I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls from Junction and
sean darcy wrote:
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls
Hi,
I have a queue with one agent added using AddQueueMember
(FAO|Local/[EMAIL PROTECTED]|0||Agent/602). My extensions.conf is
[general]
static=yes
writeprotect=yes
autofallthrough=no
clearglobalvars=no
priorityjumping=no
[from-sip]
exten = 11000,1,Dial(SIP/11000,,t)
exten =
Here's an interesting question:
If I transfer a call from Asterisk system to another with IAX, is there any way
I can get control back on the original system? Or.. do I lose control, and the
dialplan has to continue on the new system?
Scenario is we transfer calls to an Asterisk system that
Yes, the other system does not maintain control of the phone call. We're
using IAX right now to handle redundancy (going to move to SIP soon I
think) and basically, when the other system hangs up the channel, the
original server continues in it's own call pattern until it hangs up or
the user
: Friday, March 24, 2006 12:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Transferring a call with IAX
Yes, the other system does not maintain control of the phone
call. We're
using IAX right now to handle redundancy (going to move to SIP
Heh, lots of voodoo... I've got a drawer full of dolls shaped like servers
that we stick pins into when something's not working :)
Anyway... um, let's see if I can piece this together, it's kinda
scattered...
A call comes from SCM2 (the secondary call server) and it starts looking
for the
... I can see myself spending days on trying to get this to work.
Doug
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call
-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Heh, lots of voodoo... I've got a drawer full of dolls shaped
like servers
that we stick
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Hhhmmm... I missed something... You're jumping from one
calling server
through a callee server, and then from there to another server for
voicemail?
Aaron
On Fri, 24 Mar
something
similar?
Douglas.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 2:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Hhhmmm... I missed something... You're
... on the caller:
-- Hungup 'IAX2/acdserver1-3'
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
My bad, sorry
[mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
My bad, sorry, one of those days.
Change priority 4 on the ACD server to a Hangup and ignore
what I said
before
: [Asterisk-Users] Transferring a call with IAX
Hhhmmm... I missed something... You're jumping from one
calling server
through a callee server, and then from there to another
server for
voicemail?
Aaron
On Fri, 24 Mar 2006, Douglas Garstang wrote:
Thanks Aaron, but nope
! How weird.. it looks like I _AM_ getting control
back, sort of...
Doug.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call
PROTECTED]
Sent: Friday, March 24, 2006 4:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Looking at your macro, I don't have any MacroExits in mine.
I use AEL,
and it doesn't put that on the macros. Try changing
:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Looking at your macro, I don't have any MacroExits in mine.
I use AEL,
and it doesn't put that on the macros. Try changing your
MacroExit to a
NoOp(Macro Finished) and see
*
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Why do you have s-ANSWER jumping to s-OK? Try putting a NoOp
in s
, and THAT jumps to s-CHANUNAVAIL.
*sigh*
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Why do you
-CHANUNAVAIL.
*sigh*
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Why do you have s-ANSWER jumping to s-OK? Try
'
== Spawn extension (macro-DialIAX, s, 1) exited non-zero on 'SIP/2944093-9ef2'
-Original Message-
From: Douglas Garstang
Sent: Friday, March 24, 2006 4:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Nope. Still
Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Looking at your macro, I don't have any MacroExits in mine.
I use AEL
Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Ok, add g to the option list on the dial:
Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],25,wWg)
g- Proceed with dialplan execution at the current
extension
I must be missing something here. Have you tried option "g" on your
dial command to the acd server? If option g is not specified, then dial
will hangup the call when exiting regaurdless of what the other iax box
did.
-Jon
Douglas Garstang wrote:
I just changed the macro to:
exten =
Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, March 24, 2006 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Transferring a call with IAX
Ok, add g to the option list on the dial:
Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],25,wWg)
g
We are using Asterisk 1.0.7. We have this scenario:
IAX2 user comes in to Asterisk, dials an extension, and transfers to a SIP user.
The dial command is simple, looks like this:
exten = 300,1,Dial(SIP/300)
Extension 300 is a legacy PBX device operated by touchtones. The user (coming in
Title: Transferring a call
Hello,
I have successfully connected an Asterisk PBX to an old Panasonic Phone System using an AVM Fritz PCI card. But when I make a call through the Asterisk PBX to the old phone system, and the receiver wants to transfer the call to another internal number, I
.
Brian
Leyton
IT Manager
Commercial Petroleum
Equipment
From: Dennie Verstrepen
[mailto:[EMAIL PROTECTED] Sent: Wednesday, April 13,
2005 7:17 AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Transferring a call
Hello,I have successfully connected an Asterisk PBX
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