Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

[marek]

examples of "interesting" information like ICE result and howto make 
"minimal" configuration of pjproject.conf


i.e.

for  debugging app_queue.so

core set debug 5 app_queue.so

for debugging RTP

core set debug 10 rtp_engine
core set debug 10 res_rtp_asterisk

rtp set debug on

logger.conf

rtp => debug,verbose(5)


so i mean

in 
https://github.com/asterisk/asterisk/blob/master/configs/samples/pjproject.conf.sample


by few examples try to explain  what usefull info i can get


set

[startup]
log_level=6
type=startup

and dig what's usefull is not very productive

btw we are using tools like sipcapture.org,voipmonitor.org, 
callstats.io, elasticsearch+filebeat, ... but without informations whats 
happening inside asterisk is harder to solve problems



Dne 12/12/2019 v 16:00 Joshua C. Colp napsal(a):
On Thu, Dec 12, 2019 at 10:57 AM marek > wrote:


thank you very much. this is exactly whats needed for debug

example output for your info

[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .Added new remote candidate from the request:
2.2.2.2:57536 
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .New triggered check added: 1
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 ..Sending connectivity check for check 1: [1]
1.1.1.1:17728-->2.2.2.2:57536 
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 ...Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536
: state changed from Waiting to In Progress
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536
 (nominated): connectivity check SUCCESS
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536
: state changed from In Progress to Succeeded
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .Check 1 is successful  and nominated
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .Cancelling check 0: [1]
1.1.1.1:17728-->10.128.3.150:57536  (In
Progress)
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .Check 0: [1]
1.1.1.1:17728-->10.128.3.150:57536 :
state changed from In Progress to Failed
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .ICE process complete, status=Success
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .Valid list
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 . 0: [1] 1.1.1.1:17728-->2.2.2.2:57536
 (nominated, state=Succeeded)

1.1.1.1 is asterisk on "public" ip

2.2.2.2 is router on "public" ip (jssip is behind it on private ip
10.128.3.150)


our specific case

we found problem in customers internet provider

we dont know yet what technology is the problem but "sometimes" 
respond ip of some core router ( ISP - isp core/edge router ip -
customers router ip - customers private ip ) to stun request


pjsproject debug config

pjproject.conf

[startup]
log_level=4
type=startup

btw some examples will be very helpfull


Examples of what?

--
Joshua C. Colp
Senior Software Developer
Sangoma Technologies
Check us out at www.sangoma.com  and 
www.asterisk.org 


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

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

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Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

[Joshua C. Colp]

On Thu, Dec 12, 2019 at 10:57 AM marek  wrote:

> thank you very much. this is exactly whats needed for debug
>
> example output for your info
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .Added new remote candidate from the request:
> 2.2.2.2:57536
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .New triggered check added: 1
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 ..Sending connectivity check for check 1: [1]
> 1.1.1.1:17728-->2.2.2.2:57536
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 ...Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536: state
> changed from Waiting to In Progress
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536
> (nominated): connectivity check SUCCESS
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536: state
> changed from In Progress to Succeeded
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .Check 1 is successful  and nominated
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .Cancelling check 0: [1] 1.1.1.1:17728-->
> 10.128.3.150:57536 (In Progress)
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .Check 0: [1] 1.1.1.1:17728-->10.128.3.150:57536:
> state changed from In Progress to Failed
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .ICE process complete, status=Success
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .Valid list
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 . 0: [1] 1.1.1.1:17728-->2.2.2.2:57536 (nominated,
> state=Succeeded)
>
> 1.1.1.1 is asterisk on "public" ip
>
> 2.2.2.2 is router on "public" ip (jssip is behind it on private ip
> 10.128.3.150)
>
>
> our specific case
>
> we found problem in customers internet provider
>
> we dont know yet what technology is the problem but "sometimes"  respond
> ip of some core router ( ISP - isp core/edge router ip - customers router
> ip - customers private ip ) to stun request
>
>
> pjsproject debug config
>
> pjproject.conf
>
> [startup]
> log_level=4
> type=startup
>
> btw some examples will be very helpfull
>

Examples of what?

-- 
Joshua C. Colp
Senior Software Developer
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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Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

[marek]

thank you very much. this is exactly whats needed for debug

example output for your info

[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .Added new remote candidate from the request: 
2.2.2.2:57536
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .New triggered check added: 1
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 ..Sending connectivity check for check 1: [1] 
1.1.1.1:17728-->2.2.2.2:57536
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 ...Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536: state 
changed from Waiting to In Progress
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536 
(nominated): connectivity check SUCCESS
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .Check 1: [1] 1.1.1.1:17728-->2.2.2.2:57536: state 
changed from In Progress to Succeeded
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .Check 1 is successful  and nominated
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .Cancelling check 0: [1] 
1.1.1.1:17728-->10.128.3.150:57536 (In Progress)
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .Check 0: [1] 1.1.1.1:17728-->10.128.3.150:57536: 
state changed from In Progress to Failed
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .ICE process complete, status=Success
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 .Valid list
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: : 
icess0x7f5d44081e88 . 0: [1] 1.1.1.1:17728-->2.2.2.2:57536 (nominated, 
state=Succeeded)


1.1.1.1 is asterisk on "public" ip

2.2.2.2 is router on "public" ip (jssip is behind it on private ip 
10.128.3.150)



our specific case

we found problem in customers internet provider

we dont know yet what technology is the problem but "sometimes" respond 
ip of some core router ( ISP - isp core/edge router ip - customers 
router ip - customers private ip ) to stun request



pjsproject debug config

pjproject.conf

[startup]
log_level=4
type=startup

btw some examples will be very helpfull

Marek


Dne 12/12/2019 v 14:05 Joshua C. Colp napsal(a):
On Thu, Dec 12, 2019 at 8:57 AM marek > wrote:


Asterisk is on public IP (as described in the first email)

i have 10 years experience in voip, 4 years webrtc in production.
i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the
basic mechanism

but i confess. i dont understand WHY Asterisk SOMETIMES switches
destination IP in RTP. this is not only about ICE. its about RTP
engine too which is Asterisk specific

and Asterisk DEBUG is not helping


RTP traffic is given to pjnath to send using ICE, if this fails then 
it uses the c= line. If you don't see (via ICE) then the fallback has 
occurred and pjnath didn't send it via ICE, which most likely means 
ICE negotiation failed for some reason. ICE and STUN is not encrypted 
in Wireshark, so it can be seen there easily. You can enable debug in 
logger.conf to go to console, and also increase the log_level in 
pjproject.conf to a high amount to see some pjnath messages.


The learning phase doesn't impact outgoing. It's for locking on to a 
source of media so other sources can be ignored, preventing hijacking.


--
Joshua C. Colp
Senior Software Developer
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

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

[Joshua C. Colp]

On Thu, Dec 12, 2019 at 8:57 AM marek  wrote:

> Asterisk is on public IP (as described in the first email)
>
> i have 10 years experience in voip, 4 years webrtc in production. i know
> about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism
>
> but i confess. i dont understand WHY Asterisk SOMETIMES switches
> destination IP in RTP. this is not only about ICE. its about RTP engine too
> which is Asterisk specific
>
> and Asterisk DEBUG is not helping
>

RTP traffic is given to pjnath to send using ICE, if this fails then it
uses the c= line. If you don't see (via ICE) then the fallback has occurred
and pjnath didn't send it via ICE, which most likely means ICE negotiation
failed for some reason. ICE and STUN is not encrypted in Wireshark, so it
can be seen there easily. You can enable debug in logger.conf to go to
console, and also increase the log_level in pjproject.conf to a high amount
to see some pjnath messages.

The learning phase doesn't impact outgoing. It's for locking on to a source
of media so other sources can be ignored, preventing hijacking.

-- 
Joshua C. Colp
Senior Software Developer
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

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

[marek]

Asterisk is on public IP (as described in the first email)

i have 10 years experience in voip, 4 years webrtc in production. i know 
about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism


but i confess. i dont understand WHY Asterisk SOMETIMES switches 
destination IP in RTP. this is not only about ICE. its about RTP engine 
too which is Asterisk specific


and Asterisk DEBUG is not helping


... going back to read res_rtp_asterisk.c & decrypting pcaps with wireshark


Dne 12/12/2019 v 13:02 Joshua C. Colp napsal(a):
On Thu, Dec 12, 2019 at 7:57 AM marek > wrote:


with wireshark i need decrypt traffic every call which is time
consuming. get debug from pjnat through asterisk is not possible
because of technical reasons or nobody did it?


in my case its strange that ice candidates are the same

good call

v=0
o=- 3669976329745317845 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo
m=audio 52421 RTP/SAVPF 8 0 101
c=IN IP4 10.2.152.36
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host
generation 0 network-id 1 network-cost 10
a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host
tcptype active generation 0 network-id 1 network-cost 10

bad call

v=0
o=- 2602173234285924157 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS aDrO7zRNTqNWKodpSG62Co1IDoHReEpT8Ga3
m=audio 63249 RTP/SAVPF 8 0 101
c=IN IP4 10.2.152.36
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3607370648 1 udp 2122260223 10.2.152.36 63249 typ host
generation 0 network-id 1 network-cost 10
a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host
tcptype active generation 0 network-id 1 network-cost 10


but RTP looks like

bad call (1.1.1.1 is "public" ip of PSTN SIP GW)

Got  RTP packet from 1.1.1.1:13460  (type
08, seq 002433, ts 000160, len 000160)
Sent RTP packet to 10.2.152.36:63249 
(type 08, seq 022470, ts 000160, len 000160)
Got  RTP packet from 1.1.1.1:13460  (type
08, seq 002434, ts 000320, len 000160)
Sent RTP packet to 10.2.152.36:63249 
(type 08, seq 022471, ts 000320, len 000160)
Got  RTP packet from 1.1.1.1:13460  (type
08, seq 002435, ts 000480, len 000160)

good call (1.1.1.1 is "public" ip of PSTN SIP GW, 2.2.2.2 is
public IP of router)

Got  RTP packet from 1.1.1.1:15026  (type
08, seq 021197, ts 000160, len 000160)
Sent RTP packet to *10.2.152.36:52421 
(type 08, seq 032328, ts 000160, len 000160)*

[Dec 11 16:59:53] DEBUG[44360]: res_rtp_asterisk.c:6049
ast_rtp_remote_address_set: Setting RTCP address on RTP instance
'0x7faa14005408'

Got  RTP packet from 1.1.1.1:15026  (type
08, seq 021198, ts 000320, len 000160)
Sent RTP packet to 2.2.2.2:52421  (via ICE)
(type 08, seq 032329, ts 000320, len 000160)
Got  RTP packet from 1.1.1.1:15026  (type
08, seq 021199, ts 000480, len 000160)
Sent RTP packet to 2.2.2.2:52421  (via ICE)
(type 08, seq 032330, ts 000480, len 000160)
Got  RTP packet from 1.1.1.1:15026  (type
08, seq 021200, ts 000640, len 000160)
Sent RTP packet to 2.2.2.2:52421  (via ICE)
(type 08, seq 032331, ts 000640, len 000160)
Got  RTP packet from 1.1.1.1:15026  (type
08, seq 021201, ts 000800, len 000160)

looking for the part where RTP engine switch from *10.2.152.36 to
**2.2.2.2*

it looks like**its somewhere in the learning phase


You need to look at the ICE candidates given by Asterisk as well, and 
ensure that if it is behind NAT it is configured in rtp.conf to do 
some mapping of candidates, as well as ensuring the firewall is open. 
The wireshark capture like I said will provide insight into what ICE 
is doing.


ICE is what is used to figure out the path and determine the IP 
address/port to use. If that fails, then it won't work.


I would also urge you to learn more about the lower level details of 
WebRTC if you plan on deploying it. You really need some understanding 
of ICE/STUN/DTLS-SRTP if deploying, as those are fundamental aspects 
and stuff doesn't just work in all cases. Digging into why it's not 
working takes you down to those.


--
Joshua C. Colp
Senior Software Developer
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 

Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

[Joshua C. Colp]

On Thu, Dec 12, 2019 at 7:57 AM marek  wrote:

> with wireshark i need decrypt traffic every call which is time consuming.
> get debug from pjnat through asterisk is not possible because of technical
> reasons or nobody did it?
>
>
> in my case its strange that ice candidates are the same
>
> good call
>
> v=0
> o=- 3669976329745317845 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo
> m=audio 52421 RTP/SAVPF 8 0 101
> c=IN IP4 10.2.152.36
> a=rtcp:9 IN IP4 0.0.0.0
> a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host
> generation 0 network-id 1 network-cost 10
> a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host tcptype
> active generation 0 network-id 1 network-cost 10
>
> bad call
>
> v=0
> o=- 2602173234285924157 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=msid-semantic: WMS aDrO7zRNTqNWKodpSG62Co1IDoHReEpT8Ga3
> m=audio 63249 RTP/SAVPF 8 0 101
> c=IN IP4 10.2.152.36
> a=rtcp:9 IN IP4 0.0.0.0
> a=candidate:3607370648 1 udp 2122260223 10.2.152.36 63249 typ host
> generation 0 network-id 1 network-cost 10
> a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host tcptype
> active generation 0 network-id 1 network-cost 10
>
>
> but RTP looks like
>
> bad call (1.1.1.1 is "public" ip of PSTN SIP GW)
>
> Got  RTP packet from1.1.1.1:13460 (type 08, seq 002433, ts 000160,
> len 000160)
> Sent RTP packet to  10.2.152.36:63249 (type 08, seq 022470, ts
> 000160, len 000160)
> Got  RTP packet from1.1.1.1:13460 (type 08, seq 002434, ts 000320,
> len 000160)
> Sent RTP packet to  10.2.152.36:63249 (type 08, seq 022471, ts
> 000320, len 000160)
> Got  RTP packet from1.1.1.1:13460 (type 08, seq 002435, ts 000480,
> len 000160)
>
> good call (1.1.1.1 is "public" ip of PSTN SIP GW, 2.2.2.2 is public IP of
> router)
>
> Got  RTP packet from1.1.1.1:15026 (type 08, seq 021197, ts 000160,
> len 000160)
> Sent RTP packet to  *10.2.152.36:52421 
> (type 08, seq 032328, ts 000160, len 000160)*
>
> [Dec 11 16:59:53] DEBUG[44360]: res_rtp_asterisk.c:6049
> ast_rtp_remote_address_set: Setting RTCP address on RTP instance
> '0x7faa14005408'
>
> Got  RTP packet from1.1.1.1:15026 (type 08, seq 021198, ts 000320,
> len 000160)
> Sent RTP packet to  2.2.2.2:52421 (via ICE) (type 08, seq 032329, ts
> 000320, len 000160)
> Got  RTP packet from1.1.1.1:15026 (type 08, seq 021199, ts 000480,
> len 000160)
> Sent RTP packet to  2.2.2.2:52421 (via ICE) (type 08, seq 032330, ts
> 000480, len 000160)
> Got  RTP packet from1.1.1.1:15026 (type 08, seq 021200, ts 000640,
> len 000160)
> Sent RTP packet to  2.2.2.2:52421 (via ICE) (type 08, seq 032331, ts
> 000640, len 000160)
> Got  RTP packet from1.1.1.1:15026 (type 08, seq 021201, ts 000800,
> len 000160)
>
> looking for the part where RTP engine switch from *10.2.152.36 to *
> *2.2.2.2*
>
> it looks like its somewhere in the learning phase
>

You need to look at the ICE candidates given by Asterisk as well, and
ensure that if it is behind NAT it is configured in rtp.conf to do some
mapping of candidates, as well as ensuring the firewall is open. The
wireshark capture like I said will provide insight into what ICE is doing.

ICE is what is used to figure out the path and determine the IP
address/port to use. If that fails, then it won't work.

I would also urge you to learn more about the lower level details of WebRTC
if you plan on deploying it. You really need some understanding of
ICE/STUN/DTLS-SRTP if deploying, as those are fundamental aspects and stuff
doesn't just work in all cases. Digging into why it's not working takes you
down to those.

-- 
Joshua C. Colp
Senior Software Developer
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
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Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

[marek]

with wireshark i need decrypt traffic every call which is time 
consuming. get debug from pjnat through asterisk is not possible because 
of technical reasons or nobody did it?



in my case its strange that ice candidates are the same

good call

v=0
o=- 3669976329745317845 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo
m=audio 52421 RTP/SAVPF 8 0 101
c=IN IP4 10.2.152.36
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host 
generation 0 network-id 1 network-cost 10
a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host tcptype 
active generation 0 network-id 1 network-cost 10


bad call

v=0
o=- 2602173234285924157 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS aDrO7zRNTqNWKodpSG62Co1IDoHReEpT8Ga3
m=audio 63249 RTP/SAVPF 8 0 101
c=IN IP4 10.2.152.36
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3607370648 1 udp 2122260223 10.2.152.36 63249 typ host 
generation 0 network-id 1 network-cost 10
a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host tcptype 
active generation 0 network-id 1 network-cost 10



but RTP looks like

bad call (1.1.1.1 is "public" ip of PSTN SIP GW)

Got  RTP packet from    1.1.1.1:13460 (type 08, seq 002433, ts 000160, 
len 000160)
Sent RTP packet to  10.2.152.36:63249 (type 08, seq 022470, ts 
000160, len 000160)
Got  RTP packet from    1.1.1.1:13460 (type 08, seq 002434, ts 000320, 
len 000160)
Sent RTP packet to  10.2.152.36:63249 (type 08, seq 022471, ts 
000320, len 000160)
Got  RTP packet from    1.1.1.1:13460 (type 08, seq 002435, ts 000480, 
len 000160)


good call (1.1.1.1 is "public" ip of PSTN SIP GW, 2.2.2.2 is public IP 
of router)


Got  RTP packet from    1.1.1.1:15026 (type 08, seq 021197, ts 000160, 
len 000160)
Sent RTP packet to *10.2.152.36:52421 (type 08, seq 032328, ts 000160, 
len 000160)*


[Dec 11 16:59:53] DEBUG[44360]: res_rtp_asterisk.c:6049 
ast_rtp_remote_address_set: Setting RTCP address on RTP instance 
'0x7faa14005408'


Got  RTP packet from    1.1.1.1:15026 (type 08, seq 021198, ts 000320, 
len 000160)
Sent RTP packet to  2.2.2.2:52421 (via ICE) (type 08, seq 032329, ts 
000320, len 000160)
Got  RTP packet from    1.1.1.1:15026 (type 08, seq 021199, ts 000480, 
len 000160)
Sent RTP packet to  2.2.2.2:52421 (via ICE) (type 08, seq 032330, ts 
000480, len 000160)
Got  RTP packet from    1.1.1.1:15026 (type 08, seq 021200, ts 000640, 
len 000160)
Sent RTP packet to  2.2.2.2:52421 (via ICE) (type 08, seq 032331, ts 
000640, len 000160)
Got  RTP packet from    1.1.1.1:15026 (type 08, seq 021201, ts 000800, 
len 000160)


looking for the part where RTP engine switch from *10.2.152.36 to **2.2.2.2*

it looks like**its somewhere in the learning phase*
*


Dne 12/12/2019 v 11:51 Joshua C. Colp napsal(a):
On Thu, Dec 12, 2019 at 6:39 AM marek > wrote:


hi,

i have following topology

PSTN - Asterisk  internet -  router - jssip client (wss)

Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp
for SIP
connection to PSTN

router - public IP/private IP (NAT)

jssip client - private IP - sip over websocket to Asterisk PJSIP


~30% of calls has problem with no audio. reason is that Asterisk is
sending RTP to private IP of jssip

SDP looks the same for good call and bad call too


i searched through res_rtp_asterisk.c but i'm not sure where to put
DEBUG info about which IP and why Asterisk pick for RTP

any hint?


is it possible debug Asterisk STUN request/response ? or is it
hidden in
pjsip internals?


ICE is performed using pjnath, which is part of pjproject and not 
Asterisk itself. Looking at the SDP and the ICE candidates can tell 
you the possibilities for the paths, and a wireshark capture can show 
you the actual traffic going back and forth and what is being attempted.


--
Joshua C. Colp
Senior Software Developer
Sangoma Technologies
Check us out at www.sangoma.com  and 
www.asterisk.org 


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Re: [asterisk-users] asterisk pjsip webrtc rtp to private IP

[Joshua C. Colp]

On Thu, Dec 12, 2019 at 6:39 AM marek  wrote:

> hi,
>
> i have following topology
>
> PSTN - Asterisk  internet -  router - jssip client (wss)
>
> Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP
> connection to PSTN
>
> router - public IP/private IP (NAT)
>
> jssip client - private IP - sip over websocket to Asterisk PJSIP
>
>
> ~30% of calls has problem with no audio. reason is that Asterisk is
> sending RTP to private IP of jssip
>
> SDP looks the same for good call and bad call too
>
>
> i searched through res_rtp_asterisk.c but i'm not sure where to put
> DEBUG info about which IP and why Asterisk pick for RTP
>
> any hint?
>
>
> is it possible debug Asterisk STUN request/response ? or is it hidden in
> pjsip internals?
>

ICE is performed using pjnath, which is part of pjproject and not Asterisk
itself. Looking at the SDP and the ICE candidates can tell you the
possibilities for the paths, and a wireshark capture can show you the
actual traffic going back and forth and what is being attempted.

-- 
Joshua C. Colp
Senior Software Developer
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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[asterisk-users] asterisk pjsip webrtc rtp to private IP

[marek]

hi,

i have following topology

PSTN - Asterisk  internet -  router - jssip client (wss)

Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP 
connection to PSTN


router - public IP/private IP (NAT)

jssip client - private IP - sip over websocket to Asterisk PJSIP


~30% of calls has problem with no audio. reason is that Asterisk is 
sending RTP to private IP of jssip


SDP looks the same for good call and bad call too


i searched through res_rtp_asterisk.c but i'm not sure where to put 
DEBUG info about which IP and why Asterisk pick for RTP


any hint?


is it possible debug Asterisk STUN request/response ? or is it hidden in 
pjsip internals?



Marek





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[asterisk-users] Asterisk 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5 Now Available (Security)

[Asterisk Development Team]

The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.

These releases are available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk/releases
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases

The following security vulnerabilities were resolved in these versions:

* AST-2019-006: SIP request can change address of a SIP peer.
  A SIP request can be sent to Asterisk that can change a SIP peer’s IP
  address. A REGISTER does not need to occur, and calls can be hijacked as a
  result. The only thing that needs to be known is the peer’s name;
  authentication details such as passwords do not need to be known. This
  vulnerability is only exploitable when the “nat” option is set to the
  default, or “auto_force_rport”.

* AST-2019-007: AMI user could execute system commands.
  A remote authenticated Asterisk Manager Interface (AMI) user without
  “system” authorization could use a specially crafted “Originate” AMI
  request to execute arbitrary system commands.

* AST-2019-008: Re-invite with T.38 and malformed SDP causes crash.
  If Asterisk receives a re-invite initiating T.38 faxing and has a port of 0
  and no c line in the SDP, a crash will occur.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
https://downloads.asterisk.org/pub/security/AST-2019-008.pdf

Thank you for your continued support of Asterisk!-- 
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Re: [asterisk-users] Asterisk 16.6.1: PJSIP: delayed action of core since update to 16.6.1

[Joshua C. Colp]

On Wed, Nov 20, 2019 at 5:40 AM O. Hartmann  wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA256
>
> Am Sat, 16 Nov 2019 07:39:08 -0400
> "Joshua C. Colp"  schrieb:
>
> > On Sat, Nov 16, 2019 at 4:07 AM O. Hartmann 
> wrote:
> >
> > > -BEGIN PGP SIGNED MESSAGE-
> > > Hash: SHA256
> > >
> > >
> > > Hello,
> > >
> > > we're running a small Asterisk appliance on a PCengine APU2C4. Base
> > > operating system is
> > > FreeBSD 12-STABLE, most recent incarnation as of today.
> > >
> > > Since update of port net/asterisk16 to the latest bug fix revision
> 16.6.1,
> > > we face a severe
> > > "slowdown" of everything that the Asterisk core performs, i.e. outgoing
> > > calls are delayed ~ 20
> > > seconds and I guess incoming calls suffer the same until they gett
> patched
> > > through to an
> > > endpoint/telephone. We also register a higher load on idle asterisk
> > > process since the last
> > > update.
> > >
> > > Here is an example when calling two attached physical phones directly,
> > > which performed prior
> > > to 16.6.1 almost immediately and now takes up to 30 seconds to make the
> > > called ednpoint ring.
> > >
> > > The calling phone/endpoint sinals by callsound that it is calling, and
> the
> > > sound changes then
> > > (some kind of different octave/tune, don't know) when the asterisk core
> > > reports
> > >
> > > [Nov 15 13:21:24]   == Using SIP RTP Audio TOS bits 184
> > >
> > > (see below). It is here approx 10 seconds, but there are situations
> were
> > > it might more (as
> > > observed). the host has no further load so far!
> > >
> > > Incoming testcalls we made from wireless/mobile show the same. It
> seems,
> > > asterisk is acting as
> > > a black hole delaying device for approx 10 seconds until it decides to
> > > pass the call through
> > > to an endpoint and then it takes another 10 seconds until the endpoint
> > > starts ringing (it is
> > > in fact a group of phones ringing alltogether).
> > >
> > > I can not see anything unusual with the underlying OS or some critical
> > > debug messages from
> > > asterisk itself.
> > >
> > > Any ideas?
> > >
> >
> > Do you have a STUN server configured in rtp.conf? If you do, is it
> > reachable, does the problem go away if you remove it?
> >
>
> Is there anything wrong/buggy with the implementation of the STUN service
> in 16.6.1?
>

In that specific version? No. That code itself hasn't been touched in
years, so the problem applies to every version. Using the defined STUN
server to get a server reflexive ICE candidate is a blocking process. If
the server isn't reachable or is extremely slow then it has to wait until a
timeout occurs, causing a delay.

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.sangoma.com & www.asterisk.org
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Re: [asterisk-users] Asterisk 16.6.1: PJSIP: delayed action of core since update to 16.6.1

[Joshua C. Colp]

On Sat, Nov 16, 2019 at 4:07 AM O. Hartmann  wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA256
>
>
> Hello,
>
> we're running a small Asterisk appliance on a PCengine APU2C4. Base
> operating system is
> FreeBSD 12-STABLE, most recent incarnation as of today.
>
> Since update of port net/asterisk16 to the latest bug fix revision 16.6.1,
> we face a severe
> "slowdown" of everything that the Asterisk core performs, i.e. outgoing
> calls are delayed ~ 20
> seconds and I guess incoming calls suffer the same until they gett patched
> through to an
> endpoint/telephone. We also register a higher load on idle asterisk
> process since the last
> update.
>
> Here is an example when calling two attached physical phones directly,
> which performed prior
> to 16.6.1 almost immediately and now takes up to 30 seconds to make the
> called ednpoint ring.
>
> The calling phone/endpoint sinals by callsound that it is calling, and the
> sound changes then
> (some kind of different octave/tune, don't know) when the asterisk core
> reports
>
> [Nov 15 13:21:24]   == Using SIP RTP Audio TOS bits 184
>
> (see below). It is here approx 10 seconds, but there are situations were
> it might more (as
> observed). the host has no further load so far!
>
> Incoming testcalls we made from wireless/mobile show the same. It seems,
> asterisk is acting as
> a black hole delaying device for approx 10 seconds until it decides to
> pass the call through
> to an endpoint and then it takes another 10 seconds until the endpoint
> starts ringing (it is
> in fact a group of phones ringing alltogether).
>
> I can not see anything unusual with the underlying OS or some critical
> debug messages from
> asterisk itself.
>
> Any ideas?
>

Do you have a STUN server configured in rtp.conf? If you do, is it
reachable, does the problem go away if you remove it?

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.sangoma.com & www.asterisk.org
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[asterisk-users] Asterisk 16.6.1: PJSIP: delayed action of core since update to 16.6.1

[O. Hartmann]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256


Hello,

we're running a small Asterisk appliance on a PCengine APU2C4. Base operating 
system is
FreeBSD 12-STABLE, most recent incarnation as of today.

Since update of port net/asterisk16 to the latest bug fix revision 16.6.1, we 
face a severe
"slowdown" of everything that the Asterisk core performs, i.e. outgoing calls 
are delayed ~ 20
seconds and I guess incoming calls suffer the same until they gett patched 
through to an
endpoint/telephone. We also register a higher load on idle asterisk process 
since the last
update.

Here is an example when calling two attached physical phones directly, which 
performed prior
to 16.6.1 almost immediately and now takes up to 30 seconds to make the called 
ednpoint ring.

The calling phone/endpoint sinals by callsound that it is calling, and the 
sound changes then
(some kind of different octave/tune, don't know) when the asterisk core reports

[Nov 15 13:21:24]   == Using SIP RTP Audio TOS bits 184

(see below). It is here approx 10 seconds, but there are situations were it 
might more (as
observed). the host has no further load so far!

Incoming testcalls we made from wireless/mobile show the same. It seems, 
asterisk is acting as
a black hole delaying device for approx 10 seconds until it decides to pass the 
call through
to an endpoint and then it takes another 10 seconds until the endpoint starts 
ringing (it is
in fact a group of phones ringing alltogether).

I can not see anything unusual with the underlying OS or some critical debug 
messages from
asterisk itself.

Any ideas?

Kind regards,

O. Hartmann

[...]
==>> START [Nov 15 13:21:06]   == Setting global variable 'SIPDOMAIN' to 
'192.168.2.1'
[Nov 15 13:21:15]   == Using SIP RTP Audio TOS bits 184
[Nov 15 13:21:15] -- Executing [511@internalsip_o2:1] 
NoOp("PJSIP/501-0008", "") in
new stack
[Nov 15 13:21:15] -- Executing [511@internalsip_o2:2] 
Progress("PJSIP/501-0008", "")
in new stack
[Nov 15 13:21:15] -- Executing [511@internalsip_o2:3] 
Gosub("PJSIP/501-0008",
"subSetChannelLocale,start,1(abschnitt211,de,de_DE)") in new stack
[Nov 15 13:21:15] -- Executing [start@subSetChannelLocale:1] 
Verbose("PJSIP/501-0008",
"2, subSetChannelLocale: ARG1: musicclass=abschnitt211, ARG2: tonezone=de, ARG3:
language=de_DE") in new stack [Nov 15 13:21:15]> 0x807c82000 -- Strict 
RTP learning
after remote address set to: 192.168.2.50:17702
[Nov 15 13:21:15]   ==  subSetChannelLocale: ARG1: musicclass=abschnitt211, 
ARG2: tonezone=de,
ARG3: language=de_DE
[Nov 15 13:21:15] -- Executing [start@subSetChannelLocale:2]
Progress("PJSIP/501-0008", "") in new stack
[Nov 15 13:21:15] -- Executing [start@subSetChannelLocale:3] 
Set("PJSIP/501-0008",
"CHANNEL(musicclass)=abschnitt211") in new stack
[Nov 15 13:21:15] -- Executing [start@subSetChannelLocale:4] 
Set("PJSIP/501-0008",
"CHANNEL(tonezone)=de") in new stack
[Nov 15 13:21:15] -- Executing [start@subSetChannelLocale:5] 
Set("PJSIP/501-0008",
"CHANNEL(language)=de_DE") in new stack
[Nov 15 13:21:15] -- Executing [start@subSetChannelLocale:6] 
Return("PJSIP/501-0008",
"") in new stack
[Nov 15 13:21:15] -- Executing [511@internalsip_o2:4] 
Dial("PJSIP/501-0008",
"PJSIP/511,45,Ttr") in new stack
[Nov 15 13:21:15] -- Called PJSIP/511
[Nov 15 13:21:15]> 0x807c82000 -- Strict RTP switching to RTP target 
address
192.168.2.50:17702 as source
[Nov 15 13:21:20]> 0x807c82000 -- Strict RTP learning complete
- - - Locking on source address 192.168.2.50:17702

===>> CHANGE OF RINGING/CONNECTING [Nov 15 13:21:24]   == Using SIP RTP Audio 
TOS bits 184
[Nov 15 13:21:25] -- PJSIP/511-0009 is ringing
[Nov 15 13:21:25] -- PJSIP/511-0009 is ringing
[Nov 15 13:21:32] -- PJSIP/511-0009 answered PJSIP/501-0008
[Nov 15 13:21:32]> 0x807c85000 -- Strict RTP learning after remote 
address set to:
192.168.2.51:24094
[Nov 15 13:21:32] -- Channel PJSIP/511-0009 joined 'simple_bridge' 
basic-bridge

[Nov 15 13:21:32] -- Channel PJSIP/501-0008 joined 'simple_bridge' 
basic-bridge

[Nov 15 13:21:32]> 0x807c85000 -- Strict RTP switching to RTP target 
address
192.168.2.51:24094 as source
[Nov 15 13:21:37] -- Channel PJSIP/511-0009 left 'simple_bridge' 
basic-bridge

[Nov 15 13:21:37] -- Channel PJSIP/501-0008 left 'simple_bridge' 
basic-bridge

[Nov 15 13:21:37]   == Spawn extension (internalsip_o2, 511, 4) exited non-zero 
on
'PJSIP/501-0008'
[...]

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[asterisk-users] Asterisk 17.0.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
17.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.0.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
  declined stream causes crash
  (Reported by Alexei
  Gradinari)
 * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
  no body causes crash
  (Reported by Gil Richard)
 * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
  reINVITE
  (Reported by Francesco Castellano)
 * ASTERISK-28260 - Asterisk segfault when rtp negotiation is
  wrong or fails
  (Reported by Sotiris Ganouris)
 * ASTERISK-28127 - Buffer overflow for DNS SRV/NAPTR records
  
  (Reported by Jan Hoffmann)
 * ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
  Upgrade requests
  (Reported by Sean Bright)

New Features made in this release:
---
 * ASTERISK-28403 - Add native Prometheus support to Asterisk
  
  (Reported by Matt Jordan)
 * ASTERISK-28375 - res_pjsip: New configuration setting to
  allow disabling norefersub
  (Reported by Dan Cropp)
 * ASTERISK-28320 - Added ARI resource
  /ari/channels/{channelid}/rtp_statistics
  (Reported by
  sungtae kim)
 * ASTERISK-28267 - res_stasis: Add ability to switch
  applications
  (Reported by Benjamin Keith Ford)
 * ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
  in Contact header in chan_pjsip
  (Reported by Torrey
  Searle)
 * ASTERISK-27971 - res_pjsip: Implement additional SIP RFCs for
  Google Voice trunk compatability
  (Reported by Nick French)

Bugs fixed in this release:
---
 * ASTERISK-28561 - Asterisk Deadlocks
  (Reported by
  Aheliotech)
 * ASTERISK-28575 - MWI Send Notify Crash on 16.6
 
  (Reported by Joshua Elson)
 * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
  16.5
  (Reported by Niklas Larsson)
 * ASTERISK-28521 - pjsip: Memory Leak
  (Reported by Mark)
 * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
  (Reported
  by Cyril Ramière)
 * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
 
  (Reported by Joshua C. Colp)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
  FreeBSD
  (Reported by Guido Falsi)
 * ASTERISK-28499 - translate: Crash when frame does not have a
  "src" field set
  (Reported by Gregory Massel)
 * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
  re-register
  (Reported by Chris Savinovich)
 * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
  characters, NEC only supports up to 32 characters
 
  (Reported by Dan Cropp)
 * ASTERISK-28505 - app_voicemail/IMAP: segfault in
  leave_voicemail because not checking mailstream
  (Reported
  by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
  (Reported
  by Kilburn)
 * ASTERISK-28472 - Asterisk occasionally passes a NULL as
  srtp->session to srtp_protect/unprotect causing SEGV
 
  (Reported by Jonas Swiatek)
 * ASTERISK-28498 - cel / cdr: Event times may be incorrect

  (Reported by Joshua C. Colp)
 * ASTERISK-28480 - json integer overflow in ssrc and timestamp

  (Reported by Salah Ahmed)
 * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
  entries
  (Reported by Ian Jones)
 * ASTERISK-28483 - packet lost on UDPTL wrap around
 
  (Reported by Torrey Searle)
 * ASTERISK-28477 - Crash when not specifying "dbfile" in
  res_config_sqlite3.conf
  (Reported by Dennis)
 * ASTERISK-28478 - Crash performing "core reload" with modified
  res_config_sqlite3.conf
  (Reported by Dennis)
 * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
  deadlocks (in chan_sip)
  (Reported by Walter Doekes)
 * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
 
  (Reported by Sergej Kasumovic)
 * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
  systems caused by ASTERISK-28317
  (Reported by abelbeck)
 * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable
  
  (Reported by Michael Maier)
 * ASTERISK-26006 - Show offending IP for TLS setup failures in
  logs
  (Reported by Oleksandr Natalenko)
 * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
  not logged
  (Reported by Bernhard Schmidt)
 * ASTERISK-26968 - chan_pjsip: Transfer() does not result in
  TRANSFERSTATUS reflecting SIP response to transfer
 
  (Reported by Dan Cropp)
 * ASTERISK-28419 - app_amd: Does 

[asterisk-users] [asterisk-app-dev] Proposed change to External Media API

[George Joseph]

When we created the External Media addition to ARI we created an
ExternalMedia object to be returned from the channels/externalMedia REST
endpoint.  This object contained the channel object that was created plus
local_address and local_port attributes (which are also in the Channel
variables).  At the time, we thought that creating an ExternalMedia object
would give us more flexibility in the future but as we created the sample
speech to text application, we discovered that it doesn't work so well with
ARI client libraries that a) don't have the ExternalMedia object defined
and/or b) can't promote the embedded channel structure to a first-class
Channel object.

Example:

A common pattern using the node-ari-client is to create a new Channel
object, attach an event handler to it, then call originate on it like so...

chan = ari.channels.Channel();
chan.on('StasisStart', );
chan.originate(...);

With the current ExternalMedia API:

chan = ari.channels.Channel();
chan.on('StasisStart', );
chan.externalMedia(...);

This doesn't work however because the return from channels/externalMedia
isn't a Channel.  It's an ExternalMedia object with an chlld object that
looks like a Channel but has no Channel behavior attached to it.  The event
handler added to chan will never get called and you can't attach handlers
or perform any operations on ExternalMedia.channel because it's just a
plain object, not an instance of Channel.

Realistically, it doesn't make sense to force client library
implementations to create special logic to promote the
ExternalMedia.channel object into an instance of Channel and since External
Media is a new capability anyway, it seems that the least painful solution
is to remove the ExternalMedia object and have channels/externalMedia
return a Channel object directly, just like channels/create and
channels/originate.  As I described above, the only other attributes of
ExternalMedia were the local address and port and they're already available
in the Channel variables anyway.

I would think that this change would make things easier for ARI developers
but I wanted to make sure that you knew about it in advance and had a
chance to comment.  There will be a Gerrit review up for this change later
this morning.

Also...  I mentioned the "sample speech to text application" above.  It's
working and will be published next week.

Comments?  Questions?

-- 
*George Joseph*
Digium - A Sangoma Company | Software Developer | Software Engineering
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct/fax: +1 256 428 6012
Check us out at: https://digium.com · https://sangoma.com
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Re: [asterisk-users] Asterisk and CentOS 8

[Carlos Chavez]

    They only problem I have found so far is while trying to install 
Alembic for SQLAlchemy (for realtime configs).  Those are the only 
packages that I cannot get working properly.  Vanilla Asterisk works 
fine  with the only extra package needed being libedit-devel that is not 
included in any "official" repo.  You need to download the Fedora Core 
29 packages to in order to successfully compile Asterisk.  That being 
said, I would not recommend trying to put this in production any time soon.


On 10/17/2019 11:19 AM, George Joseph wrote:
At the current time, we do not recommend attempting to build Asterisk 
on CentOS 8.  Many packages Asterisk uses are not yet available and 
would require building from their sources.  The Asterisk packages are 
also not available in the EPEL 8 or CentOS 8 repositories yet for the 
same reason.


We'll update you when we think it's safe.


--
*George Joseph*
Digium - A Sangoma Company | Software Developer | Software Engineering
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct/fax: +1 256 428 6012
Check us out at: https://digium.com  · 
https://sangoma.com 



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[asterisk-users] Asterisk and CentOS 8

[George Joseph]

At the current time, we do not recommend attempting to build Asterisk on
CentOS 8.  Many packages Asterisk uses are not yet available and would
require building from their sources.  The Asterisk packages are also not
available in the EPEL 8 or CentOS 8 repositories yet for the same reason.

We'll update you when we think it's safe.


-- 
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Digium - A Sangoma Company | Software Developer | Software Engineering
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct/fax: +1 256 428 6012
Check us out at: https://digium.com · https://sangoma.com
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Re: [asterisk-users] [asterisk-app-dev] ARI Channel recording

[Joshua C. Colp]

On Wed, Oct 16, 2019, at 3:03 PM, Marcelo Garay wrote:
> Thank you for your answer!! 
> 
> Unfortunately I'm using the CEF browser based on Chromium and it 
> doesn't support H264 because license isn't free so renegotiation is not 
> an option.
> 
> I've noticed when recording a channel with video asterisk automatically 
> tries to save the video feed to a separate file besides the .wav. In my 
> case I can see "file.c:1484 ast_writefile: No such format 'vp9' " error 
> in the logs, so I would assume is just that the code for VP9 encoding 
> hasn't been added to Asterisk yet. Do you know if this is due to any 
> other reason besides nobody taking the time to implement it (reasons 
> like VP9 licensing, performance hit, etc.)? It seems like VP9 is 
> royalty-free and the encoder source code is on GitHub. I might try to 
> look into making a PR for this sometime in the future if I have some 
> time, but I don't want to waste my time if this idea has already been 
> discussed among developers and discarded for some reason.

Encoding is not the same as file recording and playback. It's how the data is 
stored in a file and retrieved, which doesn't involve any conversion. I don't 
think anyone has discussed working on such a thing or thought about it really.

-- 
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

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[asterisk-users] Asterisk 16.6.1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
16.6.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.6.1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
  16.5
  (Reported by Niklas Larsson)
 * ASTERISK-28575 - MWI Send Notify Crash on 16.6
 
  (Reported by Joshua Elson)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.1

Thank you for your continued support of Asterisk!
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[asterisk-users] Asterisk 13.29.1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
13.29.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.29.1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
  16.5
  (Reported by Niklas Larsson)
 * ASTERISK-28575 - MWI Send Notify Crash on 16.6
 
  (Reported by Joshua Elson)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.1

Thank you for your continued support of Asterisk!
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Re: [asterisk-users] [asterisk-app-dev] ARI Channel recording

[Joshua C. Colp]

On Tue, Oct 15, 2019, at 5:14 PM, Marcelo Garay wrote:
> Hello,
> 
> I’m trying to record video on a channel (from webrtc) using ARI (POST 
> /channels/{channelId}/record), but when I specify h264 for the format I 
> get error: ast_writestream: Unable to translate to format h264, source 
> format vp9
> 
> So my specific questions are:
> 
> 1) Can video be recorded using ARI?

I don't know if it's ever really been tried, recording/playback of video is not 
something anyone has put a focus n.
 
> 2) What file formats can be used to record video?

Only really h264 and h263 I believe.
 
> 3) If H264 is possible, how do I solve the error shown above?

You'd need to negotiate h264 on the channel. As it is you're using VP9, and 
Asterisk does not do video transcoding.

-- 
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

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[asterisk-users] Asterisk 16.6.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
16.6.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.6.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
  declined stream causes crash
  (Reported by Alexei
  Gradinari)

Bugs fixed in this release:
---
 * ASTERISK-28521 - pjsip: Memory Leak
  (Reported by Mark)
 * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
  (Reported
  by Cyril Ramière)
 * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
 
  (Reported by Joshua C. Colp)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
  FreeBSD
  (Reported by Guido Falsi)
 * ASTERISK-28511 - codec_resample: Bad sound quality when up
  sampling from SLIN16 to SLIN32
  (Reported by Ruddy G)
 * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
  PRI channel hangs up
  (Reported by Frederic LE FOLL)
 * ASTERISK-28527 - ChanIsAvail() creates a CDR if
  unanswered=yes is set in cdr.conf
  (Reported by Frederic LE
  FOLL)
 * ASTERISK-28499 - translate: Crash when frame does not have a
  "src" field set
  (Reported by Gregory Massel)
 * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
  type not at end of a struct
  (Reported by Alexander Traud)
 * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
  re-register
  (Reported by Chris Savinovich)
 * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
  characters, NEC only supports up to 32 characters
 
  (Reported by Dan Cropp)
 * ASTERISK-28505 - app_voicemail/IMAP: segfault in
  leave_voicemail because not checking mailstream
  (Reported
  by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
  (Reported
  by Kilburn)
 * ASTERISK-28472 - Asterisk occasionally passes a NULL as
  srtp->session to srtp_protect/unprotect causing SEGV
 
  (Reported by Jonas Swiatek)
 * ASTERISK-28498 - cel / cdr: Event times may be incorrect

  (Reported by Joshua C. Colp)
 * ASTERISK-28480 - json integer overflow in ssrc and timestamp

  (Reported by Salah Ahmed)
 * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
  entries
  (Reported by Ian Jones)
 * ASTERISK-28483 - packet lost on UDPTL wrap around
 
  (Reported by Torrey Searle)
 * ASTERISK-28477 - Crash when not specifying "dbfile" in
  res_config_sqlite3.conf
  (Reported by Dennis)
 * ASTERISK-28478 - Crash performing "core reload" with modified
  res_config_sqlite3.conf
  (Reported by Dennis)
 * ASTERISK-26968 - chan_pjsip: Transfer() does not result in
  TRANSFERSTATUS reflecting SIP response to transfer
 
  (Reported by Dan Cropp)
 * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
  deadlocks (in chan_sip)
  (Reported by Walter Doekes)

New Features made in this release:
---
 * ASTERISK-17808 - [patch] Unregister a realtime moh class

  (Reported by Byron Clark)
 * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
  chan_pjsip to setup From header URI domain
  (Reported by
  Stas Kobzar)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0

Thank you for your continued support of Asterisk!
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[asterisk-users] Asterisk 13.29.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
13.29.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.29.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-28521 - pjsip: Memory Leak
  (Reported by Mark)
 * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
  (Reported
  by Cyril Ramière)
 * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
 
  (Reported by Joshua C. Colp)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
  FreeBSD
  (Reported by Guido Falsi)
 * ASTERISK-28527 - ChanIsAvail() creates a CDR if
  unanswered=yes is set in cdr.conf
  (Reported by Frederic LE
  FOLL)
 * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
  PRI channel hangs up
  (Reported by Frederic LE FOLL)
 * ASTERISK-28511 - codec_resample: Bad sound quality when up
  sampling from SLIN16 to SLIN32
  (Reported by Ruddy G)
 * ASTERISK-28499 - translate: Crash when frame does not have a
  "src" field set
  (Reported by Gregory Massel)
 * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
  type not at end of a struct
  (Reported by Alexander Traud)
 * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
  re-register
  (Reported by Chris Savinovich)
 * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
  characters, NEC only supports up to 32 characters
 
  (Reported by Dan Cropp)
 * ASTERISK-28505 - app_voicemail/IMAP: segfault in
  leave_voicemail because not checking mailstream
  (Reported
  by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
  (Reported
  by Kilburn)
 * ASTERISK-28472 - Asterisk occasionally passes a NULL as
  srtp->session to srtp_protect/unprotect causing SEGV
 
  (Reported by Jonas Swiatek)
 * ASTERISK-28498 - cel / cdr: Event times may be incorrect

  (Reported by Joshua C. Colp)
 * ASTERISK-28483 - packet lost on UDPTL wrap around
 
  (Reported by Torrey Searle)
 * ASTERISK-28480 - json integer overflow in ssrc and timestamp

  (Reported by Salah Ahmed)
 * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
  entries
  (Reported by Ian Jones)
 * ASTERISK-28477 - Crash when not specifying "dbfile" in
  res_config_sqlite3.conf
  (Reported by Dennis)
 * ASTERISK-28478 - Crash performing "core reload" with modified
  res_config_sqlite3.conf
  (Reported by Dennis)
 * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
  deadlocks (in chan_sip)
  (Reported by Walter Doekes)

New Features made in this release:
---
 * ASTERISK-17808 - [patch] Unregister a realtime moh class

  (Reported by Byron Clark)
 * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
  chan_pjsip to setup From header URI domain
  (Reported by
  Stas Kobzar)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.0

Thank you for your continued support of Asterisk!
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Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints

[Joshua C. Colp]

On Fri, Oct 4, 2019, at 1:45 AM, Andreas Wehrmann wrote:
> 
> On 03/10/2019 16:24, Joshua C. Colp wrote:
> > In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately 
> > codec negotiation is not written or implemented in the way you need. There 
> > are some hints provided internally for outgoing legs but the result is 
> > still ultimately independent. That is: Each leg is negotiated from Asterisk 
> > to the endpoint, not endpoint to endpoint via Asterisk. This works for the 
> > vast majority of users as they have media flowing through Asterisk (by 
> > choice or via use of features) and are fine with transcoding (generally 
> > using codecs which aren't that costly or low channel count).
> >
> > Asterisk 16 has some of the foundational work to improve this through the 
> > implementation of streams but noone has worked on extending the codec 
> > negotiation support.
> >
> > [1] 
> > https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_MEDIA_OFFER
> >
> 
> Hey Joshua,
> 
> do you think it might be possible to achieve this by writing a 
> supplement for the PJ part?

PJSIP is only part of the equation, the information still has to transition 
across the core.

> 
> What happens when the other side answers, but before the incoming call 
> is answered.
> Is there a place in the code where, at that point, I have information 
> about both channels
> and could theoretically influence the answer for the incoming call?

Nope. That information is not currently exchanged or available, which is part 
of the problem.

-- 
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

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Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints

[Andreas Wehrmann]

On 03/10/2019 16:24, Joshua C. Colp wrote:

In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately 
codec negotiation is not written or implemented in the way you need. There are 
some hints provided internally for outgoing legs but the result is still 
ultimately independent. That is: Each leg is negotiated from Asterisk to the 
endpoint, not endpoint to endpoint via Asterisk. This works for the vast 
majority of users as they have media flowing through Asterisk (by choice or via 
use of features) and are fine with transcoding (generally using codecs which 
aren't that costly or low channel count).

Asterisk 16 has some of the foundational work to improve this through the 
implementation of streams but noone has worked on extending the codec 
negotiation support.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_MEDIA_OFFER



Hey Joshua,

do you think it might be possible to achieve this by writing a 
supplement for the PJ part?


What happens when the other side answers, but before the incoming call 
is answered.
Is there a place in the code where, at that point, I have information 
about both channels

and could theoretically influence the answer for the incoming call?


All the best,
Andy


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Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints

[Joshua C. Colp]

On Thu, Oct 3, 2019, at 11:10 AM, Andreas Wehrmann wrote:
> 
> On 03.10.19 15:08, Administrator TOOTAI wrote:
> 
> > Before calling the gatreway add
> >
> > same = n,set(SIP_CODEC=alaw)
> >
> > [...]
> >
> 
> Hey there,
> 
> that doesn't work as it seems to be implemented for chan_sip only;
> I'm using chan_pjsip; sorry if I didn't explain myself properly.
> 
> Anyway, in my case that would not really be an acceptable solution anyway,
> because I need the called party to be able to pick from the range of 
> codecs presented to it
> because the codec chosen by the destination might change (my example is 
> a simplified version).
> 
> I don't think putting the burden of worrying about audio codecs on the 
> dialplan writer is a good idea,
> since this should be dealt with automatically with respect to what is 
> configured and negotiated.
> This is also because in the systems I have to work with, the 'engineers' 
> usual provide the configuration (endpoints, NAT config and the like)
> while the technicians implement the dialplan (or the business logic so 
> to speak) according to customer needs.
> They (the technicians) usually don't know (much) about codecs or how the 
> channels techs work exactly...

In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately 
codec negotiation is not written or implemented in the way you need. There are 
some hints provided internally for outgoing legs but the result is still 
ultimately independent. That is: Each leg is negotiated from Asterisk to the 
endpoint, not endpoint to endpoint via Asterisk. This works for the vast 
majority of users as they have media flowing through Asterisk (by choice or via 
use of features) and are fine with transcoding (generally using codecs which 
aren't that costly or low channel count).

Asterisk 16 has some of the foundational work to improve this through the 
implementation of streams but noone has worked on extending the codec 
negotiation support.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_MEDIA_OFFER

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Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints

[Andreas Wehrmann]

On 03.10.19 15:08, Administrator TOOTAI wrote:


Before calling the gatreway add

same = n,set(SIP_CODEC=alaw)

[...]



Hey there,

that doesn't work as it seems to be implemented for chan_sip only;
I'm using chan_pjsip; sorry if I didn't explain myself properly.

Anyway, in my case that would not really be an acceptable solution anyway,
because I need the called party to be able to pick from the range of 
codecs presented to it
because the codec chosen by the destination might change (my example is 
a simplified version).


I don't think putting the burden of worrying about audio codecs on the 
dialplan writer is a good idea,
since this should be dealt with automatically with respect to what is 
configured and negotiated.
This is also because in the systems I have to work with, the 'engineers' 
usual provide the configuration (endpoints, NAT config and the like)
while the technicians implement the dialplan (or the business logic so 
to speak) according to customer needs.
They (the technicians) usually don't know (much) about codecs or how the 
channels techs work exactly...


Thanks,
Andy


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Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints

[Administrator TOOTAI]

Hi

Le 03/10/2019 à 13:13, Andreas Wehrmann a écrit :

[...]


- Even if direct_media is disabled: Is there a way to make Asterisk 
always use a common codec between SIP endpoints,

   so it doesn't need to transcode?


Before calling the gatreway add

same = n,set(SIP_CODEC=alaw)

[...]

--
Daniel

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[asterisk-users] Asterisk not using common codec between (SIP) endpoints

[Andreas Wehrmann]

Hello people,

I've ran into two problem that I can't seem to be able to solve on my own.
Here's my scenario (running Asterisk 13.28.1):

In short: - Asterisk behaves unexpectedly (at least to me) when 
negotiating between endpoints
    that have a different but intersecting set of codecs 
(preventing direct media flow).


  - Also, when an endpoint sends RTP with an unexpected (but 
internally hardcoded) payload type,
    Asterisk seems to get "confused" with what codec is 
supposed to be active (which seems to break transcoding).


I've got a couple of phones (supported codecs are: g722,alaw,ulaw (in 
that order))

and a "special" media gateway which supports alaw, ulaw only.
What I mean by "special" is, that it sends status information via RTP 
payload type 102;
this is hardcoded and I cannot change it; also that thing sends it 
regardless

of what was negotiated via SDP (yes, it's stupid and annoying to deal with).

What I actually wanted to test was to see whether Asterisk would 
correctly negotiate codecs
when making calls between the different kinds of participants (phones 
and gateways).


Here's what I tested:

- Calls between phones (OK)
-- phone01 calls phone02 (phone01 offer: g722, alaw, ulaw)
-- phone02 accepts (Asterisk/phone02 offer/answer: g722, alaw, ulaw)
-- Asterisk sends reinvite to both to establish direct media flow 
(codec: g722)

-- OKAY - this is exactly what i would expect

- Phone calls mediagateway; mediagateway sends special RTP (NOK: 
Asterisk attempts to transcode and RTP engine seems to get confused)

-- phone01 calls mediagateway (phone01 offer: g722, alaw, ulaw)
-- mediagateway accepts call (Asterisk offer: alaw, ulaw | mediagateway 
answer: alaw)

-- Asterisk accepts call from phone01 (Asterisk answer: g722, alaw, ulaw)
-- Asterisk attempts to transcodes between g722 and alaw

--> My expectation here would be for Asterisk to reduce the set of 
codecs in the answer to phone01 to alaw only,

    since this would enable direct media flow between the endpoints.
    But even if it accepts the call like I described above,
    I would expect Asterisk to reinvite the calls later
    to use alaw only and get out of the way to enable direct media flow.

-- After mediagateway accepts the call, it sends a few status messages 
via RTP PT 102;
   this seems to confuse the RTP engine in Asterisk (see the console 
output at the end of this mail)
   which also seems to break transcoding, because audio doesn't work in 
either direction.



My questions are:

- Is there a way to make Asterisk always try to get out of the media path?
  I thought direct_media=yes and an overlapping set of codecs would 
suffice.


- Even if direct_media is disabled: Is there a way to make Asterisk 
always use a common codec between SIP endpoints,

  so it doesn't need to transcode?

- When Asterisk receives unexpected RTP packets or RTP packets of an 
unexpected type in a session,

  shouldn't it just ignore them?
  This is what I observed in older Asterisk versions (very old, like 1.4)
  where it would simply print a warning about an unknown/unhandled 
payload type.


I'll attach my pjsip.conf, extensions.ael and a tcpdump that shows my 
phone (10.137.8.20)

calling the mediagateway (10.254.0.221) via Asterisk (10.137.8.19).
The rest of the configs are the default configs created with "make 
basic-pbx".


I do hope, I'm not missing something obvious here...

A few additions: I ran the tests with Asterisk 16.5.1 also, the results are:

- Calls between phones work as expected, G722 is used and Asterisks 
reinvites

  to establish direct media flow between the phones.

- When a phone calls a mediagateway, the results are as above with the 
exception of having audio in both directions.
  The status messages sent via RTP PT 102 do not seem to confuse 
Asterisk 16.5.1.


- I also ran this test: I have two endpoints (direct_media=yes) each 
with allowed codecs
  set to ulaw, alaw on one endpoint and alaw, ulaw (reverse order) set 
on the other endpoint.
  When endpoint A calls B and offers codecs (in that order) ulaw, alaw 
and endpoint B accepts alaw only,

  Asterisk uses ulaw for the call from A and alaw in the call to B.

Like I said above: I really hope I'm not missing something obvious.
I need Asterisk to use the same codec in both calls (if there is an 
overlapping set)
and try to get out of the way by establishing a direct media path if 
somehow possible.



Best Regards,
Andreas



NOTE: The following is the console output from Asterisk 13 when the 
mediagateway answers and sends RTP PT 102 at the beginning of the call.


[Oct  2 07:24:55] WARNING[23961][C-]: translate.c:490 
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-]: translate.c:490 
ast_translator_build_path: No translator path: (ending codec is not valid)
[Oct  2 07:24:55] WARNING[23961][C-]: translate.c:490 
ast_translator_build_path: 

Re: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

[Stefan Viljoen]

Hi Patrick

Wow ok thanks. I was not aware of this.

As far as I can determine the Azure VM we're using was set up on the "stock" 
Centos 7 option Azure offers. So you're correct, it then won't be an official 
Centos 7 image.

The one we are running on our bare-metal hosts IS installed from an official 
Centos 7 ISO burned to DVD... then yum updated fully once the base install is 
done.

This changes things a bit. THANKS!

I'll try and investigate further. Still wonder what the "RogueWave" or 
"OpenLogic" repos then have different that -only- GSM encoding doesn't work in 
Ast 13.22.0.

Thanks for the help.

Kind regards,

-Original Message-
From: Patrick Laimbock  
Sent: Friday, 13 September 2019 13:37
To: viljo...@verishare.co.za; asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance 
cannot encode gsm via MixMonitor

Hi Stefan,

> Hi all
> 
> I maintain the above - it was set up by an external party with whom relations 
> have now been severed by my employer.
> 
> Quite early after the deployment it became evident that all .gsm audio files 
> produced on this virtual instance at Azure via MixMonitor are corrupt.
[snip]

Is the CentOS 7 installation/image the same across your bare-metal hosts and 
the one on azure? AFAIK there is still no official CentOS 7 image provided by 
the CentOS Project on the azure marketplace. Instead it's created by a third 
party [1]. So there may be differences that could cause issues. On your azure 
host, check the repo files in /etc/yum.repos.d/. If the mirrorlist/basurl 
points to openlogic or roguewave than it's a third-party image. IIRC Amazon and 
GCP have official CentOS 7 images provided by the CentOS Project. Maybe try one 
of those to see if the issue persists? Alternatively create your own CentOS 7 
VM from the official CentOS 7 repositories using kickstart and try that on 
azure.

Best, Patrick

[1] 
https://azuremarketplace.microsoft.com/en-us/marketplace/apps/RogueWave.CentOS76?tab=Overview


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Re: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

[Patrick Laimbock]

Hi Stefan,

> Hi all
> 
> I maintain the above - it was set up by an external party with whom relations 
> have now been severed by my employer.
> 
> Quite early after the deployment it became evident that all .gsm audio files 
> produced on this virtual instance at Azure via MixMonitor are corrupt.
[snip]

Is the CentOS 7 installation/image the same across your bare-metal hosts and 
the one on azure? AFAIK there is still no official CentOS 7 image provided by 
the CentOS Project on the azure marketplace. Instead it's created by a third 
party [1]. So there may be differences that could cause issues. On your azure 
host, check the repo files in /etc/yum.repos.d/. If the mirrorlist/basurl 
points to openlogic or roguewave than it's a third-party image. IIRC Amazon and 
GCP have official CentOS 7 images provided by the CentOS Project. Maybe try one 
of those to see if the issue persists? Alternatively create your own CentOS 7 
VM from the official CentOS 7 repositories using kickstart and try that on 
azure.

Best, Patrick

[1] 
https://azuremarketplace.microsoft.com/en-us/marketplace/apps/RogueWave.CentOS76?tab=Overview

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[asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

[Stefan Viljoen]

Hi all

I maintain the above - it was set up by an external party with whom relations 
have now been severed by my employer.

Quite early after the deployment it became evident that all .gsm audio files 
produced on this virtual instance at Azure via MixMonitor are corrupt. If you 
play back the file in an audio player the audio almost sounds reversed and is 
severely distorted to the extent of being completely unintelligble. You can 
still convert the file to .wav with SOX, and if you look at the waveform in 
something like audacity the amplitude all over is near maximum and visually 
there IS no waveform to see... e. g. apparently the .gsm file itself is valid, 
but the GSM audio data itself is effectively gibbrerish.

But anyway:

The Asterisk config files and dialplan are identical to 17 other sites where 
Asterisk 13 is used in the same way as at Azure.

All the other sites are physical bare-metal hardware, and they work fine on the 
exact same Centos 7 based setup.

No errors are emitted at any verbosity level (in the Asterisk CLI or the log 
files) when the Azure Centos 7 Linux instance is writing corrupt .gsm files.

The problem most definitely is the fact that Asterisk runs in a  Centos 7 
instance on a virtual machine in the Azure cloud. 

Move the exact same config with the same updated Centos 7 kernel to a physical 
box, and the corrupt .gsm file problem disappears.

What can possibly be the problem that deploying Asterisk 13 in an Azure 
instance / VM breaks .gsm encoding?

(I have managed to do a workaround by encoding conversations via MixMonitor to 
.wav, and then using the MixMonitor hook to convert the .wav back to .gsm with 
SOX - which works fine.)

Why can SOX on the same Azure instance happily encode .wav to .gsm, but 
Asterisk "live" via MixMonitor produced corrupt .gsm files, with no errors 
emitted during encoding?

How can one fix this?

Thanks

Stefan


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[asterisk-users] Asterisk 13.28.1, 15.7.4 and 16.5.1 Now Available (Security)

[Asterisk Development Team]

The Asterisk Development Team would like to announce security releases for
Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1,
15.7.4 and 16.5.1.

These releases are available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk/releases

The following security vulnerabilities were resolved in these versions:

* AST-2019-004: Crash when negotiating for T.38 with a declined stream
  When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint
  responds with a declined media stream a crash will then occur in Asterisk.

* AST-2019-005: Remote Crash Vulnerability in audio transcoding
  When audio frames are given to the audio transcoding support in Asterisk the
  number of samples are examined and as part of this a message is output to
  indicate that no samples are present. A change was done to suppress this
  message for a particular scenario in which the message was not relevant. This
  change assumed that information about the origin of a frame will always exist
  when in reality it may not.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
https://downloads.asterisk.org/pub/security/AST-2019-005.pdf

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Re: [asterisk-users] asterisk-users Digest, Vol 181, Issue 3

[Tony Mountifield]

In article <874506323.2924334.1567645810...@mail.yahoo.com>,
bilal ghayyad  wrote:
> 
>  Thank you a lot for your kindly help and reply. Actually it helped me a 
> lot.I was using _X. in the extensions.conf at
> the trunkinbound context.Can you advise me what is the difference between _X. 
> and s? In other words, when it is better
> to use s and when it is better to use _X.?
> Again, I am fully thanks for you.RegardsBilal

They do different things.

_X. will match any extension number beginning with a digit. This is what
you would normally use to match incoming calls that specify a number,
and is presumably what you have already.

s will only match is no extension number is given. This would be the case
for an analogue line, for example, or a SIP connection that didn't give
a destination number. It is also matched for OPTIONS requests used to
handle "qualify".

So in your [trunkinbound] context, just add a line like this:

exten => s,1,Hangup

And leave everything else in that context unchanged.

Cheers
Tony
-- 
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Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] asterisk-users Digest, Vol 181, Issue 3

[bilal ghayyad]

 Thank you a lot for your kindly help and reply. Actually it helped me a lot.I 
was using _X. in the extensions.conf at the trunkinbound context.Can you advise 
me what is the difference between _X. and s? In other words, when it is better 
to use s and when it is better to use _X.?
Again, I am fully thanks for you.RegardsBilal
> Hello;
> 
> I am facing a trouble with the SIP service provider, they are saying 
> that there is a problem related to message option 200 (the heartbeat), 
> so what is required to add this for the sip configuration? Below is my 
> sip debug trace log with the them and the sip peer configuration:

OPTIONS is treated as if it were an INVITE, so it looks up the extension in the 
dialplan. The following shows what extension and context:

[Sep 4 12:42:20] Looking for s in trunkinbound (domain 10.240.147.26)

If you add an "s" extension to the "trunkinbound" context it should then 
respond 200 OK.

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Re: [asterisk-users] Asterisk manager : core show hints

[Joshua C. Colp]

On Thu, Aug 22, 2019, at 5:33 AM, Jonas Kellens wrote:
> Hello 
> 
> I see on the CLI :
> 
> tst*CLI> core show hints
>  -= Registered Asterisk Dial Plan Hints =-
>  50@blf : SIP/testacc7 State:Idle Watchers 3
>  6001@blf : Custom:q-6001 State:Idle Watchers 1
>  5@blf : SIP/testacc6 State:Unavailable Watchers 1
> 
> 
> 
> Is there a way to get this info through the manager API ?

There is an ExtensionStateList action[1]. You can also get individual extension 
state, or device state, or you could use the AMI action which allows you to 
execute a CLI command. I'd suggest looking through the manager action 
documentation to find the action that does precisely what you want if 
ExtensionStateList isn't it.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+ManagerAction_ExtensionStateList

-- 
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[asterisk-users] Asterisk manager : core show hints

[Jonas Kellens]

Hello

I see on the CLI :

tst*CLI> core show hints
    -= Registered Asterisk Dial Plan Hints =-
 50@blf  : SIP/testacc7 
State:Idle    Watchers  3
   6001@blf   : Custom:q-6001 State:Idle    
Watchers  1
  5@blf  : SIP/testacc6 
State:Unavailable Watchers  1



Is there a way to get this info through the manager API ?



Kind regards

Jonas.

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

[Joshua C. Colp]

On Sat, Aug 17, 2019, at 3:07 AM, Michael Maier wrote:
> Hello!
> 



> 
> 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)?

Taskprocessors aren't recurring things individually, they are a work queue item 
that is always executed. It may be a problem with the fact that it is TLS, and 
perhaps the act of trying to establish the TLS connection is taking a long time 
to fail causing things to build up. I'd suggest collecting a backtrace[1] and 
providing complete information on an issue[2].

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issues.asterisk.org/jira

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

[Michael Maier]

Hello!

I encountered an outage of asterisk which showed like that:


- 2019-08-10 07:22:21 Asterisk start
- 2019-08-15 19:39:33 WARNING taskprocessor.c: The 
'pjsip/outreg/ispPJSIP-0060' task processor queue reached 500 scheduled 
tasks.

- 2019-08-15 19:39:34 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'
- 2019-08-15 19:57:19 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'
- 2019-08-15 22:59:59 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'
- 2019-08-16 08:28:04 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'
- 2019-08-16 08:29:04 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'
- 2019-08-16 08:30:04 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'
- 2019-08-16 08:31:04 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'
- 2019-08-16 08:32:04 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'

- 2019-08-16 08:33:04 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'
- 2019-08-16 08:34:04 WARNING pjproject: SSL STATUS_FROM_SSL_ERR (status): 
Level: 0 err: <33558552>  len: 0 
peer: ISP-IP:5061
- 2019-08-16 08:34:04 ERROR pjproject: ssl0x7fbf92d096f0 Error loading CA list 
file '/etc/pki/tls/certs/ca-bundle.crt': Too many open files
- 2019-08-16 08:34:04 WARNING res_pjsip_outbound_registration.c: No response 
received from 'sip:isp.de' on registration attempt to '..', retrying in '60'

Inound calls via other ISP have been dropped - only those logentries can be 
seen:
- 2019-08-16 09:25:43 WARNING[27924] res_rtp_asterisk.c: Unable to allocate 
RTCP socket: Too many open files
- 2019-08-16 09:25:44 WARNING[27924] res_rtp_asterisk.c: Unable to allocate 
RTCP socket: Too many open files



limits of asterisk:
Limit Soft Limit   Hard Limit   Units
Max cpu time  unlimitedunlimitedseconds
Max file size unlimitedunlimitedbytes
Max data size unlimitedunlimitedbytes
Max stack size8388608  unlimitedbytes
Max core file sizeunlimitedunlimitedbytes
Max resident set  unlimitedunlimitedbytes
Max processes 7767 7767 processes
Max open files1024 4096 files
Max locked memory 6553665536bytes
Max address space unlimitedunlimitedbytes
Max file locksunlimitedunlimitedlocks
Max pending signals   7767 7767 signals
Max msgqueue size 819200   819200   bytes
Max nice priority 00
Max realtime priority 00
Max realtime timeout  unlimitedunlimitedus

Memory of the device:
Disk:
7866 MB / (SD card)
771  MB Swap

phys. RAM:
2GB


Memory consumption of the asterisk process 'ps -C asterisk u':

 USER PID%CPU %MEM  VSZ RSS TTYSTAT START 
TIME   COMMAND
15.08.2019 19:01 asterisk 15910  2.2  12.8  2237024 258960  ?  Ssl  Aug10 
173:59 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
15.08.2019 20:01 asterisk 15910  2.2  12.9  2237024 260016  ?  Ssl  Aug10 
175:06 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
15.08.2019 21:01 asterisk 15910  2.2  12.9  2237024 260280  ?  Ssl  Aug10 
176:11 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
15.08.2019 22:01 asterisk 15910  2.2  12.9  2237024 260544  ?  Ssl  Aug10 
177:16 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
15.08.2019 23:01 asterisk 15910  2.2  12.9  2237024 261072  ?  Ssl  Aug10 
178:21 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 00:01 asterisk 15910  2.2  16.0  2302560 323984  ?  Ssl  Aug10 
179:36 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 01:01 asterisk 15910  2.2  21.4  2368096 431076  ?  Ssl  Aug10 
180:51 /usr/sbin/asterisk -vvv -mqf -C /etc/asterisk/asterisk.conf
16.08.2019 02:01 asterisk 15910  2.2  26.6  2499168 535784  ?  Ssl  Aug10 
182:06 /usr/sbin/asterisk -vvv -mqf -C 

Re: [asterisk-users] [asterisk-app-dev] Migrating ast_call_feature from Asterisk 11 to Asterisk 16

[Joshua C. Colp]

On Mon, Aug 5, 2019, at 8:35 AM, Fernando Pardo wrote:
> Hello, everybody. I'm migrating a module I've developed for Asterisk 11 
> to use it on Asterisk 16. One of the trickiest parts I haven't found a 
> way around is the removal of the ast_call_feature struct, which I used 
> to execute a function on transfers.
> 
> This is what I do on Asterisk 11:
> 
> - Define a global 'ast_call_feature *my_feature_transfer'
> - Define a callback function 'int 
> TransferFromFeatureOperation(ast_channel *chan, ast_channel *peer, 
> ast_bridge_config*, const char*, int sense, void*)'
> - On module load, I initialize my_feature_transfer with:
>  - feature_mask = AST_FEATURE_REDIRECT
>  - fname = ast_strdup("Attended Transfer")
>  - sname = "my_att_transfer"
>  - exten = 'T'
>  - default_exten = 'T'
>  - operation = 
>  - flags = AST_FEATURE_FLAG_NEEDSDTMF
>  ... and register it with 'ast_register_feature(icc_feature_transfer);'
> - On module unload, I call 
> 'ast_unregister_feature(icc_feature_transfer)' and free fname.
> 
> What would be the way to migrate this to Asterisk 16? I think it has 
> something to do with aco_option_register, but I'm not sure.

The aco_option_register function is used by modules to register some 
configuration handling logic.

The bridge feature API is defined in bridge_features.h[1] but I'm not sure 
functionality you require (arbitrary adding of features) was ever added to it.

[1] 
https://github.com/asterisk/asterisk/blob/master/include/asterisk/bridge_features.h

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Re: [asterisk-users] Asterisk 13.28.0 Now Available

[Frank Vanoni]

Thank you, dear Asterisk Development Team, for this great software!


> The Asterisk Development Team would like to announce the release of
> Asterisk 13.28.0.

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[asterisk-users] Asterisk 16.5.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
16.5.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.5.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
  no body causes crash
  (Reported by Gil Richard)
 * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
  reINVITE
  (Reported by Francesco Castellano)

Bugs fixed in this release:
---
 * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
  systems caused by ASTERISK-28317
  (Reported by abelbeck)
 * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable
  
  (Reported by Michael Maier)
 * ASTERISK-26006 - Show offending IP for TLS setup failures in
  logs
  (Reported by Oleksandr Natalenko)
 * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
  not logged
  (Reported by Bernhard Schmidt)
 * ASTERISK-28419 - app_amd: Does not work with silence
  suppression
  (Reported by Nasir Iqbal)
 * ASTERISK-28018 - IP Fragmentation happening instead of DTLS
  fragmentation on handshake server hello certificate
 
  (Reported by vijay kumar)
 * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
  Asterisk attempts to generate hangup event
  (Reported by
  Abhay Gupta)
 * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
 
  (Reported by Dmitry Svyatogorov)
 * ASTERISK-27981 - res_fax: Fax session leak with fax
  gatewaying
  (Reported by pasandev)
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
  source files, causes build failure
  (Reported by Guido
  Falsi)
 * ASTERISK-28421 - Wrong type used for timestamp in
  res_rtp_asterisk
  (Reported by Morten Tryfoss)
 * ASTERISK-27994 - PJSIP: Early media ringback not indicated
  after Progress()
  (Reported by Gregory Massel)

Improvements made in this release:
---
 * ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support
  for DUNDi
  (Reported by Kirsty Tyerman)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0

Thank you for your continued support of Asterisk!
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[asterisk-users] Asterisk 13.28.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
13.28.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.28.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
  no body causes crash
  (Reported by Gil Richard)
 * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
  reINVITE
  (Reported by Francesco Castellano)

Bugs fixed in this release:
---
 * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
  systems caused by ASTERISK-28317
  (Reported by abelbeck)
 * ASTERISK-26006 - Show offending IP for TLS setup failures in
  logs
  (Reported by Oleksandr Natalenko)
 * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
  not logged
  (Reported by Bernhard Schmidt)
 * ASTERISK-28460 - res_pjsip_sdp_rtp: Fix ICE candidate leak
  with specific usage
  (Reported by Joshua C. Colp)
 * ASTERISK-28018 - IP Fragmentation happening instead of DTLS
  fragmentation on handshake server hello certificate
 
  (Reported by vijay kumar)
 * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
  Asterisk attempts to generate hangup event
  (Reported by
  Abhay Gupta)
 * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
 
  (Reported by Dmitry Svyatogorov)
 * ASTERISK-27981 - res_fax: Fax session leak with fax
  gatewaying
  (Reported by pasandev)
 * ASTERISK-28419 - app_amd: Does not work with silence
  suppression
  (Reported by Nasir Iqbal)
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
  source files, causes build failure
  (Reported by Guido
  Falsi)
 * ASTERISK-27994 - PJSIP: Early media ringback not indicated
  after Progress()
  (Reported by Gregory Massel)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.28.0

Thank you for your continued support of Asterisk!
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[asterisk-users] Asterisk 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4 Now Available (Security)

[Asterisk Development Team]

The Asterisk Development Team would like to announce security releases for
Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.

These releases are available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk/releases
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases

The following security vulnerabilities were resolved in these versions:

* AST-2019-002: Remote crash vulnerability with MESSAGE messages
  A specially crafted SIP in-dialog MESSAGE message can cause Asterisk to crash.

* AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver
  When T.38 faxing is done in Asterisk a T.38 reinvite may be sent to an
  endpoint to switch it to T.38. If the endpoint responds with an improperly
  formatted SDP answer including both a T.38 UDPTL stream and an audio or video
  stream containing only codecs not allowed on the SIP peer or user a crash will
  occur. The code incorrectly assumes that there will be at least one common
  codec when T.38 is also in the SDP answer.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
https://downloads.asterisk.org/pub/security/AST-2019-003.pdf

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Re: [asterisk-users] Asterisk and pulseaudio Console/dsp

[Jerry Geis]

>> I had that issue at a previous employer and got around it by using ALSA
instead.

Thanks but the other piece I need is forcing me to use pulseaudio. That's
what it connects to.

Jerry

>
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Re: [asterisk-users] Asterisk and pulseaudio Console/dsp

[Doug Lytle]

>>> I setup and extension to connect me with Console/Dsp.   I am hearing the 
>>> audio but its warbly or does not sound right.  Any thoughts on what I need 
>>> to do for that  ?

I had that issue at a previous employer and got around it by using ALSA instead.

Doug

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[asterisk-users] Asterisk and pulseaudio Console/dsp

[Jerry Geis]

Hi All,

I am running pulseaudio on my asterisk server.
I setup and extension to connect me with Console/Dsp.   I am hearing the
audio but its warbly or does not sound right.  Any thoughts on what I need
to do for that  ?

Another thought is I tried to setup the Musiconhold to be custom and the
application be a bash script that that plays pulseaudio.
[default]
mode=custom
directory=/var/lib/asterisk/moh
application=/usr/local/bin/playconsole

and playconsole is:
/usr/bin/parecord --format=s16le

But I hear nothing. The script is executable. With no file for the parecord
it should be going to stdout which should go into asterisk. I also just
tried /usr/bin/parecord and no --format=s16le

How can I get correct audio from pulseaudio into asterisk ?
Thanks,

Jerry
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[asterisk-users] Asterisk and Airplay

[Jerry Geis]

Hi All,

Is there a way to get Airplay music into asterisk ?
I have used Shairport-sync to get Airplay to play audio on my pulseaudio
computer - but was wanting that to come into Asterisk ?

Thanks

Jerry
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Re: [asterisk-users] Asterisk and Linphone

[Doug Lytle]

My self-compiled Asterisk also shows that speex dependencies are not installed

Speex Coder/Decoder

Depends on: speex(E), speex_preprocess(E)
Can use: speexdsp(E)

You'll need to installed the dependencies and re-compile.

Doug



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Re: [asterisk-users] Asterisk and Linphone

[Antony Stone]

On Friday 05 July 2019 at 16:33:56, Jerry Geis wrote:

> I have no speex translation

> core show translation paths speex
> --- Translation paths SRC Codec "speex" sample rate 8000 ---

> speex:8000   To slin:8000   : No Translation Path

> Does not look good. no paths...  Did something not get compiled ?

I suspect you're right, but this is now beyond my expertise.

I have an Asterisk 13.14.1 system here installed from Debian packages and I 
have the following:

speex:8000 To amr:8000 : (speex@8000)->(slin@8000)->(amr@8000) 
speex:8000 To amrwb:16000 : (speex@8000)->(slin@8000)->(slin@16000)-
>(amrwb@16000) 
speex:8000 To g723:8000 : No Translation Path 
speex:8000 To ulaw:8000 : (speex@8000)->(slin@8000)->(ulaw@8000) 
speex:8000 To alaw:8000 : (speex@8000)->(slin@8000)->(alaw@8000) 
speex:8000 To gsm:8000 : (speex@8000)->(slin@8000)->(gsm@8000) 
speex:8000 To g726:8000 : (speex@8000)->(slin@8000)->(g726@8000) 
speex:8000 To g726aal2:8000 : (speex@8000)->(slin@8000)->(g726aal2@8000) 
speex:8000 To adpcm:8000 : (speex@8000)->(slin@8000)->(adpcm@8000) 
speex:8000 To slin:8000 : (speex@8000)->(slin@8000) 
speex:8000 To slin:12000 : (speex@8000)->(slin@8000)->(slin@12000) 
speex:8000 To slin:16000 : (speex@8000)->(slin@8000)->(slin@16000) 
speex:8000 To slin:24000 : (speex@8000)->(slin@8000)->(slin@24000) 
speex:8000 To slin:32000 : (speex@8000)->(slin@8000)->(slin@32000) 
speex:8000 To slin:44100 : (speex@8000)->(slin@8000)->(slin@44100) 
speex:8000 To slin:48000 : (speex@8000)->(slin@8000)->(slin@48000) 
speex:8000 To slin:96000 : (speex@8000)->(slin@8000)->(slin@96000) 
speex:8000 To slin:192000 : (speex@8000)->(slin@8000)->(slin@192000) 
speex:8000 To lpc10:8000 : (speex@8000)->(slin@8000)->(lpc10@8000) 
speex:8000 To g729:8000 : No Translation Path 
speex:8000 To speex:16000 : (speex@8000)->(slin@8000)->(slin@16000)-
>(speex@16000) 
speex:8000 To speex:32000 : (speex@8000)->(slin@8000)->(slin@32000)-
>(speex@32000) 
speex:8000 To ilbc:8000 : No Translation Path 
speex:8000 To g722:16000 : (speex@8000)->(slin@8000)->(g722@16000) 
speex:8000 To siren7:16000 : No Translation Path 
speex:8000 To siren14:32000 : No Translation Path 
speex:8000 To testlaw:8000 : (speex@8000)->(slin@8000)->(testlaw@8000) 
speex:8000 To g719:48000 : No Translation Path 
speex:8000 To opus:48000 : No Translation Path 
speex:8000 To none:8000 : No Translation Path 
speex:8000 To silk:8000 : No Translation Path 
speex:8000 To silk:12000 : No Translation Path 
speex:8000 To silk:16000 : No Translation Path 
speex:8000 To silk:24000 : No Translation Path 


Antony.

-- 
"There has always been an underlying argument that we should open up our 
source code more broadly. The fact is that we are learning from open source 
and we are opening our code more broadly through Shared Source.

Is there value to providing source code? The answer is unequivocally yes."

 - Jason Matusow, head of Microsoft's Shared Source Program, in response to 
leaks of Windows source code on the Internet.

   Please reply to the list;
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Re: [asterisk-users] Asterisk and Linphone

[Jerry Geis]

I have no speex translation
  ulaw  alaw   gsm  g726 g726aal2 adpcm slin8 slin12 slin16 slin24
slin32 slin44 slin48 slin96 slin192 lpc10  ilbc  g722 testlaw
 ulaw -  9150 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000 15000 17250   15000
 alaw  9150 - 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000 15000 17250   15000
  gsm 15000 15000 - 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000 15000 17250   15000
 g726 15000 15000 15000 -15000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000 15000 17250   15000
 g726aal2 15000 15000 15000 15000- 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000 15000 17250   15000
adpcm 15000 15000 15000 1500015000 -  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000 15000 17250   15000
slin8  6000  6000  6000  6000 6000  6000 -   8000   8000   8000
  8000   8000   8000   80008000  6000  6000  82506000
   slin12 14500 14500 14500 1450014500 14500  8500  -   8000   8000
  8000   8000   8000   80008000 14500 14500 14000   14500
   slin16 14500 14500 14500 1450014500 14500  8500   8500  -   8000
  8000   8000   8000   80008000 14500 14500  6000   14500
   slin24 14500 14500 14500 1450014500 14500  8500   8500   8500  -
  8000   8000   8000   80008000 14500 14500 14500   14500
   slin32 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
 -   8000   8000   80008000 14500 14500 14500   14500
   slin44 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
  8500  -   8000   80008000 14500 14500 14500   14500
   slin48 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
  8500   8500  -   80008000 14500 14500 14500   14500
   slin96 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
  8500   8500   8500  -8000 14500 14500 14500   14500
  slin192 14500 14500 14500 1450014500 14500  8500   8500   8500   8500
  8500   8500   8500   8500   - 14500 14500 14500   14500
lpc10 15000 15000 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 - 15000 17250   15000
 ilbc 15000 15000 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000 - 17250   15000
 g722 15600 15600 15600 1560015600 15600  9600  17500   9000  17000
 17000  17000  17000  17000   17000 15600 15600 -   15600
  testlaw 15000 15000 15000 1500015000 15000  9000  17000  17000  17000
 17000  17000  17000  17000   17000 15000 15000 17250   -

core show translation paths speex
--- Translation paths SRC Codec "speex" sample rate 8000 ---
speex:8000   To g723:8000   : No Translation Path

speex:8000   To ulaw:8000   : No Translation Path

speex:8000   To alaw:8000   : No Translation Path

speex:8000   To gsm:8000: No Translation Path

speex:8000   To g726:8000   : No Translation Path

speex:8000   To g726aal2:8000   : No Translation Path

speex:8000   To adpcm:8000  : No Translation Path

speex:8000   To slin:8000   : No Translation Path

speex:8000   To slin:12000  : No Translation Path

speex:8000   To slin:16000  : No Translation Path

speex:8000   To slin:24000  : No Translation Path

speex:8000   To slin:32000  : No Translation Path

speex:8000   To slin:44100  : No Translation Path

speex:8000   To slin:48000  : No Translation Path

speex:8000   To slin:96000  : No Translation Path

speex:8000   To slin:192000 : No Translation Path

speex:8000   To lpc10:8000  : No Translation Path

speex:8000   To g729:8000   : No Translation Path

speex:8000   To speex:16000 : No Translation Path

speex:8000   To speex:32000 : No Translation Path

speex:8000   To ilbc:8000   : No Translation Path

speex:8000   To g722:16000  : No Translation Path

speex:8000   To siren7:16000: No Translation Path

speex:8000   To siren14:32000   : No Translation Path

speex:8000   To testlaw:8000: No Translation Path

speex:8000   To g719:48000  : No Translation Path

speex:8000   To opus:48000  : No Translation Path

speex:8000   To none:8000   : No Translation Path

speex:8000   To silk:8000   : No Translation Path

speex:8000   To silk:12000  : No Translation Path

speex:8000   To silk:16000  : No Translation Path

speex:8000   To silk:24000  : No Translation Path

Does not look good. no paths...  Did something not get compiled ?

Jerry

>
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Check 

Re: [asterisk-users] Asterisk and Linphone

[Antony Stone]

On Friday 05 July 2019 at 16:03:42, Jerry Geis wrote:

> I think this is what your looking for:

> [Jul  5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find a
> codec translation path: (speex) -> (speex32)

Indeed, it was.

> My linphone side only has speex@32K enabled.
> 
> My extension definition has:
> disallow=all
> allow=speex
> allow=speex16
> allow=speex32
> allow=g722
> allow=ulaw
> allow=alaw
> allow=gsm
> 
> It looks like its the codec translation ?   So then I enabled speex and
> speex32 on Linphone Got past that - I presume it will use speex32 for
> audio...

You can always see which codec is in use by doing a SIPpacket capture and 
looking at the above negotiation exchange to see what got agreed on.

> But then I am trying to place that call in a conference (confbridge) and I
> get this error:
> Unable to find a codec translation path: (slin) -> (speex)
> so I think then it hangs up.

Try "core show translation" on your Asterisk command line and check that the 
table has an entries in both directions for speex (left) to slin (top) and 
slin (left) to speex (top).

The numbers tell you how many microseconds *your* server takes to transcode 1 
second of audio between the two codecs.


You can also try "core show translation paths speex" to get a list of the 
codecs which can and cannot be converted to, with a guide to the method used 
for trancoding that combination where possible.


Antony.

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All matter in the Universe can be placed into one of two categories:

1. Things which need to be fixed.
2. Things which need to be fixed once you've had a few minutes to play with 
them.

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Re: [asterisk-users] Asterisk and Linphone

[Jerry Geis]

I think this is what your looking for:

Found RTP audio format 119
Found audio description format speex for ID 119
Capabilities: us - (speex|speex16|speex32|g722|ulaw|alaw|gsm), peer -
audio=(speex32)/video=(nothing)/text=(nothing), combined - (speex32)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.176:7078
[Jul  5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find
a codec translation path: (speex32) -> (speex)
^M^[[Kdevgeis*CLI> ^M^[[0K[Jul  5 09:55:34] WARNING[19832]: channel.c:5751
set_format: Unable to find a codec translation path: (speex) -> (speex32)

My linphone side only has speex@32K enabled.

My extension definition has:
disallow=all
allow=speex
allow=speex16
allow=speex32
allow=g722
allow=ulaw
allow=alaw
allow=gsm

It looks like its the codec translation ?   So then I enabled speex and
speex32 on Linphone Got past that - I presume it will use speex32 for
audio...

But then I am trying to place that call in a conference (confbridge) and I
get this error:
Unable to find a codec translation path: (slin) -> (speex)
so I think then it hangs up.

What do I do about that ? - thanks

Jerry

On Fri, Jul 5, 2019 at 8:22 AM Jerry Geis  wrote:

> Hi all - I am using asterisk 13.27.0 with Linphone.
> I turned off all codes on linphone except the one I want to try. For
> example:
> opus and speex (so only one enabled at a time).
> Then did this same on asterisk for the linphone extension.
> disallow=all
> allow=speex
>
> (for example).
>
> Then I place my call and the call fails.   if I enable something like gsm,
> ulaw, alaw the call works fine. Why does the call fail with opus and speex ?
> Thanks,
>
> Jerry
>
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Re: [asterisk-users] Asterisk and Linphone

[Antony Stone]

On Friday 05 July 2019 at 14:22:22, Jerry Geis wrote:

> Hi all - I am using asterisk 13.27.0 with Linphone.
> I turned off all codes on linphone except the one I want to try. For
> example:
> opus and speex (so only one enabled at a time).
> Then did this same on asterisk for the linphone extension.
> disallow=all
> allow=speex
> 
> (for example).
> 
> Then I place my call and the call fails.   if I enable something like gsm,
> ulaw, alaw the call works fine. Why does the call fail with opus and speex?

Show us the SIP INVITE from Linphone and the response from Asterisk where they 
negotiate codecs -  that should tell us why they disagree.


Antony.

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The lottery is a tax for people who can't do maths.

   Please reply to the list;
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[asterisk-users] Asterisk and Linphone

[Jerry Geis]

Hi all - I am using asterisk 13.27.0 with Linphone.
I turned off all codes on linphone except the one I want to try. For
example:
opus and speex (so only one enabled at a time).
Then did this same on asterisk for the linphone extension.
disallow=all
allow=speex

(for example).

Then I place my call and the call fails.   if I enable something like gsm,
ulaw, alaw the call works fine. Why does the call fail with opus and speex ?
Thanks,

Jerry
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Re: [asterisk-users] [asterisk-app-dev] phone is in dialing state but receiving 'StasisStart' event

[Joshua C. Colp]

On Tue, Jul 2, 2019, at 6:34 AM, Mahipal Singh wrote:
> Hello all,
> I am using node ari client library and, when i dial call using mobile 
> and my mobile is showing that it is in dialling state, but i receive 
> 'StasisStart' event.
> Actually my code is according that whenever i received 'StasisStart' 
> then do 'answer' this call and play an 'announcement' but i found above 
> problem so unable to play 'announcement' .
> 
> And i found a strange behave that whenever i found above problem and i 
> restart ARI connection to same app name then above problem solved and 
> after some call i found same above issue,
> and this issue rise when call load is high, so i don't understand what 
> is problem, so please give solution of this problem.

Have you looked at the actual ARI events that are occurring and the (presumably 
SIP) signaling for the call when it occurs?

-- 
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

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Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1

[Jason N]

We are not allowed to insert anything into the call path.  So somehow we have 
get S included into call without adding anything into the call path.  That’s 
why I thought a SIP JOIN would work (where device C would handle the multiparty 
call) – but it sounds like Asterisk doesn’t support that.

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Israel Gottlieb
Sent: Monday, July 1, 2019 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1

 

how about sticking in a pbx between [c] and [h]

so when [h] hangsup you send to [s] if that is 3rd party else i dont see how 
you could redirect [c] at all 

 

else maybe ask them to have [h] redirect [c] to [s] then [h] will also be out 
of the call

 

On Mon, Jul 1, 2019, 20:03 mailto:asterisk-users-requ...@lists.digium.com>  wrote:

Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com 
<mailto:asterisk-users@lists.digium.com> 

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When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."


Today's Topics:

   1. Re: Second Asterisk server SIP JOIN a call to conduct a
  post-call survey (Joshua C. Colp)
   2. Re: Second Asterisk server SIP JOIN a call to conduct a
  post-call survey (Jason N)
   3. Re: Second Asterisk server SIP JOIN a call to conduct a
  post-call survey (Joshua C. Colp)


--

Message: 1
Date: Mon, 01 Jul 2019 11:15:01 -0300
From: "Joshua C. Colp" mailto:jc...@digium.com> >
To: asterisk-users@lists.digium.com <mailto:asterisk-users@lists.digium.com> 
Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
to conduct  a post-call survey
Message-ID: mailto:be3a1911-7870-4039-9a35-39f7b5be8...@www.fastmail.com> >
Content-Type: text/plain;charset=utf-8

On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:
> I am designing a solution for a hotel booking call center with the 
> following (mandatory) design: After the call from the customer with the 
> booking agent is complete (and the Hotel PBX disconnects from the 
> call), a second PBX takes over to conduct a survey of how the call 
> went. Both PBX’s are Asterisk based. 
> 
> 
> So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, 
> the survey PBX [S] grabs the call and conducts the survey. [H] must 
> completely disconnect from the call before [S] can start the survey. 
> [H] cannot transfer/forward the call to [S]. 
> 
> 
> At a high level the solution seems to be: On [C] connection to [H], [H] 
> sends call information to [S]. [S] issues a SIP JOIN to [C] and joins 
> the call. [S] somehow detects that [H] has disconnected and then begins 
> the survey.
> 
> 
> Would the above work conceptually? If so, how do I tell Asterisk [S] to 
> contact [C] and join the call already in progress? (I can get call info 
> from [H] to [S]).

It would be easiest for H to just Dial S after the first call leg is done. This 
can be done using the 'g' option to Dial[1] which continues dialplan 
application after the outgoing call leg hangs up. You could even send 
information as SIP headers if need be so S sees the info.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial

-- 
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 <http://www.digium.com>  & www.asterisk.org 
<http://www.asterisk.org> 



--

Message: 2
Date: Mon, 1 Jul 2019 14:53:47 +
From: "Jason N" mailto:supp...@telium.io> >
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
mailto:asterisk-users@lists.digium.com> >
Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
to  conduct a post-call survey
Message-ID:

<0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000...@email.amazonses.com
 
<mailto:0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000...@email.amazonses.com>
 >

Content-Type: text/plain;   charset="utf-8"

Unfortunately I am not allowed any changes to H's PBX / dialplan.The 
restriction I have is that upon H's total disconnection from C, that S 
continue

Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1

[Israel Gottlieb]

how about sticking in a pbx between [c] and [h]
so when [h] hangsup you send to [s] if that is 3rd party else i dont see
how you could redirect [c] at all

else maybe ask them to have [h] redirect [c] to [s] then [h] will also be
out of the call

On Mon, Jul 1, 2019, 20:03  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: Second Asterisk server SIP JOIN a call to conduct a
>   post-call survey (Joshua C. Colp)
>2. Re: Second Asterisk server SIP JOIN a call to conduct a
>   post-call survey (Jason N)
>3. Re: Second Asterisk server SIP JOIN a call to conduct a
>   post-call survey (Joshua C. Colp)
>
>
> --
>
> Message: 1
> Date: Mon, 01 Jul 2019 11:15:01 -0300
> From: "Joshua C. Colp" 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
> to conduct  a post-call survey
> Message-ID: 
> Content-Type: text/plain;charset=utf-8
>
> On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:
> > I am designing a solution for a hotel booking call center with the
> > following (mandatory) design: After the call from the customer with the
> > booking agent is complete (and the Hotel PBX disconnects from the
> > call), a second PBX takes over to conduct a survey of how the call
> > went. Both PBX’s are Asterisk based.
> >
> >
> > So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects,
> > the survey PBX [S] grabs the call and conducts the survey. [H] must
> > completely disconnect from the call before [S] can start the survey.
> > [H] cannot transfer/forward the call to [S].
> >
> >
> > At a high level the solution seems to be: On [C] connection to [H], [H]
> > sends call information to [S]. [S] issues a SIP JOIN to [C] and joins
> > the call. [S] somehow detects that [H] has disconnected and then begins
> > the survey.
> >
> >
> > Would the above work conceptually? If so, how do I tell Asterisk [S] to
> > contact [C] and join the call already in progress? (I can get call info
> > from [H] to [S]).
>
> It would be easiest for H to just Dial S after the first call leg is done.
> This can be done using the 'g' option to Dial[1] which continues dialplan
> application after the outgoing call leg hangs up. You could even send
> information as SIP headers if need be so S sees the info.
>
> [1]
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial
>
> --
> 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
>
>
>
> --
>
> Message: 2
> Date: Mon, 1 Jul 2019 14:53:47 +
> From: "Jason N" 
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
> to  conduct a post-call survey
> Message-ID:
> <
> 0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000...@email.amazonses.com
> >
>
> Content-Type: text/plain;   charset="utf-8"
>
> Unfortunately I am not allowed any changes to H's PBX / dialplan.The
> restriction I have is that upon H's total disconnection from C, that S
> continues the call with C.  That's why I thought that if I could get S to
> SIP JOIN the call from C, that once H disconnects S can continue.   I can
> extract the SIP call info on H and pass that to S (so it can join the
> call).
>
> I'm just not sure if this concept is possible/practical.
>
>
> -Original Message-
> From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
> Behalf Of Joshua C. Colp
> Sent: Monday, July 1, 2019 10:15 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to
> conduct a post-call survey
>
> On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:
> > I am designing a solution for a hotel booking call center with the
> > following (mandatory) design: After the call from the customer with
> > the booking agent is complete (and the Hotel PBX disconnects from the
> > call), a second PBX takes over to conduct a survey of how the call
> > went. Both PBX’s are Asterisk based.
> >
> >
> > So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects,
> > the survey PBX [S] grabs the call and conducts the survey. [H] must
> > completely disconnect from the call 

[asterisk-users] Asterisk 16.4.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
16.4.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.4.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
---
 * ASTERISK-28375 - res_pjsip: New configuration setting to
  allow disabling norefersub
  (Reported by Dan Cropp)
 * ASTERISK-28320 - Added ARI resource
  /ari/channels/{channelid}/rtp_statistics
  (Reported by
  sungtae kim)

Bugs fixed in this release:
---
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
  source files, causes build failure
  (Reported by Guido
  Falsi)
 * ASTERISK-28412 - GCC 9 catches more string formatting issues

  (Reported by George Joseph)
 * ASTERISK-28379 - pjsip: show channelstats incorrect
  information output
  (Reported by Vyrva Igor)
 * ASTERISK-28399 - channel.c: Exceptionally long queue length
  queuing
  (Reported by Abhay Gupta)
 * ASTERISK-28392 - The no-partial-inlining flag isn't passed to
  the bundled pjproject or jansson builds
  (Reported by
  George Joseph)
 * ASTERISK-28402 - res_pjsip_registrar: SEGV in
  registrar_find_contact
  (Reported by Ross Beer)
 * ASTERISK-27756 - bridge: Failure to impart a channel results
  in bad data causing crash
  (Reported by Abhay Gupta)
 * ASTERISK-26718 - ARI: Bridge destroying doesn't work as
  expected
  (Reported by Marin Odrljin)
 * ASTERISK-28143 - app_amd: Infinite loop on silent calls 

  (Reported by Abhay Gupta)
 * ASTERISK-28353 - stasis: Crash at shutdown when statistics
  enabled
  (Reported by Joshua C. Colp)
 * ASTERISK-28374 - latest asterisk unconditionally launch gcc
  --version, even if the compiler is different
  (Reported by
  Guido Falsi)
 * ASTERISK-28391 - res_indications: Crash requesting
  autocomplete on indications cli command
  (Reported by Lucas
  Mendes)
 * ASTERISK-27935 - app_voicemail: emailbody per user can't
  contain commas
  (Reported by Sébastien Duthil)
 * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find
  extensions with '-' in them
  (Reported by test011)
 * ASTERISK-17799 - AEL reload causes loss of control in a
  macro
  (Reported by Kirill Katsnelson)
 * ASTERISK-18593 - AEL for loops use Macro app and pipe
  delimiter
  (Reported by Luke-Jr)
 * ASTERISK-14939 - AEL parsers does not find existing label
   
  (Reported by klaus3000)
 * ASTERISK-20182 - Parsing a label beginning with a numeric
  character in all Goto/GotoIf/GotoIfTime application causes
  unexpected behavior
  (Reported by Janu)
 * ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323
  Disabled
  (Reported by Dmitry Shubin)
 * ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
  lead to both inband and info
  (Reported by Salah Ahmed)
 * ASTERISK-28319 - musl: Crash on startup when loading modules

  (Reported by Sebastian Kemper)
 * ASTERISK-28362 - strtok_r() makes gcc compile warning
 
  (Reported by sungtae kim)
 * ASTERISK-28255 - res_rtp_asterisk: REMB RTCP packet sending
  may be incorrect
  (Reported by Joshua C. Colp)

Improvements made in this release:
---
 * ASTERISK-28401 - app_confbridge: Add *_all remb behavior
  variants
  (Reported by Joshua C. Colp)
 * ASTERISK-28400 - res_rtp_asterisk / res_pjsip_sdp_rtp: Add
  support for transport-cc
  (Reported by Joshua C. Colp)
 * ASTERISK-28363 - Millisecond-resolution call stats including
  PDD in channel variables
  (Reported by Antoni Goldstein)
 * ASTERISK-20207 - Asterisk should clear out any .lock files in
  the voice mail directory on startup.
  (Reported by Steven
  Wheeler)
 * ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
  work with.
  (Reported by Corey Farrell)
 * ASTERISK-28343 - Added app_name, app_data to channel type
   
  (Reported by sungtae kim)
 * ASTERISK-28264 - Added topic_all container
  (Reported by
  sungtae kim)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0

Thank you for your continued support of Asterisk!
-- 
_
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New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 13.27.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
13.27.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.27.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
---
 * ASTERISK-28375 - res_pjsip: New configuration setting to
  allow disabling norefersub
  (Reported by Dan Cropp)
 * ASTERISK-28320 - Added ARI resource
  /ari/channels/{channelid}/rtp_statistics
  (Reported by
  sungtae kim)

Bugs fixed in this release:
---
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
  source files, causes build failure
  (Reported by Guido
  Falsi)
 * ASTERISK-28412 - GCC 9 catches more string formatting issues

  (Reported by George Joseph)
 * ASTERISK-28392 - The no-partial-inlining flag isn't passed to
  the bundled pjproject or jansson builds
  (Reported by
  George Joseph)
 * ASTERISK-28402 - res_pjsip_registrar: SEGV in
  registrar_find_contact
  (Reported by Ross Beer)
 * ASTERISK-28143 - app_amd: Infinite loop on silent calls 

  (Reported by Abhay Gupta)
 * ASTERISK-28353 - stasis: Crash at shutdown when statistics
  enabled
  (Reported by Joshua C. Colp)
 * ASTERISK-28374 - latest asterisk unconditionally launch gcc
  --version, even if the compiler is different
  (Reported by
  Guido Falsi)
 * ASTERISK-28391 - res_indications: Crash requesting
  autocomplete on indications cli command
  (Reported by Lucas
  Mendes)
 * ASTERISK-27935 - app_voicemail: emailbody per user can't
  contain commas
  (Reported by Sébastien Duthil)
 * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find
  extensions with '-' in them
  (Reported by test011)
 * ASTERISK-17799 - AEL reload causes loss of control in a
  macro
  (Reported by Kirill Katsnelson)
 * ASTERISK-18593 - AEL for loops use Macro app and pipe
  delimiter
  (Reported by Luke-Jr)
 * ASTERISK-14939 - AEL parsers does not find existing label
   
  (Reported by klaus3000)
 * ASTERISK-20182 - Parsing a label beginning with a numeric
  character in all Goto/GotoIf/GotoIfTime application causes
  unexpected behavior
  (Reported by Janu)
 * ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323
  Disabled
  (Reported by Dmitry Shubin)
 * ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
  lead to both inband and info
  (Reported by Salah Ahmed)
 * ASTERISK-28362 - strtok_r() makes gcc compile warning
 
  (Reported by sungtae kim)

Improvements made in this release:
---
 * ASTERISK-28363 - Millisecond-resolution call stats including
  PDD in channel variables
  (Reported by Antoni Goldstein)
 * ASTERISK-20207 - Asterisk should clear out any .lock files in
  the voice mail directory on startup.
  (Reported by Steven
  Wheeler)
 * ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
  work with.
  (Reported by Corey Farrell)
 * ASTERISK-28343 - Added app_name, app_data to channel type
   
  (Reported by sungtae kim)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.27.0

Thank you for your continued support of Asterisk!
-- 
_
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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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Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video

[Joshua C. Colp]

On Thu, May 30, 2019, at 11:30 AM, Jonas Kellens wrote:
> Hello
> 
> is this mailing list still active ?

Seems like it. :D I responded previously. Many people have moved to 
Discourse[1] though and it sees more activity.

[1] https://community.asterisk.org/

-- 
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

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Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video

[Jonas Kellens]

Hello

is this mailing list still active ?




Op 10-05-19 om 14:10 schreef Jonas Kellens:


Hello

I am trying to set up webRTC video calls from my Chrome webbrowser 
(Fedora) to my Chrome webbrowser (Windows 10).


There is local video input (I can see myself), but never video on the 
receiving side.


This is the case in both directions (so it makes no difference which 
peer is calling which peer).



Both webRTC SIP peers have opus and H264 codec in their peer definition :

  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (75 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (47 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


In general sip.conf I have :

videosupport=yes
disallow=all
allow=alaw
allow=opus
allow=h264


When one peer makes a SIP INVITE for a video call, it is clear to me 
that the necessary codec information is present (this all looks fine 
to me) :


(calling webRTC client)

SIP Debugging Enabled for IP: 99.99.255.55
[May 10 10:45:24]
[May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 --->
[May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0
[May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692
[May 10 10:45:24] Max-Forwards: 70
[May 10 10:45:24] To: 
[May 10 10:45:24] From: "WC User Chrome" 
;tag=sdmbqkquhe

[May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu
[May 10 10:45:24] CSeq: 4132 INVITE
[May 10 10:45:24] Contact: 
[May 10 10:45:24] Allow: 
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

[May 10 10:45:24] Supported: outbound
[May 10 10:45:24] User-Agent: SIP.js/0.12.0
[May 10 10:45:24] Content-Type: application/sdp
[May 10 10:45:24] Content-Length: 5098
[May 10 10:45:24]
[May 10 10:45:24] v=0
[May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1
[May 10 10:45:24] s=-
[May 10 10:45:24] t=0 0
[May 10 10:45:24] a=group:BUNDLE audio video
[May 10 10:45:24] a=msid-semantic: WMS 
I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 
106 105 13 110 112 113 126

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 
192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 
network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 
192.168.1.110 9 typ host tcptype active generation 0 network-id 1 
network-cost 10

[May 10 10:45:24] a=ice-ufrag:y8md
[May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le
[May 10 10:45:24] a=ice-options:trickle
[May 10 10:45:24] a=fingerprint:sha-256 
C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B

[May 10 10:45:24] a=setup:actpass
[May 10 10:45:24] a=mid:audio
[May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[May 10 10:45:24] a=sendrecv
[May 10 10:45:24] a=rtcp-mux
[May 10 10:45:24] a=rtpmap:111 opus/48000/2
[May 10 10:45:24] a=rtcp-fb:111 transport-cc
[May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1
[May 10 10:45:24] a=rtpmap:103 ISAC/16000
[May 10 10:45:24] a=rtpmap:104 ISAC/32000
[May 10 10:45:24] a=rtpmap:9 G722/8000
[May 10 10:45:24] a=rtpmap:0 PCMU/8000
[May 10 10:45:24] a=rtpmap:8 PCMA/8000
[May 10 10:45:24] a=rtpmap:106 CN/32000
[May 10 10:45:24] a=rtpmap:105 CN/16000
[May 10 10:45:24] a=rtpmap:13 CN/8000
[May 10 10:45:24] a=rtpmap:110 telephone-event/48000
[May 10 10:45:24] a=rtpmap:112 telephone-event/32000
[May 10 10:45:24] a=rtpmap:113 telephone-event/16000
[May 10 10:45:24] a=rtpmap:126 telephone-event/8000
[May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ
[May 10 10:45:24] a=ssrc:401971016 
msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E 
f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] a=ssrc:401971016 
mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] a=ssrc:401971016 
label:f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 
102 123 127 122 125 107 108 109 124

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 
192.168.1.110 48086 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 
network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 
192.168.1.110 9 typ 

Re: [asterisk-users] asterisk-users Digest, Vol 177, Issue 11

[Joshua C. Colp]

On Sat, May 25, 2019, at 2:34 PM, Saint Michael wrote:
> 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.

Any such functionality would be documented on the wiki[1].

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip

-- 
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

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Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video

[Joshua C. Colp]

On Tue, May 28, 2019, at 9:56 AM, Jonas Kellens wrote:
> Hello
> 
> is this mailing list still active ?

It is still active. Video under chan_sip, however, is not something many do and 
in particular it is possible with WebRTC that something has changed and caused 
problems or there is a bug in a case. The chan_sip module is community 
supported so it does not see a lot of change.

The chan_pjsip module is maintained and in regards to video is something that 
the team at Sangoma who work on Asterisk daily use for video meetings.

-- 
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

Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video

[Jonas Kellens]

Hello

is this mailing list still active ?




Op 10-05-19 om 14:10 schreef Jonas Kellens:


Hello

I am trying to set up webRTC video calls from my Chrome webbrowser 
(Fedora) to my Chrome webbrowser (Windows 10).


There is local video input (I can see myself), but never video on the 
receiving side.


This is the case in both directions (so it makes no difference which 
peer is calling which peer).



Both webRTC SIP peers have opus and H264 codec in their peer definition :

  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (75 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (47 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


In general sip.conf I have :

videosupport=yes
disallow=all
allow=alaw
allow=opus
allow=h264


When one peer makes a SIP INVITE for a video call, it is clear to me 
that the necessary codec information is present (this all looks fine 
to me) :


(calling webRTC client)

SIP Debugging Enabled for IP: 99.99.255.55
[May 10 10:45:24]
[May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 --->
[May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0
[May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692
[May 10 10:45:24] Max-Forwards: 70
[May 10 10:45:24] To: 
[May 10 10:45:24] From: "WC User Chrome" 
;tag=sdmbqkquhe

[May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu
[May 10 10:45:24] CSeq: 4132 INVITE
[May 10 10:45:24] Contact: 
[May 10 10:45:24] Allow: 
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

[May 10 10:45:24] Supported: outbound
[May 10 10:45:24] User-Agent: SIP.js/0.12.0
[May 10 10:45:24] Content-Type: application/sdp
[May 10 10:45:24] Content-Length: 5098
[May 10 10:45:24]
[May 10 10:45:24] v=0
[May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1
[May 10 10:45:24] s=-
[May 10 10:45:24] t=0 0
[May 10 10:45:24] a=group:BUNDLE audio video
[May 10 10:45:24] a=msid-semantic: WMS 
I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 
106 105 13 110 112 113 126

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 
192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 
network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 
192.168.1.110 9 typ host tcptype active generation 0 network-id 1 
network-cost 10

[May 10 10:45:24] a=ice-ufrag:y8md
[May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le
[May 10 10:45:24] a=ice-options:trickle
[May 10 10:45:24] a=fingerprint:sha-256 
C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B

[May 10 10:45:24] a=setup:actpass
[May 10 10:45:24] a=mid:audio
[May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[May 10 10:45:24] a=sendrecv
[May 10 10:45:24] a=rtcp-mux
[May 10 10:45:24] a=rtpmap:111 opus/48000/2
[May 10 10:45:24] a=rtcp-fb:111 transport-cc
[May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1
[May 10 10:45:24] a=rtpmap:103 ISAC/16000
[May 10 10:45:24] a=rtpmap:104 ISAC/32000
[May 10 10:45:24] a=rtpmap:9 G722/8000
[May 10 10:45:24] a=rtpmap:0 PCMU/8000
[May 10 10:45:24] a=rtpmap:8 PCMA/8000
[May 10 10:45:24] a=rtpmap:106 CN/32000
[May 10 10:45:24] a=rtpmap:105 CN/16000
[May 10 10:45:24] a=rtpmap:13 CN/8000
[May 10 10:45:24] a=rtpmap:110 telephone-event/48000
[May 10 10:45:24] a=rtpmap:112 telephone-event/32000
[May 10 10:45:24] a=rtpmap:113 telephone-event/16000
[May 10 10:45:24] a=rtpmap:126 telephone-event/8000
[May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ
[May 10 10:45:24] a=ssrc:401971016 
msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E 
f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] a=ssrc:401971016 
mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] a=ssrc:401971016 
label:f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 
102 123 127 122 125 107 108 109 124

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 
192.168.1.110 48086 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 
network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 
192.168.1.110 9 typ 

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:

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> 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
>
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> End of asterisk-users Digest, Vol 177, Issue 11
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[asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video

[Jonas Kellens]

Hello

I am trying to set up webRTC video calls from my Chrome webbrowser 
(Fedora) to my Chrome webbrowser (Windows 10).


There is local video input (I can see myself), but never video on the 
receiving side.


This is the case in both directions (so it makes no difference which 
peer is calling which peer).



Both webRTC SIP peers have opus and H264 codec in their peer definition :

  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (75 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


  Video Support: Yes
  Prim.Transp. : WS
  Allowed.Trsp : WSS
  SIP Options  : (none)
  Codecs   : (opus|h264)
  Status   : OK (47 ms)
  Useragent    : SIP.js/0.12.0
  Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss
  RTP Engine   : asterisk
  Encryption   : Yes
  RTCP Mux : Yes


In general sip.conf I have :

videosupport=yes
disallow=all
allow=alaw
allow=opus
allow=h264


When one peer makes a SIP INVITE for a video call, it is clear to me 
that the necessary codec information is present (this all looks fine to 
me) :


(calling webRTC client)

SIP Debugging Enabled for IP: 99.99.255.55
[May 10 10:45:24]
[May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 --->
[May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0
[May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692
[May 10 10:45:24] Max-Forwards: 70
[May 10 10:45:24] To: 
[May 10 10:45:24] From: "WC User Chrome" 
;tag=sdmbqkquhe

[May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu
[May 10 10:45:24] CSeq: 4132 INVITE
[May 10 10:45:24] Contact: 
[May 10 10:45:24] Allow: 
ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

[May 10 10:45:24] Supported: outbound
[May 10 10:45:24] User-Agent: SIP.js/0.12.0
[May 10 10:45:24] Content-Type: application/sdp
[May 10 10:45:24] Content-Length: 5098
[May 10 10:45:24]
[May 10 10:45:24] v=0
[May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1
[May 10 10:45:24] s=-
[May 10 10:45:24] t=0 0
[May 10 10:45:24] a=group:BUNDLE audio video
[May 10 10:45:24] a=msid-semantic: WMS I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 
105 13 110 112 113 126

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 
34197 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 network-id 
1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 
9 typ host tcptype active generation 0 network-id 1 network-cost 10

[May 10 10:45:24] a=ice-ufrag:y8md
[May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le
[May 10 10:45:24] a=ice-options:trickle
[May 10 10:45:24] a=fingerprint:sha-256 
C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B

[May 10 10:45:24] a=setup:actpass
[May 10 10:45:24] a=mid:audio
[May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[May 10 10:45:24] a=sendrecv
[May 10 10:45:24] a=rtcp-mux
[May 10 10:45:24] a=rtpmap:111 opus/48000/2
[May 10 10:45:24] a=rtcp-fb:111 transport-cc
[May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1
[May 10 10:45:24] a=rtpmap:103 ISAC/16000
[May 10 10:45:24] a=rtpmap:104 ISAC/32000
[May 10 10:45:24] a=rtpmap:9 G722/8000
[May 10 10:45:24] a=rtpmap:0 PCMU/8000
[May 10 10:45:24] a=rtpmap:8 PCMA/8000
[May 10 10:45:24] a=rtpmap:106 CN/32000
[May 10 10:45:24] a=rtpmap:105 CN/16000
[May 10 10:45:24] a=rtpmap:13 CN/8000
[May 10 10:45:24] a=rtpmap:110 telephone-event/48000
[May 10 10:45:24] a=rtpmap:112 telephone-event/32000
[May 10 10:45:24] a=rtpmap:113 telephone-event/16000
[May 10 10:45:24] a=rtpmap:126 telephone-event/8000
[May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ
[May 10 10:45:24] a=ssrc:401971016 
msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E 
f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] a=ssrc:401971016 
mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E
[May 10 10:45:24] a=ssrc:401971016 
label:f8eee8bd-dd47-4c14-866d-07069cab255f
[May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 
102 123 127 122 125 107 108 109 124

[May 10 10:45:24] c=IN IP4 99.99.255.55
[May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0
[May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 
48086 typ host generation 0 network-id 1 network-cost 10
[May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 
48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 network-id 
1 network-cost 10
[May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 
9 typ host tcptype active generation 0 network-id 1 network-cost 10

[May 10 10:45:24] 

[asterisk-users] Asterisk 16.3.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
16.3.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.3.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-28260 - Asterisk segfault when rtp negotiation is
  wrong or fails
  (Reported by Sotiris Ganouris)

New Features made in this release:
---
 * ASTERISK-28267 - res_stasis: Add ability to switch
  applications
  (Reported by Benjamin Keith Ford)

Bugs fixed in this release:
---
 * ASTERISK-27541 - app_queue: Queue paused reason was (big
  number) secs ago when reason is set
  (Reported by César
  Benjamín García Martínez)
 * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate
  
  (Reported by Olivier Krief)
 * ASTERISK-28350 - manager: Stasis backed up due to locking
   
  (Reported by Joshua C. Colp)
 * ASTERISK-25792 - chan_sip: qualifygap bounds checking
 
  (Reported by Paul Sandys)
 * ASTERISK-28341 - res_config_odbc eliminates empty custom (“@”
  prefix) variables 
  (Reported by Alexei Gradinari)
 * ASTERISK-28333 - StasisEnd event makes wrong timestamp value

  (Reported by sungtae kim)
 * ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
  minutes to be sent
  (Reported by Jared Hull)
 * ASTERISK-28332 - Variable ALTCONF ignored when service is
  used in Debian
  (Reported by Cirillo Ferreira)
 * ASTERISK-28314 - ARI: API changed but "apiVersion" in
  rest-api\resources.json did not
  (Reported by Stefan Repke)
 * ASTERISK-28335 - stasis: Make topic and maybe subscription
  names unique and more useful
  (Reported by Joshua C. Colp)
 * ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
  zero for rtcp stat calculation
  (Reported by sungtae kim)
 * ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
  183 without SDP
  (Reported by Torrey Searle)
 * ASTERISK-28328 - MeetMe global non-admin mute is muting
  admins that subsequently join
  (Reported by Philip Mott)
 * ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
  without channel lock or reference
  (Reported by Francisco
  Seratti)
 * ASTERISK-28168 - app_queue: Adding a blank entry into sql
  queue_members crashes asterisk.
  (Reported by Michael)
 * ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion
  script fails
  (Reported by Guido Weckwerth)
 * ASTERISK-28272 - The basic-pbx config samples don't produce a
  running asterisk
  (Reported by George Joseph)
 * ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
  field after handling a 302 redirect
  (Reported by Alex
  Odrov)
 * ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
  license header
  (Reported by Jeremy Lainé)
 * ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
  multiple UDP interfaces
  (Reported by Nikolay shakin)
 * ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
  pjsip_wizard.conf  causes crash
  (Reported by Jonathan
  Harris)
 * ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
  changing voicemail password with ODBC
  (Reported by
  Michael)
 * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
  AOR is blocked
  (Reported by Ross Beer)
 * ASTERISK-28301 - Allow voicemail boxes to be subscribed to
  with a presence event package
  (Reported by George Joseph)
 * ASTERISK-28303 - res_rtp_asterisk: Interaction between
  smoother and DTMF can cause out of order timestamps
 
  (Reported by Torrey Searle)
 * ASTERISK-28302 - ARI: "Error destroying mutex" when listing
  all ARI applications
  (Reported by Stefan Repke)
 * ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
  applications
  (Reported by George Joseph)
 * ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
  events when GETting causes overload of events
  (Reported by
  George Joseph)
 * ASTERISK-28284 - switching between native_bridge and
  simple_bridge can cause one way audio
  (Reported by Torrey
  Searle)
 * ASTERISK-28251 - CI: Fix CI so it reverifies commit message
  changes
  (Reported by George Joseph)
 * ASTERISK-28277 - database: Add some basic logging
 
  (Reported by Joshua C. Colp)
 * ASTERISK-28181 - ari: Originating overwrites channel start
  time
  (Reported by sungtae kim)

Improvements made in this release:
---
 * ASTERISK-28326 - ari: Added timestamp for some ari events.
  
  (Reported by sungtae 

[asterisk-users] Asterisk 13.26.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 
13.26.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.26.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
---
 * ASTERISK-28267 - res_stasis: Add ability to switch
  applications
  (Reported by Benjamin Keith Ford)

Bugs fixed in this release:
---
 * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate
  
  (Reported by Olivier Krief)
 * ASTERISK-28350 - manager: Stasis backed up due to locking
   
  (Reported by Joshua C. Colp)
 * ASTERISK-25792 - chan_sip: qualifygap bounds checking
 
  (Reported by Paul Sandys)
 * ASTERISK-28341 - res_config_odbc eliminates empty custom (“@”
  prefix) variables 
  (Reported by Alexei Gradinari)
 * ASTERISK-28333 - StasisEnd event makes wrong timestamp value

  (Reported by sungtae kim)
 * ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
  minutes to be sent
  (Reported by Jared Hull)
 * ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
  without channel lock or reference
  (Reported by Francisco
  Seratti)
 * ASTERISK-28314 - ARI: API changed but "apiVersion" in
  rest-api\resources.json did not
  (Reported by Stefan Repke)
 * ASTERISK-28335 - stasis: Make topic and maybe subscription
  names unique and more useful
  (Reported by Joshua C. Colp)
 * ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
  zero for rtcp stat calculation
  (Reported by sungtae kim)
 * ASTERISK-28332 - Variable ALTCONF ignored when service is
  used in Debian
  (Reported by Cirillo Ferreira)
 * ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
  183 without SDP
  (Reported by Torrey Searle)
 * ASTERISK-28328 - MeetMe global non-admin mute is muting
  admins that subsequently join
  (Reported by Philip Mott)
 * ASTERISK-28168 - app_queue: Adding a blank entry into sql
  queue_members crashes asterisk.
  (Reported by Michael)
 * ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion
  script fails
  (Reported by Guido Weckwerth)
 * ASTERISK-28272 - The basic-pbx config samples don't produce a
  running asterisk
  (Reported by George Joseph)
 * ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
  field after handling a 302 redirect
  (Reported by Alex
  Odrov)
 * ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
  license header
  (Reported by Jeremy Lainé)
 * ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
  changing voicemail password with ODBC
  (Reported by
  Michael)
 * ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
  multiple UDP interfaces
  (Reported by Nikolay shakin)
 * ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
  pjsip_wizard.conf  causes crash
  (Reported by Jonathan
  Harris)
 * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
  AOR is blocked
  (Reported by Ross Beer)
 * ASTERISK-28301 - Allow voicemail boxes to be subscribed to
  with a presence event package
  (Reported by George Joseph)
 * ASTERISK-28303 - res_rtp_asterisk: Interaction between
  smoother and DTMF can cause out of order timestamps
 
  (Reported by Torrey Searle)
 * ASTERISK-28302 - ARI: "Error destroying mutex" when listing
  all ARI applications
  (Reported by Stefan Repke)
 * ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
  applications
  (Reported by George Joseph)
 * ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
  events when GETting causes overload of events
  (Reported by
  George Joseph)
 * ASTERISK-28284 - switching between native_bridge and
  simple_bridge can cause one way audio
  (Reported by Torrey
  Searle)
 * ASTERISK-28288 - Resources (udptl fd) leaking for T.38 calls

  (Reported by Paulo Vicentini)
 * ASTERISK-28251 - CI: Fix CI so it reverifies commit message
  changes
  (Reported by George Joseph)
 * ASTERISK-28277 - database: Add some basic logging
 
  (Reported by Joshua C. Colp)
 * ASTERISK-28181 - ari: Originating overwrites channel start
  time
  (Reported by sungtae kim)

Improvements made in this release:
---
 * ASTERISK-28326 - ari: Added timestamp for some ari events.
  
  (Reported by sungtae kim)
 * ASTERISK-28317 - Add logical group at DAHDIChannel event and
  create "dahdi_group" at CHANNEL function
  (Reported by
  Cirillo Ferreira)
 * ASTERISK-28279 - Added creation timestamp for bridge
 
  (Reported by sungtae kim)
 

Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Joshua C. Colp]

On Tue, Apr 2, 2019, at 9:06 PM, Sungtae Kim wrote:
> 
> On 4/3/19 1:29 AM, Joshua C. Colp wrote:
> > On Tue, Apr 2, 2019, at 8:26 PM, Matthew Jordan wrote:
> >>
> >> On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp  wrote:
> >>> On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> >>>   > I get the desired use case to run app_amd from within a Stasis
> >>>   > application, but I’m not sure about app_queue. You have everything at
> >>>   > your disposal within ARI itself to replicate all of the functionality
> >>>   > of app_queue and beyond.
> >>>
> >>>   Yes, there are certain applications which are logically building blocks 
> >>> to bigger applications. AMD is one of those which would be best if it 
> >>> were its own functionality within ARI, but allowing execution of the 
> >>> application is a good enough option. I don't think applications such as 
> >>> Queue, Dial, ConfBridge, Playback, Record or some others really make 
> >>> sense.
> >>>
> >> Assuming the TALK_DETECTION function isn't sufficient, it's worth
> >> noting that the information that AMD uses to make its decisions are
> >> available to the parts of Asterisk that make up ARI. I wonder if it
> >> would be better to simply wrap up the existing talk detection events
> >> under some other HTTP resource rather than open up this entire concept.
> > Ideally for AMD I think this would be preferred.
> >   
> >> While I'm pretty far removed from the guts of Asterisk these days, the
> >> notion of having dialplan applications be executed from within ARI just
> >> fills me with some fear. You can certainly open up some nightmare
> >> scenarios where people invoke Stasis from within Stasis recursively, or
> >> invoke GoTo or other dialplan context affecting applications.
> >>
> >> For that matter, many of the monolithic dialplan applications have
> >> specific options that place channels into dialplan contexts that
> >> execute after their execution. I'm not even sure I can begin to wrap my
> >> head around what that will do to a channel in ARI.
> > Indeed, that's why I suggested bringing it up on here precisely what 
> > applications people are needing to jump into the dialplan for. Best case 
> > those could be made first class citizens under ARI, but worst case I think 
> > a small subset could be allowed to be executed from ARI. I'm personally 
> > against allowing arbitrary execution of any application. There's just too 
> > many unknowns as you say.
> 
> Now I can see the problem too. But also I can see I'm not the only one 
> having a same dilemma.
> 
> Hm... What it suppose to be? I want implement this feature, but little 
> bit lost now.
> 
> I will wait for feedback.

I think waiting for now in case there's any additional input on what people 
jump to the dialplan is good. We can revisit in a week or so once everyone has 
had a chance to think about it and provide feedback, and then go from there.

-- 
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

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Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Sungtae Kim]

On 4/3/19 1:29 AM, Joshua C. Colp wrote:

On Tue, Apr 2, 2019, at 8:26 PM, Matthew Jordan wrote:


On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp  wrote:

On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
  > I get the desired use case to run app_amd from within a Stasis
  > application, but I’m not sure about app_queue. You have everything at
  > your disposal within ARI itself to replicate all of the functionality
  > of app_queue and beyond.

  Yes, there are certain applications which are logically building blocks to 
bigger applications. AMD is one of those which would be best if it were its own 
functionality within ARI, but allowing execution of the application is a good 
enough option. I don't think applications such as Queue, Dial, ConfBridge, 
Playback, Record or some others really make sense.


Assuming the TALK_DETECTION function isn't sufficient, it's worth
noting that the information that AMD uses to make its decisions are
available to the parts of Asterisk that make up ARI. I wonder if it
would be better to simply wrap up the existing talk detection events
under some other HTTP resource rather than open up this entire concept.

Ideally for AMD I think this would be preferred.
  

While I'm pretty far removed from the guts of Asterisk these days, the
notion of having dialplan applications be executed from within ARI just
fills me with some fear. You can certainly open up some nightmare
scenarios where people invoke Stasis from within Stasis recursively, or
invoke GoTo or other dialplan context affecting applications.

For that matter, many of the monolithic dialplan applications have
specific options that place channels into dialplan contexts that
execute after their execution. I'm not even sure I can begin to wrap my
head around what that will do to a channel in ARI.

Indeed, that's why I suggested bringing it up on here precisely what 
applications people are needing to jump into the dialplan for. Best case those 
could be made first class citizens under ARI, but worst case I think a small 
subset could be allowed to be executed from ARI. I'm personally against 
allowing arbitrary execution of any application. There's just too many unknowns 
as you say.


Now I can see the problem too. But also I can see I'm not the only one 
having a same dilemma.


Hm... What it suppose to be? I want implement this feature, but little 
bit lost now.


I will wait for feedback.





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Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Joshua C. Colp]

On Tue, Apr 2, 2019, at 8:26 PM, Matthew Jordan wrote:
> 
> 
> On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp  wrote:
> > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> >  > I get the desired use case to run app_amd from within a Stasis 
> >  > application, but I’m not sure about app_queue. You have everything at 
> >  > your disposal within ARI itself to replicate all of the functionality 
> >  > of app_queue and beyond.
> > 
> >  Yes, there are certain applications which are logically building blocks to 
> > bigger applications. AMD is one of those which would be best if it were its 
> > own functionality within ARI, but allowing execution of the application is 
> > a good enough option. I don't think applications such as Queue, Dial, 
> > ConfBridge, Playback, Record or some others really make sense.
> > 
> 
> Assuming the TALK_DETECTION function isn't sufficient, it's worth 
> noting that the information that AMD uses to make its decisions are 
> available to the parts of Asterisk that make up ARI. I wonder if it 
> would be better to simply wrap up the existing talk detection events 
> under some other HTTP resource rather than open up this entire concept.

Ideally for AMD I think this would be preferred.
 
> While I'm pretty far removed from the guts of Asterisk these days, the 
> notion of having dialplan applications be executed from within ARI just 
> fills me with some fear. You can certainly open up some nightmare 
> scenarios where people invoke Stasis from within Stasis recursively, or 
> invoke GoTo or other dialplan context affecting applications.
> 
> For that matter, many of the monolithic dialplan applications have 
> specific options that place channels into dialplan contexts that 
> execute after their execution. I'm not even sure I can begin to wrap my 
> head around what that will do to a channel in ARI.

Indeed, that's why I suggested bringing it up on here precisely what 
applications people are needing to jump into the dialplan for. Best case those 
could be made first class citizens under ARI, but worst case I think a small 
subset could be allowed to be executed from ARI. I'm personally against 
allowing arbitrary execution of any application. There's just too many unknowns 
as you say.

-- 
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

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Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Matthew Jordan]

On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp  wrote:

> On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> > I get the desired use case to run app_amd from within a Stasis
> > application, but I’m not sure about app_queue. You have everything at
> > your disposal within ARI itself to replicate all of the functionality
> > of app_queue and beyond.
>
> Yes, there are certain applications which are logically building blocks to
> bigger applications. AMD is one of those which would be best if it were its
> own functionality within ARI, but allowing execution of the application is
> a good enough option. I don't think applications such as Queue, Dial,
> ConfBridge, Playback, Record or some others really make sense.
>
>
Assuming the TALK_DETECTION function isn't sufficient, it's worth noting
that the information that AMD uses to make its decisions are available to
the parts of Asterisk that make up ARI. I wonder if it would be better to
simply wrap up the existing talk detection events under some other HTTP
resource  rather than open up this entire concept.

While I'm pretty far removed from the guts of Asterisk these days, the
notion of having dialplan applications be executed from within ARI just
fills me with some fear. You can certainly open up some nightmare scenarios
where people invoke Stasis from within Stasis recursively, or invoke GoTo
or other dialplan context affecting applications.

For that matter, many of the monolithic dialplan applications have specific
options that place channels into dialplan contexts that execute after their
execution. I'm not even sure I can begin to wrap my head around what that
will do to a channel in ARI.

-- 
*Matthew Jordan*
Digium - A Sangoma Company | Senior Vice President, Software and Services |
Engineering
445 Jan Davis Drive NW - Huntsville, AL 35806 - US

direct: +1 256 428 6107
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Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Joshua C. Colp]

On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> I get the desired use case to run app_amd from within a Stasis 
> application, but I’m not sure about app_queue. You have everything at 
> your disposal within ARI itself to replicate all of the functionality 
> of app_queue and beyond.

Yes, there are certain applications which are logically building blocks to 
bigger applications. AMD is one of those which would be best if it were its own 
functionality within ARI, but allowing execution of the application is a good 
enough option. I don't think applications such as Queue, Dial, ConfBridge, 
Playback, Record or some others really make sense.

-- 
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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

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Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Sungtae Kim]

Um... Because... I can?
Tbh, no reason... app_queue is just my favorite module. :P

On 4/2/19 11:41 PM, Joshua C. Colp wrote:

On Tue, Apr 2, 2019, at 6:20 PM, Sungtae Kim wrote:

Hi Asterisk users,

I'm one of Asterisk ARI users, and trying to designing the new ARI for
application execution in Stasis().

This will be made possible for executing the applications in the
Stasis() application.

But, before going further, I would like to know which application needs
to be considered.

Because this feature will introduce new Stasis behavior, I would like to
test the applications as many as possible before submitting the code.
However, I can't test all of them, so I would like to make a priority
and list.

If you want to use your favorite application with this feature, please
reply this.
I will add to the list. :)

Btw, I'm considering the app_amd, app_queue for myself. :)

Can you explain why app_queue would be executed from ARI? What value does ARI 
bring in that regard?



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Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Sungtae Kim]

On 4/2/19 11:41 PM, Joshua C. Colp wrote:

On Tue, Apr 2, 2019, at 6:20 PM, Sungtae Kim wrote:

Hi Asterisk users,

I'm one of Asterisk ARI users, and trying to designing the new ARI for
application execution in Stasis().

This will be made possible for executing the applications in the
Stasis() application.

But, before going further, I would like to know which application needs
to be considered.

Because this feature will introduce new Stasis behavior, I would like to
test the applications as many as possible before submitting the code.
However, I can't test all of them, so I would like to make a priority
and list.

If you want to use your favorite application with this feature, please
reply this.
I will add to the list. :)

Btw, I'm considering the app_amd, app_queue for myself. :)

Can you explain why app_queue would be executed from ARI? What value does ARI 
bring in that regard?


Um... Because... I can?
Tbh, no reason... app_queue is just my favorite module. :P





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Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Joshua C. Colp]

On Tue, Apr 2, 2019, at 6:20 PM, Sungtae Kim wrote:
> Hi Asterisk users,
> 
> I'm one of Asterisk ARI users, and trying to designing the new ARI for 
> application execution in Stasis().
> 
> This will be made possible for executing the applications in the 
> Stasis() application.
> 
> But, before going further, I would like to know which application needs 
> to be considered.
> 
> Because this feature will introduce new Stasis behavior, I would like to 
> test the applications as many as possible before submitting the code.
> However, I can't test all of them, so I would like to make a priority 
> and list.
> 
> If you want to use your favorite application with this feature, please 
> reply this.
> I will add to the list. :)
> 
> Btw, I'm considering the app_amd, app_queue for myself. :)

Can you explain why app_queue would be executed from ARI? What value does ARI 
bring in that regard?

-- 
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

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[asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Sungtae Kim]

Hi Asterisk users,

I'm one of Asterisk ARI users, and trying to designing the new ARI for 
application execution in Stasis().


This will be made possible for executing the applications in the 
Stasis() application.


But, before going further, I would like to know which application needs 
to be considered.


Because this feature will introduce new Stasis behavior, I would like to 
test the applications as many as possible before submitting the code.
However, I can't test all of them, so I would like to make a priority 
and list.


If you want to use your favorite application with this feature, please 
reply this.

I will add to the list. :)

Btw, I'm considering the app_amd, app_queue for myself. :)

For more detail of this feature, you can see it here.
http://lists.digium.com/pipermail/asterisk-dev/2019-April/077270.html

Thank you.

Kind regards,
Sungtae


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Re: [asterisk-users] Asterisk Transfers

[Joshua C. Colp]

On Thu, Mar 28, 2019, at 4:56 PM, Dan Cropp wrote:
> Hi Joshua,
> 
> Unfortunately, I tried including the Refer-Sub true and also false in 
> the REFER packet and Cisco seems to ignore them.
> 
> Refer-Sub: false
> and
> Refer-Sub: true
> 
> The only thing that seems to work properly with the Cisco switch is to 
> remove the norefersub from Supported
> 
> Joshua, is it correct that the norefersub has to be in 
> res_pjsip_refer.c for Asterisk to be able to process incoming REFER 
> requests?
> 
> If I were willing to write code, how difficult would it be to make the 
> norefersub optional or used for incoming (allowing phones to REFER), 
> but not for outgoing (not send to Cisco)?

I'll answer here too for anyone interested. The RFC[1] states that it should be 
present so that the remote endpoint knows whether or not it is supported. 
Removing it may effectively remove support. It depends on whether a remote 
endpoint takes it into account or not when issuing the REFER.

You also can't limit it at that level, because generally for most things you 
don't know if you are sending or receiving a REFER. The "norefersub" Supported 
occurs before it happens. You could only limit it globally.

[1] https://www.ietf.org/rfc/rfc4488.txt

-- 
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Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Asterisk Transfers

[Dan Cropp]

Hi Joshua,

Unfortunately, I tried including the Refer-Sub true and also false in the REFER 
packet and Cisco seems to ignore them.

Refer-Sub: false
and
Refer-Sub: true

The only thing that seems to work properly with the Cisco switch is to remove 
the norefersub from Supported

Joshua, is it correct that the norefersub has to be in res_pjsip_refer.c for 
Asterisk to be able to process incoming REFER requests?

If I were willing to write code, how difficult would it be to make the 
norefersub optional or used for incoming (allowing phones to REFER), but not 
for outgoing (not send to Cisco)?

-Original Message-
From: Dan Cropp 
Sent: Thursday, March 28, 2019 12:55 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Asterisk Transfers

Thank you Joshua.

We're trying to run more tests.
We believe Cisco may not be adhering to the specification.  Unfortunately, 
we're also stuck with having to make it work.

An interesting test, I commented out the norefersub from the res_pjsip_refer.c 
code just for a test.  Without this, Cisco does sent the NOTIFY packets to let 
Asterisk know the status (Trying, Ringing, OK, 486 Busy Here, ...)

Would it make sense for me to try including the Refer-Sub in the REFER packet 
Asterisk sends to Cisco?  Perhaps Cisco sees the norefersub, but not the 
Refer-Sub header.  It may interpret the lack of a Refer-Sub header in the REFER 
incorrectly?

Dan

-Original Message-
From: asterisk-users  On Behalf Of 
Joshua C. Colp
Sent: Thursday, March 28, 2019 9:18 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Transfers

On Thu, Mar 28, 2019, at 11:10 AM, Dan Cropp wrote:
>  
> Is there no one who knows if there is a way to turn off the norefersub 
> setting?
> 
> 
> Supported: norefersub
> 
> 
> This happens in the TRYing, OK, and other commands in response to the INVITE.
> 
> 
> For chan_sip, I noticed it does not send the norefersub. As a result, 
> Cisco then sends NOTIFY packets with TRYing, Ringing, OK inside them.
> This basically gives the chan_sip code the ability to know if the 
> REFER
> (Transfer) is succeeding or not.

There is no way to configure it. It would have to be removed from 
res_pjsip_refer's code itself. The presence of "norefersub" in Supported also 
isn't supposed to enable it. It's just for stating it is supported. It's 
supposed to be enabled by the presence of the "Refer-Sub" header in the REFER 
itself. I don't believe we set that on the REFER we produce, we only care if we 
receive a REFER with it in place.

--
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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

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Re: [asterisk-users] Asterisk Transfers

[Joshua C. Colp]

On Thu, Mar 28, 2019, at 2:56 PM, Dan Cropp wrote:
> Thank you Joshua.
> 
> We're trying to run more tests.
> We believe Cisco may not be adhering to the specification.  
> Unfortunately, we're also stuck with having to make it work.
> 
> An interesting test, I commented out the norefersub from the 
> res_pjsip_refer.c code just for a test.  Without this, Cisco does sent 
> the NOTIFY packets to let Asterisk know the status (Trying, Ringing, 
> OK, 486 Busy Here, ...)
> 
> Would it make sense for me to try including the Refer-Sub in the REFER 
> packet Asterisk sends to Cisco?  Perhaps Cisco sees the norefersub, but 
> not the Refer-Sub header.  It may interpret the lack of a Refer-Sub 
> header in the REFER incorrectly?

It's entirely possible it might. It doesn't hurt to try.

-- 
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Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Asterisk Transfers

[Dan Cropp]

Thank you Joshua.

We're trying to run more tests.
We believe Cisco may not be adhering to the specification.  Unfortunately, 
we're also stuck with having to make it work.

An interesting test, I commented out the norefersub from the res_pjsip_refer.c 
code just for a test.  Without this, Cisco does sent the NOTIFY packets to let 
Asterisk know the status (Trying, Ringing, OK, 486 Busy Here, ...)

Would it make sense for me to try including the Refer-Sub in the REFER packet 
Asterisk sends to Cisco?  Perhaps Cisco sees the norefersub, but not the 
Refer-Sub header.  It may interpret the lack of a Refer-Sub header in the REFER 
incorrectly?

Dan

-Original Message-
From: asterisk-users  On Behalf Of 
Joshua C. Colp
Sent: Thursday, March 28, 2019 9:18 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Transfers

On Thu, Mar 28, 2019, at 11:10 AM, Dan Cropp wrote:
>  
> Is there no one who knows if there is a way to turn off the norefersub 
> setting?
> 
> 
> Supported: norefersub
> 
> 
> This happens in the TRYing, OK, and other commands in response to the INVITE.
> 
> 
> For chan_sip, I noticed it does not send the norefersub. As a result, 
> Cisco then sends NOTIFY packets with TRYing, Ringing, OK inside them.
> This basically gives the chan_sip code the ability to know if the 
> REFER
> (Transfer) is succeeding or not.

There is no way to configure it. It would have to be removed from 
res_pjsip_refer's code itself. The presence of "norefersub" in Supported also 
isn't supposed to enable it. It's just for stating it is supported. It's 
supposed to be enabled by the presence of the "Refer-Sub" header in the REFER 
itself. I don't believe we set that on the REFER we produce, we only care if we 
receive a REFER with it in place.

--
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

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Re: [asterisk-users] Asterisk Transfers

[Joshua C. Colp]

On Thu, Mar 28, 2019, at 11:10 AM, Dan Cropp wrote:
>  
> Is there no one who knows if there is a way to turn off the norefersub 
> setting?
> 
> 
> Supported: norefersub 
> 
> 
> This happens in the TRYing, OK, and other commands in response to the INVITE.
> 
> 
> For chan_sip, I noticed it does not send the norefersub. As a result, 
> Cisco then sends NOTIFY packets with TRYing, Ringing, OK inside them. 
> This basically gives the chan_sip code the ability to know if the REFER 
> (Transfer) is succeeding or not.

There is no way to configure it. It would have to be removed from 
res_pjsip_refer's code itself. The presence of "norefersub" in Supported also 
isn't supposed to enable it. It's just for stating it is supported. It's 
supposed to be enabled by the presence of the "Refer-Sub" header in the REFER 
itself. I don't believe we set that on the REFER we produce, we only care if we 
receive a REFER with it in place.

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Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk Transfers

[Dan Cropp]

Is there no one who knows if there is a way to turn off the norefersub setting?

Supported: norefersub

This happens in the TRYing, OK, and other commands in response to the INVITE.

For chan_sip, I noticed it does not send the norefersub.  As a result, Cisco 
then sends NOTIFY packets with TRYing, Ringing, OK inside them.  This basically 
gives the chan_sip code the ability to know if the REFER (Transfer) is 
succeeding or not.

Dan

From: asterisk-users  On Behalf Of Dan 
Cropp
Sent: Monday, March 25, 2019 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Transfers

Does anyone know if there is a way to disable the norefersub for PJSIP?
It appears this is causing problems with a test we're running with Cisco.

A wireshark trace from a system where the transfer with Cisco works versus a 
trace with Asterisk/Cisco shows one big difference being the supported: 
norefersub

The REFER Accepted response is received by Asterisk.
However, Cisco doesn't send the NOTIFY messages with 100 Trying followed by 404 
Not Found.

>From what we've been able to determine, this is a direct result of
200 OK packet including
Supported: 100rel, timer, replaces, norefersub

Specifically, the norefersub.

Dan
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[asterisk-users] Asterisk Transfers

[Dan Cropp]

Does anyone know if there is a way to disable the norefersub for PJSIP?
It appears this is causing problems with a test we're running with Cisco.

A wireshark trace from a system where the transfer with Cisco works versus a 
trace with Asterisk/Cisco shows one big difference being the supported: 
norefersub

The REFER Accepted response is received by Asterisk.
However, Cisco doesn't send the NOTIFY messages with 100 Trying followed by 404 
Not Found.

>From what we've been able to determine, this is a direct result of
200 OK packet including
Supported: 100rel, timer, replaces, norefersub

Specifically, the norefersub.

Dan
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Re: [asterisk-users] Asterisk using Path: and chan_sip

[Joel Serrano]

Hi,

I found this:
https://lists.kamailio.org/pipermail/sr-users/2019-January/104312.html

It turns out my issue was caused by a wrong *nat=* setting for the device...

After changing:

nat=force_rport,comedia

to:

nat=comedia

I can now see correct IP and RTT in asterisk `sip show peers` for devices
registering via an intermediate proxy :)

Just wanted to update this in case it helps anyone else.

Cheers,
Joel.




My problem was caused by the *nat=* setting for the device.

I found

On Wed, Mar 20, 2019 at 11:37 AM Joel Serrano  wrote:

> Hello,
>
> We have a couple asterisk11 servers behind a Kamailio4 proxy. We are in
> the process of upgrading to asterisk16 and Kamailio5 and I'm testing out
> Path: support with chan_sip (migration to PJSIP is not possible right now
> due to integrations with other systems).
>
> Functionality-wise things are working. If I do a `sip show peer XXX` I can
> see the Path header, I can also receive/send calls to/from that peer.
>
> I could swear that in earlier tests I managed to actually see the
> real-device IP instead of the proxy-IP in `sip show peers`... but now I
> have my doubts as I'm not able to reproduce it.
>
> So, the question is: when a device registers with Asterisk via a Proxy
> (using Path:) is there a way to show the original-IP instead of the
> proxy-IP in sip show peers? Is the IP shown in `sip show peers` strictly
> the IP from where the packet was received, or is the IP obtained from data
> contained in any of the headers?
>
> I'm almost sure this worked and now I'm missing something to get it
> working again, or, I just dreamed about this and in reality it never worked
> :(
>
> Thanks!
>
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[asterisk-users] Asterisk using Path: and chan_sip

[Joel Serrano]

Hello,

We have a couple asterisk11 servers behind a Kamailio4 proxy. We are in the
process of upgrading to asterisk16 and Kamailio5 and I'm testing out Path:
support with chan_sip (migration to PJSIP is not possible right now due to
integrations with other systems).

Functionality-wise things are working. If I do a `sip show peer XXX` I can
see the Path header, I can also receive/send calls to/from that peer.

I could swear that in earlier tests I managed to actually see the
real-device IP instead of the proxy-IP in `sip show peers`... but now I
have my doubts as I'm not able to reproduce it.

So, the question is: when a device registers with Asterisk via a Proxy
(using Path:) is there a way to show the original-IP instead of the
proxy-IP in sip show peers? Is the IP shown in `sip show peers` strictly
the IP from where the packet was received, or is the IP obtained from data
contained in any of the headers?

I'm almost sure this worked and now I'm missing something to get it working
again, or, I just dreamed about this and in reality it never worked :(

Thanks!
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Re: [asterisk-users] Asterisk users survey

[Joshua C. Colp]

On Tue, Mar 12, 2019, at 3:05 AM, Stefan Viljoen wrote:
> Hi Joshua
> 
> Does the survey imply that there are big changes coming for Asterisk?
> 
> E. g. features or facilities will be dropped / deprecated from the open 
> source version in  new releases, big changes to existing facilities / 
> protocols, what is supported officialy by Digium for the official 
> version, and what not, etc.
> 
> Given that you are now owned by Sangoma & all and there are different 
> people in control now than before.
> 
> Can you confirm / deny?

Nothing changing as of yet really. The survey itself stems from our inability 
to really collect data or have insight into how Asterisk is used. With it we 
can better understand use cases and potentially what we should focus more on. 
We've proposed an automated collection module in the past which was met with 
some pushback so this is another avenue.

-- 
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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

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Re: [asterisk-users] Asterisk users survey

[Stefan Viljoen]

Hi Joshua

Does the survey imply that there are big changes coming for Asterisk?

E. g. features or facilities will be dropped / deprecated from the open source 
version in  new releases, big changes to existing facilities / protocols, what 
is supported officialy by Digium for the official version, and what not, etc.

Given that you are now owned by Sangoma & all and there are different people in 
control now than before.

Can you confirm / deny?

Thanks,

> We are in the same situation as Jean Denis, running 1.4 to 16 version as 
> integrator/service provider/user/...
> 
> Difficult to replay the survey for each scenarios ;)

The most active deployment, or a few deployments, is fine.

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Digium - A Sangoma Company | Senior Software Developer


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Re: [asterisk-users] Asterisk Usage Survey

[Joshua C. Colp]

On Mon, Mar 11, 2019, at 7:16 AM, Administrator TOOTAI wrote:
> Le 11/03/2019 à 10:23, Marcelo Terres a écrit :
> > Hello Jean-Denis.
> > 
> > I believe the idea is that you answer the survey for each type of 
> > scenarios you are running.
> > 
> > So one for call centre, another one for ivr, etc...
> 
> And what for instance about exact version of asterisk?
> 
> We are in the same situation as Jean Denis, running 1.4 to 16 version as 
> integrator/service provider/user/...
> 
> Difficult to replay the survey for each scenarios ;)

The most active deployment, or a few deployments, is fine.

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Asterisk Usage Survey

[Administrator TOOTAI]

Le 11/03/2019 à 10:23, Marcelo Terres a écrit :

Hello Jean-Denis.

I believe the idea is that you answer the survey for each type of 
scenarios you are running.


So one for call centre, another one for ivr, etc...


And what for instance about exact version of asterisk?

We are in the same situation as Jean Denis, running 1.4 to 16 version as 
integrator/service provider/user/...


Difficult to replay the survey for each scenarios ;)



Regards,

Marcelo

On Mon, 11 Mar 2019, 02:10 Jean-Denis Girard, > wrote:


Hi Matt,

I would have loved to participate to the survey, but I feel it does
apply to my situation: as an integrator, I'm installing Asterisk for
call centers, PBX, IVR... so I can not answer the first question of the
survey ;) I also have dfferent versions installed.

This is not a negative comment, I just want to express that the survey
does not seem to apply to me; and many people on the Asterisk lists may
be in a situation similar as mine.


Thanks,
-- 
Jean-Denis Girard


SysNux                   Systèmes   Linux   en   Polynésie  française
https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527

Le 08/03/2019 à 05:35, Matthew Fredrickson a écrit :
 > Hey All,
 >
 > For those of you that do not know me, my name is Matthew Fredrickson
 > and I’m the project lead for the Asterisk project. First off, I
wanted
 > to thank all of you that contribute in various ways to the project –
 > whether it be at a developmental level, answering questions on forums
 > and mailing lists, contributing documentation, or just generally
 > advocating for it within your sphere of influence. It takes so many
 > people’s efforts to make the project what it is and to sustain such a
 > large and vibrant user and developer community.
 >
 > We created a general survey inquiring how people utilize Asterisk. It
 > should only take about 10-15 minutes, but would help us understand
 > better how our users are utilizing Asterisk and help us to understand
 > if there are important areas of Asterisk that we underemphasize
from a
 > development perspective. If you don’t mind filling it out, it
would be
 > greatly appreciated.
 >
 > Thanks *so* much again for your time, and best wishes to each of you
 > in your efforts.
 >
 > https://goo.gl/forms/xL1VUHRsf95saly13
 >
 > Matthew Fredrickson
 >

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Re: [asterisk-users] Asterisk Usage Survey

[Marcelo Terres]

Hello Jean-Denis.

I believe the idea is that you answer the survey for each type of scenarios
you are running.

So one for call centre, another one for ivr, etc...

Regards,

Marcelo

On Mon, 11 Mar 2019, 02:10 Jean-Denis Girard,  wrote:

> Hi Matt,
>
> I would have loved to participate to the survey, but I feel it does
> apply to my situation: as an integrator, I'm installing Asterisk for
> call centers, PBX, IVR... so I can not answer the first question of the
> survey ;) I also have dfferent versions installed.
>
> This is not a negative comment, I just want to express that the survey
> does not seem to apply to me; and many people on the Asterisk lists may
> be in a situation similar as mine.
>
>
> Thanks,
> --
> Jean-Denis Girard
>
> SysNux   Systèmes   Linux   en   Polynésie  française
> https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527
>
> Le 08/03/2019 à 05:35, Matthew Fredrickson a écrit :
> > Hey All,
> >
> > For those of you that do not know me, my name is Matthew Fredrickson
> > and I’m the project lead for the Asterisk project. First off, I wanted
> > to thank all of you that contribute in various ways to the project –
> > whether it be at a developmental level, answering questions on forums
> > and mailing lists, contributing documentation, or just generally
> > advocating for it within your sphere of influence. It takes so many
> > people’s efforts to make the project what it is and to sustain such a
> > large and vibrant user and developer community.
> >
> > We created a general survey inquiring how people utilize Asterisk. It
> > should only take about 10-15 minutes, but would help us understand
> > better how our users are utilizing Asterisk and help us to understand
> > if there are important areas of Asterisk that we underemphasize from a
> > development perspective. If you don’t mind filling it out, it would be
> > greatly appreciated.
> >
> > Thanks *so* much again for your time, and best wishes to each of you
> > in your efforts.
> >
> > https://goo.gl/forms/xL1VUHRsf95saly13
> >
> > Matthew Fredrickson
> >
>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Asterisk Usage Survey

[Jean-Denis Girard]

Hi Matt,

I would have loved to participate to the survey, but I feel it does
apply to my situation: as an integrator, I'm installing Asterisk for
call centers, PBX, IVR... so I can not answer the first question of the
survey ;) I also have dfferent versions installed.

This is not a negative comment, I just want to express that the survey
does not seem to apply to me; and many people on the Asterisk lists may
be in a situation similar as mine.


Thanks,
-- 
Jean-Denis Girard

SysNux   Systèmes   Linux   en   Polynésie  française
https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527

Le 08/03/2019 à 05:35, Matthew Fredrickson a écrit :
> Hey All,
> 
> For those of you that do not know me, my name is Matthew Fredrickson
> and I’m the project lead for the Asterisk project. First off, I wanted
> to thank all of you that contribute in various ways to the project –
> whether it be at a developmental level, answering questions on forums
> and mailing lists, contributing documentation, or just generally
> advocating for it within your sphere of influence. It takes so many
> people’s efforts to make the project what it is and to sustain such a
> large and vibrant user and developer community.
> 
> We created a general survey inquiring how people utilize Asterisk. It
> should only take about 10-15 minutes, but would help us understand
> better how our users are utilizing Asterisk and help us to understand
> if there are important areas of Asterisk that we underemphasize from a
> development perspective. If you don’t mind filling it out, it would be
> greatly appreciated.
> 
> Thanks *so* much again for your time, and best wishes to each of you
> in your efforts.
> 
> https://goo.gl/forms/xL1VUHRsf95saly13
> 
> Matthew Fredrickson
> 



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[asterisk-users] Asterisk Usage Survey

[Matthew Fredrickson]

Hey All,

For those of you that do not know me, my name is Matthew Fredrickson
and I’m the project lead for the Asterisk project. First off, I wanted
to thank all of you that contribute in various ways to the project –
whether it be at a developmental level, answering questions on forums
and mailing lists, contributing documentation, or just generally
advocating for it within your sphere of influence. It takes so many
people’s efforts to make the project what it is and to sustain such a
large and vibrant user and developer community.

We created a general survey inquiring how people utilize Asterisk. It
should only take about 10-15 minutes, but would help us understand
better how our users are utilizing Asterisk and help us to understand
if there are important areas of Asterisk that we underemphasize from a
development perspective. If you don’t mind filling it out, it would be
greatly appreciated.

Thanks *so* much again for your time, and best wishes to each of you
in your efforts.

https://goo.gl/forms/xL1VUHRsf95saly13

Matthew Fredrickson

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Re: [asterisk-users] asterisk 16.2.1 inbound route

[Gokan Atmaca]

> 1. What is the content of ${OPERATOR}?
>
> 2. What do you have for this connection in sip.conf?
>
> 3. What number/s have you been assigned by your upstream SIP provider?
>
> Antony.

Hello

The problem appeared in siptrunk. The problem is "insecure=very". This
"insecure=invite" improved.

Very thanks.

On Tue, Mar 5, 2019 at 7:29 PM Antony Stone
 wrote:
>
> On Tuesday 05 March 2019 at 17:22:16, Gokan Atmaca wrote:
>
> > > exten => _13XXX,1,dial(${OPERATOR},20)
>
> 1. What is the content of ${OPERATOR}?
>
> 2. What do you have for this connection in sip.conf?
>
> 3. What number/s have you been assigned by your upstream SIP provider?
>
> Antony.
>
> > On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle  wrote:
> > > On 3/5/19 2:46 AM, Gokan Atmaca wrote:
> > > > Asterisk can send calls, but I don't get a call. What could be the
> > > > problem?
> > > >
> > > > [from-siptrunk]
> > > > exten => 13XXX,1,dial(${OPERATOR},20)
> > >
> > > exten => _13XXX,1,dial(${OPERATOR},20)
> > >
> > > Doug
>
> --
> In science, one tries to tell people
> in such a way as to be understood by everyone
> something that no-one ever knew before.
>
> In poetry, it is the exact opposite.
>
>  - Paul Dirac
>
>Please reply to the list;
>  please *don't* CC me.
>
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>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
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>
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Re: [asterisk-users] asterisk 16.2.1 inbound route

[Antony Stone]

On Tuesday 05 March 2019 at 17:22:16, Gokan Atmaca wrote:

> > exten => _13XXX,1,dial(${OPERATOR},20)

1. What is the content of ${OPERATOR}?

2. What do you have for this connection in sip.conf?

3. What number/s have you been assigned by your upstream SIP provider?

Antony.

> On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle  wrote:
> > On 3/5/19 2:46 AM, Gokan Atmaca wrote:
> > > Asterisk can send calls, but I don't get a call. What could be the
> > > problem?
> > > 
> > > [from-siptrunk]
> > > exten => 13XXX,1,dial(${OPERATOR},20)
> > 
> > exten => _13XXX,1,dial(${OPERATOR},20)
> > 
> > Doug

-- 
In science, one tries to tell people
in such a way as to be understood by everyone
something that no-one ever knew before.

In poetry, it is the exact opposite.

 - Paul Dirac

   Please reply to the list;
 please *don't* CC me.

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Re: [asterisk-users] asterisk 16.2.1 inbound route

[Gokan Atmaca]

> exten => _13XXX,1,dial(${OPERATOR},20)

Hello

"SIP/2.0 401 Unauthorized"  Unfortunately the negative. An asterisk
indicates a 404 error.



On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle  wrote:
>
> On 3/5/19 2:46 AM, Gokan Atmaca wrote:
> > Asterisk can send calls, but I don't get a call. What could be the problem?
> >
> > [from-siptrunk]
> > exten => 13XXX,1,dial(${OPERATOR},20)
> >
>
> exten => _13XXX,1,dial(${OPERATOR},20)
>
> Doug
>
> Doug
>
>
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>
> New to Asterisk? Start here:
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>
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Re: [asterisk-users] asterisk 16.2.1 inbound route

[Doug Lytle]

On 3/5/19 2:46 AM, Gokan Atmaca wrote:

Asterisk can send calls, but I don't get a call. What could be the problem?

[from-siptrunk]
exten => 13XXX,1,dial(${OPERATOR},20)



You are trying to match a pattern, so this needs to be

exten => _13XXX,1,dial(${OPERATOR},20)

Doug


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[asterisk-users] asterisk 16.2.1 inbound route

[Gokan Atmaca]

Hello

Asterisk can send calls, but I don't get a call. What could be the problem?

[from-siptrunk]
exten => 13XXX,1,dial(${OPERATOR},20)

Thanks.

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[asterisk-users] Asterisk 15.7.2 and 16.2.1 Now Available (Security)

[Asterisk Development Team]

The Asterisk Development Team would like to announce security releases for
Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
16.2.1.

These releases are available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk/releases

The following security vulnerabilities were resolved in these versions:

* AST-2019-001: Remote crash vulnerability with SDP protocol violation
  When Asterisk makes an outgoing call, a very specific SDP protocol violation
  by the remote party can cause Asterisk to crash.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1

The security advisory is available at:

https://downloads.asterisk.org/pub/security/AST-2019-001.pdf

Thank you for your continued support of Asterisk!-- 
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