Displaying 20 results from an estimated 2000 matches similar to: "Accountcode being ignored?"
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 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 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 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 =>
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 May 07
5
Good NAT Pnp Hardphone
Hello All,
I am looking for a sip phone that is capable of automatic nat. The
Cisco ata186 for example works fine for natting with iconnecthere, but
as for asterisk, both my 7960 and polycom ip600 require you to set the
nat ip on the tftp.
Does anyone know a good phone (or ata) that can do this automatically?
For example,
I want to give a phone to my brother, who is going to europe. His ICH
2004 Dec 22
1
Asterisk billing solution
Hello.
I am looking for a simple Asterisk billing solution. I expect about
50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all
IAX).
I need something that can handle monthly fees and per call charges
(depending on destination, obviously), and should provide a web
interface for customers and administrators.
Something that can tie in to one of the existing management GUIs
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
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
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
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 May 17
3
Guest
Guys.
What do I need to configure in order to let my Asterisk receive calls from
sip phones, etc not registered with my server on my extension?
For example, let people use their asterisks or sip phones to call
blah111@server.com?
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
2005 May 11
0
Fw: pinout for"standard"telephoneheadsetrequired.?
I saw these adapters on eBay. 2.5mm stereo jack to modular RJ-9 jack. I
think original site is http://www.ciscoheadsetadapter.com
Mike
>Nabeel,
> I am very interested in what you came up with for a 2.5mm to RJ-10
> adapter.
>
>I played with every combination I could think of but the best I was able to
>come up with had a echo of the far end voice back to the far end.
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:
2005 Jan 11
1
Direct SIP calls to *
Hello.
I have my * server set up and working perfectly. I wanted to allows
calls to sip:nabeel@sip.myserver.net. In sip.conf, I have:
[general]
context=default
Also, in extensions.conf, I have:
[default]
exten => myname,1,Goto(internal,nabeel,1)
However, when I make a call using a "Direct Dial IP" account in X-Lite,
I get the following error in *:
Failed to authenticate
2005 Jan 18
0
Out of 5 Grandstream BudgeTone 101 THREE are
Ronald,
Grandstream products have a one year warrantee. If you don't have any luck
with Pulver, contact us and we can probably get your phones exchanged.
Please don't assume that your experience with Grandstream is typical. We
sell a lot of these phones and the overwhelming majority of the purchasers
are very happy with their units. The quality has improved tremendously over
the last
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
2005 May 07
2
Inexpensive FAX and 800 Number retail service
Greetings All,
I have a number of projects in the works at the moment and for one of
them, I need to locate an inexpensive and reliable service that can
provide small-office virtual services:
1. FAX to Email
2. Toll Free number with voicemail boxes for Tech Support, Billing
Inquiries, Customer Service, Abuse Reporting, etc...
I have been looking all over the Internet and there seem to be a LOT