Displaying 20 results from an estimated 1000 matches similar to: "Which numbers should be blocked?"
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${EXTEN:1})
exten =>
2004 Dec 30
1
IAXy issues
Hello.
I picked up a couple of IAXy's for testing. Unfortunately, I read the
negative comments only after I bought 'em :(
Regardless, I provisioned one unit using my local Linux computer. Now,
I'm trying to set it up to provision using the remote * server whenever
it tries to register, but it seems I need to know the "service
identifier" for the specific device. I can't
2005 Jan 10
3
Request to schedule in the past?!?!
Hello,
Ever since I started using Asterisk I always get this
error:
Jan 10 15:39:26 NOTICE[4501]: res_musiconhold.c:463
monmp3thread: Request to schedule in the past?!?!
I have a dedicated system system that really runs only
Asterisk:
- Pentium III 500Mhz
- 128MB of RAM
- 10GB of Disk Space
- SuSE v9.2
- MySQL
- Apache (only for use with Asterisk)
- NTP client for clock synch
There is no X
2005 Jan 08
4
Toronto?
Anyone in the Toronto area interested in getting together to share notes
and swap war stories?
--
Jim Van Meggelen
jim@vanmeggelen.ca
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.9 - Release Date: 06/01/2005
2005 Jan 08
3
ASTCC questions
Hello.
I have set up ASTCC properly, calling it like this:
DeadAGI(${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 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 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 05
4
Broadvoice / * re-register issues
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =>
##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234
[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
2004 Dec 28
4
DHCP, the TFTP Server setting and the Cisco 79xx phones
The thing I dislike the most about the 79xx phones is that in DHCP mode,
they expect the DHCP server to tell them their TFTP server address. They
won't let you set it manually. So if I don't have DHCP server that gives
TFTP server info, which is most of the DHCP servers at out there, then the
phone won't be able to download any updates made to the SIP000*.cnf file.
Using dhcpd on
2005 Jan 01
25
Qs about FXO/FXS cards
Hello.
I am going to be putting together my first * system using FXO/FXS
interfaces. All the systems I have set up thus far have been pure VoIP
setups.
The system I need to set up should have 3 FXO interfaces and 1 FXS
interface, as well as several SIP phones. I have noticed people
complaining about Digium's TDM cards - are these isolated incidents or
are these cards unreliable? I intend to
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 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
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 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
2004 Dec 20
7
'I'nvalid extension handling problems, even with workaround
Hello folks,
I'm having trouble configuring Asterisk to play an "invalid extension" message to
anyone dialing an undefined extension.
First I tried using the 'i' pseudo-extension, but it didn't work at all;
searching the wiki I found that page:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension
where it basically says that the 'i'
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
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 06
2
Multiple lines on Cisco 7960
I have been trying to get multiple lines on the 7960 to work for several
days. i have read all the posts I can find and have run multiple "sip
debug" and have gotten no place on this.
Here are the relevant section of the config files:
sip.conf
[scott]
type=friend
host=dynamic
username=scott
secret=scott
context=default
mailbox=6101
callerid=Scott Henderson
[scott1]
type=friend
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
---------------------------------
Do you Yahoo!?
Yahoo! Mail - Find what you need with new enhanced search. Learn more.
2005 Jan 10
3
Multiple gateways for same dial pattern
Hi,
How can I setup Asterisk to place calls if the same dial pattern can be
routed through several PRI gateways. I have one way that I tried:
exten => _9737XXXX,1,Dial(SIP/${EXTEN:1}@172.17.99.5)
exten => _9737XXXX,2,Dial(SIP/${EXTEN:1}@172.17.99.6)
exten => _9737XXXX,3,Dial(SIP/${EXTEN:1}@172.17.99.7)
exten => _9737XXXX,4,Congestion
exten => _9737XXXX,102,Busy