Please help me..
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com
On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:
Hello Experts,
I have pasted my issue in http://pastebin.com/zBGVmdcY
I Cant able to Originate call from SIp trunk..I got this [Jan
Hi,
Give the complete details about the asterisk version, and SIP trunk conf
details
On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:
Please help me..
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com
On Wed, Jan 4, 2012 at 12:51 PM, Jayesh
Hi,
I am using asterisk ver 1.8.8.1.
My SIP trunk conf details are below..
[general]
context=default ; Default context for incoming calls
realm=192.168.1.55
allowguest=yes
realmauth=yes
send_rpid=pai
register = test02:test02@192.168.1.55
[test02]
type=peer
nat=no
Hi checked your debug like.
Did you check that your SIP device ir registered with server ?
if yes then dial below command from CLI
*originate sip/test02 application dial*
On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:
Hi,
I am using asterisk ver 1.8.8.1.
My
Hi virendra,
Dialed same command.. I got below output
ast18*CLI originate sip/test02 application dial
== Using SIP RTP CoS mark 5
[Jan 4 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite:
Failed to authenticate on INVITE to 'Anonymous
On 1/4/2012 4:37 AM, Jayesh Labade wrote:
Please help me..
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com
On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com
mailto:jayesh.lab...@gmail.com wrote:
Hello Experts,
I have
I use realtime. Both information and extensions are stored in DB. It
is just a simple setting of the user with dial plan Dial([EMAIL PROTECTED]).
exten = 9003,1,Dial([EMAIL PROTECTED])
What I found is the following.
9002 --- S1 --- S2
9002 can make request to S1 and S1 forward the request to
ango,
I have been playing with connecting two * servers... I had to stop but
I do think I had it working... even with this link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers
it wasn't as straight forward as I would have liked... I used a
register on one box and a conf
hi,
I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both users to both servers, i.e.
asterisk server 1 (S1) has users 9002,9003
asterisk server 2
ango,
can you provide some sip.conf and extens.conf info?
daveC
Rilawich Ango wrote:
hi,
I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both
I have 2 asterisk boxes connected via SIP
box 1 sip peer connected
to box 2 (ip addresses intentionally removed)
[ast20]
type=friend
host=x.x.x.20
insecure=very
context=subscriber
dtmfmode=inband
qualify=no
canreinvite=no
disallow=all
allow=ulaw
box 2 sip peer connected
I have installed the first time Asterisk, (forgive me simple questions)
I have also installed the demo.
After testing demo (call 1000, call 600, ...) I changed in the
extensions.conf:
; include = demo
include = incomingsipgate
include = sipgate.de
include = sipgate.col.uk
[incomingsipgate]
Ronald Wiplinger wrote:
I have installed the first time Asterisk, (forgive me simple
questions)
I have also installed the demo.
I solved it with the newest cvs version !!!
bye
Ronald
After testing demo (call 1000, call 600, ...) I changed in the
extensions.conf:
; include = demo
include
PROTECTED]Sent: 17 September 2004
14:40To: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] Failed to authenticate on
INVITE
I am
getting this also.
I am
trying to get Asterisk to talk similarly to BT Communicator to the BT server. I
can register
NOTICE[98310]: chan_sip.c:6638 handle_response:
Failed to authenticate on INVITE to
'sip:[EMAIL PROTECTED];tag=as0f1d3429'
sip.conf
register =
1234:[EMAIL PROTECTED]
extension.conf
--
;; Own extensions;exten =
From: Stig Thune
[mailto:[EMAIL PROTECTED]Sent: 17 September 2004
12:55To: [EMAIL PROTECTED]Subject:
[Asterisk-Users] Failed to authenticate on INVITE
NOTICE[98310]: chan_sip.c:6638 handle_response:
Failed to authenticate on INVITE to
'sip:[EMAIL PROTECTED];tag=as0f1d3429'
At 16:49 16/06/2004 -0400, Eric wrote:
I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
These two boxes talk to eachother via sip, not iax. Since the upgrade, I
get the error Failed to authenticate on INVITE trying to make calls to/from
either box. Removing the secret
Hi Jason,
Thanks for your reply. I didn't really want to use the insecure option,
that defeats the purpose of using a password :)
I was, however, able to specify user= in my sip.conf entity and that
solved the problem I was having.
Thanks again.
- Eric
On Thu, 17 Jun 2004 10:17:54 +0100
Hi,
I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
These two boxes talk to eachother via sip, not iax. Since the upgrade, I
get the error Failed to authenticate on INVITE trying to make calls to/from
either box. Removing the secret from each box's sip config seems
Hi,
I need to inteconnection to VoIP Provider but I have this error message
when i try to dial external number :
-- Executing SetCallerID(SIP/491-1f64, x x ) in
new stack
-- Executing Dial(SIP/491-1f64, SIP/[EMAIL PROTECTED]|30|r)
in new stack
-- Called [EMAIL
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