Displaying 20 results from an estimated 3000 matches similar to: "Multiple gateways for same dial pattern"
2005 Jan 03
2
PSTN to VoIP FXO gateways?
Sure would like to hear experiences using various FXO to VoIP gateways
with *. It seems that any thread that has anything to do with
problematic FXO interfaces goes on forever with speculation about
everything under the sun. Unless there is someone out there with the
engineering experience to build a better one it is a waste of time, let
Digium deal with it. If the TDM400P can ever be made 99.99%
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 10
1
SIP Reorder tones
We have a strange issue here - we have the following setup:
Asterisk CVS-HEAD-12/15/04-07:42:16
40 SIP Cisco 7940 phones, linking to PSTN via EuroiSDN 30 channels.
Often, when someone tries to dial any internal extension or external
number, they get the "Reorder" message. If they try again, they get
another "Reorder" message. If they try a third time, the call gets
2005 Jan 09
2
ASTCC Trunk and Routes Configuration
Dear List members-
I am trying to configure ASTCC (Asterisk calling card application) but
having a hard time to configure it properly. My project deadline is
approaching and couldn't figure out how to make ASTCC functional. Here
are some details what I have done so far.
1) I have installed ASTCC successfully.
2) I can access astcc-admin.cgi script without any problem.
3) I have created
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 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 24
3
Sipura Behind NAT howto
I am trying to get a SPA-3000 to work behind NAT - for the sake of the
exercice.
The SPA is on the local network at the address 192.168.0.125 behind a
NATted linux router.
The machine I am trying to work with is a friend's (let's call it
lolo.dyndns.org) and I've installed Asterisk 1.0.3 on it.
I can see the SPA register but when I try to make an outbound call I get
the message:
2005 Jan 17
4
SIP IOS for cisco 7902G IP Phone
Hi all
I was looking for the SIP IOS of the Cisco IP Phone but i canĀ“t find it in the cisco web page.
I need to now the name os de file or a specific category link where i can download it.
If you can send me the file is beter ;-)
Thanks in advance
Regards
Wert
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2005 Mar 13
5
ASTCC - how to use different brands?
I just downloaded the new astcc and it includes now a new field in the
list of the cards: Brand
Great!
How can I use it in the dialplan?
bye
Ronald
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
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
2006 Mar 08
5
Cisco 7960 SIP - Displaying Time
Is there a way to display the time of the 7960 running firmware 7.4? Im
unable to find any information.
Thanks,
Ben Blakely
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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
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 10
3
Request to schedule in the past?!?!
Hello,
Ever since I started using Asterisk I always get this
error:
Jan 10 15:39:26 NOTICE[4501]: res_musiconhold.c:463
monmp3thread: Request to schedule in the past?!?!
I have a dedicated system system that really runs only
Asterisk:
- Pentium III 500Mhz
- 128MB of RAM
- 10GB of Disk Space
- SuSE v9.2
- MySQL
- Apache (only for use with Asterisk)
- NTP client for clock synch
There is no X
2006 Apr 20
4
Announcement System for a Charity
I'm putting together an Asterisk server for a local charity to use as an
announcement system. I've been thinking about how to write the dialplan to
allow different options for different groups' announcements, as well as
mailboxes for the various groups and the charity's administrators. Of
course, this would also need to include an option for the heads of the
different groups to
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 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 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?