Re: [asterisk-users] Asterisk NOT in the media path
On 02/24/2012 10:51 PM, Jared Geiger wrote:
On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls
to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call,
and based upon
some checks in the dialplan send the call through to one
of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to
stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes
in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider Asterisk A1 Asterisk B1 Asterisk B2
I want the audio to go directly from Provider to server B1
when the call
has been set up.
As long as there are no NATs involved, yes, this should work.
You will also need 'canreinvite' ('directmedia' in Asterisk
1.8 and later) in the peer definition for the provider.
Hello again,
this is currently not really working.
I see on the Asterisk CLI that the call streams through my
Asterisk A1 (which should stay out of the media path) :
[Feb 23 22:24:47] -- Called Mast/980419
[Feb 23 22:24:47] -- SIP/Mast-000e answered
SIP/VOXBONEin-000d
[Feb 23 22:24:47] -- Native bridging SIP/VOXBONEin-000d
and SIP/Mast-000e
*CLI
*CLI core show channels
Channel Location State Application(Data)
SIP/Mast-00 (None) Up AppDial((Outgoing Line))
SIP/VOXBONEin-00 980419@VOXBONEin Up Dial(SIP/Mast/980419)
2 active channels
1 active call
Peer VoxBone and peer Mast should re-invite and leave this
Asterisk out of the media path on call answer.
These are my SIP peer definitions :
[VOXBONEin]
type=peer
host=XX.XX.XX.XX
context=VOXBONEin
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
[Mast]
type=peer
host=XX.XX.XX.XX
defaultuser=Mast
secret=guessme
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
Am I missing a setting ? Using Asterisk 1.6.2.22
The Asterisk server still stays in the SIP Signaling path of the call,
just media does not flow through the server. You can verify this by
running a SIP debug and looking at the media endpoints.
What is it that I should be looking for in the SIP debug information ?
Is it in the SDP-body ?
Kind regards,
Jonas.
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Re: [asterisk-users] Asterisk NOT in the media path
On Feb 29, 2012, at 7:25 AM, Jonas Kellens wrote: The Asterisk server still stays in the SIP Signaling path of the call, just media does not flow through the server. You can verify this by running a SIP debug and looking at the media endpoints. What is it that I should be looking for in the SIP debug information ? Is it in the SDP-body ? To set up direct media, Asterisk will send a re-invite with an SDP body containing the address of the other endpoint. RTP then flows directly between endpoints. If you just want to know if Asterisk is in the media path or not, you can also use rtp set debug ... and Asterisk will log a line for each RTP packet it handles. -- v: 248.893.0738 | f: 248.893.0747 http://macprofessionals.com/ find us on facebookall -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk NOT in the media path
On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call, and based upon
some checks in the dialplan send the call through to one of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider Asterisk A1 Asterisk B1 Asterisk B2
I want the audio to go directly from Provider to server B1 when the call
has been set up.
As long as there are no NATs involved, yes, this should work. You will
also need 'canreinvite' ('directmedia' in Asterisk 1.8 and later) in the
peer definition for the provider.
Hello again,
this is currently not really working.
I see on the Asterisk CLI that the call streams through my Asterisk A1
(which should stay out of the media path) :
[Feb 23 22:24:47] -- Called Mast/980419
[Feb 23 22:24:47] -- SIP/Mast-000e answered SIP/VOXBONEin-000d
[Feb 23 22:24:47] -- Native bridging SIP/VOXBONEin-000d and
SIP/Mast-000e
*CLI
*CLI core show channels
Channel Location State Application(Data)
SIP/Mast-00 (None) Up AppDial((Outgoing Line))
SIP/VOXBONEin-00 980419@VOXBONEin Up Dial(SIP/Mast/980419)
2 active channels
1 active call
Peer VoxBone and peer Mast should re-invite and leave this Asterisk out of
the media path on call answer.
These are my SIP peer definitions :
[VOXBONEin]
type=peer
host=XX.XX.XX.XX
context=VOXBONEin
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
[Mast]
type=peer
host=XX.XX.XX.XX
defaultuser=Mast
secret=guessme
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
Am I missing a setting ? Using Asterisk 1.6.2.22
The Asterisk server still stays in the SIP Signaling path of the call, just
media does not flow through the server. You can verify this by running a
SIP debug and looking at the media endpoints.
If you don't want the call to maintain in the server at all, you need to
look into doing 302 redirects.
Regards,
Jared
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Re: [asterisk-users] Asterisk NOT in the media path
On 02/23/2012 01:48 PM, Jonas Kellens wrote:
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call, and based upon
some checks in the dialplan send the call through to one of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider Asterisk A1 Asterisk B1 Asterisk B2
I want the audio to go directly from Provider to server B1 when the call
has been set up.
As long as there are no NATs involved, yes, this should work. You will
also need 'canreinvite' ('directmedia' in Asterisk 1.8 and later) in
the peer definition for the provider.
Hello again,
this is currently not really working.
I see on the Asterisk CLI that the call streams through my Asterisk A1
(which should stay out of the media path) :
[Feb 23 22:24:47] -- Called Mast/980419
[Feb 23 22:24:47] -- SIP/Mast-000e answered SIP/VOXBONEin-000d
[Feb 23 22:24:47] -- Native bridging SIP/VOXBONEin-000d and
SIP/Mast-000e
This indicates that it *is* working. Asterisk has setup a 'native' RTP
bridge between these two call legs. If they accept the re-INVITES that
are sent, then the media will flow directly between them.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
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Re: [asterisk-users] Asterisk NOT in the media path
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call, and based upon
some checks in the dialplan send the call through to one of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider Asterisk A1 Asterisk B1 Asterisk B2
I want the audio to go directly from Provider to server B1 when the call
has been set up.
As long as there are no NATs involved, yes, this should work. You will
also need 'canreinvite' ('directmedia' in Asterisk 1.8 and later) in
the peer definition for the provider.
Hello again,
this is currently not really working.
I see on the Asterisk CLI that the call streams through my Asterisk A1
(which should stay out of the media path) :
[Feb 23 22:24:47] -- Called Mast/980419
[Feb 23 22:24:47] -- SIP/Mast-000e answered SIP/VOXBONEin-000d
[Feb 23 22:24:47] -- Native bridging SIP/VOXBONEin-000d and
SIP/Mast-000e
*CLI
*CLI core show channels
Channel Location State Application(Data)
SIP/Mast-00 (None) Up AppDial((Outgoing Line))
SIP/VOXBONEin-00 980419@VOXBONEin Up Dial(SIP/Mast/980419)
2 active channels
1 active call
Peer VoxBone and peer Mast should re-invite and leave this Asterisk out
of the media path on call answer.
These are my SIP peer definitions :
[VOXBONEin]
type=peer
host=XX.XX.XX.XX
context=VOXBONEin
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
[Mast]
type=peer
host=XX.XX.XX.XX
defaultuser=Mast
secret=guessme
disallow=all
allow=alaw
allow=gsm
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
Am I missing a setting ? Using Asterisk 1.6.2.22
Regards,
Jonas.
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[asterisk-users] Asterisk NOT in the media path
Hello, I want to place an Asterisk-server A in front of 2 other Asterisk-servers (B1 B2). This first Asterisk-server A needs to send incoming calls to one of the 2 available Asterisk-servers (B1 or B2) behind it. So I want the first Asterisk-server A to accept the call, and based upon some checks in the dialplan send the call through to one of the other Asterisk-servers (B1 or B2) which further handle the call. The first Asterisk-server A then needs to pull itself from the media-path. There's no further need for this Asterisk to stay within the audio-path. 1. Is this possible ? 2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer definition of Asterisk B1 and Asterisk B2 ? So I have : Provider Asterisk A1 Asterisk B1 Asterisk B2 I want the audio to go directly from Provider to server B1 when the call has been set up. Thank you for your input ! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk NOT in the media path
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call, and based upon
some checks in the dialplan send the call through to one of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider Asterisk A1 Asterisk B1 Asterisk B2
I want the audio to go directly from Provider to server B1 when the call
has been set up.
As long as there are no NATs involved, yes, this should work. You will
also need 'canreinvite' ('directmedia' in Asterisk 1.8 and later) in the
peer definition for the provider.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk NOT in the media path
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call, and based upon
some checks in the dialplan send the call through to one of the other
Asterisk-servers (B1 or B2) which further handle the call.
The first Asterisk-server A then needs to pull itself from the
media-path. There's no further need for this Asterisk to stay within the
audio-path.
1. Is this possible ?
2. Using Asterisk 1.6.2.22, do I just use canreinvite=yes in the peer
definition of Asterisk B1 and Asterisk B2 ?
So I have :
Provider Asterisk A1 Asterisk B1 Asterisk B2
I want the audio to go directly from Provider to server B1 when the call
has been set up.
As long as there are no NATs involved, yes, this should work. You will
also need 'canreinvite' ('directmedia' in Asterisk 1.8 and later) in
the peer definition for the provider.
There is no NAT in the path : Provider Asterisk A Asterisk B1
All public IP-adresses.
All these peer definitions will have canreinvite=no (Asterisk 1.6.2.22)
OK then ! Great !
Jonas.
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Re: [Asterisk-Users] Asterisk remains in the media path
Le lun 03/05/2004 à 18:48, Jeremy McNamara a écrit : Actually its cuz chan_h323 sucks like that. Correct me if I'm wrong, but I browsed the archives and I got the feeling that you (Jeremy) were one of the main developers of the chan_h323... aren't you a little harsh about your own work? :-) Anyway, is there any plan in the chan_h323 roadmap to support direct RTP between endpoints? Thanks, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk remains in the media path
Hi all, Just a quick question: I have an H323 terminal and some MGCP phones connected to *, and when they call each other * remains in the media path no matter what (while I'd like to have the RTP stream directly between the phones). - mgcp.conf has canreinvite=yes - extension.conf doesn't contain any Dial() instance with t or T Did I forget something? Thanks, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk remains in the media path
Can't do it because you are changing from one technology to another. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Berger Sent: Monday, May 03, 2004 10:29 AM To: Liste Asterisk Subject: [Asterisk-Users] Asterisk remains in the media path Hi all, Just a quick question: I have an H323 terminal and some MGCP phones connected to *, and when they call each other * remains in the media path no matter what (while I'd like to have the RTP stream directly between the phones). - mgcp.conf has canreinvite=yes - extension.conf doesn't contain any Dial() instance with t or T Did I forget something? Thanks, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk remains in the media path
Le lun 03/05/2004 à 17:34, brian a écrit : Can't do it because you are changing from one technology to another. Thanks for your answer. H323 and MGCP are supposed to stay on the call control level, why isn't it possible to open RTP channels between the terminals then? Again, thanks, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk remains in the media path
brian wrote: Can't do it because you are changing from one technology to another. Actually its cuz chan_h323 sucks like that. Jeremy McNamara -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Berger Sent: Monday, May 03, 2004 10:29 AM To: Liste Asterisk Subject: [Asterisk-Users] Asterisk remains in the media path Hi all, Just a quick question: I have an H323 terminal and some MGCP phones connected to *, and when they call each other * remains in the media path no matter what (while I'd like to have the RTP stream directly between the phones). - mgcp.conf has canreinvite=yes - extension.conf doesn't contain any Dial() instance with t or T Did I forget something? Thanks, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk remains in the media path
On Mon, 2004-05-03 at 12:05, jimfl wrote: So does this mean you could get direct RTP steams between a SIP client and a IAX2 client? What about inband/out of band DTMF issues? IAX doesn't use rtp and therefore it couldn't do it either. All DTMF should be OOB to be reliable. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
