Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Jerry Geis]

I found dahdi_test...

 dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.999% 99.904% 99.974% 99.814% 98.070% 97.850% 99.985% 99.887%
99.708% 99.899% 99.805% 99.708% 99.902% 100.000% 99.949% 99.883%
99.891% 99.906% 99.784% 99.719% 99.827% 99.903%
--- Results after 22 passes ---
Best: 100.000% -- Worst: 97.850% -- Average: 99.698465%
Cummulative Accuracy (not per pass): 99.991

seems like low numbers and not even running audio at this time.

I'm thinking with the PRI card removed there is no reliable timing source.

How do I get ConfBridge to have a reliable timing source?

Thanks,

Jerry
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[asterisk-users] Asterisk 11.24.1 garbled audio

[Jerry Geis]

Hi all

I am using asterisk 11.24.1 on a centos 5 machine. kernel 2.6.18 flavor.
(x86_64).
I have about SIP 150 endpoints on it.
when I send a message I'm getting garbled audio.

I used to have a single PRI card in the box - but something happened and
that connection
no longer worked. I removed the card and also removed the system.conf and
chan_dahdi entries.

I am using ConfBridge in a PA kind of mode so all endpoints are muted.

how do I monitor any dahdi timing? (dahdi 2.11.1 complete) (libpri 1.5)

Network is 1G. Devices are 100Full connected to a 100Full switch.
Server is a real server 95% idle when speaking.

Any suggestions on what to look at or how to resolve?

Thanks,

Jerry
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[asterisk-users] Asterisk 14.1.2 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 14.1.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.1.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
  calls after 2 minutes - rtptimeout behaving badly - regression
  (Reported by Michael Keuter)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.2

Thank you for your continued support of Asterisk!

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[asterisk-users] Asterisk 13.12.2 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 13.12.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.12.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
  calls after 2 minutes - rtptimeout behaving badly - regression
  (Reported by Michael Keuter)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.2

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] SIP and RTP port and IP addresses

[Matthew Jordan]

On Thu, Nov 10, 2016 at 7:15 AM, Ethy H. Brito 
wrote:

> On Thu, 10 Nov 2016 00:35:54 +0100
> Max Grobecker  wrote:
>
> > Hi Ethy,
>
> Hi Max and All.
>
> >
> >
> > Am 09.11.2016 um 17:13 schrieb Ethy H. Brito:
> >
> > > How are these parameters available from dialplan?
> > >
> > > For instance, ${SIPURI} holds the internal "IP:port" if the client is
> > > behind NAT. I need the external IP:port
> >
> >
> > You can get the peer's signalling IP address from ${CHANNEL(recvip)} and
> the
> > RTP address with ${CHANNEL(rtpsource)} resp. ${CHANNEL(rtpdest)}. If you
> need
> > more information (like the codecs used) you can find other channel
> variables
> > on https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+
> Function_CHANNEL
>
> H.
>
> ${CHANNEL(rtpsource)} is always returning something like "0.0.0.0:p"
> where
> p=[0-9]
>
>
You've bound to the 'bind all' address - hence why you get '0.0.0.0'. The
'p' values are the RTP port that was chosen for that call. RTP port ranges,
by default, are from 5000 to 31000.



> and ${CHANNEL(rtpdest)} returns the internal (not accessible) IP addr if
> the
> caller is behind NAT, therefore, not what I need.
>
>
The RTP destination is going to be what is negotiated in the SDP. If that's
a private IP address, then that's what you'd see there.

If you have symmetric RTP enabled, then this will switch to the address
that we are receiving RTP from. That may or may not be the original
negotiated address - if the remote end is behind a NAT, it will most likely
switch to the public IP address that we are receiving media from.




> Wouldn't these two variables have correct values only after the callee
> answers
> the call??
>
>
No. In fact, as Asterisk is a B2BUA, there are always going to be two sets
of RTP values:

 - The source/destination of the RTP stream to the inbound channel
 - The source/destination of the RTP stream to the outbound channel

The inbound channel will have its set of RTP addresses when Asterisk either
sends a Progress indication or Answers the inbound channel. The outbound
channel will have its set of RTP addresses when the far end sends a
Progress indication or Answers the outbound channel.

All of these RTP addresses may change due to:
 * NAT settings (symmetric RTP)
 * re-INVITEs, either due to Asterisk directmedia settings or re-INVITEs
initiated by the far endpoints (call hold, etc.)
 * ICE negotiation



> >
> > Please note that, if you have not disabled re-invites, the RTP address
> may
> > change while the call is running.
>
> Interesting observation.
>
> Thanx
>
> Ethy
>
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> org/
>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>



-- 
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] SIP and RTP port and IP addresses

[Ethy H. Brito]

On Thu, 10 Nov 2016 00:35:54 +0100
Max Grobecker  wrote:

> Hi Ethy,

Hi Max and All. 

> 
> 
> Am 09.11.2016 um 17:13 schrieb Ethy H. Brito:
> 
> > How are these parameters available from dialplan?
> > 
> > For instance, ${SIPURI} holds the internal "IP:port" if the client is
> > behind NAT. I need the external IP:port  
> 
> 
> You can get the peer's signalling IP address from ${CHANNEL(recvip)} and the
> RTP address with ${CHANNEL(rtpsource)} resp. ${CHANNEL(rtpdest)}. If you need
> more information (like the codecs used) you can find other channel variables
> on https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL

H.

${CHANNEL(rtpsource)} is always returning something like "0.0.0.0:p" where
p=[0-9]

and ${CHANNEL(rtpdest)} returns the internal (not accessible) IP addr if the
caller is behind NAT, therefore, not what I need.

Wouldn't these two variables have correct values only after the callee answers
the call??

> 
> Please note that, if you have not disabled re-invites, the RTP address may
> change while the call is running.

Interesting observation.

Thanx

Ethy

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