[asterisk-users] Inband DTMF not detected - bug or config error?

[Sebastian Damm]

Hi,

we have an Asterisk server basically passing on calls using the Dial
application. In the pjsip endpoint settings, the dtmf_mode is set to audio.
This works with most calls. However, there is a scenario where DTMF tones
don't get forwarded the way I would expect them to get forwarded.

A: Caller without RfC4733 support
B: our Asterisk, version 17.6.0
C: Another Asterisk, with RfC4733 support, running an IVR

Now when the call comes in, our Asterisk (B) sends out a new call, offering
telephone-event to C. However, since A and C use the same codec, B bridges
those two calls using a native bridge. And that's probably the reason why
it doesn't detect inband DTMF tones from A, but instead just passes on the
media stream to C. C, however, thinks DTMF should come as RTP events, not
detecting the tones coming inband, either.

The documentation says, the native bridge is not capable of doing anything
with DTMF, but the core bridge would be. (
https://wiki.asterisk.org/wiki/display/AST/Bridges)

In my opinion, Asterisk should not enter native bridge, when dtmf modes for
the two channels differ. Is this a bug? If it isn't, is there a way to
force Asterisk to use the core bridge, so DTMF tones get detected inband
and converted to rtp events?

Any hint appreciated.

Regards,
Sebastian
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Re: [asterisk-users] Need for more hangup reasons in ARI?

[Sebastian Damm]

Hi,

On Thu, Dec 6, 2018 at 2:44 PM Joshua C. Colp  wrote:
> Nope. No specific reason. The ones there are what most people would use, and 
> what those not as familiar with telephony would understand.

Thanks for the quick response. I have submitted a patch for adding
more hangup reasons.

BTW: Is there a way to have them documented on the Wiki page instead
of having to dig into the source code? I'd be happy to help.

Regards,
Sebastian

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[asterisk-users] Need for more hangup reasons in ARI?

[Sebastian Damm]

Hi,

while implementing an application based on ARI, I wanted to hangup
calls in different states with different hangup reasons. After not
finding anything in the documentation I found the possible reasons in
the 
sources:(https://github.com/asterisk/asterisk/blob/4ca709768db9bafdfa83ee1cfc6cc0639f097857/res/ari/resource_channels.c#L866)

I think, there are some important reasons missing. For example I want
to return a 408 if the callee doesn't answer after some time. I guess,
I could submit a patch for more hangup reasons, but before doing so,
is there a reason that there are only those five hangup reasons?

Regards,
Sebastian

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[asterisk-users] Asterisk registers with TLS, but sends out calls via UDP

[Sebastian Damm]

Hi,

I have an Asterisk 13.8.2, which is supposed to be only a client to an
encrypted SIP service. All local phones are connected via UDP.

Since I can't use PJSIP (see my mailing list post from yesterday), I
tried configuring chan_sip to work that way. My settings are:

[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.
tlsbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
tlsenable=yes
transport=udp
srvlookup=yes
tlscafile=/usr/local/etc/asterisk/keys/4cfd3c78.0
tlscapath=/usr/local/etc/asterisk/keys
tlsclientmethod=tlsv1
sipdebug = yes

register => tls://1234...@example.org:foo...@dev.example.org

[devtrunk]
type=peer
host=example.org
defaultuser=1234567
fromuser=1234567
remotesecret=foobar
transport=tls
outboundproxy=dev.example.org
context=carrier-in
encryption=yes

When I start up, I see my Asterisk doing a _sips._tcp SRV lookup, but
that's just for the registration, I guess. I also see it doing
_sip._udp SRV queries. I wouldn't know why it would have to do that.

The REGISTER packets are sent out via TLS, as I would expect.

When I issue a "sip show peer devtrunk" command, it tells me this:

  Prim.Transp. : TLS
  Allowed.Trsp : TLS

Looks okay to me. But when I place a call, Asterisk does this:

Reliably Transmitting (no NAT) to 2.3.4.5:5060:
INVITE sip:0123456...@example.org SIP/2.0
Via: SIP/2.0/UDP 9.8.7.6:0;branch=z9hG4bK2974d534

It sends the packet out via UDP, and to the wrong host, since it
doesn't use the correct SRV entry and instead sends it to the UDP
server.

I did not generate a certificate for my Asterisk, because it only acts
as a client. I think, this shouldn't be needed.

Can anyone point me to where I misconfigured something? Or did I
stumble upon a bug? What would I have to do to make Asterisk use the
open TLS connection used for registering for outbound calls, too?

Best Regards,
Sebastian

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[asterisk-users] Asterisk (PJSIP) registers with sips Contact URI, but why?

[Sebastian Damm]

Hi,

I'm registering an Asterisk against my TLS capable service, using
res_pjsip. My config looks like this:

[devtrunk_reg]
type=registration
outbound_auth=devtrunk_auth
server_uri=sip:example.org\;transport=tls
client_uri=sip:1234...@example.org\;transport=tls
outbound_proxy=sip:dev.example.org\;transport=tls\;lr
contact_user=1234567
retry_interval=60
expiration=600
line=yes
endpoint=222

[devtrunk_auth]
type=auth
auth_type=userpass
username=1234567
password=secret
realm=example.org


It registers fine, but this is what the REGISTER request looks like:

<--- Transmitting SIP request (903 bytes) to TLS:1.2.3.4:5061 --->
REGISTER sip:example.org;transport=tls SIP/2.0
Via: SIP/2.0/TLS
9.8.7.6:55664;rport;branch=z9hG4bKPjNlqlgmSOP7O4LqOTUqJtFZB8fTmc0ZKL;alias
Route: 
From: ;tag=vhDrzKtv9lMR53ZJFgVTnvGcACJiN6Aa
To: 
Call-ID: nzgHdLyliuBwecmae2Y..0oY2DqYjH0V
CSeq: 14861 REGISTER
Contact: 
Expires: 600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, MESSAGE, REFER, REGISTER
Max-Forwards: 70
User-Agent: Asterisk PBX 13.8.2
Content-Length:  0

What I really don't like is the Contact line. It starts with sips
instead of sip. This makes inbound calls not work because the server
sends a sip Contact header instead of sips. And Asterisk rejects that.

In the header of the 480 response I see this line:

Warning: 381 SIP "SIPS Required"

Since I can't reconfigure the server to send sips Contact URIs, I need
Asterisk to send out a contact URI in the register, that starts with
sip: as well. Then inbound calls would work.

Is there any way to get rid of this sips URI?

Interestingly, when sending out calls, the Contact URI starts with sip
instead of sips, so outbound calls work.

Best Regards,
Sebastian

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Re: [asterisk-users] constantly increasing load in Asterisk 11.14

[Sebastian Damm]

Hi,

On Thu, Feb 5, 2015 at 4:56 PM, Scott Griepentrog sgriepent...@digium.com
wrote:

 Can you tell me if the memory usage by Asterisk is also increasing with
 load over time?


Yes, the memory usage does rise a bit.

USER   PID %CPU %MEMVSZ   RSS TTY  STAT START   TIME COMMAND
11.14
root 26047  8.9  5.0 1302036 407640 ?  Sl   Jan28 1201:26 asterisk
-vvvgf
11.6
root 36336  9.9  1.1 881684 90360 ?Sl   Jan28 1328:19 asterisk
-vvvgf

Interestingly, the CPU graphs show no visible increase.

If you need any more information, just let me know.

Gareth, the slow increase of load over months was there with 11.5 or 11.6
already, but I can live with a restart every couple month. Once every week
is too much.
Here is another overview of the load of one machine: http://pbrd.co/1zeFwBy
You can see, the Asterisk was running from April to September without much
change in load, then was restarted. In November, we updated to 11.14, and
from that time, it looks a bit different (and Asterisk needed a lot more
restarts).

Best Regards,
Sebastian
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[asterisk-users] constantly increasing load in Asterisk 11.14

[Sebastian Damm]

Hi,

we have quite a few Asterisk machines running and try to keep them on a
current version of the Asterisk 11 branch. But since we upgraded to 11.14.0
a couple weeks ago, we have to restart the Asterisk process every week
because the load gets too high and our monitoring complains.

Those machines are doing only SIP-to-SIP call relay, the dialplan is quite
complex, transcoding is done only on a few percent of the calls processed.
During the daytime, there are at max around 200 SIP channels (100 calls)
running at the same time. After one week, one machine has processed about
170k calls.

I have uploaded a comparison of cacti load graphs for one week of a machine
running with 11.14.0 and one running with 11.6.0: http://pbrd.co/1v0SO3R

As you can see, after a restart, both machines have about the same load.
But after the really quiet weekend, the 11.14 Asterisk starts the new week
with a much higer load than the 11.6 Asterisk, where it stays constant.
We've had an 11.5.1 machine running for about half a year without the need
of restarting, but right now, this is not possible.

Has anyone seen this before? Or does anyone know a reason, what change
somewhere between 11.6 and 11.14 could cause this behaviour? It looks like
we have to go back to 11.6.

Best Regards,
Sebastian
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