Thanks ,it solved by adding
insecure=very
regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Tuesday, September 28, 2010 2:16 PM
To: Asterisk; Asterisk List
Subject: [asterisk-user
You shouldn't "open text" your password. Shouldn't IP on Asterisk 2 be
1.2.3.4?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun Wingrin
Sent: Monday, December 01, 2008 4:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound calls from Aste
You can use next parameter:
Fromuser = VoipDirect777821
At 04:23 p.m. 01/12/2008, Shaun Wingrin wrote:
Please help.
Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=
accountcode=5260477782
amaflags=billing
context=Incoming
On Oct 9, 2008, at 10:40 AM, [EMAIL PROTECTED]
wrote:
> Hi I have searched the mailing lists and come across similar threads,
> but no actual solution. I am trying to use a Cisco AS5300 as a
> gateway for PSTNr. I have been able to configure it to take outbound
> calls and send them to the PST
On 10/9/08, Ketema Harris <[EMAIL PROTECTED]> wrote:
>
> Hi I have searched the mailing lists and come across similar threads, but no
> actual solution. I am trying to use a Cisco AS5300 as a gateway for PSTNr.
> I have been able to configure it to take outbound calls and send them to the
> PSTN j
Not offhand / without seeing the Asterisk side.
On Thu, October 9, 2008 10:26 am, Ketema Harris wrote:
> dtmf mode was set in the sip.conf
>
> dtmfmode=rfc2833
>
> I will remove the other codecs from sip.conf and see what effect it
> has. Do you see any other potential issues in the configs?
>
>
dtmf mode was set in the sip.conf
dtmfmode=rfc2833
I will remove the other codecs from sip.conf and see what effect it
has. Do you see any other potential issues in the configs?
thanks
On Oct 9, 2008, at 9:36 AM, Alex Balashov wrote:
>
> This is due to an SDP mismatch of some sort, codec o
This is due to an SDP mismatch of some sort, codec or otherwise.
Perhaps you have not set your Asterisk SIP peers to support RFC2833
DTMF? Try dtmfmode=rfc2833. Either that, or your Asterisk SIP peers
are not accepting the gateway's offer of G.711u.
Of course, I have seen interop bugs in Aster
So it looks like the information is coming through in the SIP header.
Is there anyway to avoid the "register" command - at the end of the
day I may have 100s of DIDs and I don't want to have to set them up by
hand.
Is it possible to "fix" what Asterisk thinks the extension is be "resetting" it?
bug
of the session to the list, so we may examine it?
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Mr. Jones
Sent: Friday, August 11, 2006 10:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound Calls
nt required anymore
Sherwood
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Friday, August 11, 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound Calls & SIP/2.0 404 Not F
Title: RE: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
Hmmm...
Appears as if the SIP invite request is ill-formed. Can you send the SIP debug
of the session to the list, so we may examine it?
Nir S
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTE
st argument required anymore
Sherwood
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Friday, August 11, 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound Calls & SIP/2
I have that
Looking for s in test-context (domain 9495551212)
-- Executing NoOp("SIP/5060-b7a1aa50", "Exten is: s") in new stack
== Auto fallthrough, channel 'SIP/5060-b7a1aa50' status is 'UNKNOWN'
SIP/2.0 603 Declined
I'm just not sure how to use the "domain BLAH" to match an extensi
Actually it looks like I am getting the number but its coming through weird:
This is what sip debug gives me:
Looking for s in test-context (domain 9495551212)
So clearly I am getting the number, just not sure if its formated ok?
On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote:
Yeah...
I tr
Yeah...
I tried the NoOp function someone gave me above and I'll I'm getting is "s"
I'll go back to the provider
On 8/11/06, C F <[EMAIL PROTECTED]> wrote:
s, means that it got an incoming call, but no exten came with it.
On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote:
> I double checked th
Mr. Jones wrote:
I have 20 DIDs, some I want to send to a menu, most directly to an
extension.
sip debug is (really) your friend. It should give you the [context]
where your DID is being send to and the 404 not found error also.
A particular line to look for: "Looking for ...".
___
s, means that it got an incoming call, but no exten came with it.
On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote:
I double checked the context.
But the "Looking for s" is a bit confusing - not sure what "s" is?
On 8/11/06, Vadim Berezniker <[EMAIL PROTECTED]> wrote:
> Perhaps the context in s
You might want to take a look at 'sip debug' to see what your provider
is actually sending you. Its likely they aren't sending you the
9495551212 sting as you are expecting.
Thanks -
Just to be clear - I just replaced the real digits with - I want
to direct these to specific extensions.
Thanks Kevin -
I realized afterwards that the was a bad example.
It should be a specific number, I was just masking it.
I have 20 DIDs, some I want to send to a menu, most directly to an extension.
I've tried:
exten=>9492711234,1, Macro(druiexten,3711,SIP/3711)
and:
exten=>_9492711234,1,
I double checked the context.
But the "Looking for s" is a bit confusing - not sure what "s" is?
On 8/11/06, Vadim Berezniker <[EMAIL PROTECTED]> wrote:
Perhaps the context in sip.conf doesn't match the context in the dial plan.
From: [EMAIL PROTECTED] on beha
Thanks -
Just to be clear - I just replaced the real digits with - I want
to direct these to specific extensions. So maybe I should have used
or something else?
I tried this:
exten=>_9495551212,1, Goto(mainmenu,s,1)
But still to no avail.
On 8/11/06, Vadim Berezniker <[EMAIL PROTECT
If I am following you right, for extension matching you need to have a
"_" in front of the number.
So your example should be like this:
exten => _949927,1,Goto(mainmenu,s,1)
Also I don't know if you did this on purpose or not but N will only
match for numbers 2-9, if you want 0-9 you will
Perhaps the context in sip.conf doesn't match the context in the dial plan.
From: [EMAIL PROTECTED] on behalf of Mr. Jones
Sent: Fri 8/11/2006 2:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound Calls & SIP/2.0 404 Not Found
I'm trying
Try changing it to:
exten => _949271,1,Goto(whatever)
another way to troubleshoot and figure out what you are getting for
the DID would be:
exten => _X.,1,Noop(Exten is: ${EXTEN})
watch the CLI to see what DID is coming in.
On 8/11/06, Mr. Jones <[EMAIL PROTECTED]> wrote:
I'm trying to get i
Hi!
> Jan 6 15:49:55 WARNING[1112791984]: pbx.c:1868 ast_pbx_run: Channel
> 'CAPI[contr1/221591030]/0' sent into invalid extension 's' in context
> 'default', but no invalid handler
> [interfaces]
> msn=0221591030
> incomingmsn=221591030
> controller=1
> devices=2
> softdtmf=1
> callgroup=1
> co
Hey Dan!!
Give us a clue as to what hardware/setup & network provider you have there,
and we might be able to help :)
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 06 January 2005 06:56
To: asterisk-users@lists.digium.com
S
We will need more info on your setup.
When people call into your Asterisk system what device will they be calling
in on?
Will they call a number provided by a termination/origination provider which
is then fed into your Asterisk server using IAX or SIP?
Will they call a TDM card attached to you
I have my RX at 4.0 ant TX at 8.0,
I get slight echo for the first 5-6 seconds then all OK.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Stenton
Sent: 22 April 2004 17:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] inbound calls b
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