I made these changes in dialplan and it worked. Thanks a lot.
In most of the cases S1, S2 and C1 are in my control. But in some
cases the dialplan of C1 is not in my control. Also in some cases C1
can be any SIP client like a softphone or SIP device, so it wont work
in those case. Is there some
Deepesh D wrote:
I made these changes in dialplan and it worked. Thanks a lot.
In most of the cases S1, S2 and C1 are in my control. But in some
cases the dialplan of C1 is not in my control. Also in some cases C1
can be any SIP client like a softphone or SIP device, so it wont work
in those
This doesn't work reliably well with all all clients. I tested it
using a zoiper soft phone and it worked. But from an ATA device it
failed. On the S2 server it failed to authenticate
The console of S2 showed
[Oct 12 18:21:06] WARNING[30483]: chan_sip.c:13952 check_auth:
username mismatch, have
Deepesh D wrote:
This doesn't work reliably well with all all clients. I tested it
using a zoiper soft phone and it worked. But from an ATA device it
failed. On the S2 server it failed to authenticate
The console of S2 showed
[Oct 12 18:21:06] WARNING[30483]: chan_sip.c:13952 check_auth:
From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is
From C1 when I directly dial into S2, it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2, I define the context as
'test_context' and the default context is
From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is
From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'.
When I directly dial from C1 into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default
Discussion
Subject: Re: [asterisk-users] How to use 'Transfer' to send calls to another
asterisk?
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'.
When I directly dial from C1 into S2 it goes into the context
'test_context'. But when
] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 5:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to use 'Transfer' to send calls to another
asterisk?
In all my peer definitions on S1 and S2 I define the context as
'test_context
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to use 'Transfer
: Re: [asterisk-users] How to use 'Transfer' to send calls to another
asterisk?
From C1 when I directly dial into S2 it goes into the context 'test_context'.
But when the call is made to S1 and S1 transfers the call to S2 then the call
goes into default context.
In all my peer definitions on S1
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to use 'Transfer' to send calls to another
asterisk?
I am able to register S1 as a peer in S2 and dial from S1 to S2, but this is
not my requirement. I want to dial from C1 into S1 and S1 should
Deepesh D wrote:
If C1 dials S1 and then S1 dials S2 to transfer the call then S1 still
remains in the loop till the call is finished. What I wanted to do is
to reduce the number of calls on S1, so as soon as S1 receives a call
from C1 it redirects the call to S2 using 'Transfer' application and
Sip trunk?
2012/10/10 Deepesh D deep.d2...@gmail.com
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which registers as a
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which
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