Displaying 20 results from an estimated 1000 matches similar to: "Asterisk billing solution"
2004 Dec 18
5
Q about IAX (and IAXy)
This is somewhat related to my other query on the list regarding NAT
traversal.
I have heard many times that IAX is "NAT-transperant". I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port 5060) with whatever command. The SIP
server respends to your device's "apparent" IP and port (this
2004 Dec 18
1
One-way audio with SIP client only on certain calls
Hello.
I have an * server set up on a public IP. I have SIP clients at three
different locations, all behind NATs. I have all the SIP users set up
this way:
[user1]
type=friend
username=user1
secret=password1
callerid="User 1"<101>
host=dynamic
qualify=yes
context=outgoing
All three SIP clients are configured to use STUN (using
stun.fwdnet.net:3478).
Furthermore, I have
2004 Dec 17
0
Total newbie here looking to do a VoIPconference call?
Thanks for that. I just got rid of packet 8 and went with 100% asterisk
in my house.
But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But
would
like to have an extra FXS laying around just in case..
.o-------------------------------------------------------o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-----Original Message-----
From:
2004 Dec 17
2
Total newbie here looking to do a VoIPconfer ence call?
Come to think of it since the DTA310 uses DNS to find the SIP server, you
could setup a DNS cache and override the DNS entry for what packet8 uses
(proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your
own SIP server? Kind of a hack but it should work as long as it's running
on port 15062. I am very new to this so I don't know if there's a port
standard for SIP
2004 Dec 23
3
error starting asterisk
Just upgraded to the current stable ver. when I start asterisk with
-vvvvvcg I get the following error
[pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined
symbol: pbx_substitute_variables_varshead
Dec 23 19:25:33 WARNING[1633]: loader.c:440 load_modules: Loading module
pbx_loopback.so failed!
Asterisk
2004 Dec 18
2
External Address Books
I'm not sure if this is possible, but I was hoping to find an address book
that runs on Windows XP that will allow me to select a phone number and send
that to my Asterisk. The Asterisk system would make the call and connect
the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out
there that can do that?
Thanks,
Dave
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An HTML
2004 Dec 20
1
Problem using SPA-2000 behind NAT
Hello all,
I have a new Sipura SPA-2000 that I am trying to configure beind a
NAT. The SPA is able to register to the asterisk server without a
problem and the SPA can make calls to other extension that are not
behind a NAT. However, when I try to call the SPA from another
extension, the extension connected to the SPA rings, the user at the
SPA answers, and there is no audio in either
2004 Dec 16
4
Polycom SIP Phones
Could someone please direct me (via personal email) to a provider with
good prices on Polycom Soundpoint IP 500's with POE cables? I need 14
of them.
Thanks,
Adam
________________________________
Adam S. Robins
Executive Vice President & CIO
PHARMACENTRA, LLP
5901B Peachtree Dunwoody Road, Suite 380
Atlanta, GA 30328
Office: 770-395-0088 x34
Fax: 770-395-0989
Mobile:
2005 Jan 22
1
ASTCC: potential billing issue and "fix"
Before I start, I just want to say this is not necessarily a problem
with ASTCC, but may be a problem the way I have set up ASTCC (and
possibly the way others have set it up as well). The issue is that ASTCC
tries to match the pattern *anywhere* in the called number, not
necessarily only at the beginning.
I have set up ASTCC Routes like this:
1800 Tollfree Trunk1 0 0 100
1416 Canada Trunk2 0 0
2005 Jan 25
3
OT: pinout for"standard"telephoneheadsetrequired.?
> Many thanks Julian.
Are you looking for the pinout for a single plug 2.5mm (cellphone)
headset or a dual plug 3.5mm (computer) headset?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeel<at>jafferali.net
2005 Jan 06
3
IAX outgoing redundancy
Hello.
I am having an issue where sometimes the cheapest provider for certain
international destinations is not always reliable in completing calls.
However, there is not problem once the call is made (i.e. no lag or echo
or anything). The way I have it set up right now (for example) for Dar
es Salaam, Tanzania is:
exten => _925522XX.,1,Dial(IAX2/livevoip/011${EXTEN:1})
exten =>
2004 Dec 17
14
Call on hold disconnects...
G'Day All,
How do I fix this:
I receive a call at the extension. Press the hold button. Music on hold
starts. When I place the handset back on the cradle, the call gets hung
up/disconnected. The Phone is A GrandStream Budge Tone 100.
Thanks
2004 Dec 30
1
IAXy issues
Hello.
I picked up a couple of IAXy's for testing. Unfortunately, I read the
negative comments only after I bought 'em :(
Regardless, I provisioned one unit using my local Linux computer. Now,
I'm trying to set it up to provision using the remote * server whenever
it tries to register, but it seems I need to know the "service
identifier" for the specific device. I can't
2005 Jan 08
3
ASTCC questions
Hello.
I have set up ASTCC properly, calling it like this:
DeadAGI(${ACCOUNTCODE},${EXTEN})
It seems to be working correctly, but I have two questions:
- Although the cards' credit seems to be maintained correctly, I cannot
see the call details in astcc-admin. When I try to view information on
the card, it's just blank. Any ideas?
- When does the 2nd, 3rd and 4th trunk get used? I have
2005 Jan 21
2
Can anyone recoment T1/PRI provider in SouthOntario?
> http://www.mixdown.ca/~andrew/dump/threaded_email.png is what
> a mailing list looks like to most people, and you can see why
> replying to a message, erasing its contents and starting an
> entirely new email about a different topic is frowned upon
> (yours is the highlighted message).
I know this is OT, but can you recommend an email program for Windows
that does something like
2004 Dec 17
5
Total newbie here looking to do a VoIP conference call?
I am looking to help out my company find a more budget conscious but
reliable way to hold conference calls between 5+ people. 4x a month we hold
several hour long conference calls during non-business hours. All of the
employees have high speed internet. Currently we dial up an AT&T conf using
regular analog phones.
I don't have a great grasp as to what Asterick is capable of, but my
2005 Jan 17
1
ASTCC single stage + no access number + auth usingsip username and password
> I would like to have all SIP phones to work on prepaid basis
> and without need to dial any access number, instead I would
> like to use the phone as normal dialing only the destination
> number, for example 00464090510.
I use the AccountCode for authentication. This is how, for example:
exten => _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2})
> Once the call is
2005 May 17
1
One * server unavailable when multiple servers connected together
Hello.
I was just brainstorming for a future project and was hoping to get some
creative ideas from the list. If I have multiple * servers at multiple
locations all connected together with a nicely partitioned dialplan (2XX for
office 1, 3XX for office 2, etc.) it's pretty straightforward to link them
all using IAX and allow intra-office transfers.
Further, servers at each location are
2005 Jun 14
4
488 Not Acceptable Here
I have a whole bunch of remote devices connected to my Asterisk box,
including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only
rolled out recently and I am having a problem that is intermittent and
inconsistent.
It happens to some users but not other users on the same ISP. It happens to
users in 2 different countries where the Internet setup (NAT issues) are
completely different. It
2005 Jun 05
1
Accountcode being ignored?
I have a sip.conf entry for a customer's PBX (IP based authentication) that
reads:
[customer]
type=friend
context=customer
host=x.x.x.x
accountcode=10000
disallow=all
allow=g729
When the customer makes a call to my * server, * recognizes the peer
correctly. However, for some reason, the AccountCode is blank. I have a
NoOp(${ACCOUNTCODE}) and the CLI shows:
-- Executing