Hello All.
I am stumped, please help me out..
I have the following setup:
VOIP provider = VOIP GW (asterisk GW1) = VOIP server (asterisk - VS1)
The gateway is there to get around the limitations running on the VOIP
server.
I can call out from and receive calls VS1 no problems at all.
How about using time announments? I list of these
for each country would be great!
Jason
Ronald Wiplinger wrote:
Hi,
--snip--
I would need some testing numbers in different countries. Testing
numbers where a tape is or where a long company announcement is.
Do you know such numbers?
Armin,
Why not use OCN etc and connect directly to their sip server?
If you use OCN .office you can get multiple lines and multiple
numbers...The quality is great.
Jason
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
I think this should have been sent to Chris..
Chris,
Why not use OCN etc and connect directly to their sip server?
If you use OCN .office you can get multiple lines and multiple
numbers...The quality is great.
Jason
___
--Bandwidth and Colocation
Hello all..
In an effort to find a reasonably priced analog VOIP-TA here in Japan, I
came accross one
made by NTT for just over $100 (most are over $400). I have been able to
call out fine, but incoming
always gives me a 404 Not Found.
I don't think the different IPs are related (this is
Hello All,
I recently upgraded my asterisk, and it seems that something has changed
with regard to using domains in sip.conf host settings. Nothing
seems to work unless I use the actual IP now. I have tried
putting entries in hosts, but still no joy.
Is anybody aware of something that needs to
Hi everybody.
I have been searching and trying for an answer, but no luck, so here I go..
Is there anyway to execute multiple commands on a single priority in
extensions.conf?
eg:
exten = X.,1,Dial(SIP/) somefunction(${EXTEN})
I need the dial command to dial internal extensions, and the
Hasn't anybody ever come accross these changes before?
Jason
Jason Frisch wrote:
Can someone help me with upgrading to the lastest version. I am using the
same sip.conf file, but the headers have changed and registration fails.
Has something change in the conf file that would cause
,
ast_inet_ntoa(iabuf, sizeof(iabuf), p-ourip), ourport, p-branch);
}
Jason Frisch wrote:
Can someone help me with upgrading to the lastest version. I am using the
same sip.conf file, but the headers have changed and registration fails.
Has something change in the conf file that would cause
c: application/sdp
l: 129
Max-Forwards: 5
Proxy-Require: privacy
k: privacy,timer
Best Regards,
Jason Frisch
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
Can someone help me with upgrading to the lastest version. I am using the
same sip.conf file, but the headers have changed and registration fails.
Has something change in the conf file that would cause this?
Notice 1.2.5 has no Authoization at all...
Regards,
Jason
Version 1.0.9
Hi All,
I have been having trouble with my asterisk box since last week. It
was going fine until then and I can't remember changing anything..
nothing that I haven't put back anyway.
The issue is with that about half of the calls received or placed,
the outside party cannot hear my voice; I can
Think I have it sorted :-)
For those that have the same trouble, try replacing your NIC.
Jason
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Kohlsmith wrote:
On Monday 12 December 2005 21:55, Jason Frisch wrote:
Think I have it sorted :-)
For those that have the same trouble, try replacing your NIC.
Nonsense. How can *everyone* who have this problem have a faulty NIC at the
same time? My particular debugging has shown
. (its an old one I
got out of my junk heap) I changed it to an Intel
server NIC and it hasn't failed on 20 calls yet. It
was failing on more than half before.
But..I will indeed wait a week ;-)
Jason
Andrew Kohlsmith wrote:
On Monday 12 December 2005 21:55, Jason Frisch wrote
=45e12a1c,response=c862acc59c
3914311b52e1bad7f8f4a5,uri=sip:[EMAIL PROTECTED]
Does anybody know? :-(
I think I need a sip-proxy setting but I cannot find anything written on
how to do this.
Jason Frisch
___
Asterisk-Users mailing list
Asterisk-Users
I am trying to get * to use proxy-auth when dialing out,
to mimc what x-lite does when force proxy is set to
yes. Is there any options that can be set to do this?
This particular sip provider does not support
the username:[EMAIL PROTECTED]/number for
Dialing, so I connect as a peer. But it seems
Hi again,
I don't know if I am asking the wrong questions or just nobody knows,
but I will try
again anyway because I am quickly running out of hair to pull out...
Is there any setting in asterisk that will force proxy-authentication on
every call?
Please help :-(
Jason Frisch
Come on now children. Is this not a place to share knowledge?
Jimmy Smith wrote:
coudnt agree more.. thats exact thing i was saying the other day..
please hold my di..k while i take a leak i don't want to wet my hands.
RTFM, google and test. || Pay
On 7/6/05, Brian West [EMAIL PROTECTED]
'
== Spawn extension (sip, 05013428873, 1) exited non-zero on
'SIP/2201-49dc'
If anyone can spot anything wrong I would greatly appreciate any
assistance.
Regards,
Jason Frisch
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
Ahh...sorry. Got this bit sorted now.
Kevin P. Fleming wrote:
Jason Frisch wrote:
sip.conf
[70501956]
type=peer
useragent=M2 X-Lite
host=p10.taraba.net
username=12345
password=secret
qualify=yes
fromuser=12345
fromdomain=taraba.net
realm=taraba.net
Read the example sip.conf more closely
Hi. I am trying to connect to a SIP provider as a client but
cannot get asterisk to work the same way as the UA.
My Configs:
-
sip.conf
[70501956]
type=friend
host=taraba.net
username=70501956
password=password
insecure=very
fromuser=70501956
Jason Frisch wrote:
Hi. I am trying to connect to a SIP provider as a client but
cannot get asterisk to work the same way as the UA.
My Configs:
-
sip.conf
[70501956]
type=friend
host=taraba.net
username=70501956
password=password
insecure=very
fromuser
Has anyone tried Sipura products such as the 3000 in Japan?
Jason
Steven Critchfield wrote:
On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote:
Sorry for my ignorance, but what is J1? I actually hope to use Softbanks
fiber-based IPtel
service, but I believe they require VoIP TA so I guess
Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
finding hardware that is supported. I tried
Voicetronix but they said that they are too busy
to create a driver.
If anyone has had success with any particular
hardware please let me know. I plan to
will check and see if they offer driver support for J1, I do not know
of an analog solution.
Cory Andrews
Senior Partner
VOIPSupply.com
+
800.398.VOIP X22
[EMAIL PROTECTED]
Jason Frisch wrote:
Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
I asked Softbank and it seems that using SIP etc directly is not an option.
Something to do with theVoIP-TA being used for communications between
the providers call-agent.
Jason
Steven Critchfield wrote:
On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote:
Sorry for my ignorance, but what
Eicon Diva Server BRI = ISDN I think..
JAson
Leo Ann Boon wrote:
Jason Frisch wrote:
Hi all,
I am trying to setup Asterisk here in Japan in
my office. However I am having a hard time
finding hardware that is supported. I tried
Voicetronix but they said that they are too busy
to create a driver
28 matches
Mail list logo
