Is there a way in asterisk to configure a sip invite timeout ? It seems
to be about 30 seconds right now which is too long. I would like to
have asterisk return congestion if a host does not respond to an invite
within 5 seconds.
___
--Bandwidth
Well.. you have to be a member of www.goiax.com for someone to call the
87820 number that is assigned. Of course it is free, so that isn't a
huge barrier. It's basically like Free World Dialup, although I allow
regular US dids to be assigned for free as an option.
yours,
matthew
Rehan
anyone using a high availability server set up for Asterisk ? I saw IBM
had some kind of solution at VON but was too busy exhibiting to check it
out. :(
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
is to only allow a certain amount of calling per
month, add velocity checking, and somehow put some accountability into
the sign up process to keep the prank callers and multiple account
abusers away.
yours,
Matthew Simpson
GoIAX -- www.goiax.com
TxLink -- www.txlink.net
To address the issue about the Firefox browser, I develop [if you want
to call it that, lol] the sites on Opera so it should work fine in
Firefox. Nothing fancy at all just HTML 4.0 forms, etc. The other
poster hit the nail on the head, it was out of DIDs. I just loaded
another 200 random
Joe Stewart wrote:
On Fri, Sep 23, 2005 at 11:12:24AM -0700, Matthew Simpson wrote:
I launched www.goiax.com last week, which is intended to promote the use
of IAX as a free and open source alternative to products like skype.
There is no charge for the service. Right now I have free
Kevin Scott wrote:
I'm not sure what he/she was sending as the caller ID information, what I
was trying to do, was send a normal 10 digit number as caller ID. Is there
any solution to this? Or anything planned?
There are no plans to allow just any caller ID to be sent. Once US dids
are
Steven wrote:
Can I ask how you are providing calls to us domestic numbers for free?
goiax.com is backed by TxLink [www.txlink.net]. We terminate a lot of
minutes.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users
I launched www.goiax.com last week, which is intended to promote the use
of IAX as a free and open source alternative to products like skype.
There is no charge for the service. Right now I have free outbound to
united states toll-free and us domestic numbers working.
Currently the site
Yeah. It's a brilliant idea because I believe they would probably return
answer supervision to play these custom ring tones therefore creating
more revenue from the incoming calls.
Marko Rakar wrote:
Few weeks back local telco introduced option of custom ring tones. I am
not talking about
Actually, we have thought about that too. The problem is scheduling
so as not to conflict with any of the other shows that our major
sponsors are doing (remember, they help keep the cost of the
conference down).
Within roughly week of AstriCon on one side or ther other we have the
If you are looking for the maximum number of cheap flights from around
the world, and plenty of convention and room space, the answer is Las
Vegas :-)
quoted for truth.
$79 flights from Dallas.
___
--Bandwidth and Colocation sponsored by
So far we've had at least one person indicate that they would not want
to travel to the US at this time. All politics aside, how many out
there feel the same way? I do NOT want to start a polical flame-war,
but I am curious at the number of people who simply won't come to the
States due to
Actually, we're based out of Kansas City, Missouri (NOT Kansas - we
believe in science) so Omaha would be pretty convenient for us --
it's the other 98% of the community which would have four connections.
;-)
Atlanta is hub for Delta and Airtran
Dallas is hub for American
Chicago is
Hello all,
I have set up a free IAX calling platform similar to
FreeWorldDialup/IAXtel. You can access it at http://www.goiax.com/
The website is still very beta but it will allow you to sign up for a
virtual phone number, and you can make outgoing calls to US toll-free
numbers. There is
I will be there.
Matthew Simpson
TxLink-Commpartners
Steven Sokol wrote:
Hi,
I'm taking a straw-poll to see who out there is planning on going to
AstriCon. I would like to hear from both new members of the community
and gurus. What kinds of things would you like to see at an Asterisk
I'm trying to copy the functionality of something like this in
extensions.conf [extension matching on callerid]:
exten = _NXXNXX/7065557230,1,NoCDR
exten = _NXXNXX/7065557230,2,Dial(Zap/g1/${EXTEN})
exten = _NXXNXX/7065557230,3,Busy
in realtime:
| 16 | fb | _NXXNXX/7065557230
answered my own question
from: pbx_realtime.c
The realtime table currently does not support callerid fields.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
I have a stupid question. How do you remove line presentations on a cisco
7960 ? I have 3 line presentations I don't use anymore and I can't figure
out how to remove them.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Is there anyway to destroy a sip channel ? I get hung up channels like this
in sip show channels:
67.153.9.20 2145558260 33ae28f6088 00102/0 unknow
67.153.9.20 2145558260 52be8085005 00102/0 unknow
67.153.9.20 2145558260 4653d937578 00102/0 unknow
67.153.9.20
Hello, I have a dial plan that tries to place a call over several different
outbound gateways, like this:
exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,2,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,3,Dial(SIP/[EMAIL PROTECTED])
exten = _1X.,4,Dial(SIP/[EMAIL PROTECTED])
exten =
was sending with a protocol analyzer...
card was sending out all 1s.
A new TE-405P seems to be working okay.
Matthew Simpson
TxLink Communications
http://www.txlink.net/
+ SIP and IAX origination and termination
+ Unlimited incoming toll-free $20/LATA
+ Texas origination and termination for $0.005/min
Hello all,
I am getting console debug messages about tone detected on channel XX,
disabling echo cancelation on channel XX when using echocancel=yes with a
Digium T1 card.
does this mean that DTMF breaks the echo can? Does Asterisk permanently
disable the echo can or is it for that channel
Is anyone using the Verisign SIP7 SIP -- SS7 service with Asterisk?
Does anyone have a Verisign contact?
yours,
Matthew Simpson
TxLink Communications
www.txlink.net/
IAX and SIP Origination and Termination Services
___
Asterisk-Users mailing list
We are in the process of mating our Lucent softswitch to Level3. I am also
wanting to put some Asterisk equipment in this configuration. Has anyone
done interop testing with Level3 and Asterisk? Any issues ? I am about to
start next week.
___
Hello, I am having trouble with call files. I want my call files to attempt
only 1 time, and never retry. I am trying to bridge two calls together, one
call to my office [9726172877] and the other call to my cell [2022463521] My
call file looks like this:
Channel: IAX2/outgoing/19726172877
I thought I read somewhere on the Wiki that one could give Dial() an
argument that would first dial the extension, but not bridge the connection
until the called party hit the # key. It must have been during one of
those late night coding sessions because now I can't find anything to do
with
I have a grandstream BT-486 in the lab running 1.0.5.11 firmware.
For the past three days I've had no trouble dialing out without hitting #.
I had the setting for using # as dial key to no in the config.
Today the BT wouldn't pass outgoing calls. I turned on # as dial key and it
works now if I
Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99
retail. They have a version with a router for $89.99. We picked the
non-router version up and it seems to be a rebadged Sipura SPA-2000. The
box has a Vonage service package inside as well, but it does work with other
From your experience, could you give us the merits and demerits of
these ATA devices as well as the IAXy.
They are essentially a Sipura SPA-2000. One of my customers uses the Sipura
exclusively for his customers and they work very well. Setup is easy, and
they support the CLASS type features
Does IAX2 properly update call records for transferred calls to another IAX2
server? Or should I still be using notransfer=yes ?
Example:
SERVER1 calls SERVER2 which transfers call to SERVER3
If Call records are pulled from Server2 will that call have proper CDRs?
The Wiki says no.
I'm using Asterisk to read voicemail users out of a SQL database. I am
assigning users real phone numbers as their voicemail box. The problem is
that if I re-assign a phone number (say, 972-245-0001), the new user is
stuck with the old user's greeting and saved messages. What is the best way
to
-list to me as well.
yours,
Matthew Simpson
TxLink Communications www.txlink.net/
Asterisk PSTN Origination and Termination -- Connect your * to the public
telephone network
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman
I am trying to use chan_skinny but when loading the module I get:
[ Booting../usr/lib/asterisk/modules/chan_skinny.so: undefined symbol:
ast_pickup_call
I am using CVS 07/23
I can't get chan_sccp2 to compile, it gives me parse errors, or I'd be using
that. :-/
Are you sure you have a mailbox for this number ?
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew
Simpson
Sent: 23 July 2004 16:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] hang up when going to voicemail
I have a little menu set up
I have a little menu set up where hitting 1, 2, or 3 places the call through
to a cellular phone over IAX. That works. However, if caller hits 4 to go
into voicemail, the system hangs up. Am I doing something wrong in the dial
plan, or is this a CVS change? I had no trouble with this until I
From: Randy Bush [EMAIL PROTECTED]
[214]
disallow=all
allow=ulaw
type=friend
secret=
host=dynamic
nat=no
dtmfmode=rfc2833
canreinvite=no
incominglimit=1
mailbox=214
where is the
context=
to send it to an incoming context?
In the general part I have
Hello list,
I have a Cisco 7960 with SIP Image 7.1. I can make calls outgoing through
Asterisk, but I'm having problems with incoming calls from Asterisk. The
phone is on a public IP address, no NAT, no firewall. The phone is
registered and shows up in sip show peers.
If I place a call to the
or less
than 10 digits. Also, I've thought of a bug already, if your caller ID name
has digits in it, it'll break the regexp. Adjust accordingly
if that is true about your installation.
Yours,
Matthew Simpson
TxLink Communications
IAX/SIP Termination and Origination
Wholesale Dialup Services
I'm having trouble getting an AGI exec command to spawn app_disa. The
script executes properly, but does not spawn DISA. The CLI gives no helpful
clues. Am I doing the exec incorrectly?
I want to have a way to authenticate callers to the extension by Caller
ID... if their caller ID is in my
I have a Linux 2.6.6 box with Hyperthreading with a Digium 4 port T1 board
[TE-405P ?] Intel P4 3.2 w/ HT and the board is an Intel 875 w/ HT
support.
So far no issues. I did have a hard-lock six hours after first booting the
box, but so far it has been up since then [uptime 5 days 16:11], and
In this dialplan, the SIP user agent is a Sipura two line adapter with line
1 as SIP ID 1000 and line 2 as SIP ID 2000. Basically I have this set
up so that 1000 and 2000 are lines in hunting on incoming extension 555.
I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring
Good evening. I just wanted to take a minute and review my experiences with
some of the SIP devices out there on the market. I hope this post will help
newbies or someone considering a certain device. I would appreciate any
other input on either the devices I am reviewing or other devices that
If I turn allow=ulaw on only, asterisk tries to use it
a=rtpmap:0 PCMU/8000
but the ATA says it doesn't have it:
Answering/Requesting with root capability 4
Answering with non-codec capability 0x1(G723)
If I turn allow=alaw on only or with allow=ulaw, asterisk sends it, the ATA
says it has it
think I figured out the binary bit thing, so I am posting to list to
hopefully help someone else out
bits 15-8 are all 0 and are reserved
bit 7:value 0:numeric 8 reserved
bit 6:value 0:numeric 4 reserved
bit 5:value 0:numeric 2 dtmfmethod
bit 4:value
I'm having a horrible experience getting a Cisco ATA-186 to work with *.
I can make calls from the ATA with no problems. However, incoming calls
make the ATA ring once, and then the call is disconnected. I have no
problems with my Sipura 2000 or my Grandstream phones.
I am running 2.16.1 sip
quality,
etc.
I have ulaw, alaw, and GSM codecs enabled.
To use, just send your call via SIP to 67.153.209.214 with the username of
free secret free
yours,
Matthew Simpson
TxLink Communications
972-617-2877
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
as a
media
gateway with Digiums TE405P cards and we appreciate the work that is going
into Asterisk.
Contact [EMAIL PROTECTED] or 972-617-2877
yours,
Matthew Simpson
TxLink Communications
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
I'm having a very strange problem I've been fighting with all day. It's
2:30am, and I'm stuck. I think the problem may lie with one of my SIP
providers, but I'm not sure.
I have two ways to call into my test Grandstream. I can call a PSTN 360
area code number that will forward to my FWD
The number of codecs is overwhelming to me.
What do ya'll consider the best codec for conserving bandwidth? [I realize
at the cost of quality]
Secondly, what do you think the best codec for voice quality is?
Yours,
Matthew
___
Asterisk-Users mailing
50 matches
Mail list logo
