar to: "OT: Creating documentation using a web interface"
2003 Sep 10
9
Free World Dialup (FWD).
Hi,
Is it possible to use asterisk with Free World Dialup (FWD) ?
Did someone manage to make it work? how?
Best,
-P
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2003 Sep 26
2
Creating a SIP gateway for use behind NAT
Hi all,
Here is a graphical diagram of what I am trying to do:
<SIP> <---> <GW/NAT/*> <--IAX--> <*> <--TDM400P--> <Analog Phone>
So I have incoming SIP calls go to the * on the GW, which I then want to
forward over IAX to the second * box behind the NAT GW. If I was to
place a call on the second * box, it should then forward to the * on the
NAT GW
2003 Sep 11
1
How much to charge for Asterisk installations?
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I have a medium sized business that is interested in implementing *
as their PBX system. They currently have a Panasonic system with
Panasonic handsets that they are going to replace Asterisk with, as
the current system is maxed out, and they don't even have voicemail
capabilities.
I have been considering using an Adtran Atlas 550 with FXO and
2007 Mar 25
3
mythbackend dvb cards...need some guidance...
Trying to get a myth backend running in fedora core 6 and xend 3.0.3 but
having some issues with the DVB ATSC card, I have removed it from Dom0 and
presented it to my vm, but I am unable to scan for channels...this is what
lspci looks like for the card:
lspci -v
00:00.0 Network controller: Techsan Electronics Co Ltd B2C2 FlexCopII DVB
chip / Technisat SkyStar2 DVB card (rev 02)
Subsystem:
2010 May 20
10
Which issue is keeping you from updrading to 1.6.2 ?
Hi,
I'm evaluating what could keep me from upgrading production systems to
1.6.2.
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Have you met other issues I should include include in my checklist ?
Regards
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2003 Sep 09
2
Has the "allow=all" function changed in sip.conf?
I had posted earlier asking about a Snom200 communicating with a C7960
and lots of noise in one direction. Turned out the problem was created
by me removing the allow=all statement in sip.conf. Someone had suggested
that statement is no longer needed, and using allow=ulaw, etc, had an issue
where one or more deny's had to be used as well.
By adding allow=ulaw in the sip.conf file, the Snom
2003 Sep 11
1
newbie - sip, pxb, ata, nat
hi all,
I don't know how to setup asterix to work as PBX.
If I want just basic configuration with 2 SIP phones (snom, ata), what
all I have to write in the configuration files, or respectively in the
configuration of ata and snom ?
If there is any good documention available, send me URL too.
All (ata, snom) are behind firewall (nat) and astrix is on the public
IP, but I can move for
2005 Oct 15
7
You ASKED for an Asterisk book, you GOT an Asterisk book!
Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk
Documentation Project, in conjunction with O'Reilly Media are pleased
to announce the official release of Asterisk: The Future of Telephony
on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA.
In the true spirit of Open Source, the authors and O'Reilly Media have
published the book under the open, Creative Commons
2003 Dec 01
3
Re: Asterisk behind NAT << How to do it. (Leif Madsen)
> I'm pretty sure that is incorrect. The inside_net is the ip address of
> the asterisk server, and the inside_mask is the subnet mask. At least
> that is how I have mine setup in my sip.conf, and it works.
>
> inside_mask for the internal mask would make more sense to me as well :)
>
> --
> Leif Madsen <leif@hacklocalhost.com>
> http://www.hacklocalhost.com
2012 Sep 26
5
PLAYIN MUSIC WHILE SEARCHING MYSQL
Dear All,
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play music .
Thanks in advanced.
Regards,
Mehdi
2009 Aug 03
5
Difference between 1.4.x and 1.6.x?
Forgive me if this is a FAQ question but I didnt see anything on the website
of forum spelling out the difference between 1.4.x and 1.6.x
Obviously 1.6.x is in development. Is it stable enough for production use?
What are the new features being implemented in 1.6.x?
Will Cepstral work with 1.6.x?
Thanks,
Mike
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2003 Sep 02
4
DTMF Tones During Call
Hi,
I am receiving calls via a Netjet-S card on asterisk, and I notice that
whenever I am talkimng to someone, if their voice is loud enough, sometimes
asterisk generates a DTMF Tone as they speak. that is played to me. (Caller
doesn't hear it).
Any ideas how to stop this?
2006 Jan 16
5
Dundi Examples
Can someone show me how to set up DUNDi, I will be using it to connect
14 asterisk servers internally. I don't want to use it on the external
world. If anyone has any examples of connecting 2 or 3 (if their is a
difference) machines in a DUNDi co-operation that would be helpful.
Johnathan Falk
Network Administrator
Clinton Community Schools
2003 Sep 30
5
* not logging CDR to MySQL - anyway I can debug this?
Hi all,
I think I've run out of options in terms of what I know about this.
I have created a user called asteriskuser and granted all privileges to
the asteriskcdrdb database. Then I created the table via the
cdr_mysql.txt file. I have edited the cdr_mysql.conf file to reflect
this, and added load => cdr_addon_mysql.so after compiling it from the
latest CVS.
If I check the
2009 Jul 22
3
Inquiry abount Asterisk "extensions.conf"
Dear All
Can you please let us know how we can modify our Asterisk "extensions.conf"
file so it interprets the subscriber dialed digits in one-by-one digit
manner . At its current configuration , it interprets them in an whole
packet . I mean , say the subscriber dials as "665 0000" so we need Asterisk
to send it to the peer switch as 6,6,5,0,0,0,0 but not as one
2005 Feb 08
12
SRV lookups
Hi everyone,
I have a question concerning DNS SRV lookups. The situation is like this:
- one central Asterisk server
- many domains with SRV records, let's say we have bar.com and doe.com
Now the question is: if the SRV lookup is done for foo@bar.com the call is
mapped to foo@myasterisk.mydomain.net. Is that correct?
If so, I have a problem: if somebody calls foo@bar.com, Asterisk
2003 Nov 27
13
Asterisk behind NAT << How to do it.
Thanks to ww and his patch on bug #104, I have successfully implemented
Asterisk behind NAT without using STUN or anything crazy. It's quite
straight forward.
Until this gets tested enough and put into CVS, you will have to patch
your chan_sip.c file to do this. I'm sure within the next few days this
will get put merged into CVS if no one finds any problems.
I tried this on chan_sip.c
2003 Sep 21
2
Incoming phone line rollover / hunt?
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Hi All,
I have a simple question about incoming phone line rollovers. How are
these usually done? Is this done at the phone company usually, or is
this something that Asterisk or channel bank is capable of? I just need
someone to give me a brief explanation how it usually works, and if
someone was implementing an Asterisk system, how they would go
2003 Sep 15
3
SOME QUESTIONES (LOG, MySQL, Extensions)
Hi all.
I have some questions:
1) Is there a way to get a full log of the calls (incoming and outgoing)
2) How is the intregation of Mysql and Asterisk. At witch Aplicattions.
3) And of the Extension
a) I have a Support Call Center. Almost all the time all the
extensions are busy, and some calls at hold. Is there a way that when some
2003 Oct 06
5
Help with questions for initial Asterisk wizard (GUI)
Hey all,
I am in the middle of creating a new user wizard which will generate all
the .conf's the new Asterisk user will require to get themselves up and
running in Asterisk without having to touch a single configuration file.
This is what I have come up with as a rough draft. It is far from
complete, so I'm asking people to submit things that should be added,
changed, removed