Re: [asterisk-users] Deleting voicemail by program

[Michael Bradeen]

Hi Mike,

New AMI actions were recently added to app_voicemail to let you remotely
manipulate a mailbox:
https://github.com/asterisk/asterisk/issues/181

Hope this helps.

BR,
-Mike


On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl  wrote:

> Hi all,
>
> I need to be able to delete a voicemail message using a program.
>
> Is is sufficient to simply delete the .wav and .txt files in the spool
> directory?
> Or do I need to also renumber the remaining files?
>
> For example, let say a given mailbox has 20 messages in it and I want to
> delete message number 5.  Can I just delete the 2 files and expect that
> asterisk will renumber them?  Or do I need to?
>
> Also, is the answer the same when I migrate to storing voicemails in a
> database?
>
> Thanks in advance.
>
> Mike
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk Release 20.3.1

[Michael Maier]

At the moment, I can't see any differences here. sha512sum is identical.

Regards
Michael

On 08.07.23 at 01:50 Jean-Denis Girard wrote:

Le 07/07/2023 à 12:49, Joshua C. Colp a écrit :
On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard <mailto:jd.gir...@sysnux.pf>> wrote:


    There seems to be a problem with the tar.gz archive on github. It's
    correct on downloads.asterisk.org <http://downloads.asterisk.org>.

Can you be more specific? They are identical and the same tarball. I just 
downloaded both from each place and confirmed that, and confirmed they both 
extract fine.


Downloading from github (I tried 5 times), I get:
10353870  7 juil. 13:44 'asterisk-20.3.1(1).tar.gz'
10353870  7 juil. 13:46 'asterisk-20.3.1(2).tar.gz'
sha256sum is:
aec7271fda5eb1e185bb94f3f52977c636783bd456e9c361dd853cd0eba10203
Extracting is fine.

Downloading from asterisk.org, I get:
28176262  7 juil. 11:34  asterisk-20.3.1.tar.gz
5d7dea82b11ce97eec294ba0234c3a68fe2f05065c04a4279baa4a4442f4f628


Bien cordialement,



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

FYI: i've created a feature request to add SIP_CODEC_INBOUND equivalent 
functionality to chan_pjsip:


https://github.com/asterisk/asterisk-feature-requests/issues/9

Let's see where it goes

*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/5/23 11:58, Michael Ulitskiy wrote:


Hello,

Anyone? I have hard time to believe this is not possible with chan_pjsip.

Anyway, may I ask how people handle the following scenario which I 
imagine should be quite common:


- I have internal extensions talk to each other using g722. so their 
codec setting (with chan_sip now) is "allow=g722,ulaw"

- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its 
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence 
codec selection on calling channel and the calls set up using ulaw 
end-to-end


Can somebody please advise how to achieve the same with chan_pjsip?

Thanks,

Michael

On 6/30/23 09:30, Michael Ulitskiy wrote:


Hello,

I finally got to look at chan_sip to chan_pjsip migration again. This 
time I’m having problems with influencing codec selection on 
originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only 
works on outbound (called) channel and has no affect on calling 
channel. My experiments and function documentation (which says “Media 
and codec offerings to be set on an outbound SIP channel prior to 
dialing.”) seem to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s 
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s 
equivalent of ${SIP_CODEC_INBOUND}? Or, in other words, what are we 
supposed to do to influence /calling/ channel codec selection from 
dialplan?

I’m working with asterisk 20.3.0.

Thank you,
Michael



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

Oh, that's great. It wasn't clear from that page, at least not for me. :-(

Having it clearly stated on the document would save me (and probably 
others) lots of time.


Thanks for clarifying it. Any idea on the timeframe of implementation?

*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/6/23 12:47, Joshua C. Colp wrote:
On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy  
wrote:


Hello,

After I have re-read the "PJSIP Advanced Codec negotiation"
document, it occurred to me that the desired behavior should
actually happen automatically, just due to the codec negotiation
logic, but it looks like asterisk doesn't actually follow the
described logic which is likely a bug.


That functionality is not implemented as of this time.

--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com <http://www.sangoma.com> and 
www.asterisk.org <http://www.asterisk.org>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

Hello,

After I have re-read the "PJSIP Advanced Codec negotiation" document, it 
occurred to me that the desired behavior should actually happen 
automatically, just due to the codec negotiation logic, but it looks 
like asterisk doesn't actually follow the described logic which is 
likely a bug.


Can you please follow with me through a simple sip call and see if I'm 
missing something or asterisk actually doesn't do what it's supposed to do?


Here's the codec negotiation config:
CLI> pjsip show endpoint A
 ...
 codec_prefs_incoming_answer    : prefer:pending, 
operation:intersect, keep:all, transcode:allow
 codec_prefs_incoming_offer : prefer:pending, 
operation:intersect, keep:all, transcode:allow
 codec_prefs_outgoing_answer    : prefer:pending, 
operation:intersect, keep:all, transcode:allow
 codec_prefs_outgoing_offer : prefer:pending, operation:union, 
keep:all, transcode:allow


All endpoints have the same default config above.

Let's go over simplest scenario: A calls B.
A is configured with g722 and ulaw (allow=!all,g722,ulaw) and B is 
configured with ulaw and alaw (allow=!all,ulaw,alaw)


1. codec_prefs_incoming_offer: A sends INVITE to asterisk with codecs in 
SDP g722,g729,g711u,g711a:

...
m=audio 2266 RTP/AVP 9 18 0 8 101.
a=rtpmap:9 G722/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8

  - according to Advanced Codec Negotiation logic now we have:
    - pending=g722,g729,ulaw,alaw
    - configured=g722,ulaw
    Applying operation "intersect" the resulting resolved topology is 
"g722,ulaw" which is sent to the core


2. codec_prefs_outgoing_offer: Outgoing channel driver receives the 
offer from the core

    - pending=g722,ulaw
    - configured=ulaw,alaw
    Applying operation "union" the resulting resolved topology should 
be "g722,ulaw,alaw" which should be sent
    to B in the outgoing INVITE. What I see is actually sent in 
outgoing INVITE is "ulaw,alaw":

...
m=audio 41506 RTP/AVP 0 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
...
    So this is the 1st point where codec negotiation doesn't work as 
expected


3. codec_prefs_incoming_answer: B replies with "200 OK" which contains 
only ulaw codec:

...
m=audio 2226 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
...
   - pending: ulaw
   - configured: ulaw,alaw (it's result of step 2. it should be 
g722,ulaw,alaw but actually is ulaw,alaw as described in step 2)
   Applying operation "intersect" the resulting resolved topology is 
"ulaw" which is sent to the core


4. codec_prefs_outgoing_answer: asterisk replies "200 OK" back to A
   - pending: ulaw (from step 3)
   - configured: g722,ulaw (from step 1)
   Applying operation "intersect" the resulting resolved topology 
should be "ulaw". What I see is actually sent in

   "200 OK" is "g722,ulaw":
...
m=audio 43004 RTP/AVP 9 0 101.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
...

If I understand it correctly the result of codec negotiation in the 
above scenario should be ulaw in both call legs, thus avoiding 
transcoding, but actual asterisk behavior differs.


Am I missing something. What are your thoughts?

Thanks,

*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/5/23 11:58, Michael Ulitskiy wrote:


Hello,

Anyone? I have hard time to believe this is not possible with chan_pjsip.

Anyway, may I ask how people handle the following scenario which I 
imagine should be quite common:


- I have internal extensions talk to each other using g722. so their 
codec setting (with chan_sip now) is "allow=g722,ulaw"

- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its 
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence 
codec selection on calling channel and the calls set up using ulaw 
end-to-end


Can somebody please advise how to achieve the same with chan_pjsip?

Thanks,

Michael

On 6/30/23 09:30, Michael Ulitskiy wrote:


Hello,

I finally got to look at chan_sip to chan_pjsip migration again. This 
time I’m having problems with influencing codec selection on 
originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only 
works on outbound (called) channel and has no affect on calling 
channel. My experiments and function documentation (which says “Media 
and codec offerings to be set on an outbound SIP channel prior to 
dialing.”) seem to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s 
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s 
equivalent of ${SIP_CODEC_INBOUND}? Or, in other

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that.

It's so surprising that the issue so seemingly obvious and trivial 
hasn't been addressed yet that I wanted to query the collective wisdom 
of this list to verify my observations.


Thanks for github pointer.

Michael

On 7/5/23 16:46, aster...@phreaknet.org wrote:

On 7/5/2023 4:19 PM, Michael Ulitskiy wrote:


Hi Michael,

Thanks for the reply.

I was referring to the scenario you named as 'outbound broken'. I 
didn't get to look at inbound call behavior yet, as I got stuck with 
inability to avoid transcoding on outbound calls.


To be more specific the scenario is as follows:

1. a phone initiates a call offering g722,g711 to asterisk
2. asterisk creates outbound call to carrier offering g711 only 
(carrier only supports g711)

3. carrier accepts the call and outbound call leg is now running on g711
4. asterisk accepts a phone's call with g722 since it's allowed on 
phone's endpoint and was indicated as preferred in phone's INVITE and 
now initial call leg is running on g722, resulting in transcoding


This is very disappointing. Since developers announced their plans to 
drop chan_sip from future asterisk versions



It's already been removed and won't be in any future major releases.
If you still need chan_sip after removal, you can continue adding it 
from out of tree and building it. I maintain a working version of it 
out of tree.


I was under impression that chan_pjsip has reached feature paritiy 
with chan_sip.



It has mostly, but not completely, no.


What is needed is an ability to tell asterisk which codecs are 
allowed to be included in "200 OK" asterisk sends back to the phone. 
I guess we need to submit a feature request. How do we go about it 
these days?


I'm not sure about the particulars of this issue at all, but to answer 
the question at hand, there's a repo for it: 
https://github.com/asterisk/asterisk-feature-requests.-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

Hi Michael,

Thanks for the reply.

I was referring to the scenario you named as 'outbound broken'. I didn't 
get to look at inbound call behavior yet, as I got stuck with inability 
to avoid transcoding on outbound calls.


To be more specific the scenario is as follows:

1. a phone initiates a call offering g722,g711 to asterisk
2. asterisk creates outbound call to carrier offering g711 only (carrier 
only supports g711)

3. carrier accepts the call and outbound call leg is now running on g711
4. asterisk accepts a phone's call with g722 since it's allowed on 
phone's endpoint and was indicated as preferred in phone's INVITE and 
now initial call leg is running on g722, resulting in transcoding


This is very disappointing. Since developers announced their plans to 
drop chan_sip from future asterisk versions I was under impression that 
chan_pjsip has reached feature paritiy with chan_sip. What is needed is 
an ability to tell asterisk which codecs are allowed to be included in 
"200 OK" asterisk sends back to the phone. I guess we need to submit a 
feature request. How do we go about it these days?


Thanks,
Michael

On 7/5/23 14:59, Michael Maier wrote:

Hello Michael,

you are referring to the following behavior - did I get it correctly?:

outbound broken: asterisk offers g722 / g711 to provider (callee), 
callee answers g711. Asterisk now transcodes between caller and callee 
(g722 <-> g711).


inbound works: call from provider: g711 -> asterisk drops g722 and 
passes g711 to internal callee -> no transcoding.



As far as I know, there is no working solution as of now. I discussed 
this problem years ago already here but unfortunately nothing usable 
happened so far (which I would know off). The priority is not high 
enough. I need a solution, too. I understand that this behavior is a 
nogo if you have a lot of calls because transcoding is expensive.



Thanks
Michael



On 05.07.23 at 17:58 Michael Ulitskiy wrote:

Hello,

Anyone? I have hard time to believe this is not possible with 
chan_pjsip.


Anyway, may I ask how people handle the following scenario which I 
imagine should be quite common:


- I have internal extensions talk to each other using g722. so their 
codec setting (with chan_sip now) is "allow=g722,ulaw"

- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its 
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence 
codec selection on calling channel and the calls set up using ulaw 
end-to-end


Can somebody please advise how to achieve the same with chan_pjsip?

Thanks,

Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Maier]

Hello Michael,

you are referring to the following behavior - did I get it correctly?:

outbound broken: asterisk offers g722 / g711 to provider (callee), 
callee answers g711. Asterisk now transcodes between caller and callee 
(g722 <-> g711).


inbound works: call from provider: g711 -> asterisk drops g722 and 
passes g711 to internal callee -> no transcoding.



As far as I know, there is no working solution as of now. I discussed 
this problem years ago already here but unfortunately nothing usable 
happened so far (which I would know off). The priority is not high 
enough. I need a solution, too. I understand that this behavior is a 
nogo if you have a lot of calls because transcoding is expensive.



Thanks
Michael



On 05.07.23 at 17:58 Michael Ulitskiy wrote:

Hello,

Anyone? I have hard time to believe this is not possible with chan_pjsip.

Anyway, may I ask how people handle the following scenario which I 
imagine should be quite common:


- I have internal extensions talk to each other using g722. so their 
codec setting (with chan_sip now) is "allow=g722,ulaw"

- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its 
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence codec 
selection on calling channel and the calls set up using ulaw end-to-end


Can somebody please advise how to achieve the same with chan_pjsip?

Thanks,

Michael



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

Hello,

Anyone? I have hard time to believe this is not possible with chan_pjsip.

Anyway, may I ask how people handle the following scenario which I 
imagine should be quite common:


- I have internal extensions talk to each other using g722. so their 
codec setting (with chan_sip now) is "allow=g722,ulaw"

- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between internal extensions naturally happen over g722 as its 
their preferred codec
- for external calls I now set SIP_CODEC_INBOUND=ulaw to influence codec 
selection on calling channel and the calls set up using ulaw end-to-end


Can somebody please advise how to achieve the same with chan_pjsip?

Thanks,

Michael

On 6/30/23 09:30, Michael Ulitskiy wrote:


Hello,

I finally got to look at chan_sip to chan_pjsip migration again. This 
time I’m having problems with influencing codec selection on 
originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only 
works on outbound (called) channel and has no affect on calling 
channel. My experiments and function documentation (which says “Media 
and codec offerings to be set on an outbound SIP channel prior to 
dialing.”) seem to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s 
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s 
equivalent of ${SIP_CODEC_INBOUND}? Or, in other words, what are we 
supposed to do to influence /calling/ channel codec selection from 
dialplan?

I’m working with asterisk 20.3.0.

Thank you,
Michael


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

[Michael Ulitskiy]

Hello,

I finally got to look at chan_sip to chan_pjsip migration again. This 
time I’m having problems with influencing codec selection on originating 
(calling) channel. It looks like PJSIP_MEDIA_OFFER only works on 
outbound (called) channel and has no affect on calling channel. My 
experiments and function documentation (which says “Media and codec 
offerings to be set on an outbound SIP channel prior to dialing.”) seem 
to confirm it.
So PJSIP_MEDIA_OFFER is supposed to be (and it works) chan_pjsip’s 
equivalent of ${SIP_CODEC_OUTBOUND}, but what is chan_pjsip’s equivalent 
of ${SIP_CODEC_INBOUND}? Or, in other words, what are we supposed to do 
to influence /calling/ channel codec selection from dialplan?

I’m working with asterisk 20.3.0.

Thank you,
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TLS and NAT

[Michael Maier]

On 09.04.23 at 19:55 Steve Matzura wrote:

Thanks, Michael. A few questions:


Is [transport_name] a reserved word, or am I supposed to replace it with a name of 
my own, like '[did-transport]'?


Yes. You are free.

Some of the keywords I haven't seen before. Is ca_list_file supposed to be an 
aggregate of the public and private key?


ca_list_file is the list of all CAs the server should accept as valid (these are 
public keys - no private keys) like Let's encrypt e.g..


And what are the 'method,' 'tos' and 
'cos' keywords, which are commented out in your instructions?


Take a look here: 
https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample


Search for "tos=0"


Regards,
Michael


Otherwise, the rest is quite clear.


On 4/8/2023 12:35 PM, Michael Maier wrote:

Hello Steve,

use the following configuration for the transport and bind this transport to the 
trunk:


[transport_name]
type=transport
protocol=tls
bind=192.168.13.24 ; your bind IP
ca_list_file=/etc/pki/tls/certs/ca-bundle.crt
; method=tlsv1_2
verify_server=yes
allow_reload=no
;tos=0xb8
;cos=3
external_media_address=your.ext.host.name ; hostname pointing to your ext. IP
external_signaling_address=your.ext.host.name ; hostname pointing to your ext. 
IP
local_net=192.168.0.0/24 # your local net


Regards
Michael

On 07.04.23 at 17:25 Steve Matzura wrote:
I want to configure communication with my phone provider using TLS for all the 
obvious reasons. Since I'm behind a firewall, I'll be needing to do it with 
NAT. There are examples of UDP plus NAT in pjsip.conf, but none for TLS plus 
NAT. Would it be correct to set up the TLS transport stanza to look like the 
[transport-udp-nat] stanza example, replacing UDP with TLS in lines like 
'transport=tls' and 'protocol=tls', and including the lines for local_net, 
external_media_address and external_signaling_address?









--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TLS and NAT

[Michael Maier]

Hello Steve,

use the following configuration for the transport and bind this 
transport to the trunk:


[transport_name]
type=transport
protocol=tls
bind=192.168.13.24 ; your bind IP
ca_list_file=/etc/pki/tls/certs/ca-bundle.crt
; method=tlsv1_2
verify_server=yes
allow_reload=no
;tos=0xb8
;cos=3
external_media_address=your.ext.host.name ; hostname pointing to your 
ext. IP
external_signaling_address=your.ext.host.name ; hostname pointing to 
your ext. IP

local_net=192.168.0.0/24 # your local net


Regards
Michael

On 07.04.23 at 17:25 Steve Matzura wrote:
I want to configure communication with my phone provider using TLS for 
all the obvious reasons. Since I'm behind a firewall, I'll be needing to 
do it with NAT. There are examples of UDP plus NAT in pjsip.conf, but 
none for TLS plus NAT. Would it be correct to set up the TLS transport 
stanza to look like the [transport-udp-nat] stanza example, replacing 
UDP with TLS in lines like 'transport=tls' and 'protocol=tls', and 
including the lines for local_net, external_media_address and 
external_signaling_address?




--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] extensions.conf asterisk 18.8.0 question

[Michael Englehorn]

If you're on RHEL or CentOS or one of its descendants, I would check if SELinux 
is enforcing (`sestatus` or `cat /etc/selinux/config` and look for 
"SELINUX=enforcing"), if it is, you'll probably need to create a policy to 
allow the Asterisk context to execute rm and/or delete files.
I use `audit2why` and `audit2allow` in policycoreutils-devel (on CentOS) to 
generate SELinux policy modules.

-Michael Englehorn

‐‐‐ Original Message ‐‐‐
On Monday, January 10th, 2022 at 1:03 PM, Jerry Geis  
wrote:

> I am trying to run this command:
> exten => _4XX,n,System(/usr/bin/rm /tmp/test.incoming.txt)
> 

> From the log:
> Executing [402@smvoice-sip:7] System("SIP/103-0018", "/usr/bin/rm 
> /tmp/test.incoming.txt") in new stack
> 

> Is "rm" not an allowed command - the above file is not removed.
> -rw-rw-rw- 1 silentm silentm 3 Jan 10 14:02 /tmp/test.incoming.txt
> 

> Thanks!
> 

> Jerry

publickey - michael@englehorn.com - 0x8B2C043D.asc
Description: application/pgp-keys


signature.asc
Description: OpenPGP digital signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] voicemail message not deleted

[Michael Keuter]

> Am 26.07.2021 um 07:28 schrieb Fourhundred Thecat <400the...@gmx.ch>:
> 
> Hello,
> 
> I have this in my voicemail.conf:
> 
>  attach=yes
> 
>  delete=yes
> 
> I do get an email when new voicemail is received, and I do get the
> voicemail message as attachment.
> 
> However, the original message is not deleted from the sevber.
> 
> How do I delete the message, after it has been sent per email as
> attachment? I don't want to store messages on the server indefinitely.
> 
> thanks,
> 
> -- 

I think you need to set "delete=yes" as option per mailbox account. 

100 => 1234,Test,,,delete=yes

The global setting is only an example.

Michael

http://www.mksolutions.info




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] problems with natted phones

[Michael L. Young]

El jue, 8 de jul. de 2021 a la(s) 14:58, Marek Greško (mgres...@gmail.com)
escribió:


> The asterisk is connected to the internet with public static IP address.
>
> The pjsip config contains:
>
>
What does your transport config look like?

Take a look at this wiki page:
https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT

--
Michael L. Young
(elguero)
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

[Michael Maier]

On 30.06.21 at 23:17 Joshua C. Colp wrote:
> On Wed, Jun 30, 2021 at 1:36 PM Michael Maier  wrote:
> 
>>
>> Hello!
>>
>> Short question: Is it possible to set
>>
>> a=silenceSupp:off
>>
>> in the SDP for alaw / ulaw for fax calls?
>>
> 
> No.

Thanks for your kindly support!

Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk / pjsip: RTP - alaw - a=silenceSupp:off

[Michael Maier]

Hello!

Short question: Is it possible to set

a=silenceSupp:off

in the SDP for alaw / ulaw for fax calls?


Thanks
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk pjsip and NAT just doesn't work

[Michael Maier]

On 02.05.21 at 17:24 Michael Maier wrote:


On 02.05.21 at 10:08 Michael Maier wrote:


Hello!

I've just playing around some time to get NAT and pjsip running with asterisk 
18.3 and 18.4 (w/o any patches added). NAT should be used for connection to the 
trunk.


I wasn't able to get it working, because SDP address rewriting just doesn't work 
as it should.


The situation is like this (CentOS 7):

- Multihomed
- One small net for the trunk 10.15.13.17/32 as alias on one of the existing 
devices.

- TLS for SIP
- Added complete masquerading for this IP address
- added dnsmanager which provides the global IP address
- used direct media no
- used rewrite contact yes or no
- force rport: yes
- transport:
 external_media_address=external.mydom.org
 external_signaling_address=external.mydom.org
 bind to 10.15.13.17 (may I bind to a interface name?)


What are the problems?

Outbound calls:
- Biggest problem: even if the WAN IP is set everywhere correctly in the 
*initial* INVITE, it's *always* missing in the INVITE *after* the 407 Proxy auth 
request in the SDP. In the first Invite, the SDP was ok, in the second Invite, 
the SDP is broken (rewriting doesn't seem to happen). Such calls naturally are 
dropped by the ISP (ok, one of my providers seems to ignore the entry completely).


This problem disappeared as I tested on a simple system holding just one local IP 
address. Do you have any idea where it could be fixed in the code?


- Another problem is, that the given external IP just isn't used consistently, 
some times it's there - mostly not (always the same easy call setup). I suspect 
/ fear different behavior between reload and restart with same configuration.

>>

- I expect all IP addresses of mine in all sip headers have to be the WAN IP.


- Next finding: The via header in a simple Ack isn't rewritten, too. Seems all 
packages sent by pjsip itself don't know anything about NAT.


This problem persists even on the simple system. Where could it be fixed in the 
code?


- How can I check if NAT is involved at all?

- Maybe asterisk gets confused if it can see the WAN IP (it's on the system, too), 
but is bound to a local IP? But why are some headers written correctly and some 
wrong?


Inbound calls:
- Playing announcements doesn't work at all (no sound though rtp packages are 
flowing in both directions according tcpdump at the WAN interface).

- Calls given to local devices are working.



Could somebody maybe give me a reference configuration for a working NAT 
configuration?



Thanks
Michael



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk pjsip and NAT just doesn't work

[Michael Maier]

On 02.05.21 at 10:08 Michael Maier wrote:


Hello!

I've just playing around some time to get NAT and pjsip running with asterisk 18.3 
and 18.4 (w/o any patches added). NAT should be used for connection to the trunk.


I wasn't able to get it working, because SDP address rewriting just doesn't work 
as it should.


The situation is like this (CentOS 7):

- Multihomed
- One small net for the trunk 10.15.13.17/32 as alias on one of the existing 
devices.
- TLS for SIP
- Added complete masquerading for this IP address
- added dnsmanager which provides the global IP address
- used direct media no
- used rewrite contact yes or no
- force rport: yes
- transport:
 external_media_address=external.mydom.org
 external_signaling_address=external.mydom.org
 bind to 10.15.13.17 (may I bind to a interface name?)


What are the problems?

Outbound calls:
- Biggest problem: even if the WAN IP is set everywhere correctly in the *initial* 
INVITE, it's *always* missing in the INVITE *after* the 407 Proxy auth request in 
the SDP. In the first Invite, the SDP was ok, in the second Invite, the SDP is 
broken (rewriting doesn't seem to happen). Such calls naturally are dropped by the 
ISP (ok, one of my providers seems to ignore the entry completely).


- Another problem is, that the given external IP just isn't used consistently, 
some times it's there - mostly not (always the same easy call setup). I suspect / 
fear different behavior between reload and restart with same configuration.


- I expect all IP addresses of mine in all sip headers have to be the WAN IP.


- Next finding: The via header in a simple Ack isn't rewritten, too. Seems all 
packages sent by pjsip itself don't know anything about NAT.


- How can I check if NAT is involved at all?

- Maybe asterisk gets confused if it can see the WAN IP (it's on the system, too), 
but is bound to a local IP? But why are some headers written correctly and some wrong?




Inbound calls:
- Playing announcements doesn't work at all (no sound though rtp packages are 
flowing in both directions according tcpdump at the WAN interface).

- Calls given to local devices are working.



Could somebody maybe give me a reference configuration for a working NAT 
configuration?



Thanks
Michael



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk pjsip and NAT just doesn't work

[Michael Maier]

Hello!

I've just playing around some time to get NAT and pjsip running with asterisk 18.3 
and 18.4 (w/o any patches added). NAT should be used for connection to the trunk.


I wasn't able to get it working, because SDP address rewriting just doesn't work 
as it should.


The situation is like this (CentOS 7):

- Multihomed
- One small net for the trunk 10.15.13.17/32 as alias on one of the existing 
devices.
- TLS for SIP
- Added complete masquerading for this IP address
- added dnsmanager which provides the global IP address
- used direct media no
- used rewrite contact yes or no
- force rport: yes
- transport:
external_media_address=external.mydom.org
external_signaling_address=external.mydom.org
bind to 10.15.13.17 (may I bind to a interface name?)


What are the problems?

Outbound calls:
- Biggest problem: even if the WAN IP is set everywhere correctly in the *initial* 
INVITE, it's *always* missing in the INVITE *after* the 407 Proxy auth request in 
the SDP. In the first Invite, the SDP was ok, in the second Invite, the SDP is 
broken (rewriting doesn't seem to happen). Such calls naturally are dropped by the 
ISP (ok, one of my providers seems to ignore the entry completely).


- Another problem is, that the given external IP just isn't used consistently, 
some times it's there - mostly not (always the same easy call setup). I suspect / 
fear different behavior between reload and restart with same configuration.


- I expect all IP addresses of mine in all sip headers have to be the WAN IP.



Inbound calls:
- Playing announcements doesn't work at all (no sound though rtp packages are 
flowing in both directions according tcpdump at the WAN interface).

- Calls given to local devices are working.



Could somebody maybe give me a reference configuration for a working NAT 
configuration?



Thanks
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] pjsip transport and dynamic WAN IP address (not: NAT)

[Michael Maier]

On 01.05.21 at 07:13 Michael Maier wrote:


Hello all!

I'm actually wondering about how to achieve fast fail handling for the pjsip 
transport if underlying WAN IP address changes.


Forgot to mention:

- I'm using TLS!
- pjsip tries every 92s to send the Registration until the timeout comes up. I 
don't know where the 92s are coming from. The tls_keep_alive_interval is 90s - but 
that's not a keep alive package. The "normal" configured retry time if a 
Registration is not answered, is 60s (which is working fine).





Thanks
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] pjsip transport and dynamic WAN IP address (not: NAT)

[Michael Maier]

Hello all!

I'm actually wondering about how to achieve fast fail handling for the pjsip 
transport if underlying WAN IP address changes.



Following scenario:
Asterisk runs on a device holding ppp0, which provides the interface for outbound 
registration to ISP trunks (transport acts as client) and therefore the WAN IP.


If WAN IP changes after restarting ppp0 (and ppp0 device completely disappeared 
for some time), it takes pretty long until the transport realizes, that the WAN IP 
changed.


I tried dnsmanager to always provide correct external IP address in SIP requests 
like this


external_media_address : external.mydom.org
external_signaling_address : external.mydom.org
local_net  : 192.168.0.0/255.255.0.0

and I can see, that after IP address change, the SIP requests (Register e.g.) 
provide the correct WAN IP - but transport didn't realize the change on low level.


NOTICE[4061] dnsmgr.c: dnssrv: host 'external.mydom.org' changed from 46.r.x.y:0 
to 79.t.a.b:0


It takes quite some long time (~13 minutes), until pjsip / asterisk detects a 
timeout or a network unreachable.


tlsc0x7ffa800203a8 TLS transport destroyed with reason 120110: Connection timed 
out

After this point has been reached, the transport is destroyed and a new transport 
is generated - from now on, things are working fine again.


I added a global logging rule in iptables at first place in OUTPUT to see any 
packages which are handled - but I couldn't see any - therefore I think things are 
breaking even earlier.



My question: is there any possibility to programmatically force the restart of a 
given transport (I don't want to restart all transports)? This could be easily 
done in ppp up scripts. Or is there a configuration option to reduce the timeout 
until restart of the individual transport? Maybe in pjsip (connection timeout 
detection)?


Another idea would be to restart the relevant transports based on dnsmgr detecting 
new external IP.




Thanks
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Channel Variable inheritance

[Michael Munger]

I have a blacklisting system to deal with robocalls. Transferring a 
given call to extension *88 will add the CALLERID(num) to astdb, and 
when that number calls back, it goes straight to tt-monkeys.


Problem:

With Polycom phones, transfer -> blind -> *88 works just fine. But, 
transfer -> *88 (attended transfer) does not work. I assume that's 
because an attended transfer is creating a new channel, and the 
CALLERID(num) of the attended transfer is the CID of the station making 
the call. So, I decided to save the incoming CID into 
__ORIGINAL_CALLER_ID, with the assumption that the variable would be 
inherited into subsequent channels, but that does not work either.


What am I missing?

Asterisk: 13.14.1~dfsg-2+deb9u4

OS: Debian 9.13 (Stretch).

--

    
Michael Munger, dCAP, MCPS, MCNPS, MBSS
*Microsoft Certified Professional*
*Microsoft Certified Small Business Specialist*
*Digium Certified Asterisk Professional*
*High Powered Help, Inc.*
p:  678-905-8569
w:  hph.io <https://hph.io> e: m...@hph.io <mailto:m...@hph.io>



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

[Michael Maier]

On 18.02.21 at 20:01 Luca Bertoncello wrote:

Am 18.02.2021 um 18:59 schrieb Michael Maier:

On 17.02.21 at 21:46 Luca Bertoncello wrote:

Am 16.02.2021 um 22:32 schrieb Michael Maier:

Hi Michael


Maybe could you send me an abstract of your configuration?


Take a look here [1]


So, maybe I got it...
I tested the configuration with my Fax number and it seems to work (= I
can call the fax and can call my mobile phone from the fax with
"originate...").


Congrats!


So, it seems it does NOT work as expected...
I tried to activate the FAX and it works, then I activated my number and
it works, too.
Finally I activated the number of my wife and it does not work anymore...
If I call the number I can only see (verbose 42):

[Feb 18 19:57:12] NOTICE[19379] res_pjsip/pjsip_distributor.c: Request
'INVITE' from ''
failed for '217.0.21.64:5060' (callid: p65550t1613674632m753568c93349s2)
- No matching endpoint found


You have to do all of the configuration mentioned here[1] for *each* number. 
Afterwards, you have to route the incoming call to an internal device. As I'm 
using FreePBX, I don't know how to do it *correctly*.



and no phone rings...
After that, even if I restore the single number to SIP I only get the
error and nothing work, until I restored _ALL_ numbers to SIP.

Do someone has an explanation and (better!) a solution to the problem?


Solution:
You have the choice between: programming your PBX yourself (and have the struggle 
and pain) or let this pretty difficile job do others for you - they provide 
extremely good solutions for a lot of telephony features - it makes no sense to 
reinvent those features without having the required knowledge - so, use FreePBX. 
But it's of course your decision.



[1] 
https://www.ip-phone-forum.de/threads/hilfe-f%C3%BCr-grundeinstellung-asterisk-telekom-ben%C3%B6tigt.307115/post-2374234


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

[Michael Maier]

On 17.02.21 at 21:46 Luca Bertoncello wrote:
> Am 16.02.2021 um 22:32 schrieb Michael Maier:
> 
> Hi Michael
> 
>>> Maybe could you send me an abstract of your configuration?
>>
>> Take a look here [1]
> 
> So, maybe I got it...
> I tested the configuration with my Fax number and it seems to work (= I
> can call the fax and can call my mobile phone from the fax with
> "originate...").

Congrats!

> On the registration I have:
> 
> [pbxfax]
> type = registration
> retry_interval = 20
> max_retries = 10
> contact_user = 00493514977291
> expiration = 120
> transport = transport-udp
> outbound_auth = pbxfax
> client_uri = sip:03514977...@tel.t-online.de
> server_uri = sip:tel.t-online.de
> 
> First: can I use tel.t-online.de or _MUST_ I change it?

No, you mustn't change it. You must use tel.t-online.de.

> If I understand
> your previous E-Mail, I'd say that I can leave tel.t-online.de...

Correctly!

> Then I have a question by the Dialplan... Currently I have:
> 
> [fax-out]
> exten => _X.,1,NoOp()
> exten => _X.,n,Verbose(2,Call from FAX)
> exten => _X.,n,Dial(SIP/pbxfax/${EXTEN},,R)
> 
> And I'll replace it with:
> 
> [fax-out]
> exten => _X.,1,NoOp()
> exten => _X.,n,Verbose(2,Call from FAX)
> exten => _X.,n,Dial(PJSIP/pbxfax/sip:${EXTEN}@tel.t-online.de,,R)
> 
> Is it correct? I tried with
> "PJSIP/pbxfax/pjsip:${EXTEN}@tel.t-online.de,,R" and it does NOT work...
> Is it correct, that I have to leave "sip:..."?

Don't know - I don't care about dialplan - I'm using FreePBX :-)


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

[Michael Maier]

On 16.02.21 at 20:33 Luca Bertoncello wrote:

Am 16.02.2021 um 19:56 schrieb Michael Maier:

Hi Michael,


Do I use pjsip?


pjsip show registrations


gw*CLI> pjsip show registrations
No objects found.

So I don't use pjsip... :(


Yes.


Maybe could you send me an abstract of your configuration?


Take a look here [1]


You mean, I have to create a "fake" Zone tel.t-online.de in my Bind with
these settings? Looks like dangerous, if they changes something...


If you do that statically -> yes, you're right. You have to do it
dynamically. I attached a script, which can be used to dynamically build
a rpz each 15 minutes e.g. It directly asks the telekom nameserver for
naptr and srv entries. It looks like this:

server 192.168.62.13
zone rpz-tonline
update delete tel.t-online.de.rpz-tonline.
update delete _sips._tcp.tel.t-online.de.rpz-tonline.
update delete _sip._tcp.tel.t-online.de.rpz-tonline.
update add tel.t-online.de.rpz-tonline. 60  NAPTR   10 0 "s"
"SIPS+D2T" "" _sips._tcp.tel.t-online.de.
update add tel.t-online.de.rpz-tonline. 60  NAPTR   30 0 "s"
"SIP+D2T" "" _sip._tcp.tel.t-online.de.
update add _sips._tcp.tel.t-online.de.rpz-tonline.  60 SRV  10 0
5061 s-eps-110.edns.t-ipnet.de.
update add _sip._tcp.tel.t-online.de.rpz-tonline.   60 SRV  10 0
5060 s-epp-110.edns.t-ipnet.de.
send


So if I undestand what you mean, you check the NAPTR and SRV für
_sips._tcp.tel.t-online.de and save the record in a "virtual domain"
rpz-tonline, is it correct?
Then I suppose you use this domain instead of tel.t-online.de in the SIP
configuratione as "host", "outboundproxy" and "fromdomain", is it correct?


No - you have to use the correct domain name in asterisk.  Only bind knows about 
the fake domain. You have to configure bind correctly.


You have to create the fake domain in the bind config like this:
options {
...
response-policy {
zone "rpz-tonline";
};
};

...

zone "rpz-tonline" {
type master;
file "/var/named/rpz-tonline-override";
allow-query { any; };
allow-transfer { any; };
allow-update { any; };
};

All other things: take a look at the script! It's not that complicated.




The script unregisters and registers the telekom trunks, if a change is
detected. This is done as long as there is no call active. This works
for me - but may not wort for others - feel free to change the code.


OK, I'll check it...


Independently you have to add your own trunk names to get it working
(telekomPJSIP-a, ...).


Could you explain me that? I'm not an expert of Asterisk... :(


Well, if you want to use it, you really should engage yourself a bit more to get 
it solved. It's not that easy. Or you may forget about the DNS fake and live with 
the problem, that asterisk could partly switch sometimes to another server - 
breaking the telephony. I don't think it would happen that often, because Telekom 
usually is extremely stable. Try at first to get a running pjsip configuration. 
The DNS theme could be done later on.



Regards
Michael

[1] 
https://www.ip-phone-forum.de/threads/hilfe-f%C3%BCr-grundeinstellung-asterisk-telekom-ben%C3%B6tigt.307115/post-2374234


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

[Michael Maier]

Hi Luca,

On 15.02.21 at 21:48 Luca Bertoncello wrote:
> Am 15.02.2021 um 21:40 schrieb Michael Maier:
> 
> Hi Michael,
> 
>> They're switching to DNS NAPTR / SRV[1]. If you are using Asterisk /
>> pjsip and hostnames (tel.t-online.de e.g. for the AllIP service), you
> 
> Mmm... I'm using tel.t-online.de, but I'm not sure I'm using pjsip...
> 
> module show say me:
> 
> res_pjsip.so   Basic SIP resource
> 46 Running  core
> 
> Do I use pjsip?

pjsip show registrations

> You mean, I have to create a "fake" Zone tel.t-online.de in my Bind with
> these settings? Looks like dangerous, if they changes something...

If you do that statically -> yes, you're right. You have to do it
dynamically. I attached a script, which can be used to dynamically build
a rpz each 15 minutes e.g. It directly asks the telekom nameserver for
naptr and srv entries. It looks like this:

server 192.168.62.13
zone rpz-tonline
update delete tel.t-online.de.rpz-tonline.
update delete _sips._tcp.tel.t-online.de.rpz-tonline.
update delete _sip._tcp.tel.t-online.de.rpz-tonline.
update add tel.t-online.de.rpz-tonline. 60  NAPTR   10 0 "s"
"SIPS+D2T" "" _sips._tcp.tel.t-online.de.
update add tel.t-online.de.rpz-tonline. 60  NAPTR   30 0 "s"
"SIP+D2T" "" _sip._tcp.tel.t-online.de.
update add _sips._tcp.tel.t-online.de.rpz-tonline.  60 SRV  10 0
5061 s-eps-110.edns.t-ipnet.de.
update add _sip._tcp.tel.t-online.de.rpz-tonline.   60 SRV  10 0
5060 s-epp-110.edns.t-ipnet.de.
send

You have to configure bind to use the rpz for all lookup calls resolving
*.tel.t-online.de. I assume that the individual t-ipnet.de entries are
"statically" and therefore resolved directly (w/o rpz). But this could
be added to the script, too (would be a new rpz).

At the moment, I'm using only one DNS server for digging the NAPTR and
SRV entries - this could be enhanced to use 2 servers if the first
fails. If the first fails, the script currently stops and does nothing.
I assume, that the DNS server is stable.

The script unregisters and registers the telekom trunks, if a change is
detected. This is done as long as there is no call active. This works
for me - but may not wort for others - feel free to change the code.
Independently you have to add your own trunk names to get it working
(telekomPJSIP-a, ...).

You can verify if it's working by checking for entries like this in
journalctl:
Feb 16 19:35:46 myfw named[1516]: client @0x7ff574027bd0
192.168.62.13#25869 (tel.t-online.de): rpz QNAME NODATA rewrite
tel.t-online.de via tel.t-online.de.rpz-tonline
They are appearing at the moment asterisk starts a lookup.


Hope this helps!


Thanks
Michael


check-t-online.pl
Description: Perl program
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

[Michael Maier]

Hi!

On 15.02.21 at 08:43 Luca Bertoncello wrote:
> Hi list!
> 
> I received a letter from Deutsche Telekom where they say me, that I need
> to change "something" on my router until 28.02.2021, otherwise I cannot
> phone anymore.
> Since I use Asterisk and I don't have a router, I'm not sure what I need
> to do...
> In the letter there is an URL to "explain" how to change the
> configuration if I use a VoIP-phone, but they only say, that I don't
> have to use Port 5060, but Port 0...
> 
> Surely there are in this list someone other using Deutsche Telekom...
> Does someone of them understand what I should change in the Asterisk
> configuration?

They're switching to DNS NAPTR / SRV[1]. If you are using Asterisk /
pjsip and hostnames (tel.t-online.de e.g. for the AllIP service), you
won't have any problem (using asterisk 14 or higher), because it's
default. But you may have problems with the handling of the calls,
because Telekom needs the client always to use the same server for all
activities after the register has been done (the SRV entries contain 3
servers and asterisk will use them "randomly" if it detects a problem -
regardless which server of the list has been used for registration -
this won't work with Telekom and will lead to not working outbound calls
/ interrupted calls e.g.). This won't happen very often (because they
have been extremely stable in the past), but I could see it nevertheless
already. If you want to be really sure to not face this problem, you
have to create a workaround by adding a rpz zone e.g. with an own bind,
which is fed by an own job and presents asterisk just one server when
looking up the SRV entries after the NAPTR call. NAPTR / SRV works like
this (example for tel.t-online.de):

1. Search for the service names
dig noall +answer tel.t-online.de NAPTR
tel.t-online.de.5   IN  NAPTR   10 0 "s" "SIPS+D2T" ""
_sips._tcp.tel.t-online.de.
tel.t-online.de.5   IN  NAPTR   30 0 "s" "SIP+D2T" ""
_sip._tcp.tel.t-online.de.

2. Take the answer of the NAPTR output (TCP/TLS, TCP)
dig +noall +answer _sips._tcp.tel.t-online.de SRV
_sips._tcp.tel.t-online.de. 2234 IN SRV 10 0 5061
s-eps-110.edns.t-ipnet.de.
_sips._tcp.tel.t-online.de. 2234 IN SRV 20 0 5061
h2-eps-100.edns.t-ipnet.de.
_sips._tcp.tel.t-online.de. 2234 IN SRV 30 0 5061
d-eps-100.edns.t-ipnet.de.

dig +noall +answer _sip._tcp.tel.t-online.de SRV
_sip._tcp.tel.t-online.de. 3600 IN  SRV 30 0 5060
d-epp-100.edns.t-ipnet.de.
_sip._tcp.tel.t-online.de. 3600 IN  SRV 10 0 5060
s-epp-110.edns.t-ipnet.de.
_sip._tcp.tel.t-online.de. 3600 IN  SRV 20 0 5060
h2-epp-100.edns.t-ipnet.de.

Asterisk now must use always the same server for all activities to
Telekom - like register, invite, options - but that's not yet supported
by Asterisk - therefore you have to ensure, that asterisk always uses
the same server. Easiest way is to provide just one in the DNS answer ... .


Regards
Michael

[1]
https://geschaeftskunden.telekom.de/hilfe-und-service/online-services/hilfe-internetanschluss/telefonieanpassung#telekom

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink

[Michael L. Young]

- On Feb 5, 2021, at 11:18 AM, Michael L. Young  wrote: 

> - On Feb 4, 2021, at 4:26 PM, Social Boh  wrote:

>> The problem is with this CentOS 7 glibc version:

>> 2.17-317.el7

>> After the library update and system reboog,
>> gotoif Asterisk application, stop to working

>> Any hint to solve?

> Until it is resolved, you can do a 'yum history' and note the transaction ID 
> of
> the update. Then try running 'yum history undo [transaction id]'. That should
> roll you back to the previous glibc.

> Looks like Red Hat is already working on it:
> https://access.redhat.com/solutions/5778071

Here is the Bugzilla report for anyone on RHEL / CentOS 7: 
https://bugzilla.redhat.com/show_bug.cgi?id=1925204 

-- 
Michael Young 
(elguero) 
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink

[Michael L. Young]

- On Feb 4, 2021, at 4:26 PM, Social Boh  wrote: 

> The problem is with this CentOS 7 glibc version:

> 2.17-317.el7

> After the library update and system reboog,
> gotoif Asterisk application, stop to working

> Any hint to solve?

Until it is resolved, you can do a 'yum history' and note the transaction ID of 
the update. Then try running 'yum history undo [transaction id]'. That should 
roll you back to the previous glibc. 

Looks like Red Hat is already working on it: 
https://access.redhat.com/solutions/5778071 

-- 
Michael Young 
(elguero) 
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

[Michael Maier]

On 29.01.21 at 22:33 Ruisheng Peng wrote:

Thanks for the detailed explanation Michael.

I stop the current asterisk process (started by systemd), and restart it as
asterisk:

[asterisk@voip1 ~]$ strace -f -o /home/asterisk/strace.log asterisk -fmq
-vvv -C /etc/asterisk/asterisk.conf


from the log there was no attempt to even open the cert file.  I edited
/etc/asterisk/pjsip.conf to add a "method = tlsv1" line to the
transport-tls section. Rerun the strace command, and here the part re cert
files:

8189  stat("/home/asterisk/certs/asterisk.crt", {st_mode=S_IFREG|0640,
st_size=1

212, ...}) = 0

8189  geteuid() = 1002

8189  getegid() = 1002

8189  getuid()  = 1002

8189  getgid()  = 1002

8189  access("/home/asterisk/certs/asterisk.crt", R_OK) = 0

8189  stat("/home/asterisk/certs/asterisk.key", {st_mode=S_IFREG|0640,
st_size=8

91, ...}) = 0

8189  geteuid() = 1002

8189  getegid() = 1002

8189  getuid()  = 1002

8189  getgid()  = 1002

8189  access("/home/asterisk/certs/asterisk.key", R_OK) = 0

8189  socket(AF_INET, SOCK_STREAM, IPPROTO_IP) = 16

8189  setsockopt(16, SOL_SOCKET, 0x /* SO_??? */, [1], 4) = -1
ENOPROTOOPT (


I'm missing the "open" (or "openat") and the following "read" call - weren't there 
any or didn't you post them? These are the important calls! They will show, if the 
file is used at all or not (and possibly the reason, why it is not used - EACCESS 
e.g.).



Thanks
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

[Michael Maier]

On 29.01.21 at 06:41 Michael Maier wrote:


On 27.01.21 at 22:57 Ruisheng Peng wrote:

Thanks Michael for the suggestion!  I've installed strace and assigned one
of the endpoints (SOFTPHONE_B) to use transport-tls. Then run strace (as
user asterisk):

[asterisk@voip1 ~]$ strace asterisk -rx "module reload res_pjsip.so"


You should use strace like this as root and from the very beginning of the start 
of asterisk:


Sorry - my wrong - not necessarily as root - it should be started the same way and 
in the same context as it runs normally.




strace -f -o /tmp/strace.log asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf

-f means, to follow even forked processes, ... (see man page)
-o writes all the output to a file. You can search afterwards pretty easily for 
the file (or the open call).


You shouldn't do this in production but in the test environment!

You have to run it as long as the error has happened.


Thanks
Michael




--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

[Michael Maier]

On 27.01.21 at 22:57 Ruisheng Peng wrote:

Thanks Michael for the suggestion!  I've installed strace and assigned one
of the endpoints (SOFTPHONE_B) to use transport-tls. Then run strace (as
user asterisk):

[asterisk@voip1 ~]$ strace asterisk -rx "module reload res_pjsip.so"


You should use strace like this as root and from the very beginning of the start 
of asterisk:


strace -f -o /tmp/strace.log asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf

-f means, to follow even forked processes, ... (see man page)
-o writes all the output to a file. You can search afterwards pretty easily for 
the file (or the open call).


You shouldn't do this in production but in the test environment!

You have to run it as long as the error has happened.


Thanks
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 16.14.0 pjsip transport-tls cert parsing error

[Michael Maier]

On 26.01.21 at 21:12 Ruisheng Peng wrote:
> Hi,
> 
>   I'm experimenting with Asterisk-16.14.0 on a CentOS7 box, and run into
> problems loading the SSL certificate to establish transport-tls.  Tried
> self-signed certificate generated with ast_tls_cert under contrib/scripts
> and the one issued by Letsencrypt, both would bomb out with a parsing error:
> 
> [Dec  3 15:47:50] ERROR[11233] res_pjsip/config_transport.c: Transport:
> transport-tls: cert_file /home/asterisk/certs/asterisk.crt is either
> missing or not readable

It's missing or not readable! Take care, that the file access rights of
the file and the complete path are ok. Do a strace to verify, if the
file is really loaded at all.


Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk-users Digest, Vol 197, Issue 17

[Saint Michael]

Re: Get a SHAKEN Identity Token (Alexander Perkins)

Saint Michael 
1:06 PM (0 minutes ago)
to Asterisk
Please look at this
https://issues.asterisk.org/jira/browse/ASTERISK-28924
I have a solution that works for any version of Asterisk, if interested
contact me at venefax at the Google mail service.


On Sun, Jan 24, 2021 at 1:00 PM 
wrote:

> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Re: Get a SHAKEN Identity Token (Alexander Perkins)
>
>
> --
>
> Message: 1
> Date: Sat, 23 Jan 2021 20:30:42 -0500
> From: Alexander Perkins 
> To: Markus 
> Cc: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Get a SHAKEN Identity Token
> Message-ID:
> <
> callktp0j5hfj1ou+rhrxaevab_wvxqzcoon9s-kfpvzwt6m...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Markus.  Thanks a bunch!  I will try that out!
>
> On Fri, Jan 22, 2021 at 8:06 AM Markus  wrote:
>
> > Am 07.01.2021 um 23:49 schrieb Alexander Perkins:
> > > Hi All.  We have old Asterisk servers, 1,89, (we cannot upgrade because
> > > of several reasons) and we are now implementing SHAKEN via our
> > > provider.  We place a SIP call to our provider and they return a 302
> > > (information below).  I am trying to get the X-Identity information
> > > below, but I do not seem to be able to do so.  Can somebody help me
> with
> > > this?  Any suggestions on how to get it?
> >
> > I use SIP_HEADER to extract information from inbound SIP packets and
> > SIPAddHeader to copy that info to the outgoing call leg. Maybe this
> > helps you?
> >
> > Example:
> >
> > exten => _+X.,1,NoOp(${CALLERID(num)})
> > exten => _+X.,n,Set(PAI=${SIP_HEADER(P-Asserted-Identity)})
> > exten => _+X.,n,Set(PAI=${CUT(PAI,:,2)})
> > exten => _+X.,n,Set(PAI=${CUT(PAI,@,1)})
> > exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "anonymous"]?anonymous:cli)
> > exten => _+X.,n(anonymous),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> > )
> > exten => _+X.,n,SIPAddHeader(Privacy: user\;id)
> > exten => _+X.,n,Goto(dial)
> > exten => _+X.,n(cli),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> > )
> > exten => _+X.,n,SIPAddHeader(Privacy: id)
> > exten => _+X.,n,Goto(dial)
> >
> >
> >
> -- next part --
> An HTML attachment was scrubbed...
> URL: <
> http://lists.digium.com/pipermail/asterisk-users/attachments/20210123/a26b80f1/attachment-0001.html
> >
>
> --
>
> Subject: Digest Footer
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
>
> End of asterisk-users Digest, Vol 197, Issue 17
> ***
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Get a SHAKEN Identity Token (Alexander Perkins)

[Saint Michael]

Please look at this
https://issues.asterisk.org/jira/browse/ASTERISK-28924
I have a solution that works for any version of Asterisk, if interested
contact me at venefax at the Google mail service.

On Sun, Jan 24, 2021 at 1:00 PM 
wrote:

> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Re: Get a SHAKEN Identity Token (Alexander Perkins)
>
>
> --
>
> Message: 1
> Date: Sat, 23 Jan 2021 20:30:42 -0500
> From: Alexander Perkins 
> To: Markus 
> Cc: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Get a SHAKEN Identity Token
> Message-ID:
> <
> callktp0j5hfj1ou+rhrxaevab_wvxqzcoon9s-kfpvzwt6m...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Markus.  Thanks a bunch!  I will try that out!
>
> On Fri, Jan 22, 2021 at 8:06 AM Markus  wrote:
>
> > Am 07.01.2021 um 23:49 schrieb Alexander Perkins:
> > > Hi All.  We have old Asterisk servers, 1,89, (we cannot upgrade because
> > > of several reasons) and we are now implementing SHAKEN via our
> > > provider.  We place a SIP call to our provider and they return a 302
> > > (information below).  I am trying to get the X-Identity information
> > > below, but I do not seem to be able to do so.  Can somebody help me
> with
> > > this?  Any suggestions on how to get it?
> >
> > I use SIP_HEADER to extract information from inbound SIP packets and
> > SIPAddHeader to copy that info to the outgoing call leg. Maybe this
> > helps you?
> >
> > Example:
> >
> > exten => _+X.,1,NoOp(${CALLERID(num)})
> > exten => _+X.,n,Set(PAI=${SIP_HEADER(P-Asserted-Identity)})
> > exten => _+X.,n,Set(PAI=${CUT(PAI,:,2)})
> > exten => _+X.,n,Set(PAI=${CUT(PAI,@,1)})
> > exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "anonymous"]?anonymous:cli)
> > exten => _+X.,n(anonymous),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> > )
> > exten => _+X.,n,SIPAddHeader(Privacy: user\;id)
> > exten => _+X.,n,Goto(dial)
> > exten => _+X.,n(cli),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> > )
> > exten => _+X.,n,SIPAddHeader(Privacy: id)
> > exten => _+X.,n,Goto(dial)
> >
> >
> >
> -- next part --
> An HTML attachment was scrubbed...
> URL: <
> http://lists.digium.com/pipermail/asterisk-users/attachments/20210123/a26b80f1/attachment-0001.html
> >
>
> --
>
> Subject: Digest Footer
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
>
> End of asterisk-users Digest, Vol 197, Issue 17
> ***
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk-users Digest, Vol 197, Issue 7

[Saint Michael]

Stir Shaken
Asterisk cannot do that, but my company can give you Stir Shaken for
Asterisk, via ODBC, any version.
Please contact me via email venefax at the google mail system
Philip Orleans

On Fri, Jan 8, 2021 at 1:00 PM 
wrote:

> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Get a SHAKEN Identity Token (Alexander Perkins)
>2. Re: Get a SHAKEN Identity Token (Joshua C. Colp)
>
>
> --
>
> Message: 1
> Date: Thu, 7 Jan 2021 17:49:27 -0500
> From: Alexander Perkins 
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Get a SHAKEN Identity Token
> Message-ID:
>  hxaeuyw+lx0guee9smj-0su1hqge8s...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi All.  We have old Asterisk servers, 1,89, (we cannot upgrade because of
> several reasons) and we are now implementing SHAKEN via our provider.  We
> place a SIP call to our provider and they return a 302 (information
> below).  I am trying to get the X-Identity information below, but I do not
> seem to be able to do so.  Can somebody help me with this?  Any suggestions
> on how to get it?
>
> Thank you, All.  Very much appreciated!
>
> <--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
> SIP/2.0 302 STIR/SHAKEN
> Via: SIP/2.0/UDP
>
> XXX.XXX.XXX.XXX:5066;received=XXX.XXX.XXX.XXX;branch=z9hG4bK37b49c97;rport=5066
> From: "12125551212" ;tag=as0026c4e3
> To:  >;tag=bcaa-20103108495689bb4065d39c43badb69
> Call-ID: 6755a4484427f12b0e56d6903fe50...@xxx.xxx.xxx.xxx:5066
> CSeq: 102 INVITE
> X-Identity:
>
> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9jZXJ0aWZpY2F0ZXMuY2xlYXJpcC5jb20vZmRmYjMzMjgtYjc1NC00YTBkLThiMzQtZGUzMGIwOGFkYWMyLzQ3NmMyODliYTgxY2QxNWU3MjBmNzkxOWM5NGU5MzU2LmNydCJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMjU2NTI3NjIwMSJdfSwiaWF0IjoxNjEwMDU2MDE3LCJvcmlnIjp7InRuIjoiMTI1NjkwNjQ5NTUifSwib3JpZ2lkIjoiMGFlODFjZWQtYzhlZS00ZWFiLTliNjAtMDY3OWM0Y2Q1MjUwIn0.lr3uj0fmlHbSori-msdbvKu5SQrVnLA-ZMswCY_dLk79jrpr1yFhWmL4GiAr16VtMKVSamQ-0bi3Pptoi7TUfw;info=<
>
> https://certificates.clearip.com/fdfb3328-b754-4a0d-8b34-de30b08adac2/476c289ba81cd15e720f7919c94e9356.crt
> >;alg=ES256;ppt=shaken
> Server: TILTX Technology Innovation Lab SHAKEN
> Content-Length: 0
> -- next part --
> An HTML attachment was scrubbed...
> URL: <
> http://lists.digium.com/pipermail/asterisk-users/attachments/20210107/566f39cf/attachment-0001.html
> >
>
> --
>
> Message: 2
> Date: Thu, 7 Jan 2021 19:00:13 -0400
> From: "Joshua C. Colp" 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Get a SHAKEN Identity Token
> Message-ID:
> <
> cam0a2z2fyurcpi7_bh3gzjhkvx2jlqqcbxnzcrhhulbbbhj...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> On Thu, Jan 7, 2021 at 6:50 PM Alexander Perkins <
> alexanderhenryperk...@gmail.com> wrote:
>
> > Hi All.  We have old Asterisk servers, 1,89, (we cannot upgrade because
> of
> > several reasons) and we are now implementing SHAKEN via our provider.  We
> > place a SIP call to our provider and they return a 302 (information
> > below).  I am trying to get the X-Identity information below, but I do
> not
> > seem to be able to do so.  Can somebody help me with this?  Any
> suggestions
> > on how to get it?
> >
> > Thank you, All.  Very much appreciated!
> >
> > <--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
> > SIP/2.0 302 STIR/SHAKEN
> > Via: SIP/2.0/UDP
> >
> XXX.XXX.XXX.XXX:5066;received=XXX.XXX.XXX.XXX;branch=z9hG4bK37b49c97;rport=5066
> > From: "12125551212"  :5066>;tag=as0026c4e3
> > To:  > >;tag=bcaa-20103108495689bb4065d39c43badb69
> > Call-ID: 6755a4484427f12b0e56d6903fe50...@xxx.xxx.xxx.xxx:5066
> > CSeq: 102 INVITE
> > X-Identity:
> >
> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9jZXJ0aWZpY2F0ZXMuY2xlYXJpcC5jb20vZmRmYjMzMjgtYjc1NC00YTBkLThiMzQtZGUzMGIwOGFkYWMyLzQ3NmMyODliYTgxY2QxNWU3MjBmNzkxOWM5NGU5MzU2LmNydCJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMjU2NTI3NjIwMSJdfSwiaWF0IjoxNjEwMDU2MDE3LCJvcmlnIjp7InRuIjoiMTI1NjkwNjQ5NTUifSwib3JpZ2lkIjoiMGFlODFjZWQtYzhlZS00ZWFiLTliNjAtMDY3OWM0Y2Q1MjUwIn0.lr3uj0fmlHbSori-msdbvKu5SQrVnLA-ZMswCY_dLk79jrpr1yFhWmL4GiAr16VtMKVSamQ-0bi3Pptoi7TUfw;info=<
> >
> https://certificates.clearip.com/fdfb3328-b754-4a0d-8b34-de30b08adac2/476c289ba81cd15e720f7919c94e9356.crt
> > >;alg=ES256;ppt=shaken
> > 

Re: [asterisk-users] Asterisk 18.0.0 Now Available

[Michael Maier]

On 21.10.20 at 12:49 Joshua C. Colp wrote:
> On Wed, Oct 21, 2020 at 7:46 AM Michael Maier  wrote:
> 
>> Hello!
>>
>> On 20.10.20 at 14:00 Asterisk Development Team wrote:
>>> The Asterisk Development Team would like to announce the release of
>> Asterisk 18.0.0.
>>> This release is available for immediate download at
>>> https://downloads.asterisk.org/pub/telephony/asterisk
>>
>> I just tested the new codec negotiation feature and unfortunately wasn't
>> able to get it working as expected. I tried several configurations - but
>> none has been working - the result
>> has always been the same.
>>
> 
> This is expected right now. Foundational aspects were put in, but there is
> still work to be done for PJSIP which will land in a future release.

Oh - thanks for the information - I missed this :-(. How do I know if this 
feature is finally enabled? Will it be in asterisk 18 - or will it come in some 
later major version?

> The
> complexity of it and the investigation of how things work, interactions,
> etc took considerably longer than expected. If there's specific scenarios
> that you'd like to ensure are met you can reach out on the asterisk-dev
> mailing list and George Joseph will add them to the list if not already
> present.

Well, I think the scenario I have should be a very easy and basic scenario. I 
already discussed it in the past. Therefore I think it's not necessary to add 
it again.


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 18.0.0 Now Available

[Michael Maier]

Hello!

On 20.10.20 at 14:00 Asterisk Development Team wrote:
> The Asterisk Development Team would like to announce the release of Asterisk 
> 18.0.0.
> This release is available for immediate download at
> https://downloads.asterisk.org/pub/telephony/asterisk

I just tested the new codec negotiation feature and unfortunately wasn't able 
to get it working as expected. I tried several configurations - but none has 
been working - the result
has always been the same.

Use case:
Alice calls Bob - sends INVITE  G722 / alaw / ulaw

Configured in Asterisk for this device: G722 / alaw / ulaw / gsm
A:
codec_prefs_incoming_offer = prefer: configured, operation: intersect, keep: 
all, transcode: prevent


Bob:
Configured in Asterisk for this device: alaw / ulaw
B:
codec_prefs_outgoing_offer = prefer: configured or pending, operation: 
intersect, keep: first or all, transcode: prevent
Asterisk sends INVITE to Bobalaw / ulaw


Asterisk receives OK from Bob   alaw
B:
codec_prefs_incoming_answer = prefer: configured or pending, operation: 
intersect, keep: first or all, transcode: prevent

Asterisk sends OK to Alice  G722 / alaw / ulaw
A:
codec_prefs_outgoing_answer = prefer: pending, operation: intersect, keep: 
first or all, transcode: prevent

=> I would have expected alaw to be sent to A - but G722 / alaw / ulaw is sent 
and transcoding is active!


What did I do wrong?
Could you please add the correct configuration you expect to get the expected 
result alaw?



Thanks
Kind regards
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] linphone calls not missed due to cause not 487

[Michael Maier]

On 16.10.20 at 11:07 sergio wrote:
> On 16/10/2020 10:11, Michael Maier wrote:
>>> Sometimes, linphone shows missed calls as missed.
>> You could try to reproduce it
> 
> I can't reproduce it, it happens less than once a month.

Then you should enable the tracing as I wrote in the previous post.

Regards
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] linphone calls not missed due to cause not 487

[Michael Maier]

Hi Sergio



On 16.10.20 at 07:54 sergio wrote:

> Sometimes, linphone shows missed calls as missed. Look like asterisk

> replies with cause=487 that time, but I can't understand why.

>

> Grandstream always shows calls as missed ones.

>

> How should I investigate this?

You could try to reproduce it while activating pcap traces and analyze
it afterwards - or you could enable pcap traces on asterisk[1] itself
and just wait for the different issues and compare them.





Regards

Michael



[1] activating pcap traces on asterisk / pjsip

pjsip set logger on

pjsip set logger verbose off

pjsip set logger pcap trace.pcap

You can find the trace.pcap in /var/lib/asterisk/trace.pcap (maybe it's
elsewhere on your installation)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] BLF support in Asterisk and early/confirmed/terminated/proceeding NOTIFY states.

[Michael Keuter]

> Am 03.10.2020 um 03:26 schrieb Jobst Schmalenbach :
> 
> I have a setup with Yealink phones & Asterisk Server (all latest patches).
> 
> I am using BLF to display the states of other phones. While this works MOST 
> of the time (busy, being called) it does NOT work when a phone is NOT 
> regisstered at all, the yealink phones display a green dot EVEN if a phone is 
> turned off (try explain this to users, they are shaking their heads!!!)
> 
> I can see on the Asterisk server it shows correctly in the logs when a phone 
> is disconnected.
> It also advises the otehr phones correctly when a phone is busy, even if a 
> person starts dialing - the red dot shows up milliseconds later on ALL of the 
> other phones.
> 
> I have asked a question about this before Green Dot
> 
> So I went and asked Yealink about this. The reply was something like this: 
> “Currently the phone can only support the BLF LED display with 
> early/confirmed/terminated/proceeding NOTIFY states.”
> 
> My question now is can I implement this properly on an Asterisk server?
> I.e. when a phone gets disconnected that all other phones are advised “hey I 
> am not available” and actually show a RED dot.
> thanks
> 
> -- 

Hi Jobst,

there is a setting for the Yealink phones so that the BLF keys are "off" 
instead of "green" when idle:

BLF LED Mode = 1

in the provisioning files:

features.blf_led_mode = 1

Hope that helps.

Michael

http://www.mksolutions.info




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk-users Digest, Vol 193, Issue 15

[Saint Michael]

memory vs disk cache

> This is an issue that has plagued Asterisk since day one. Basically there
>> is no solution available because there is no way to set aside memory to be
>> kept from a growing disk cache. I did some research and this looks like a
>> bad design from the Kernel people. Meanwhile all you can do us every 60
>> seconds:
>
> echo 3 | sudo tee /proc/sys/vm/drop_caches
> Asterisk should be able to reserve memory and force it to stay locked in
> memory, exactly like Mariadb does with
> memlock=1
> Would the Asterisk developers consider something like this?
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] 1. memory issues (hw)

[Saint Michael]

>
> This is an issue that has plagued Asterisk since day one. Basically there
> is no solution available because there is no way to set aside memory to be
> kept from a growing disk cache. I did some research and this looks like a
> bad design from the Kernel people. Meanwhile all you can do us every 60
> seconds:

echo 3 | sudo tee /proc/sys/vm/drop_caches
Asterisk should be able to reserve memory and force it to stay locked in
memory, exactly like Mariadb does with
memlock=1
Would the Asterisk developers consider something like this?



>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts

[Michael Maier]

On 05.09.20 at 15:22 sean darcy wrote:
> asterisk-16.13.0-rc2. Fedora 32
> 
> pjsip won't load because of undefined symbols:

This means, that your pjsip library doesn't match the asterisk binary. It's 
best to remove the independent pjsip library and compile asterisk[1] with the 
bundled pjsip library. Doing
it this way ensures that pjsip and asterisk match for sure (and some additional 
patches are applied to pjsip on top regarding usage of pjsip in asterisk).


Greetings
Michael

[1] https://downloads.asterisk.org/pub/telephony/asterisk

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Stir Shaken

[Saint Michael]

I need to point out the this is factually misleading and materially false:
"I think this, being the basis of your whole argument, is the fallacy.
S/S is forcing people to take responsibility, for sure, but carriers
won't just let their customers leave because they don't want to sign
calls.  It will force them to make sure they know who their customers
are, and make it impossible for those customers to escape consequences if
they misbehave."

There is Law of The Land that is about to take effect. Use google and
search "stir shaken" Whoever thinks I am exaggerated is dreaming. Also: it
is true that my service is the only one for asterisk --worldwide. The model
proposed by Transexus (302 redirect with a new header) can't be used by
Asterisk.
But don't take my word for it:
https://issues.asterisk.org/jira/browse/ASTERISK-28924




>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Stir Shaken

[Saint Michael]

>
> There is a big confusion here about Stir Shaken. It is NOT a provider
> issue. Un fact, all providers are whasing their hands and modifying their
> swihtches to pass-through the Signature. They cannot sign the call because
> then the become the responsible party for the call before the FCC, and
> liable for any illegal call. Every owner of a PBX that sends calls to the
> network, except if you use a trunk for the likes of Vonage, needs to sign
> their calls. So if you send calls with any kind of dialer and use DIDs,
> real or "borrowed", you need to get the signature service urgently or your
> business will stop terminating calls. You cannot self-sign, you cannot get
> around it, the calls will either go to straight to voicemail or fail. Even
> worse, the carries wil play a fake voicemail and charge you a fee,
> something that some already a are doing when they detect robocallig.

Don't even think about Transnexus, because they use 302 Redirect with a
header, and no version of Asterisk supports it.  I am the only game in the
world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is
literally true. If you need to sign your calls to get through, with
Asterisk, you need to connect to my service. I am an approved Service
Provider from the FCC. If you keep thinking this is not happening, it is,
and your business will disappear overnight.
The issue is that Vicidial, for example, does not provide res_odbc and
func_odbc, so you need to solve that first with Vicidial. Then you can
apply the code I provided earlier and your calls with have a legal, binding
signature. The carriers verify each signature and discard the ones that
fail the cryptography test.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stir Shaken is upon us

[Michael Maier]

On 13.07.20 at 10:54 Joshua C. Colp wrote:
> On Sun, Jul 12, 2020 at 11:37 PM Michael Maier  wrote:
>> One more question,
>> what about the pjsip pcap support? Will it be backported to Asterisk 16,
>> too? Would be absolutely cool! Debugging encrypted SIP is really a pain.
>>
> 
> It can't be backported ... because it already is. :D This support is
> actually in the latest releases of 13, 16, and 17.

This is perfectly good news! How often would I have it already needed in the 
past! Thanks!

Just to be sure:

pjsip set logger pcap  (written to /var/lib/asterisk/)
pjsip set logger on (switches on logging to file and console)
pjsip set logger off (switches off logging to file and console)

Is it possible to log only to the file and not to the console?

>>
>> BTW: what about the (encrypted) RTP packets? Will they be dumped, too?
>>
> 
> Not yet supported but certainly something we'd like to see as well as the
> RTCP, ICE, STUN, TURN, and DTLS packets.

Would be absolutely necessary to debug broken encrypted packets.


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Stir Shaken is upon us

[Michael Maier]

On 13.07.20 at 00:17 Joshua C. Colp wrote:
> On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins  wrote:
> 
>> Asterisk 18 will have support based on this asterisk update Matt F did for
>> CommCon's sponsor slots
>>
>> https://youtu.be/eas1csaX-wc
>>
>>
> As well support will go into Asterisk 16 and 17 as well. It's just been
> under active development so we've been waiting for that to finish before
> bringing it back into those versions.

One more question,
what about the pjsip pcap support? Will it be backported to Asterisk 16, too? 
Would be absolutely cool! Debugging encrypted SIP is really a pain.

BTW: what about the (encrypted) RTP packets? Will they be dumped, too?


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Stir Shaken is upon us

[Saint Michael]

WORLDWIDE EMERGENCY
The code below needs to be executed before any SIP or PJSIP call destined
to the US network, or soon no call will terminate. This is called
Stir-Shaken, a new law from the FCC.
If this is not working the whole Asterisk industry will crash, vanish, be
gone. I am assuming that the caller ID and the Destination Number are in
the variables "${CALLERID(num):-10}" "${EXTEN:-11}"

;Dialplan section to execute before any Dial
[strshk]
exten =>
_X.,1,Set(ARRAY(Token)=${MYSQL_STRSHK(${CALLERID(num):-10},${EXTEN:-11})})
;same=n,Verbose(0,Token ${Token})
;same=n,SIPAddHeader(Identity:${Token}) ;OLD SIP CHANNEL
same=n,Set(PJSIP_HEADER(add,Identity)=${Token}) ; NEW PJSIP CHANNEL
 same=n,Return()

/etc/odbcinst.ini or /etc/unixODBC/odbcinst.ini
[ODBC]
Trace=No
Trace File=/tmp/sql.log
Pooling=yes

[maria]
Description=ODBC for MySQL
Driver=/usr/lib64/libmaodbc.so
FileUsage=1
Threading=0

/etc/odbc.ini or /etc/unixODBC/odbc.ini
[strshk]
Description = MySQL ODBC Driver Testing
Driver = maria
Server = 208.73.232.47
#free testing service
User = anonymous
Password =
Database = strshk
Option = 3

res_odbc.conf
[strshk]
enabled=yes
dsn=strshk
sanitysql => select 1
isolation => read_uncommitted
username=anonymous
password=
pre-connect => yes
forcecommit => yes
connect_timeout => 10
negative_connection_cache => 300
max_connections=100
database=strshk

func_odbc.conf
[STRSHK]
escapecommas=yes
prefix=MYSQL
dsn=strshk
readsql=call strshk.stir_shaken_signature('${ARG1}','${ARG2}')
escapecommas=yes

Of course, you need to compile the modules res_odbc and func_odbc, which I
have done for Vicidial using Asterisk 13. But any Asterisk 11 and up can
use unixODBC.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice broken during calls (again...)

[Michael Maier]

On 03.07.20 at 19:57 Luca Bertoncello wrote:
[...]
>> On the Gateway (Banana PI), where the Asterisk server also runs, the
>> load is about 0.50 during calls and it has a Gbps LAN.
>> I can't believe, the problem is here...
> 
> So, now I know what was the problem and I solved it...
> 
> The problem was: the Banana PI... :(

Glad you could find and solve the problem.

> I checked it with mtr and I see really bad times to communicate with
> other devices im same networks (~2 - 380 ms!!).
> Many tries with other Switch ports and so on didn't solved the problem.

Yeah, that's what I already thought for myself. VoIP is (based on its realtime 
nature) extremely picky about network interfaces (or even complete hardware of 
the system) and their
drivers and the corresponding configuration. But most of the people can't / 
won't believe it.

Many of them (NICs) are pretty broken (sometimes the nic hardware, sometimes 
both hard- and software). Even APU 1 or 2 devices don't work reliably for VoIP 
with the standard in
kernel drivers or with default configuration here. I always had / have to use 
other drivers / kernels / configurations to get a proper and reliable rtp 
stream - even over hours.


Regards
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice broken during calls (again...)

[Michael Maier]

Am 24.06.20 um 08:10 schrieb Luca Bertoncello:

Am 24.06.2020 05:05, schrieb Michael Maier:

Hi


Your basic architecture looks good to me - now you have to start the


Nice to hear it...


analysis of the problem with pcapsipdump and wireshark as I wrote
before to get an idea what actually happens at
all. You most probably won't come any further without doing any
analyzing. You have to learn it. It will take some, or even more,
time. You can't do it in just few hours or maybe
even days or weeks. It is work or even hard work to learn and to do it.


Well, that's the very problem...
I don't know *how* to analyze it... Or, better, how to read the data...
I know, I can use tcpdump, sngrep and many other tools, but I don't know what I have to expect and how to decide, that a 
paket is wrong...

Can someone help me to learn it?

Google is your friend as usual. Try *for example* those search patterns as 
*entry point*:
wireshark rtp stream analysis
wireshark voip mitschneiden

https://support.yeastar.com/hc/en-us/articles/360007606533-How-to-Analyze-SIP-Calls-in-Wireshark
https://www.innosoft.at/news/169/voip-grundlagen-wireshark-analyse-von-sip-telefonie
https://sharkfestus.wireshark.org/sharkfest.12/presentations/BI-7_VoIP_Analysis_Fundamentals.pdf


Regards
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice broken during calls (again...)

[Michael Maier]

On 23.06.20 at 21:10 Luca Bertoncello wrote:
> Am 23.06.2020 um 21:08 schrieb Michael Maier:
>> On 23.06.20 at 08:05 Luca Bertoncello wrote:
>>> Am 23.06.2020 07:27, schrieb Luca Bertoncello:
>>>
>>> I again
>>>
>>>>> Do not change MTU. Probably there will be another problem. I expect
>>>>> packet size 1466 would pass and higher will have the same result. It
>>
>> RTP-VoIP-packets never reach this size. Size is about 214 bytes.
> 
> OK, so it must be something other...
> 
> But I really don't have any idea what... :(

Your basic architecture looks good to me - now you have to start the analysis 
of the problem with pcapsipdump and wireshark as I wrote before to get an idea 
what actually happens at
all. You most probably won't come any further without doing any analyzing. You 
have to learn it. It will take some, or even more, time. You can't do it in 
just few hours or maybe
even days or weeks. It is work or even hard work to learn and to do it.

That's my problem: It's impossible for me to assist because I can't see any 
effort on your side to learn. I won't fix your problem. You have to fix it 
yourself. All I can do, is, to
show you a way to *find* your problem (I can't know your problem) and may be to 
give some hints how to fix it (once you've found it). Finding / localizing 
problems and fixing them
are two completely different things. But before you fix a problem, you have to 
know the problem. Therefore: go and find your problem by starting the analysis. 
That's the first thing
to do.


Regards
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice broken during calls (again...)

[Michael Maier]

On 23.06.20 at 08:05 Luca Bertoncello wrote:
> Am 23.06.2020 07:27, schrieb Luca Bertoncello:
> 
> I again
> 
>>> Do not change MTU. Probably there will be another problem. I expect
>>> packet size 1466 would pass and higher will have the same result. It

RTP-VoIP-packets never reach this size. Size is about 214 bytes.


Regards
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice broken during calls (again...)

[Michael Maier]

Am 22.06.20 um 16:48 schrieb Luca Bertoncello:

Hi list!

So, now I have a business contract and a technician was here to check
the DSL...
Nothing found, except that for 50Mbps I need now vectoring. Really
nice... A couple of years ago I could get 50Mbps without vectoring.
Of course, Deutsche Telekom said nothing about this change...

Well, I got it working, and now I have 48Mbps down and 10Mbps up.
I _REALLY CAN'T_ believe, that this is not enough...


This is enough if you're doing it correctly. But that's your job to do it 
correctly - not Telekom's one.


The problem with many little disruptions during calls is always here.


Not surprising. That's most probably not a problem of the provider. VoIP of Deutsche Telekom mostly is pretty perfect 
regarding voice quality and availability.



I tried changing the codecs and changing some settings in the SIP
configuration of the peers.
No changes...


Not surprising.

Did you check to prevent transcoding?


On the Gateway (Banana PI), where the Asterisk server also runs, the
load is about 0.50 during calls and it has a Gbps LAN.


What's running on this device on parallel? What about other network traffic - 
not necessarily to the internet interface?


I can't believe, the problem is here...


That's irrelevant. You have to ensure, that the driver doesn't have any problems. Reducing the queue sizes of the 
interface may help.



@all german users using Telekom: how did you configured your Asterisk?


- At first, you have to trace down the problem and analyze those traces when the problem occurred. This could be done 
with pcapsipdump[1] on both sides (internal and external).

Example:

pcapsipdump -i ppp0 -p -d /tmp/pcapsipdump &

will trace the connection to Telekom. You have to add another process to 
another device to trace the internal call.
Use Wireshark to analyze the dumps. Wireshark understands VoIP. (I assume you are using SIP / RTP on all legs.) Now you 
can see on which side the problem happens and how it looks like.

- Are you using NAT or is asterisk running on the device which runs the 
ppp-interface?
- What's the modem you are using? What about the wiring between APL and modem? 
Is it done correctly? [2]
- Did you configure prioritization for the up-stream regarding RTP and SIP? 
This is done with the tc tool.
- Did you correctly configure tos? For Deutsche Telekom you may use tos=0xb8 (pjsip). You have to verify it with 
Wireshark with your traces. You have to set it to the same value as the packages which are received from their server.
- You have to use the DNS of Deutsche Telekom which they provide during the ppp-login because they usually provide 
optimal sip servers for you (regarding distance). You're RTT of ping (18 ms) is pretty bad. I'm having here 5 ms to the 
primary server (Telekom provides 3). See


dig +noall +answer _sip._udp.tel.t-online.de SRV

e.g. (don't know the hostname for the business infrastructure)


Regards,
Michael


[1] https://sourceforge.net/projects/pcapsipdump/
[2] 
https://telekomhilft.telekom.de/t5/Telefonie-Internet/Das-richtige-Kabel-zwischen-APL-und-TAE-Dose/ta-p/3499089

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice broken during calls (again...)

[Michael Keuter]

You could also use the 'mtr' command under Linux.

> Am 22.06.2020 um 17:41 schrieb Marek Greško :
> 
> Hello,
> 
> try pinging your sip peer ip address following way:
> 
> ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress}
> 
> Post several lines and the statistics.
> 
> Were you also thinking about MTU problems? Not very probable, but one
> never knows.
> 
> Marek
> 
> 
> 2020-06-22 17:18 GMT+02:00, Luca Bertoncello :
>> Am 22.06.2020 um 17:01 schrieb Telium Technical Support:
>>> I don't know if there was a prior email with more details, but
>>> 
>>> Latency is as important as speed.  Have you checked latency between your
>>> device and pop?  What about QoS at your location, and does your ITSP
>>> support/respect QoS?
>> 
>> That's a very good idea...
>> Could you suggest me how can I check it?
>> The Gateway is a Linux with Debian 9.
>> 
>>> Could problem be inside your network?  Have you tested/optimized internal?
>> 
>> Really difficult to believe... If I call another VoIP-phone in my
>> network (using the "internal number") the quality is excellent.
>> 
>> If I call my wife using the "external number", the quality is very bad...
>> 
>> Thanks
>> Luca Bertoncello
>> (lucab...@lucabert.de)


Michael

http://www.mksolutions.info




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice "broken" during calls

[Michael Keuter]

> Am 14.06.2020 um 16:38 schrieb Luca Bertoncello :
> 
> Am 13.06.2020 um 22:56 schrieb Antony Stone:
> 
> Hi again,
> 
>> 2b. Take your Thomson telephone to some other location with Internet access, 
>> let it register to your home Asterisk server, and them make a call to the 
>> same 
>> number yet again.  I'm sure you can get the Thomson to connect to Asterisk 
>> via 
>> some external network, since you say you can do this from your Android 
>> phone.  
>> Again, check the call quality.
> 
> I tried it on the network of a friend.
> Not possible to establish a connection at all...
> I *suppose* Deutsche Telekom just allow a logon on their servers from
> the IP of the user, who tries to log on (with other words: my VoIP login
> can just log on from my current IP)...

Hi Luca,

the standard Deutsche Telekom SIP-account (former ISDN Mehrgeräteanschluß PTMP 
with 3-10 numbers) is always tied to your DSL account.

There is a special "DeutschlandLAN SIP-Trunk Pure" where it does not depend on 
your DSL account (as it is standard with most other VoIP providers).

> This would explain why I didn't got my mobile phone connecting to the
> Telekom's server and establish a call...
> 
> I also tried to stop Asterisk and all other network services on my
> Linux-Box Firewall/Gateway, including the traffic shaper (in the case,
> this was the problem), then connect my Thomson phone to the Telekom's
> server and call my father in law.
> Always the same problem...
> 
> So, tomorrow I'll get another VoIP phone from a colleque (Elmeg IP 290).
> I'll connect it to my network and my Asterisk and will try to call my
> father in law for a test.
> 
> I really do *not* expect any change in the situation... I think, the
> problem should be somewhere by Deutsche Telekom...
> 
> What is your opinion?
> 
> Btw: I did all tests with my father in law, since he had time for me
> today, but the problem exists an almost all calls, incoming or outgoing,
> no matter from/to which network provider...
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)

Michael

http://www.mksolutions.info




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice "broken" during calls

[Michael Keuter]

So the call used Alaw as Codec.

> Am 13.06.2020 um 17:23 schrieb Luca Bertoncello :
> 
> Am 13.06.2020 um 13:47 schrieb Michael Keuter:
> 
> Hi
> 
>> Try "sip show peer " for a phone.
> 
> So:
> 
> mobile phone:
> bpi*CLI> sip show peer 0049177xxx
> 
> 
> 
> 
>  * Name   : 0049177xxx
> 
> 
>  Description  :
> 
> 
>  Secret   : 
> 
> 
>  MD5Secret: 
> 
> 
>  Remote Secret: 
> 
> 
>  Context  : default
> 
> 
>  Record On feature : automon
> 
> 
>  Record Off feature : automon
> 
> 
>  Subscr.Cont. : 
> 
> 
>  Language : de
> 
> 
>  Tonezone : 
>  AMA flags: Unknown
>  Transfer mode: open
>  CallingPres  : Presentation Allowed, Not Screened
>  Callgroup: 1
>  Pickupgroup  : 1
>  Named Callgr :
>  Nam. Pickupgr:
>  MOH Suggest  :
>  Mailbox  :
>  VM Extension : asterisk
>  LastMsgsSent : 0/0
>  Call limit   : 2147483647
>  Max forwards : 0
>  Dynamic  : Yes
>  Callerid : "0049177xxx" <>
>  MaxCallBR: 384 kbps
>  Expire   : -1
>  Insecure : no
>  Force rport  : Yes
>  Symmetric RTP: Yes
>  ACL  : No
>  DirectMedACL : No
>  T.38 support : Yes
>  T.38 EC mode : FEC
>  T.38 MaxDtgrm: 4294967295
>  DirectMedia  : No
>  PromiscRedir : No
>  User=Phone   : No
>  Video Support: No
>  Text Support : No
>  Ign SDP ver  : No
>  Trust RPID   : No
>  Send RPID: Yes
>  Path support : No
>  Path : N/A
>  TrustIDOutbnd: Legacy
>  Subscriptions: Yes
>  Overlap dial : No
>  DTMFmode : rfc2833
>  Timer T1 : 500
>  Timer B  : 32000
>  ToHost   :
>  Addr->IP : (null)
>  Defaddr->IP  : (null)
>  Prim.Transp. : UDP
>  Allowed.Trsp : UDP
>  Def. Username:
>  SIP Options  : (none)
>  Codecs   :
> (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk)
>  Auto-Framing : No
>  Status   : UNKNOWN
>  Useragent:
>  Reg. Contact :
>  Qualify Freq : 6 ms
>  Keepalive: 0 ms
>  Sess-Timers  : Refuse
>  Sess-Refresh : uac
>  Sess-Expires : 1800 secs
>  Min-Sess : 90 secs
>  RTP Engine   : asterisk
>  Parkinglot   :
>  Use Reason   : No
>  Encryption   : No
> 
> VoIP-phone (Thomson ST2022):
> bpi*CLI> sip show peer 0049351xxx
> 
> 
> 
> 
>  * Name   : 0049351xxx
> 
> 
>  Description  :
> 
> 
>  Secret   : 
> 
> 
>  MD5Secret: 
> 
> 
>  Remote Secret: 
> 
> 
>  Context  : default
> 
> 
>  Record On feature : automon
> 
> 
>  Record Off feature : automon
>  Subscr.Cont. : 
>  Language : de
>  Tonezone : 
>  AMA flags: Unknown
>  Transfer mode: open
>  CallingPres  : Presentation Allowed, Not Screened
>  Callgroup: 1
>  Pickupgroup  : 1
>  Named Callgr :
>  Nam. Pickupgr:
>  MOH Suggest  :
>  Mailbox  :
>  VM Extension : asterisk
>  LastMsgsSent : 0/0
>  Call limit   : 2147483647
>  Max forwards : 0
>  Dynamic  : Yes
>  Callerid : "0049351xxx" <>
>  MaxCallBR: 384 kbps
>  Expire   : 3111
>  Insecure : no
>  Force rport  : Yes
>  Symmetric RTP: Yes
>  ACL  : Yes
>  DirectMedACL : No
>  T.38 support : Yes
>  T.38 EC mode : FEC
>  T.38 MaxDtgrm: 4294967295
>  DirectMedia  : No
>  PromiscRedir : No
>  User=Phone   : No
>  Video Support: No
>  Text Support : No
>  Ign SDP ver  : No
>  Trust RPID   : No
>  Send RPID: Yes
>  Path support : No
>  Path : N/A
>  TrustIDOutbnd: Legacy
>  Subscriptions: Yes
>  Overlap dial : No
>  DTMFmode : rfc2833
>  Timer T1 : 500
>  Timer B  : 32000
>  ToHost   :
>  Addr->IP : 192.168.200.10:25572
>  Defaddr->IP  : (null)
>  Prim.Transp. : UDP
>  Allowed.Trsp : UDP
>  Def. Username: 0049351xxx
>  SIP Options  : (none)
>  Codecs   : (alaw|ulaw|ilbc|g729|g723|gsm)
>  Auto-Framing : No
>  Status   : OK (17 ms)
>  Useragent: THOMSON ST2022 hw2 fw3.56 00-26-44-31-10-23
>  Reg. Contact : sip:0049351xxx@192.168.200.10:25572;user=phone
>  Qualify Freq : 6 ms
>  Keepalive: 0 ms
>  Sess-Timers  : Refuse
>  Sess-Refresh : uac
>  Sess-Expires : 1800 secs
>  Min-Sess : 90 secs
>  RTP Engine   : asterisk
>  Parkinglot   :
>  Use Reason   : No
>  Encryption   : No
> 
> 
>> Then "sip show cha

Re: [asterisk-users] Voice "broken" during calls

[Michael Keuter]

> Am 13.06.2020 um 13:36 schrieb Luca Bertoncello :
> 
> Am 13.06.2020 09:30, schrieb Luca Bertoncello:
> 
> Hi again (again)
> 
> I noticed right now another strange detail...
> I made a call using my mobile phone (connected to the Asterisk). The quality 
> was top...
> Maybe is the problem in a codec used from our phones at homes?
> Could someone suggest me how to check the codec used by my mobile phone and 
> the codec used by the phones at home?
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)

Try "sip show peer " for a phone.
Then "sip show channels" during an existing call.
And "sip show channel " for more info.

Michael

http://www.mksolutions.info




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CLI color prompt

[Michael Maier]

On 31.05.20 at 19:26 Jeff LaCoursiere wrote:
> Hi,
> 
> I had posted this a few hours ago, but got caught in moderation for size.  I 
> trimmed down the pic and attached.
> 
> I am on an Ubuntu 16 workstation, in an Ubuntu terminal window, ssh'ed to the 
> PBX (amazon instance).  You can see my term type matches yours.

Same problem here (asterisk 16.x / CentOS 7).
# echo $TERM
xterm-256color

=> CLI prompt looks the same as described by the OP:
[1;31m[myhostname]:

But to make it even more strange: colored output on the asterisk CLI works 
pretty fine for DEBUG output like
core set debug 5

Maybe some link time functionality not enabled? Don't know ... . Or some other 
additional asterisk switch needed?


Thanks
Regards
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Extracting a SIP Header from a 302 Response

[Saint Michael]

I got the response below from a provider. How do I extract the Identity
header and apply it to the next INVITE? Is it possible at all with PJSIP?
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 172.16.7.254:52169
;rport=52169;received=XX.205.172.89;branch=z9hG4bK-524287-1---129f4244aaba9f04
Call-ID: 102650Mzg4NmFiNTQzOGY5NDJmNjM3OTYzNmE5MzNlZDIwZmI
From: "Peter Perez" ;tag=81a25c36
To: ;tag=9e198dc4-7ce8-433d-ae23-05b9bc14d55a
Identity:
eyJhbGciOiJFUzI1NiIsInR5cCI6InBhc3Nwb3J0IiwicHB0Ijoic2hha2VuIiwieDV1IjoiaHR0cDovL2NlcnQtYXV0aC5wb2Muc3lzLmNbWNhc3QubmV0L2V4YW1wbGUuY2VydCJ9eyJhdHRlc3QiOiJBIiwiZGVzdC6eyJ0biI6IisxMjE1NTU1MTIxMyJ9LCJpYXQiOiIxNDcxMzc1NDE4Iiwib3JpZyI6eyJ0biI64oCdKzEyMTU1NTUxMjEyIn0sIm9yaWdpZCI6IjEyM2U0NTY3LWU4OWItMTJkMy1hNDU2LTQyNjY1NTQ0MDAwMCJ9._28kAwRWnheXyA6nY4MvmK5JKHZH9hSYkWI4g75mnq9Tj2lW4WPm0PlvudoGaj7wM5XujZUTb_3MA4modoDtCA
;info=;alg=ES256  CSeq: 1 INVITE
Server: Asterisk PBX 16.10.0
Contact: 
Reason: Q.850;cause=0
Content-Length:  0
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Stir-Shaken clarified

[Saint Michael]

https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN
The Wiki above is misleading in what Stir-Shaken means and how it works.
End users cannot get a certificate, they cannot self-certify their calls.
Somebody completely misunderstood the model. I am afraid the moment will
come and thousands of Asterisk operators will be unable to terminate calls.
To start with, the model is a hierarchical one: there is an FCC
designated central authority, which appoints (so far two) Certification
Authorities, allowed to issue Certificates for Service Providers ONLY,
which themselves are ALSO pre-approved by then GA (Governance Authority),
and they need to have an OCN, they need to be a CLEC, have their own block
of numbers. So the idea that an Asterisk operator can have its own
certificate and somehow calculate the signature, is ridiculous. Once the
call arrives a the last mile, let's say VZ or ATT, the carrier will open
the signature added to each call and verify it with the Certification
Authority that issued the certificate. They will check if the caller-ID and
destination number match the actual call. Each signature is valid only for
60 seconds and each call has a different signature, even for the same
caller-ID and destination number, so it cannot be stored.
As you can see, this is a new world and we need to prepare for its arrival,
or our calls will simply fail and we shall be out of business. My company
is an approved Service Provider and we are waiting for the certificate,
which is in itself complicated paperwork.
Our model to solve this riddle for Asterisk is simple: Add a
res_odbc.so-connection pointed to our MySQL database. Create a func_odbc
function that executes our stored procedure. For each call, you send us the
pair Caller-ID and Destination number, and we send you back the signature.
In the next line in the dialplan, you add a SIP-header called Identity, and
our signature becomes the content.
Identity:
eyJhbGciOiJFUzI1NiIsInR5cCI6InBhc3Nwb3J0IiwicHB0Ijoic2hha2VuIiwieDV1IjoiaHR0cHM6Ly9jZXJ0LmV4YW1wbGUub3JnL3Bhc3Nwb3J0LmNlciJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxOTU0NDQ0NzQwOCJdfSwiaWF0IjoxNTkwNjcyNDc2LCJvcmlnIjp7InRuIjoiMjE1OTE0MDQyMSJ9LCJvcmlnaWQiOiIxMjNlNDU2Ny1lODliLTEyZDMtYTQ1Ni00MjY2NTU0NDAwMDAifQ.X7noevZGawXv1Jw1wkaqunTMFVE9FLt7sEX1QSgk0GMJmAHJWnbF5PCdj-Mc7UD2JY_5xvuJU3UlhSvswfK7SQ;info=<
https://cert.example.org/passport.cer>;alg="ES256";ppt="shaken"

With two lines of code in the dialplan, you solve the FCC requirements.
BUT, the caller-ID must be either verifiable associated with the company
that owns Asterisk, or we can supply one for you, from our pool of numbers.
Wireless numbers are not allowed. We check each and call return an error if
the conditions are not met. What happens if you send a random but valid
caller-ID? We still sign it, BUT, with Attestation level "C", which means
we don't know anything about the caller-ID. At some point, carriers will
decline to terminate those calls. It is up to them to terminate or not
those calls.
So what I am doing for the Asterisk community is helping everybody to stay
in business. If you delay the interconnection with me and pretend it is not
urgent, you will end-up in the fauces of nexus, which acts double as a
Certification Authority and Service Provider and charges huge fees. I mean
HUGE.
This wiki should be erased, for it is misleading:

> https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN




>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] STIR-Shaken

[Saint Michael]

>
> My company is one if the six service providers approved. We are not ready
> yet, probbably next week, since the certificate needs to be issued by the
> Certification Authority. As I said, we are the ONLY provider that  you may
> use with Asterisk remotely, via UnixODBC. The rest of the other providers
> will force you to send a call to them.

 Here is some material for you to read. Rest assured that this is real.
https://www.fcc.gov/call-authentication
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Stir-Shaken for asterisk

[Saint Michael]

In a few weeks, no SIP call is going to terminate unless they are signed
properly, as mandated by law.  We are in the business of Stir-Shaken,
signing calls, as an FCC-approved provider. A big differentiator between
our service and the rest: we are the only ones who don't need to receive
the calls in our servers to sign them. We do this over a MySQL call,
easily connectable to Asterisk via res_odbc, so you never have to send us
your calls. This is a sample of how we do this so you may test now:
mysql -u anonymous -h 208.73.232.47 -e "call
strshk.stir_shaken_signature('7274433019','1957408')".
If your caller-ID is a valid US number and not a wireless number (that is a
NO-NO for the FCC), we sign the call as 'C', if you use your own DIDs,
something we can verify as legit, then we sign as 'B', and if you use our
DID as caller ID, we sign as 'A', full attestation.
Please email to venefax at g mail if you have any questions. Do not think
you can do business as usual. The wild west of VOIP is coming to an end.
But we can keep you in business if you follow the rules.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] PJSIP sending RTP to private address

[Saint Michael]

About this case: the old SIP channel behaves correctly.

On Sun, May 17, 2020 at 2:44 AM Saint Michael  wrote:

> My phone is located behind a NAT, 172.16.0.0/21.
> Asterisk 16 is on a public IP.
> PJSIP has the config below:
> force_rport=yes
> direct_media=yes
> disable_direct_media_on_nat = yes
> direct_media_method=invite
>
> But when I send a call I see the RTP being sent to my private address, vs
> the public IP. This only happens when Asterisk  has dialed the call to
> another carrier. If instead of Dial I choose Answer() and MusicOnHold, then
> the RTP gets shipped to the right address.
> This is a sample of the erroneous behavior:
> Got  RTP packet fromXX.XX.XX.XX:17510 (type 00, seq 024786, ts 017440,
> len 000160)
> Sent RTP packet to  172.16.7.254:50798 (type 00, seq 010736, ts
> 017440, len 000160)
>
> 172.16.7.254 is my private address.
> What am I missing? Should I open a bug?
> Asterisk should never, ever send RTP to a private address when Asterisk
> itself is on a public IP.
> Before you ask, the dialplan is 3 lines,
> '_X.' =>  1. NoOP()
> 2. Dial(PJSIP/${EXTEN}@carrier)
> 3. Hangup()
>
>
>
>
>
>
>
>
>
>
>
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Meaning of RTT in channelstats

[Michael Maier]

On 17.05.20 at 01:28 Joshua C. Colp wrote:
> On Sat, May 16, 2020 at 10:58 AM Michael Maier  wrote:
> 
>> => How are the RTT values exactly calculated? Which values are actually
>> used for?
>>
> 
> The value is calculated according to the logic in the RFC[1]. Specifically
> using embedded timestamps in the RTCP packets and calculated delays. The
> value is presented in seconds I believe in the output.

Thanks Joshua!

>> => What about the processing time between the inbound leg and the outbound
>> leg (transcoding, ...)?
>>
> 
> That has no impact within this, since it's calculated using the RTCP
> traffic which is not used for media. It's really just for round trip time
> of a UDP packet, not for end to end time of a single audio packet/frame
> through the system.

Let's try to sum it up on base of the given easy example how to get the 
complete delay between those two speakers:

A calls B:
 ...Receive. 
.Transmit..

 BridgeId ChannelId  UpTime.. Codec.   CountLost Pct  Jitter   
CountLost Pct  Jitter   RTT
 
===

 c8137221 327-0004   03:22:42 g722  608K  00   0.000
608K  00   0.000   0.000
 c8137221 providePJSIP-xxx-0 03:22:42 alaw  608K  00   0.000
608K  00   0.000   0.023

A says something.

1. quantization:20 ms
2. processing time on the phone base / DECT:?
3. way from phone base to asterisk: 0 ms
4. processing time on asterisk / transcoding:   ?
5. transport to destination:11.5 ms
6. processing time on destination side: ?

Therefore it would take about 35 ms until B can here A. Is this basically a 
correct estimation or did I miss(understand) something?


Thanks
Michael


> [1] https://tools.ietf.org/html/rfc3550

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Help missing

[Saint Michael]

I want to see the help when I type core show application , and it's not
available. This is asterisk 16 from sources. I have libxml2-dev  installed.
Ubuntu 19
What am I missing?
Philip
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PJSIP does not stop sending invites after call is canceled

[Saint Michael]

Endpoint sends an INVITE
Asterisk send an INVITE to the Carrier
Carrier is down, does not even sends ACK
PJSIP sends  several INVITES
End point sends
<--- Received SIP request (397 bytes) from UDP ::50187 --->
CANCEL sip:xxx@xxx SIP/2.0
Via: SIP/2.0/UDP xxx
:50187;branch=z9hG4bK-524287-1---fbad0437cf02653d;rport
Max-Forwards: 70
To: 
From: "x";tag=a0acbb3e
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
CSeq: 1 CANCEL
User-Agent: Bria 5 release 5.8.3 stamp 102650
Content-Length: 0

PJSIP responds to endpoint

<--- Transmitting SIP response (403 bytes) to UDP:xx:50187 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
X:50187;rport=50187;received=;branch=z9hG4bK-524287-1---fbad0437cf02653d
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
From: "xx" ;tag=a0acbb3e
To: ;tag=5d2fe4a1-b7b1-4868-9696-356511924c60
CSeq: 1 CANCEL
Server: Asterisk PBX 13.33.0
Content-Length:  0

the PJISP sends an additional response to endpoint
<--- Transmitting SIP response (419 bytes) to UDP::50187 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.16.7.254:50187
;rport=50187;received=x;branch=z9hG4bK-524287-1---fbad0437cf02653d
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
From: "x" ;tag=a0acbb3e
To: ;tag=5d2fe4a1-b7b1-4868-9696-356511924c60
CSeq: 1 INVITE
Server: Asterisk PBX 13.33.0
Content-Length:  0

to make a long story short, the endpoint sends back an ACK, but after that,
PJSIP keeps sending INVITES to the carrier, which means it did not close
the second leg of the call. If the carrier sends back a 200 OK, there will
be a billing charge, which in case of Mexico is minimum 60 seconds, and the
endpoint will not agree with the charge, resulting in a financial loss for
the Asterisk owner. This is absurd. The second leg must close as soon as a
CANCEL has been received.

The dialplan is only one line
Dial(PJSIP/${EXTEN}@carrier)

Kindly tell me what am interpreting wrong.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Meaning of RTT in channelstats

[Michael Maier]

On 15.05.20 at 14:31 Doug Lytle wrote:
> Google says Round Trip Time
> 
> https://www.voip-info.org/asterisk-rtcp/

That doesn't answer my question (I know the abbreviation RTT). Therefore I'm 
trying again:

I'm just wondering what the RTT *exactly* means. Where are the exact measuring 
points located?

=> How are the RTT values exactly calculated? Which values are actually used 
for?

=> What about the processing time between the inbound leg and the outbound leg 
(transcoding, ...)?


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Meaning of RTT in channelstats

[Michael Maier]

Hello!

I'm just wondering what the RTT exactly means. Where are the exact measuring 
points located?

> pjsip show channelstats

 ...Receive. 
.Transmit..
 BridgeId ChannelId  UpTime.. Codec.   CountLost Pct  Jitter   
CountLost Pct  Jitter   RTT
 
===

 c8137221 327-0004   03:22:42 g722  608K  00   0.000
608K  00   0.000   0.000
 c8137221 providePJSIP-xxx-0 03:22:42 alaw  608K  00   0.000
608K  00   0.000   0.023


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?

[Michael L. Young]

> From: "John Hughes" 
> To: "Asterisk Users Mailing List, Non-Commercial Discussion"
> 
> Sent: Thursday, May 14, 2020 2:10:45 AM
> Subject: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and
> alaw; asterisk wants to translate g729 -> alaw. WHY?

> I am having a problem with one of my callers who is using either g729 or 
> alaw. I
> can do alaw but not g729 so asterisk should negotiate alaw right? In fact from
> the sip debug it looks like it does, but then I get the dreaded 
> "channel.c:5630
> set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the
> call hangs up. Why?

> Last minute thought: Is it possible that the caller is sending g729 in RTP 
> even
> though the SIP negotiation clearly chooses alaw? Maybe I need some RTP
> debugging.

> Asterisk 13.14.1 on Debian, using chan_sip.
Hi John, 

Maybe a newer version of Asterisk would help? The latest release for 13 is 
version 13.33. The version you are on was released 3 years ago. 

Here is an issue which looks like what you describe and was fixed in 13.16 
[ https://issues.asterisk.org/jira/browse/ASTERISK-26143 | 
https://issues.asterisk.org/jira/browse/ASTERISK-26143 ] 

Not sure if this is the answer to your problem but thought that I would throw 
that out there. 

Michael L. Young 

(elguero) 
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] New RTP engine

[Saint Michael]

>
> Asterisk needs urgently to push the RTP engine to the Kernel, away from
> userland, like professional and commercial softwares do. I measured the
> cost of passing call from a public IP to a private IP, like typically a
> Session Border Controller may do. In Asterisk, ulaw, no transcoding, it
> takes 1.7% of a 3 Ghz core. If the packets where flowing through the
> kernel, like iptables does, it would take 10% if the CPU. Asterisk then
> could be used in hundreds of different roles in the enterprise.  PJSIP has
> no importance at all, this is the big issue. I suggest the developers look
> at an open-source package and adapt the code, is called rtpengine. It uses
> a kernel module to do the job.

Philip
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PJSIP crashes

[Saint Michael]

>
> I have no control over the SIP calls I receive. PJSIP should log a warting
> and continue. It is causing the CPU usage to spike dramatically.
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PJSIP crashes

[Saint Michael]

PJISP cannot handle the From  field when it does not contain a number.
Can this be fixed?

[Feb 25 12:35:43] ERROR[7143]: pjproject: : sip_transport.c
Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax
error exception when parsing 'From' header on line 4 col 40:
CANCEL sip:14408785990@162.255.138.102:5060 SIP/2.0
Via: SIP/2.0/UDP 8.38.43.67:5060;branch=z9hG4bK1sansay261086943rdb109274
To: 
From: "Radefeld Dental" ;tag=sansay261086943rdb109274
Call-ID: 1001880886-0-2320154044@8.38.43.49
CSeq: 1 CANCEL
Max-Forwards: 66
Reason: Q.850;cause=34;text="No Ring Timeout"
Content-Length: 0
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] avoiding any media proxy with PJSIP

[Saint Michael]

Is there a guide on how to use PJSIP and never have the media travel inside
Asterisk? No matter what I do, I cannot make this work.
Philip Orleans
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] permission woes on systemd

[Michael L. Young]

- Original Message -
> From: "sean darcy" 
> To: "Asterisk Users Mailing List, Non-Commercial Discussion" 
> 
> Sent: Tuesday, January 21, 2020 9:22:28 PM
> Subject: [asterisk-users] permission woes on systemd

[..]

> So why would starting asterisk as user asterisk work, but fail using
> systemd ?
> 

Have you checked SELinux?  After creating the configuration files, did you run 
'restorecon' on the appropriate asterisk directories?  If not, the files are 
not labeled correctly and SELinux might be denying access.

Just a thought.

Michael

(elguero)

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to: send dtmf back to the calling channel from post-answer subroutine executed on outbound channel

[Saint Michael]

I have a customer who wants me to send a DTMF on the calling channel if the
called channel says any word. So I am using
[my_gosub_routine]

exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
 same => n,Playback(hello)
 same => n,Return()

[default]

exten => _X.,1,NoOp()
 same =>
n,Dial(PJSIP/alice,,U(my_gosub_routine^my_gosub_arg1^my_gosub_arg2))
 same => n,Hangup()

Is there a way to send DTMF back to the caller from [my_gosub_routine]?
If I use sendDTMF at the moment, it will be heard only by the callee, and
only the caller must hear it.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Own MOH incorrectly kicking in instead of the MOH of the callee

[Michael Maier]

Hello all!

I'm reproducibly getting my *own MOH* if I should get the MOH of the Callee 
instead. I can see this with asterisk 13 and 16 (and probably before, too). The 
reason of the
wrong MOH is an in dialog reInvite received from trunk, which sends a SDP 
containing

a=sendonly

After this reInvite, I can hear own MOH instead of the MOH of the Caller. The 
situation is cleared by another reInvite received from the trunk containing

a=sendrecv


Is this expected behavior? I don't think it should act like this.

BTW: I'm additionally using FreePBX. Maybe it's a problem of FreePBX?


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FREEPBX Mailinglist

[Michael Maier]

On 11.09.19 at 15:24 Joshua C. Colp wrote:
> On Wed, Sep 11, 2019, at 10:18 AM, basti wrote:
>> Hallo,
>> is there a Freepbx mailinglist? or can this be posted here?
> 
> FreePBX does not have a mailing list. People use the community forum[1] 
> instead.
> 
> [1] https://community.freepbx.org/

This forum of freepbx is quite a PITA (and therefore quite unusable), at least 
from (some (don't know if all) providers of) Germany. It takes 1 minute or more 
to load
(each! site). The IP 199.102.239.92 produces packet loss on a huge scale. Even 
reducing the mss size to 900(!) doesn't have any effect. I can see this since 
years or
better: I never saw it w/o package loss.

# ping 199.102.239.92
PING 199.102.239.92 (199.102.239.92) 56(84) bytes of data.
64 bytes from 199.102.239.92: icmp_seq=1 ttl=56 time=124 ms
64 bytes from 199.102.239.92: icmp_seq=2 ttl=56 time=124 ms


# traceroute -T 199.102.239.92
traceroute to 199.102.239.92 (199.102.239.92), 30 hops max, 52 byte packets
 1  * * *
 2  x.x.x.x (62.155.245.70)  7.841 ms  7.874 ms  7.868 ms
 3  y.y.y.y (217.5.116.234)  10.882 ms  10.916 ms  10.912 ms
 4  z.z.z.z (217.5.116.234)  10.908 ms  10.904 ms  10.909 ms
 5  62.157.249.186 (62.157.249.186)  22.582 ms  22.609 ms  23.037 ms
 6  ae-1.r25.frnkge08.de.bb.gin.ntt.net (129.250.4.16)  25.279 ms  24.443 ms  
24.543 ms
 7  ae-8.r22.asbnva02.us.bb.gin.ntt.net (129.250.4.96)  103.701 ms  104.829 ms  
117.578 ms
 8  ae-0.r23.asbnva02.us.bb.gin.ntt.net (129.250.3.85)  102.750 ms  102.776 ms  
115.097 ms
 9  ae-3.r21.chcgil09.us.bb.gin.ntt.net (129.250.2.139)  122.340 ms  121.706 ms 
 119.258 ms
10  ae-16.r08.chcgil09.us.bb.gin.ntt.net (129.250.2.204)  122.853 ms 
ae-2.r07.chcgil09.us.bb.gin.ntt.net (129.250.4.214)  114.721 ms  117.954 ms
11  ae-1.a01.chcgil09.us.bb.gin.ntt.net (129.250.5.94)  119.707 ms 
ae-0.a01.chcgil09.us.bb.gin.ntt.net (129.250.4.218)  122.066 ms  120.186 ms
12  129.250.199.66 (129.250.199.66)  121.861 ms  125.719 ms  122.840 ms
13  static.66.185.29.254.cyberlynk.net (66.185.29.254)  125.742 ms  126.253 ms  
124.522 ms
14  * wi-mke1-dc2-c5-47-haf-7578.cyberlynk.net (66.185.29.26)  118.165 ms *
15  199.102.239.92 (199.102.239.92)  122.352 ms  126.759 ms  123.944 ms
16  199.102.239.92 (199.102.239.92)  123.890 ms  124.069 ms  126.670 ms


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] asterisk 16.5 / pjsip outage because of task processor queue >= 500 tasks and too many open files later on

[Michael Maier]

C /etc/asterisk/asterisk.conf
16.08.2019 03:01 asterisk 15910  2.2  31.0  2564704 626192  ?  Ssl  Aug10 
183:22 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 04:01 asterisk 15910  2.2  36.6  2695776 738540  ?  Ssl  Aug10 
184:38 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 05:01 asterisk 15910  2.2  41.1  2761312 827892  ?  Ssl  Aug10 
185:55 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 06:01 asterisk 15910  2.2  46.2  2892384 932244  ?  Ssl  Aug10 
187:12 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 07:01 asterisk 15910  2.2  51.4  2957920 1036004 ?  Ssl  Aug10 
188:31 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 08:01 asterisk 15910  2.2  56.2  3088992 1133328 ?  Ssl  Aug10 
189:49 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 09:01 asterisk 15910  2.2  58.3  3088992 1175064 ?  Ssl  Aug10 
191:07 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
restart
16.08.2019 10:01 asterisk   629  4.9  3.5   2103380 71700   ?  Ssl  09:34   
1:19 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf

VSZ   virtual memory size of the process in KiB (1024-byte units)
RSS   resident set size, the non-swapped physical memory that a task has 
used (in kiloBytes).

=> Memory usage had been rapidly growing since 2019-08-16 00:00:01


Few words about the usage of asterisk:
- 2 registered endpoints
- 4 SIPS / SRTP trunks
- 46 calls at 2019-08-15
- the sip:isp.de trunk hadn't been used


Some findings:

- The problem seems to be triggered by the "task processor queue reached 500 
scheduled tasks" problem. I saw this message in June, too.
  Context each time was the same ISP (-> not Deutsche Telekom!) as described 
above in conjunction with registration retries, too.

- Registration configuration of this provider:

 

==

 ispPJSIP/sip:isp.de ispPJSIP  
Registered

 ParameterName: ParameterValue
 ==
 auth_rejection_permanent : true
 client_uri   : sip:0049...@isp.de
 contact_user : +49...
 endpoint : ispPJSIP
 expiration   : 3600
 fatal_retry_interval : 0
 forbidden_retry_interval : 10
 line : true
 max_retries  : 1
 outbound_auth: ispPJSIP
 outbound_proxy   :
 retry_interval   : 60
 server_uri   : sip:isp.de
 support_path : false
 transport: 0.0.0.0-tls


The expiration value given above is not true. isp.de forces ReRegistration each 
90s (asterisk does it each 60s if I remember correctly)!
Contact: ;expires=90

- After performing the restart of asterisk, registration to the isp.de hadn't 
any problem any more. Therefore I think,
  the reregistration problem wasn't a problem of the provider not answering but 
in fact a problem of asterisk being unable to correctly perform the 
ReRegistration.




The final question:
===
Is there a problem with taskprocessors probably not being canceled on some 
conditions (maybe unanswered or hanging registrations?) and therefore steadily 
growing up and using more and more open files (and memory) until asterisk can't 
do
anything anymore because some limits are exceeded as a result?
Could there be a problem with the retry interval 60s and the real ReRegister 
done each 60s, too (performing a "fork" bomb)?



Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anyone ever experienced a crash where Asterisk debug output a line with all nulls

[Michael Maier]

On 14.08.19 at 18:12 Dan Cropp wrote:



Maybe because the machine is performing a file system check on some other 
partitions in parallel and it's slowed down therefore?


Wouldn't /var/log/syslog show something like this if it's happening in parallel?


Well, it was just speculation. Is it even now reproducible after the 3. or even later reboot? I would try to log in via ssh as fast as possible 
(or directly via a shell) and try to find the responsible process. Try to examine the IO.



syslog has items before asterisk is starting, but once the Asterisk log files 
show it's starting to completed, /var/log/syslog has nothing.
Syslog has nothing for about 10 minutes after that.

The strange thing is the customer did a manual restart of Ubuntu 20 minutes 
later and the same slow startup time happened.


If the machine is shut down manually, did it perform a correct shutdown (= all file systems have been correctly unmounted and all processes have 
been stopped before unmounting)?


Is the startup slow even if it's done w/o reboot before?


Regards
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anyone ever experienced a crash where Asterisk debug output a line with all nulls

[Michael Maier]

er.conf
> 
> 
> 
> I believe this was bad enough that Ubuntu actually crashed, but there is 
> nothing in the syslog indicating anything until 15:32:42 where it appears 
> Linux is starting up.
> 
> 
> After this situation happens, every time Asterisk starts up, it was taking 
> significantly longer to load.  Normally 1-2 seconds, became 26-28 seconds.
> [08/12 15:33:03.240] NOTICE[1385] loader.c: 286 modules will be loaded.
> [08/12 15:33:23.844] VERBOSE[1385] loader.c: Loading extconfig.
> 
> 
> Loading the modules is taking 20 seconds after this incident occurred.  
> Looking at the debug logs, I see the modules loading loader.c PASS.  There 
> all seem to load fine, just much slower than it was previously.

Maybe because the machine is performing a file system check on some other 
partitions in parallel and it's slowed down therefore?


Regards
Michael
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Wanted: professional softphone

[Michael Maier]

Hello!

Does anybody by chance know of a softphone which additionally has a management 
suite to fully configure it userID based for Windows clients? Any idea is 
appreciated!


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PJSIP / tcp: define local port to use on base of trunk definition

[Michael Maier]

Hello!

Following problem: If there are different trunks (-> different numbers and 
users / passwords) to the same destination, asterisk (16.x) always uses the 
same local tcp port for each connection. This is a problem with Deutsche 
Telekom (they
want to have different local ports for different users).

Besides the possibility to use different IP addresses (aliases) - is there a 
generic way to define on trunk base which local port to use for each user / 
number?


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?

[Michael Maier]

On 06.07.19 at 22:16 hwilmer wrote:

Is there an advantage to using pjsip?  What's needed for easybell with pjsip?


For easybell, I don't know of any advantage. But that's not very reliable, because I'm using easybell for dedicated requirements only. I'm 
considering chan_sip legacy. I wouldn't build any new installation on chan_sip (if there are no technical contradictions).


Easybell does have a pretty fine documentation for FreePBX and pjsip:
https://www.easybell.de/nc/hilfe/ergebnis/freepbx-130124-mit-asterisk-13.html


[why encryption?]

I consider it a requirement for when employees end up using their mobile phones over foreign wireless networks, which is something that would be 
virtually impossible to prevent should the asterisk server be made reachable from the outside.


That's true. But don't forget to encrypt RTP at this point! This must be done 
additionally.
BTW: easybell doesn't support full RTP encryption. It's supported for outgoing calls only 
(https://en.easybell.de/nc/help/questions/questions-concerning-the-telephone-connection/answer/does-easybell-support-the-data-encryption-srtp-for-voip.html).


That's an example for an inbound call - there isn't any support for RTP 
encryption:

<--- Received SIP request (869 bytes) from TLS:195.185.37.60:5061 --->
INVITE sip:+4912345678989@93.165.22.128:5061;transport=TLS;line=xxx SIP/2.0
Via: SIP/2.0/TLS 195.185.37.60:5061;branch=z9hG4bKcu8ZTaf6c4iPU;rport
From: ;tag=2DC25244-5D21B65400044D02-6A194700
To: 
CSeq: 1891 INVITE
Call-ID: SBC4115eaf5-5d21b654-67c091e3-a60cca0-924117e-01_b2b-1
Max-Forwards: 68
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK
Supported: histinfo
Content-Type: application/sdp
Content-Length: 286
Contact: 


v=0
o=- 1667528048 2824605765 IN IP4 195.185.37.60
s=-
c=IN IP4 195.185.37.60
t=0 0
m=audio 30934 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:both
a=sendrecv
a=maxptime:150




And before that, why shouldn't phone calls always be encrypted for just in 
case?  They are always genuinely private, and it doesn't hurt anything.


True - no contradiction of mine.




Setting 'tlscapath' to /etc/pki or to /etc/pki/ca-trust/source/ didn't seem to


I'm sorry - I don't know how to handle ca bundles with chan_sip. With pjsip it's

ca_list_file=/etc/pki/tls/certs/ca-bundle.crt >
in pjsip.transports.conf.


Thanks, setting 'tlscafile=/etc/pki/tls/certs/ca-bundle.crt' seems to do the 
trick.  However:


Happy to here that it's working now.



First I set 'tlsdontverifyserver=no' and issued a 'sip reload'.  There was no error message.  I found that suspicious and restarted asterisk, 
and the error message came back.


There are some config changes which need a complete restart AFAIK (FreePBX 
explicitly warns about some changes requiring a restart).


Only then I added 'tlscafile=/etc/pki/tls/certs/ca-bundle.crt' (which was unset 
before), and after a 'sip reload', the error message was gone.
So far, it hasn't come back even when restarting asterisk.


That's how I'm handling it: After each change concerning the transport I'm restarting asterisk. Just to be sure. But that's a question which 
could be answered by Joshua much better.


This shows that 'sip reload' doesn't really do a reload in that a certificate which hasn't been verified continues to be accepted after the 
configuration changed to now require verifying the certificate.


The certificate check is done on starting the connection (SYN) by openssl. sip reload most probably doesn't restart the connection (because all 
running calls would be disconnected - that's most of the time not a good idea - sip reload usually doesn't destroy running sessions / calls).


This might be a security problem, and if not, it is certainly good for surprises 
and can create much confusion.


Just handle it like this: each transport relevant change requires a complete 
restart!



Is it supposed to be like this, or should I make a bug report?


I think it's supposed to behave like this, because it would mean to disconnect all running calls on sip reload. That's probably not what most of 
the people expect / want to have.



Regards
Michael

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?

[Michael Maier]

On 06.07.19 at 12:16 hwilmer wrote:
> On 7/6/19 10:40 AM, Michael Maier wrote:
>> On 05.07.19 at 22:02 hw wrote:
>>>
>>> openssl verify -CAfile ca.pem asterisk.pem
>>> asterisk.pem: OK
>>>
>>>
>>> When I set tlsdontverifyserver=yes, it works (i. e. asterisk registers
>>> to the SIP provider and there is no error message).  Otherwise I'm
>>> getting the error message and asterisk does not register.
>>>
>>> Reading the comments in sip.conf.sample, I would assume that asterisk
>>> can not verify the certificate of the SIP provider.  Yet
>>>
>>>
>>> openssl s_client -connect secure.sip.easybell.de:5061

I'm using easybell via tls, too - but with pjsip - I had never any problem.

>>
>> You know that you don't need an own certificate to connect via tls to the 
>> ISP?
> 
> No, I didn't know that.  However, there are local clients connecting to 
> asterisk
> using encryption, so I suppose my own certificate is required.

That's true - but why do you need encryption on your own LAN? Just for fun or 
are there any particular requirements?

>> To be able to verify the certificate of the ISP, asterisk has to know the 
>> local CA database. For CentOS 7, this is /etc/pki/tls/certs/ca-bundle.crt.
> 
> How did you know I'm doing this on Centos? :)

This was just meant as an example - chance :-)

> Setting 'tlscapath' to /etc/pki or to /etc/pki/ca-trust/source/ didn't seem to

I'm sorry - I don't know how to handle ca bundles with chan_sip. With pjsip it's

ca_list_file=/etc/pki/tls/certs/ca-bundle.crt

in pjsip.transports.conf.


Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?

[Michael Maier]

On 05.07.19 at 22:02 hw wrote:
> 
> openssl verify -CAfile ca.pem asterisk.pem
> asterisk.pem: OK
> 
> 
> When I set tlsdontverifyserver=yes, it works (i. e. asterisk registers
> to the SIP provider and there is no error message).  Otherwise I'm
> getting the error message and asterisk does not register.
> 
> Reading the comments in sip.conf.sample, I would assume that asterisk
> can not verify the certificate of the SIP provider.  Yet
> 
> 
> openssl s_client -connect secure.sip.easybell.de:5061

You know that you don't need an own certificate to connect via tls to the ISP?

To be able to verify the certificate of the ISP, asterisk has to know the local 
CA database. For CentOS 7, this is /etc/pki/tls/certs/ca-bundle.crt.



Regards
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] High delay and some echo

[Michael Maier]

On 11.06.19 at 20:32 Luca Bertoncello wrote:
> Hi list!
> 
> I use Asterisk 13.14.1 from Debian repository on a DSL from Deutsche
> Telekom.
> 
> Asterisk works well, but I have really often an high delay (I understand
> it since the other party speak some seconds before he hears my question
> and answer) and sometimes I hear an echo.

First of all: I'm using Deutsche Telekom, too (with pjsip on CentOS 7) and 
don't have this problem.

Let me sum up at first what I understand at the moment:
- Only VoIP
- The problem isn't new.
- The problem doesn't happen always, but often.
- Asterisk uses the internet IP and doesn't do NAT.
- You're using chan_sip - not pjsip
- DSL-Line: 50/10 MBit


My questions to analyze the problem:

- What's the real usable DSL sync (can be seen at the modem)?
- Are there any (CRC) errors on the DSL side? How many and in which time?
- Deutsche Telekom reports the usable bandwidth during pppoe login. In 
messages, you can see
  something like
  SRU=37868#SRD=102957# (it's an example for a 100 MBit line)
  (grep messages for "SRU=" after a successful pppoe login)
  It contains the upload and download bandwidth in kbit/s
- Did you configure traffic shaping with tc to be sure that voice packages are 
always sent at first?
- Problem can be seen with different callees or just with one?
- Are there any callees the problem never occurred?
- Is it "just" a delay or is it choppy, too?
- You're using Banana PI - which one exactly? RAM? eth interface manufacturer? 
What about the load
  (uptime) of the system when the problem occurs? Is it swapping (what says 
"free")?
- What about the temperature of the device if the problem occurs / not occurs?
- Is there any other outbound traffic at the same time? Check with the tool 
bmon at the ppp0
  device and take a look at the upstream. One call creates 50 packages/s (pps) 
on each direction (if there is no other traffic). It shouldn't fluctuate.
- Did you set the correct QoS-type for the outgoing sip and rtp packages? In 
pjsip, the options are:
  tos=cs3
  cos=3
  You can check it with wireshark. The DSCP must be expedited forwarding (or 
the same you can see for incoming voice packages).
- asterisk has an own console, that can be reached with asterisk -r as root.
  At this point, you can get some information about the quality of a running 
call. For pjsip it's reporting the following e.g.:

  *CLI> pjsip show channelstats

 ...Receive. 
.Transmit..
 BridgeId ChannelId  UpTime.. Codec.   CountLost Pct  Jitter   
CountLost Pct  Jitter RTT
 
===

 5d67cd0b x-007e   00:00:39 g722 1296   00   0.000   1299   
00   0.000   0.000
 5d67cd0b y-007f   00:00:39 alaw 1299   00   0.000   1296   
00   0.000   0.000

 Instead of "pjsip show channelstats" you have to use something like sip show 
[press 2 times tab key] to get the possible commands.

 Each call generates two entries: one for the call from your local phone to 
asterisk and the other from asterisk to the ISP.



Hope this helps to locate the problem.
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk-users Digest, Vol 177, Issue 11

[Saint Michael]

Joshua
Is there a way in PJSIP to send the audio between the parties always,
unless one of the parties is behind a NAT?
A session refresh would work.
That my only problem with PJSIP. This is routine in the old sip channel.

On Sat, May 25, 2019 at 1:03 PM 
wrote:

> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Re: Is there a way to make asterisk send a INVITE in-dialog
>   to re-establish the audio (Dan Cropp)
>
>
> --
>
> Message: 1
> Date: Fri, 24 May 2019 17:02:56 +
> From: Dan Cropp 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Is there a way to make asterisk send a
> INVITE in-dialog to re-establish the audio
> Message-ID:
> 
> Content-Type: text/plain; charset="utf-8"
>
> Thank you Joshua
>
>
> -Original Message-
> From: asterisk-users  On Behalf
> Of Joshua C. Colp
> Sent: Friday, May 24, 2019 9:53 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Is there a way to make asterisk send a
> INVITE in-dialog to re-establish the audio
>
> On Fri, May 24, 2019, at 9:47 AM, Dan Cropp wrote:
> >
> > We are working with an Avaya switch.
> >
> >
> > We send them a REFER. If the transfer is successful, everything is
> > great. If it fails (busy), they send an INVITE in-dialog with a media
> > attribute of inactive. After that, they send a 486 busy.
> >
> > The problem is Avaya basically put the call on hold so audio is not
> active.
> >
> > The Avaya rep is indicating we need to send in dialog invite to get
> > the call audio back? They are essentially saying they put the call on
> > hold because we told them to transfer and it’s our responsibility to
> > take the call off hold.
> >
> >
> > Is there a way to do this?
>
> I don't think there is. We provide the ability in PJSIP to do a session
> refresh[1] but there's no ability to set the stream state like that, so I'm
> not sure what we would specify in that scenario automatically.
>
> [1]
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_SEND_SESSION_REFRESH
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
> www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
>
> Subject: Digest Footer
>
> ___
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
>
> End of asterisk-users Digest, Vol 177, Issue 11
> ***
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] pjsip and tls client: How to decrypt Wireshark trace?

[Michael Maier]

Hello!

I'm just wondering if it's possible to decrypt sips packages in Wireshark while 
asterisk runs as sips client (connecting to the provider w/
tls 1.2)? I don't use an own certificate.


Thanks
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Forking AGI or GoSub

[Michael Munger]

I am very much a PHP person. PHP doesn't really have an elegant way to handle 
forking / threading, which can make it non-trivial to implement and can be 
unreliable if the implementation is not exact. PHP must also be compiled to be 
thread safe in order to do this properly.

Granted, the last time I looked into this with PHP was under PHP 5.6, and tests 
at that time did not yield the results we wanted. We ultimately moved to Python 
when we needed multi-threading, which is extremely elegant and reliable for 
this application.


[cid:image001.png@01D4F6B9.8C1499D0]

Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

Microsoft Certified Professional

Microsoft Certified Small Business Specialist

Digium Certified Asterisk Professional

High Powered Help, Inc.

p:

678-905-8569

w:

hph.io<https://hph.io>  e: m...@hph.io<mailto:m...@hph.io>




From: asterisk-users  On Behalf Of 
Mark Wiater
Sent: Friday, April 19, 2019 2:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Forking AGI or GoSub

On 4/19/2019 1:49 PM, Dovid Bender wrote:

Mark,

I am using PHP agi and when forking the call does not continue util the forked 
process is done. Am I doing it wrong?


On Wed, Apr 10, 2019 at 4:27 PM Mark Wiater 
mailto:mark.wia...@greybeam.com>> wrote:
On 4/10/2019 3:54 PM, Dovid Bender wrote:

I have an AGI that can sometimes take time complete. I don't want the dialplan 
to be held up by the agi. Is there any way to call it and have Asterisk 
continue with the dialplan?


Is there a reason you can't fork in the AGI and just return to the dialplan in 
the parent?

Dovid,

I'm not much of a PHP person, but in perl, i check the process id that's 
returned from fork() and exit if it's 1 (parent) and keep processing if it's 
the child (greater than 1).

I think php uses pcntl_fork().

Is that how you're doing it?
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Forking AGI or GoSub

[Michael Munger]

What’s the purpose of the URL? Does it assist operators who handle the 
emergency services call?

Off the top of my head, I am not sure you can fork an AGI call from asterisk. 
Seems it would defeat the purpose of AGI, which should handle the call flow 
when it has control of the call. Have you considered have the AGI write to a 
socket of a secondary application, which will then perform the web call from 
its own process?

Seems relatively simple to write the call information to a socket as it’s 
getting processed, and then the AGI will complete in milliseconds and the call 
can continue. The secondary application can then do the URL POST. (You can 
probably whip this up in python very easily).

Not sure this solve the delay issue, however. If the secondary application 
stops running, you’d have to see if / how the AGI fails. I think it would just 
fail and the dialplan would just continue. Would need testing so it fails 
gracefully.


[cid:image001.png@01D4F6B8.F3CEF8F0]

Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

Microsoft Certified Professional

Microsoft Certified Small Business Specialist

Digium Certified Asterisk Professional

High Powered Help, Inc.

p:

678-905-8569

w:

hph.io<https://hph.io>  e: m...@hph.io<mailto:m...@hph.io>




From: asterisk-users  On Behalf Of 
Dovid Bender
Sent: Friday, April 19, 2019 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Forking AGI or GoSub

Steve,

In my case this is for emergency services. The AGI calls a web URL with the 
callers information. The call passes through Asterisk and we don't want to 
delay the call at all if the API takes 1-2 extra seconds.


On Wed, Apr 10, 2019 at 10:01 PM Steve Edwards 
mailto:asterisk@sedwards.com>> wrote:
On Wed, 10 Apr 2019, Dovid Bender wrote:

> I have an AGI that can sometimes take time complete. I don't want the
> dialplan to be held up by the agi. Is there any way to call it and have
> Asterisk continue with the dialplan?

On Wed, 10 Apr 2019, Dovid Bender wrote:

> I have an AGI that can sometimes take time complete. I don't want the
> dialplan to be held up by the agi. Is there any way to call it and have
> Asterisk continue with the dialplan?

I had a situation that required this functionality -- processing a credit
card could take a second or two and we didn't want 'dead air' for our user
experience.

I created a pthread to play 'Please hold on while we process your card and
get ready for a good time...' while the main program continued with the
card authorization.

Most of the time the auth completed before the audio finished so it
appeared to be instantaneous to the caller.

The only caveat is to not interact (stdin/stdout) with Asterisk until
'stream file' in the thread completed.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com<mailto:sedwa...@sedwards.com>  
Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Paging systems?

[Michael Munger]

Excellent point.

This is it: https://www.valcom.com/pdf/v-1109rthf.pdf

Get Outlook for Android<https://aka.ms/ghei36>




On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack" 
mailto:jnov...@comcast.net>> wrote:



Michael Munger wrote:
Does anyone have an (overhead) paging system that they like that works with SIP?

We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.

Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply battery TO 
the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections have 
existed through the last 30 + years, and since you haven't revealed the brand 
and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack


--
Dog is my Co-Pilot
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Paging systems?

[Michael Munger]

It worked on the old system.

I am open to suggestions, but don't want (or have the option) to add a
TDM card.



Michael Munger, dCAP, MCPS, MCNPS, MBSS
*Microsoft Certified Professional*
*Microsoft Certified Small Business Specialist*
*Digium Certified Asterisk Professional*
*High Powered Help, Inc.*
p:  678-905-8569
w:  hph.io <https://hph.io>  e: m...@hph.io <mailto:m...@hph.io>



On 3/21/19 3:01 PM, Sebastian Nielsen wrote:
>
> How did the page system answer the call when it was used with the
> analog system?
>
> You could propably ”fake” those signals from inside asterisk, and
> cause it to answer.
>
>  
>
> *Från:* asterisk-users  *För
> *Michael Munger
> *Skickat:* den 21 mars 2019 20:00
> *Till:* asterisk-users@lists.digium.com
> *Ämne:* [asterisk-users] Paging systems?
>
>  
>
> Does anyone have an (overhead) paging system that they like that works
> with SIP?
>
>  
>
> We’ve got a client with an old paging system that (supposedly) just
> takes an rj11 POTS connection, but when we put an SPA Cisco adapter on
> it, it doesn’t auto-answer the call, so paging never happens.
>
>  
>
>  
>
>   
>
> Michael J. Munger, dCAP, MCPS, MCNPS, MBSS
>
> *Microsoft Certified Professional*
>
> *Microsoft Certified Small Business Specialist*
>
> *Digium Certified Asterisk Professional*
>
> *High Powered Help, Inc.*
>
> p:
>
>   
>
> 678-905-8569
>
> w:
>
>   
>
> hph.io <https://hph.io>  e: m...@hph.io <mailto:m...@hph.io>
>
>  
>
>  
>
>


pEpkey.asc
Description: application/pgp-keys
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Paging systems?

[Michael Munger]

Does anyone have an (overhead) paging system that they like that works with SIP?

We've got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn't 
auto-answer the call, so paging never happens.


[cid:image001.png@01D4DFF6.9C1F1AA0]

Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

Microsoft Certified Professional

Microsoft Certified Small Business Specialist

Digium Certified Asterisk Professional

High Powered Help, Inc.

p:

678-905-8569

w:

hph.io<https://hph.io>  e: m...@hph.io<mailto:m...@hph.io>




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Cannot change astdatadir?

[Michael Munger]

I've installed asterisk from the debian repo, but, despite changing 
`astdatadir` in asterisk.conf, it insists on using `/usr/share/asterisk` 
instead of `/var/lib/asterisk`, which creates user permission problems since 
it's running as the user `asterisk`. The Record() application instantly fails 
due to permissions problems.

Instead of giving the asterisk user read / write permissions for 
/usr/share/asterisk, I was trying to change it to /var/lib/asterisk/, but it 
continuously ignores my changes.

Is this a Debian issue or am I changing the wrong thing in asterisk.conf? (I 
normally compile from source, which uses /var/lib/asterisk by default. First 
time using a repo package)

Asterisk version: 13.14.1~dfsg-2+deb9u4
OS: Debian 9.8 (stretch). Fully updated.

[cid:image001.png@01D4D272.0AFF3870]

Michael J. Munger, dCAP, MCPS, MCNPS, MBSS

Microsoft Certified Professional

Microsoft Certified Small Business Specialist

Digium Certified Asterisk Professional

High Powered Help, Inc.

p:

678-905-8569

w:

hph.io<https://hph.io>  e: m...@hph.io<mailto:m...@hph.io>




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Outbound call: caller gets no ringback on session progress

[Michael Maier]

On 17.12.18 at 11:52 Joshua C. Colp wrote:
> On Sun, Dec 16, 2018, at 4:43 AM, Michael Maier wrote:
> 
> 
> 
>>
>> Another question: is there any use case for 183 Session Progress w/o 
>> SDP? IOW: Is a 183 Session
>> Progress just a bug of the ISP? If so, problem could be solved by 
>> dropping each 183 package w/o SDP.
> 
> Nothing really comes to mind that would be accomplished by sending a 183 
> without SDP but there may be cases on the internet.
> 

As long as [1] aren't fixed, the attached patch could be used as workaround. 
This workaround drops 183 Session Progress w/o SDP.

The attached workaround assumes, that 183 Session Progress w/o SDP is followed 
by 180 Ringing. If there isn't any 180 Ringing following, this patch won't fix 
anything. In
the latter case, it would be necessary to generate inband ringing by asterisk 
or send 180 Ringing instead of the broken 183 Session Progress.

The workaround applies to 13.24.1.



Caller  AsteriskISP
-
Invite >
...
<---183 Session Progress 
w/o SDP
<---180 Ringing
<---180 Ringing
...



Another 183 Situation with *P-Early-Media* header
=

Caller  AsteriskISP
-
Invite >
...
<-- 183 Session Progress w 
SDP / P-Early-Media:sendonly
<-  183 Session Progress w SDP sendrecv
<-  RTP
--> RTP
<-- 180 Session Progress 
w/o SDP / P-Early-Media:sendonly
<-- RTP (inband ringback)
...

=> the sendonly-value of P-Early-Media seems to be ignored. Asterisk doesn't 
apply the sendonly value of the P-Early-Media header of the 183 Session 
Progress package
received from the ISP to the 183 Session Progress sent to the caller.
But this is not a problem, as the ringback can still be transferred.


Regards,
Michael


[1] https://issues.asterisk.org/jira/browse/ASTERISK-28208, 
https://issues.asterisk.org/jira/browse/ASTERISK-27994
--- a/res/res_pjsip_session.c	2019-01-20 16:37:16.98300 +0100
+++ b/res/res_pjsip_session.c	2019-01-21 02:34:06.23500 +0100
@@ -2485,6 +2485,17 @@
 {
 	struct ast_sip_session_supplement *supplement;
 	struct pjsip_status_line status = rdata->msg_info.msg->line.status;
+	pjsip_rdata_sdp_info *sdp_info;
+
+	ast_debug(3, "Response is %d %.*s\n", status.code, (int) pj_strlen(),
+			pj_strbuf());
+
+	/* Michael: ignore 183 Session Progress if there is no MEDIA in package */
+	if (status.code == 183 && (! (sdp_info = pjsip_rdata_get_sdp_info(rdata)) || sdp_info->sdp_err != PJ_SUCCESS || ! sdp_info->sdp)) {
+		ast_debug(3, "Ignore response %d %.*s because of missing SDP\n", status.code, (int) pj_strlen(),
+pj_strbuf());
+		return;
+	}
 
 	AST_LIST_TRAVERSE(>supplements, supplement, next) {
 		if (!(supplement->response_priority & response_priority)) {
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MWI Delayed on Polycom VVX phones

[Michael Keuter]

> Am 15.01.2019 um 15:23 schrieb Doug Lytle :
> 
> Hi all,
> 
> When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has 
> resulted in a MWI clearing delay of around 5 minutes.
> 
> After listening to a voicemail and deleting it, the Polycom VVX 601's MWI 
> light is left on for around five minutes, before clearing.
> 
> Installing Asterisk 13.24.1 did not fix this.
> 
> Moving back to 13.23.1 allows the MWI to clear immediately.  I see a note in 
> the change logs for 13.24.0
> 
> [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead 
> of mailboxes=##@default
> 
> Any suggestions on what to look at to diagnose?
> 
> Doug

Hi Doug,

applying this patch helped in my case (with AstLinux 1.3.x + Asterisk 13.24.1):

https://github.com/astlinux-project/astlinux/commit/3bfd9f0400e990a42e1317f4aa2bad51a3ef9f17

I am using "mailboxes=##@default" and had the issue as well (before).

Michael

http://www.mksolutions.info




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Compiling error

[Saint Michael]

>
> when compiling the latest version, it fails here

./configure   LDFLAGS="-z muldefs" --libdir=/usr/lib64
--with-unixodbc=$(odbc_config --include-prefix)/ --disable-asteriskssl
-enable-xmldoc NOISY_BUILD=yes


> gcc -o res_pjsip/config_transport.o -c res_pjsip/config_transport.c -MD
> -MT res_pjsip/config_transport.o -MF .res_pjsip_config_transport.o.d -MP
> -pthread -I/usr/src/asterisk/include-I/usr/include/libxml2  -pipe -Wall
> -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations   -g3  -O3
> -march=native -fPIC -DAST_MODULE=\"res_pjsip\"

res_pjsip/config_transport.c: In function ‘cipher_name_to_id’:

res_pjsip/config_transport.c:982:24: error: ‘PJ_SSL_SOCK_MAX_CIPHERS’
> undeclared (first use in this function)

  pj_ssl_cipher ciphers[PJ_SSL_SOCK_MAX_CIPHERS];

^

res_pjsip/config_transport.c:982:24: note: each undeclared identifier is
> reported only once for each function it appears in

res_pjsip/config_transport.c:982:16: warning: unused variable ‘ciphers’
> [-Wunused-variable]

  pj_ssl_cipher ciphers[PJ_SSL_SOCK_MAX_CIPHERS];

^

res_pjsip/config_transport.c: In function ‘handle_pjsip_list_ciphers’:

res_pjsip/config_transport.c:1106:24: error: ‘PJ_SSL_SOCK_MAX_CIPHERS’
> undeclared (first use in this function)

  pj_ssl_cipher ciphers[PJ_SSL_SOCK_MAX_CIPHERS];

^

res_pjsip/config_transport.c:1106:16: warning: unused variable ‘ciphers’
> [-Wunused-variable]

  pj_ssl_cipher ciphers[PJ_SSL_SOCK_MAX_CIPHERS];

^

make[1]: *** [res_pjsip/config_transport.o] Error 1

make[1]: Leaving directory `/usr/src/asterisk/res'

make: *** [res] Error 2
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voice mail: MWI problem / pjsip (13.24.0)

[Michael Maier]

On 28.12.18 at 13:20 Doug Lytle wrote:
>>>> Before I'm opening an issue, I would like to prove my expectations - maybe 
>>>> it isn't a problem at all or it's a problem of the phone.
> 
> Michael,
> 
> Just a side note.  I've had reports of MWI not turning off after a message 
> has been listened to under both 13.24.0 and 13.24.1.  It will typically turn 
> off after a few minutes.  We're just using chan_sip.  I haven't had any 
> reports when we were under 13.23.1

Turning off isn't a problem here: Any time I'm requesting VMs on the phone, the 
blinking MWI is turned off and no voice mails are shown on the display any more 
(I think
it's done by the phone itself) - until next REGISTER / NOTIFY is retrieved: if 
there is any unrequested / new voice mail left, it will be signaled again by 
blinking MWI and
on the display.


Regards,
Michael

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users