Re: [asterisk-users] Asterisk 13 on CentOS 6

[albert zhang]

http://linoxide.com/tools/install-setup-asterisk-13-pbx-centos-7/

2017-04-27 2:15 GMT+08:00 Jerry Geis :

> > yum install jansson*
>
> This works for CentOS 7 but not CentOS 6.
>
> Thanks,
>
> Jerry
>
>
>
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Re: [asterisk-users] ** in extensions.conf

[Steve Edwards]

On Wed, 26 Apr 2017, Jerry Geis wrote:


dialplan show testing-sip
  '**' =>           1. Noop(Testing)                              [pbx_config]
                    2. Playback(demo-congrats)                    [pbx_config]

Looks like its there.

if I do ** "Dial" it works, but if I do "New Call" ** then "Dial" it 
does not work. Weird. How do I get it to work for both cases. (glad I 
tried the other)


I never use 'New Call' -- just 'Dial' and 'Redial,'

I suspect you'll need to fiddle with the Polycom dialplan. As soon as I 
press the first '*' my Poly sends the INVITE.


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Re: [asterisk-users] ** in extensions.conf

[Jerry Geis]

dialplan show testing-sip
  '**' =>   1. Noop(Testing)
 [pbx_config]
2. Playback(demo-congrats)
 [pbx_config]

Looks like its there.

if I do ** "Dial" it works, but if I do "New Call" ** then "Dial" it does
not work. Weird.
How do I get it to work for both cases. (glad I tried the other)

Jerry
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Re: [asterisk-users] ** in extensions.conf

[Steve Edwards]

On Wed, 26 Apr 2017, Jerry Geis wrote:


I just tried this in my extensions.conf

exten => **,1,Noop(Testing)
exten => **,n,Playback(demo-congrats)

Did a reload... and the above does not happen.
I created as 12 instead of the ** and that works fine.

Is there anyway to get the ** to work?  I also am using a polycom phone 
if that affects things. I'm using asterisk 13.15.0


Coincidentally, this is exactly how I exercise test code:

; test something
; (changes frequently)
exten = **,1,   verbose(1,[${EXTEN}@${CONTEXT}!${ANI}])
same = n,   answer()

I use an ancient Polycom IP 501 just fine.

Does 'dialplan show **@' yield any clues?

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Re: [asterisk-users] ** in extensions.conf

[Richard Mudgett]

On Wed, Apr 26, 2017 at 2:28 PM, Jerry Geis  wrote:

> I just tried this in my extensions.conf
>
> exten => **,1,Noop(Testing)
> exten => **,n,Playback(demo-congrats)
>
> Did a reload... and the above does not happen.
> I created as 12 instead of the ** and that works fine.
>
> Is there anyway to get the ** to work?  I also am using a polycom phone if
> that affects things. I'm using asterisk 13.15.0
>

A ** extension should work just fine.  I expect it is the dialplan in the
polycom phone
that doesn't allow it.

Richard
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[asterisk-users] ** in extensions.conf

[Jerry Geis]

I just tried this in my extensions.conf

exten => **,1,Noop(Testing)
exten => **,n,Playback(demo-congrats)

Did a reload... and the above does not happen.
I created as 12 instead of the ** and that works fine.

Is there anyway to get the ** to work?  I also am using a polycom phone if
that affects things. I'm using asterisk 13.15.0

Thanks

Jerry
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Re: [asterisk-users] Asterisk 13 on CentOS 6

[Jerry Geis]

> yum install jansson*

This works for CentOS 7 but not CentOS 6.

Thanks,

Jerry
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Re: [asterisk-users] pjsip direct_media=yes and "unknown" endpoints

[Daniel Tryba]

> > Anybody got an idea why the last scenario fails to work?
> 
> If you turn up core debug (core set debug 2) and ensure it is going to
> the CLI then the bridge_native_rtp module will tell you why exactly it
> can't native bridge. You might also want to do a core show channel on
> both channels to see what the codecs are.

Thanks for the hint, I wasn't seeing any debug since it wasn't getting
send to console. I'll take a better look and report back.


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Re: [asterisk-users] pjsip direct_media=yes and "unknown" endpoints

[Joshua Colp]

On Wed, Apr 26, 2017, at 01:25 PM, Daniel Tryba wrote:



> The request now gets routed based on a fully qualified domainname (with
> NAPTR/SRV records), which ultimately resolves to an ip that is matched in
> the
> endpoint SBC used above to originate a call.  But now the asterisk stays
> in the
> loop regarding RTP, a simple bridge is created but never switches to
> direct
> media.
> 
> SIP: enduser <-> uplink <-> asterisk 13 <-> pathfinder (302 redirect)
> 
> SIP: enduser <-> uplink <-> asterisk 13 <-> sip.xx.nl
> RTP: enduser <-> uplink <-> asterisk 13 <-> sip.xx.nl
> 
> Anybody got an idea why the last scenario fails to work?

If you turn up core debug (core set debug 2) and ensure it is going to
the CLI then the bridge_native_rtp module will tell you why exactly it
can't native bridge. You might also want to do a core show channel on
both channels to see what the codecs are.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] pjsip direct_media=yes and "unknown" endpoints

[Daniel Tryba]

I'm trying to implement direct_media between multiple peers and an
uplink provider, all of whom have direct_media=yes configures.

For originating calls to the uplink provider direct_media=yes works like
expected. SIP flows through asterisk, rtp doesn't

SIP: enduser <-> SBC <-> asterisk 13 <-> uplink
RTP: enduser <-> SBC <-> uplink

SBC matches an endpoint based on ip and dials the uplink:

-- Executing [+31x@outgoingrr:9] Dial("PJSIP/sbcs-0092", 
"PJSIP/+31x@uplink") in new stack
-- Called PJSIP/+31x@uplink
-- PJSIP/uplink-0093 is making progress passing it to PJSIP/sbcs-0092

-- PJSIP/uplink-0093 answered PJSIP/sbcs-0092
-- Channel PJSIP/uplink-0093 joined 'simple_bridge' basic-bridge 
<3b25c543-13a3-4d74-b2fe-7122a1cfe4a4>
-- Channel PJSIP/sbcs-0092 joined 'simple_bridge' basic-bridge 
<3b25c543-13a3-4d74-b2fe-7122a1cfe4a4>
   > Bridge 3b25c543-13a3-4d74-b2fe-7122a1cfe4a4: switching from simple_bridge 
technology to native_rtp
   > Remotely bridged 'PJSIP/sbcs-0092' and 'PJSIP/uplink-0093' - media 
will flow directly between them
   > Remotely bridged 'PJSIP/sbcs-0092' and 'PJSIP/uplink-0093' - media 
will flow directly between them

Whoever when a terminating call comes in from the uplink provider, a sip
request is send to a redirector. The redirector has redirect_method=uri_core
configured (the only method that works for me).

-- Executing [+31x@incoming:11] Dial("PJSIP/uplink-0094", 
"PJSIP/+31x@pathfinder") in new stack
-- Called PJSIP/+31x@pathfinder
-- Now forwarding PJSIP/uplink-0094 to 
'PJSIP/pathfinder/sip:+31xx...@sip.xx.nl' (thanks to 
PJSIP/pathfinder-0095)
...
-- PJSIP/pathfinder-0096 answered PJSIP/uplink-0094
-- Channel PJSIP/pathfinder-0096 joined 'simple_bridge' basic-bridge 
<1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a>
-- Channel PJSIP/uplink-0094 joined 'simple_bridge' basic-bridge 
<1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a>
...
-- Channel PJSIP/pathfinder-0096 left 'simple_bridge' basic-bridge 
<1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a>
-- Channel PJSIP/uplink-0094 left 'simple_bridge' basic-bridge 
<1bf02059-ea8f-4f9c-bc33-8ae99ba45c9a>

The request now gets routed based on a fully qualified domainname (with
NAPTR/SRV records), which ultimately resolves to an ip that is matched in the
endpoint SBC used above to originate a call.  But now the asterisk stays in the
loop regarding RTP, a simple bridge is created but never switches to direct
media.

SIP: enduser <-> uplink <-> asterisk 13 <-> pathfinder (302 redirect)

SIP: enduser <-> uplink <-> asterisk 13 <-> sip.xx.nl
RTP: enduser <-> uplink <-> asterisk 13 <-> sip.xx.nl

Anybody got an idea why the last scenario fails to work?


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Re: [asterisk-users] Asterisk 13 on CentOS 6

[albert zhang]

yum install jansson*
Jerry Geis 于2017年4月26日 周三下午8:32写道：

> >It can't be disabled. jansson is a required dependency for Asterisk 13
> >as JSON is used internally for things.
>
>
> Ok thanks - that is a little confusing since there are entries in the 
> configure script that lead one to think it can be a configure time switch.
>
> I'll go the other route and install the library, was hoping not to.
>
>
> Jerry
>
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>
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Re: [asterisk-users] Asterisk 13 on CentOS 6

[Jerry Geis]

>It can't be disabled. jansson is a required dependency for Asterisk 13
>as JSON is used internally for things.


Ok thanks - that is a little confusing since there are entries in the
configure script that lead one to think it can be a configure time
switch.

I'll go the other route and install the library, was hoping not to.


Jerry
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Re: [asterisk-users] Asterisk 13 on CentOS 6

[Joshua Colp]

On Wed, Apr 26, 2017, at 09:24 AM, Jerry Geis wrote:
> Trying to install asterisk 13 on CentOS 6.
> 
> The ./configure tells me:
> configure: error: *** JSON support not found (this typically means the
> libjansson development package is missing)
> 
> I don't really need JSON so I thought I would just disable it.
> 
> ./configure --with-jansson=no does not work
> ./configure --without-jansson does not work
> 
> How do I use a configure switch to disable it?

It can't be disabled. jansson is a required dependency for Asterisk 13
as JSON is used internally for things.

-- 
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] Asterisk 13 on CentOS 6

[Jerry Geis]

Trying to install asterisk 13 on CentOS 6.

The ./configure tells me:
configure: error: *** JSON support not found (this typically means the
libjansson development package is missing)

I don't really need JSON so I thought I would just disable it.

./configure --with-jansson=no does not work
./configure --without-jansson does not work

How do I use a configure switch to disable it?

Thanks,

Jerry
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