Re: [asterisk-users] No audio for 10 seconds and then comfort noise

2022-05-19 Thread David Cunningham 
Hi Joshua,

You're right, it was a firewall problem. One of those things where testing
a change in one place throws up a previously unseen problem somewhere else!
Thanks for the tip.


On Thu, 19 May 2022 at 21:18, Joshua C. Colp  wrote:

> On Thu, May 19, 2022 at 6:04 AM David Cunningham <
> dcunning...@voisonics.com> wrote:
>
>> Hi Dovid and Joshua,
>>
>> The PSTN is sending RTP immediately after the 200 OK, on both legs of the
>> call. Since the PCAP taken on the Asterisk server itself shows this RTP
>> from the PSTN then presumably it can't be a network issue preventing the
>> RTP.
>>
>> Having said that, the problem is not reproduced when the peer is another
>> Asterisk server on the same network, and that does point to a network
>> difference.
>>
>> Is there any other way in which the RTP keepalive might affect Asterisk's
>> behaviour?
>>
>
> No, the option only does anything if no RTP has been sent for a period of
> time. It doesn't fundamentally alter the behavior of RTP in general.
>
> Another thing to consider is that a PCAP is taken before any local
> firewall rules are applied, which can give a false impression that the
> firewall on the system is not an issue when in reality it can be. That's
> something to check.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and 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



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[asterisk-users] Asterisk 19.4.1 Now Available

2022-05-19 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
19.4.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.4.1 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1

Thank you for your continued support of Asterisk!
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[asterisk-users] Asterisk 18.12.1 Now Available

2022-05-19 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
18.12.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.12.1 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1

Thank you for your continued support of Asterisk!
-- 
_
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[asterisk-users] Asterisk 16.26.1 Now Available

2022-05-19 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
16.26.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.26.1 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
  reused for outgoing requests
  (Reported by LA)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1

Thank you for your continued support of Asterisk!
-- 
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Re: [asterisk-users] [External] [External] [External] Asterisk 18.12.0 question

2022-05-19 Thread Joshua C. Colp 
On Thu, May 19, 2022 at 1:18 PM Dan Cropp  wrote:

> After further testing, not sure this is chan_sip related.
>
>
>
> I can disable chan_sip.so from loading in modules.conf and that does solve
> the startup/loading for res_pjsip_transport_websocket.
>
> However, there is some issue with the wss transport.  Seeing this in both
> 16.26.0 (not in 16.25.0) and 18.12.0 (not in 18.11.2).
>
>
>
> REGISTER message comes in, is accepted.  However, when it goes to send the
> OPTIONS, it’s outputting the Unsupported transport.
>
>
>
>
>
> [05/19 10:11:41.992] VERBOSE[2456] res_pjsip_logger.c: <--- Received SIP
> request (907 bytes) from WSS:192.168.32.27:56443 --->
>
> REGISTER sip:mybox.mydomain.com SIP/2.0
>
> Via: SIP/2.0/WSS c2537bthsnvo.invalid;branch=z9hG4bK2816987
>
> Max-Forwards: 69
>
> To: 
>
> From: ;tag=24ipeon952
>
> Call-ID: lshogr91tba8r5f335c1g5
>
> CSeq: 2 REGISTER
>
> Authorization: Digest algorithm=MD5, username="1234", realm="asterisk",
> nonce="1652973101/72159fe10d9432b64a16fec84fc414e7", uri="sip:
> mybox.mydomain.com", response="f46f710af7db6e2e86ec2fabe38325e8",
> opaque="06a146a816d699e2", qop=auth, cnonce="meehpb38l93l", nc=0001
>
> Contact:  ;transport=ws>;+sip.ice;reg-id=1;+sip.instance="";expires=600
>
> Expires: 600
>
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
>
> Supported: path,gruu,outbound
>
> User-Agent: JsSIP 3.3.6
>
> Content-Length: 0
>
>
>
>
>
> [05/19 10:11:41.993] VERBOSE[2456] res_pjsip_logger.c: <--- Transmitting
> SIP response (482 bytes) to WSS:192.168.32.27:56443 --->
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/WSS
> c2537bthsnvo.invalid;rport=56443;received=192.168.32.27;branch=z9hG4bK2816987
>
> Call-ID: lshogr91tba8r5f335c1g5
>
> From: ;tag=24ipeon952
>
> To: ;tag=z9hG4bK2816987
>
> CSeq: 2 REGISTER
>
> Date: Thu, 19 May 2022 15:11:41 GMT
>
> Contact: ;expires=599
>
> Expires: 600
>
> Server: Asterisk PBX 18.12.0
>
> Content-Length:  0
>
>
>
>
>
> [05/19 10:11:41.994] ERROR[2456] res_pjsip.c: Error 171060 'Unsupported
> transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 1234
>

Already a known issue[1] which has been fixed, with a regression release in
progress. Not caused by the PJSIP issue you mentioned.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-30065

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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_
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Re: [asterisk-users] [External] [External] [External] Asterisk 18.12.0 question

2022-05-19 Thread Dan Cropp 
After further testing, not sure this is chan_sip related.

I can disable chan_sip.so from loading in modules.conf and that does solve the 
startup/loading for res_pjsip_transport_websocket.
However, there is some issue with the wss transport.  Seeing this in both 
16.26.0 (not in 16.25.0) and 18.12.0 (not in 18.11.2).

REGISTER message comes in, is accepted.  However, when it goes to send the 
OPTIONS, it’s outputting the Unsupported transport.


[05/19 10:11:41.992] VERBOSE[2456] res_pjsip_logger.c: <--- Received SIP 
request (907 bytes) from WSS:192.168.32.27:56443 --->
REGISTER sip:mybox.mydomain.com SIP/2.0
Via: SIP/2.0/WSS c2537bthsnvo.invalid;branch=z9hG4bK2816987
Max-Forwards: 69
To: 
From: ;tag=24ipeon952
Call-ID: lshogr91tba8r5f335c1g5
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="1234", realm="asterisk", 
nonce="1652973101/72159fe10d9432b64a16fec84fc414e7", 
uri="sip:mybox.mydomain.com", response="f46f710af7db6e2e86ec2fabe38325e8", 
opaque="06a146a816d699e2", qop=auth, cnonce="meehpb38l93l", nc=0001
Contact: 
;+sip.ice;reg-id=1;+sip.instance="";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound
User-Agent: JsSIP 3.3.6
Content-Length: 0


[05/19 10:11:41.993] VERBOSE[2456] res_pjsip_logger.c: <--- Transmitting SIP 
response (482 bytes) to WSS:192.168.32.27:56443 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 
c2537bthsnvo.invalid;rport=56443;received=192.168.32.27;branch=z9hG4bK2816987
Call-ID: lshogr91tba8r5f335c1g5
From: ;tag=24ipeon952
To: ;tag=z9hG4bK2816987
CSeq: 2 REGISTER
Date: Thu, 19 May 2022 15:11:41 GMT
Contact: ;expires=599
Expires: 600
Server: Asterisk PBX 18.12.0
Content-Length:  0


[05/19 10:11:41.994] ERROR[2456] res_pjsip.c: Error 171060 'Unsupported 
transport (PJSIP_EUNSUPTRANSPORT)' sending OPTIONS request to endpoint 1234


Identical behavior happening with Asterisk 16.26.0, but not on Asterisk 16.25.0
Configuration files are same for between Asterisk versions.

[transport3]
type = transport
bind = 0.0.0.0
protocol = wss
allow_reload = no

[1234]
type = aor
max_contacts = 1
remove_existing = yes
qualify_frequency = 60

[1234]
type = auth
auth_type = userpass
username = 1234
password = mypassword

[1234]
type = endpoint
context = IS
auth = 1234
aors = 1234
dtmf_mode = rfc4733
webrtc = yes
disallow = all
allow = ulaw
transport = transport3
acl = acl5


Might this be because PJSIP 2.12 changes to the
“WebRTC updates with AEC3 & AGC2”



From: Dan Cropp
Sent: Friday, May 13, 2022 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: RE: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 
question

Thank you Joshua!!!

Not loading chan_sip module resolved the problem.

Hope you have an awesome weekend.

From: asterisk-users 
mailto:asterisk-users-boun...@lists.digium.com>>
 On Behalf Of Joshua C. Colp
Sent: Friday, May 13, 2022 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Subject: Re: [External] [asterisk-users] [External] [External] Asterisk 18.12.0 
question

On Fri, May 13, 2022 at 3:19 PM Dan Cropp 
mailto:d...@amtelco.com>> wrote:
Thanks Joshua.

I didn’t describe that very well.

When I first noticed the res_http_transport_websocket wasn’t loading on that 
box, I compared the modules folder on both boxes.  My thought was I forgot some 
module that was required.

I noticed I forgot to include these files, so I added them to the package.  
Rolled back the VM and re-installed.  Didn’t make a difference whether they 
were present or not.
/usr/lib/asterisk/modules/codec_g729a.*
/usr/lib/asterisk/modules/codec_silk.*
/usr/lib/asterisk/modules/codec_siren14.*
/usr/lib/asterisk/modules/codec_siren7.*
/usr/lib/asterisk/modules/format_ogg_opus.so

Comparing the menuselect-tree between the two versions, only changes I see are
func_evalexten
res_aeap
res_speech_aeap
and four test_aeap_... added to the TEST_FRAMEWORK.

Would it make sense for me to modify my bash script to disable those settings, 
compile, and try installing?  Bash script configures the menuselect options and 
compiles asterisk.
Seems like that would be a better apples to apples comparison.  Eliminating the 
new features.

You can. It would also make sense as a test to just not load chan_sip and see 
what happens.

--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and 
www.asterisk.org
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_
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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

2022-05-19 Thread Joshua C. Colp 
On Thu, May 19, 2022 at 6:04 AM David Cunningham 
wrote:

> Hi Dovid and Joshua,
>
> The PSTN is sending RTP immediately after the 200 OK, on both legs of the
> call. Since the PCAP taken on the Asterisk server itself shows this RTP
> from the PSTN then presumably it can't be a network issue preventing the
> RTP.
>
> Having said that, the problem is not reproduced when the peer is another
> Asterisk server on the same network, and that does point to a network
> difference.
>
> Is there any other way in which the RTP keepalive might affect Asterisk's
> behaviour?
>

No, the option only does anything if no RTP has been sent for a period of
time. It doesn't fundamentally alter the behavior of RTP in general.

Another thing to consider is that a PCAP is taken before any local firewall
rules are applied, which can give a false impression that the firewall on
the system is not an issue when in reality it can be. That's something to
check.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and 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/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

2022-05-19 Thread David Cunningham 
Hi Dovid and Joshua,

The PSTN is sending RTP immediately after the 200 OK, on both legs of the
call. Since the PCAP taken on the Asterisk server itself shows this RTP
from the PSTN then presumably it can't be a network issue preventing the
RTP.

Having said that, the problem is not reproduced when the peer is another
Asterisk server on the same network, and that does point to a network
difference.

Is there any other way in which the RTP keepalive might affect Asterisk's
behaviour?

Thanks for your help on this.


On Thu, 19 May 2022 at 20:40, Joshua C. Colp  wrote:

> On Thu, May 19, 2022 at 3:52 AM Dovid Bender  wrote:
>
>> David,
>>
>> Are you getting any RTP from the PSTN for either leg? If not it could be
>> that they assume you are behind NAT and want to see where the SRC of the
>> RTP before they send it back. We had a few carriers that did this. The
>> easiest way to get around it was to play a 0.5 second audio clip to the
>> incoming leg. This will send RTP to the inbound carrier, causing them to
>> send RTP back to you which would then hit the terminating carrier, which
>> then sends you back RTP completing the loop. The dialplan looks
>> something like this.
>>
>> same =>n, Progress()
>> same =>n,
>> Playback(/var/lib//asterisk_custom/sounds/xc,noanswer)
>> same =>n, Dial(SIP/+${EXTEN}@carrier,,)
>>
>
> I've also seen this happen due to networking equipment, specifically the
> equipment wanting Asterisk to send packets before allowing packets in. If
> both sides of the call are in this state, then you reach a stalemate and
> media won't flow. Since rtp_keepalive is generated by Asterisk, it gets
> sent, and media starts flowing.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and 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



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
-- 
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/

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Re: [asterisk-users] No audio for 10 seconds and then comfort noise

2022-05-19 Thread Joshua C. Colp 
On Thu, May 19, 2022 at 3:52 AM Dovid Bender  wrote:

> David,
>
> Are you getting any RTP from the PSTN for either leg? If not it could be
> that they assume you are behind NAT and want to see where the SRC of the
> RTP before they send it back. We had a few carriers that did this. The
> easiest way to get around it was to play a 0.5 second audio clip to the
> incoming leg. This will send RTP to the inbound carrier, causing them to
> send RTP back to you which would then hit the terminating carrier, which
> then sends you back RTP completing the loop. The dialplan looks
> something like this.
>
> same =>n, Progress()
> same =>n,
> Playback(/var/lib//asterisk_custom/sounds/xc,noanswer)
> same =>n, Dial(SIP/+${EXTEN}@carrier,,)
>

I've also seen this happen due to networking equipment, specifically the
equipment wanting Asterisk to send packets before allowing packets in. If
both sides of the call are in this state, then you reach a stalemate and
media won't flow. Since rtp_keepalive is generated by Asterisk, it gets
sent, and media starts flowing.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and 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

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