[asterisk-users] [asterisk13] Multiple transport objects of same protocol in pjsip.conf

[O. Hartmann]

Scenario:

Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) 
is behind
a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is 
server for
various VoIP telephones. 

Acoording to Asterisk's wiki, the transport section of pjsip.conf is configured 
as
follows:

; Transport via UDP
[transport-nat-udp]
type=   transport
protocol=   udp
local_net=  192.168.254.0/24
local_net=  127.0.0.1/32
bind=   192.168.254.1:5060
external_media_address= ddns.gdr
external_signaling_address= ddns.gdr

The VoIP phones are also residing on network 192.168.254.0/24, bu they are 
separated and
strictly prohibited by firewall rules to act with the outer net. The Asterisk 
PBX is
acting as the transition point between our VoIP phones and the ITSPs SIP server.

So, my understanding is that for NAT, the transport is recommended to be 
configured as
shown above. But what is about the transport with the phones "inside"?

I'm new to Asterisk, and from the "naiv" understanding of what I extracted from 
the
sparse documentation on that subject, for each endpoint associated with a 
phone, I need
transport. The protocol used by the phones is UDP, but no NAT. So, I did create 
a new
transport section, like

[transport-udp]
type=   transport
protocol=   udp
bind=   0.0.0.0

This results immediately in an error due to the bind= attribute. Asterisk bails 
out at
havind the address already in use. In fact, it is the trunk/endpoint consuming 
the
192.168.254.1:5060, and since the VoIP phones are all in 192.168.254.0/24, this 
results
obviously in an error. This is surprising me :-(

How to deal with this without adding more network complexity like routing (by 
putting the
phones into a subnet or other network)?

Kind regards,

oh
-- 
O. Hartmann


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Re: [asterisk-users] PJSIP, NAT and STUN/ICE

[O. Hartmann]

Am Tue, 10 Oct 2017 09:32:54 +0200
Guido Falsi <m...@madpilot.net> schrieb:

> On 10/09/2017 23:56, O. Hartmann wrote:
> > I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. 
> > Asterisk is
> > behind a NAT router, the physical setup is very much a trivial one. The 
> > Asterisk PBX
> > is supposed to act as the telephone gateway for several VoIP/SIP phones.
> > 
> > I'm using throughout pjsip as configuration, I have no experience with 
> > chan_sip since
> > I started recently using Asterisk for several SoHo and lab's projects. So 
> > be alarmed,
> > there may come some noobish questions.
> > 
> > When planning and setting up the Asterisk, I had to deal with NAT. The 
> > Asterisk config
> > object of type=transport knows about essential entries:
> > 
> > local_net=  192.168.254.1/24
> > bind=   192.168.254.1:5060
> > external_media_address= dyndns FQDN
> > external_signaling_address= dyndns FQDN
> > direct_media=   no
> > rtp_symmetric=  yes
> > force_rport=yes
> > 
> > dyndns FQDN is the FQDN of my broadband access point provided by some 
> > dynamical DNS
> > provider.
> > 
> > This setup is not working properly with when external_media_address= and
> > external_signaling_address= are set that way, but commenting out both makes 
> > all of the
> > ITSP which provides me with service happy.
> > 
> > I think, at this point I have no idea of the concept or, there is simply a 
> > missing
> > link, even a (dangerously) misconfigured router - on which the Asterisk 
> > runs. When
> > external__ are not set, I suppose they're set to 0.0.0.0/0.0.0.0, 
> > aren't they,
> > implying that they listen on all configured IPs?  
> 
> the external_xxx settings are not configuring where to listen. Simply
> tell asterisk what IP to put in the SDP data. You don't need to suppose
> or guess the content of the SIP/SDP packets, you can look at them by
> enabling asterisk debug output.
> 
> you can enable it following this guide:
> 
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide
> 
Thank you very much.

That is the way I figured out that the DDNS has been inserted into the packets 
and not,
as expected, my "outbound" IP (I expected the IP to be resolved by giving the 
DDNS FQDN).
Commenting out the options in question made Asterisk then work properly with 
the ITSP.
Obviously, Asterisk 13.17.2 is capable of figuring this out by itself, I try to
understand.


-- 
O. Hartmann

Ich widerspreche der Nutzung oder Übermittlung meiner Daten für
Werbezwecke oder für die Markt- oder Meinungsforschung (§ 28 Abs. 4 BDSG).


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Re: [asterisk-users] PJSIP, NAT and STUN/ICE

[O. Hartmann]

Am Mon, 9 Oct 2017 15:07:46 -0700
John Kiniston <johnkinis...@gmail.com> schrieb:

> Well, I can answer one of your questions.
> 
> To see the current external_media_address and external_signalling_address
> from the CLI you can issue a 'pjsip show transport '
> 
> Your router could have a built in sip helper that is rewriting the contact
> for your packets.
> 
> On Mon, Oct 9, 2017 at 2:56 PM, O. Hartmann <ohartm...@walstatt.org> wrote:
> 
> > I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current.
> > Asterisk is behind a
> > NAT router, the physical setup is very much a trivial one. The Asterisk
> > PBX is supposed
> > to act as the telephone gateway for several VoIP/SIP phones.
> >
> > I'm using throughout pjsip as configuration, I have no experience with
> > chan_sip since I
> > started recently using Asterisk for several SoHo and lab's projects. So be
> > alarmed, there
> > may come some noobish questions.
> >
> > When planning and setting up the Asterisk, I had to deal with NAT. The
> > Asterisk config
> > object of type=transport knows about essential entries:
> >
> > local_net=  192.168.254.1/24
> > bind=   192.168.254.1:5060
> > external_media_address= dyndns FQDN
> > external_signaling_address= dyndns FQDN
> > direct_media=   no
> > rtp_symmetric=  yes
> > force_rport=yes
> >
> > dyndns FQDN is the FQDN of my broadband access point provided by some
> > dynamical DNS
> > provider.
> >
> > This setup is not working properly with when external_media_address= and
> > external_signaling_address= are set that way, but commenting out both
> > makes all of the
> > ITSP which provides me with service happy.
> >
> > I think, at this point I have no idea of the concept or, there is simply a
> > missing link,
> > even a (dangerously) misconfigured router - on which the Asterisk runs.
> > When
> > external__ are not set, I suppose they're set to 0.0.0.0/0.0.0.0,
> > aren't they,
> > implying that they listen on all configured IPs?
> >
> > If so, is there a way to show the setting of external_media_address= and
> > external_signaling_address= on the CLI?
> >
> > When using PJSIP with the setting excluding attributes external_xxx_xxx,
> > does pjsip do
> > some magic to traverse the NAT in a proper way? At this very moment, I do
> > not understand
> > how things would work avoiding setting external_media_address= and
> > external_signaling_address=. The Asterisk 13 I'm running is supposed to be
> > bound to IP
> > 192.168.254.1, which is routed to the gateway.
> >
> > My problem is, I don not understand why the communication is working
> > fluently with both
> > external_xxx_xxx atrributes commeneted out which I suppose to be crucial -
> > following the
> > Asterisk 13 documentation!
> >
> > Maybe someone can sched some light onto this. I think, my view on the
> > matter is
> > completely confused.
> >
> > Thanks in advance,
> >
> > oh
> >
> > Another point
> >
> > --
> > _
> > -- 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
> >  
> 
> 
> 
Thank you for your answer.

This is the output of 

pjsip show transport transport-nat-udp (name of the transport object defined):

 external_media_address : 
 external_signaling_address : 
 external_signaling_port: 0

It is set, as expected, to nothing since I didn't set anything to make it work. 
I guess,
from this point, Asterisk figures somehow out itself how to traverse the NAT 
(in my
simple standard setup). 

I'm just curious. Maybe I'm on the wrong page here and the setup I have is that 
common,
that those extra options doesn't matter and NAT is easily circumferenced. As I 
said in my
introduction, Asterisk and pjsip are new to me (and I have no experience with
chan_sip) ;-).

Kind regards,

oh
-- 
O. Hartmann

Ich widerspreche der Nutzung oder Übermittlung meiner Daten für
Werbezwecke oder für die Markt- oder Meinungsforschung (§ 28 Abs. 4 BDSG).


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[asterisk-users] asterisk13: no voicemail prompt in German

[O. Hartmann]

According to the instructions given at 

https://www.asterisksounds.org/de

I converted and installed German prompts successfully and for numbers, I can 
successfully
listen to a German female voice counting or telling the date/time.

But unlikily, somehow the voicemail prompt is still English, although my 
general language
settings are "de".

I use pjsip.conf, not sip.conf.
In asterisk.conf, there is the follwoing defined:

[options]
[...]
defaultlanguage=de_DE

In extensions.conf,

I use:

[general]
format= gsm
[...]

and for the inbound endpoint:

[inbound]
exten   =>  THENUMBER,1,SetGlobalVar(CHANNEL(tonezone)=de)
same=>  n,SetGlobalVar(CHANNEL(language)=de_DE)
same=>  n,Set(DIALGROUP(group_isp2,add) = PJSIP/502)
same=>  n,Set(DIALGROUP(group_isp2,add) = PJSIP/512)
same=>  n,Dial(${DIALGROUP(group_isp2)},30)
same=>  n,VoiceMail(502@my-vmmbox)
same=>  n,Hangup()


and in voicemail.conf I have

[general]
[...]
tz= german  ; Timezone from zonemessages below. Irrelevant if 
envelope=no.
locale= de_DE.UTF-8 ; set the locale for generation of the 
date/time strings
(make ; sure the locales are installed in your operating system; e.g
german= Europe/Berlin|'vm-received' a d b Y 'digits/at' kN


So far.

In asterisk.conf, the directory for sounds data points to

/pool/asterisk/share/asterisk/sounds

which is comprised of 

./en/
./de_DE/

de_DE contails all the .sln16 and .gsm files, owned by asterisk:asterisk, from 
the source
above. Since the prompting of numbers and the date works well, but not the 
voicemail
prompt, there is something fishy I can not fathom.

For your help I'd like to thank in advance,

Oliver

-- 
O. Hartmann

Ich widerspreche der Nutzung oder Übermittlung meiner Daten für
Werbezwecke oder für die Markt- oder Meinungsforschung (§ 28 Abs. 4 BDSG).


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Re: [asterisk-users] asterisk13: no voicemail prompt in German

[O. Hartmann]

Am Sun, 27 Aug 2017 16:47:06 +0200
Guido Falsi <m...@madpilot.net> schrieb:

> On 08/27/2017 15:53, O. Hartmann wrote:
> 
> > and in voicemail.conf I have
> > 
> > [general]
> > [...]
> > tz= german  ; Timezone from zonemessages below. Irrelevant if 
> > envelope=no.
> > locale= de_DE.UTF-8 ; set the locale for generation of the 
> > date/time
> > strings  
> 
> Are you sure this should no be simply "de_DE"?
> 

Not sure, but according to the documents, this referes to encoding and needs to 
meet base
system's locales, so it doesn't have to do anything with the delegation of the 
correct
prompting in the proper language, I guess.

-- 
O. Hartmann

Ich widerspreche der Nutzung oder Übermittlung meiner Daten für
Werbezwecke oder für die Markt- oder Meinungsforschung (§ 28 Abs. 4 BDSG).


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[asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

[O. Hartmann]

It might sound stupid and a kind of "noobish", but I have serious trouble with
registering one of my ITSP to Asterisk 13, running on a FreeBSD 12-CURRENT box.

The following is seen in the log and anything seems somehow "normal", my PBX 
tries to
REGISTER, receives 401, and then  nothing more!

I can't see why the REGISTER attempt dies that early (reason?). The only hint 
is:

SIP/2.0 401 Unauthorized 1103003032F

Can someone shed some light/help onto this?

Thanks in advance,

Oliver

[...]
[Sep 1 17:32:06] VERBOSE[100189] res_pjsip_logger.c: <--- Transmitting SIP 
request (829
bytes) to UDP:213.20.127.47:5060 ---> REGISTER sip:sip.alice-voip.de SIP/2.0
Via: SIP/2.0/UDP
XXX.XXX.XXX.XXX:5060;rport;branch= From:
;tag= 
To:
 Call-ID: yxyxyxyxyxyxyxyxyxyxyxyxyxyxy
CSeq: 15095 REGISTER
Contact: 
Expires: 1800
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK,
REGISTER, REFER, MESSAGE Max-Forwards: 70
User-Agent: Asterisk13
Authorization: Digest username="491234567890", realm="ims.telefonica.de",
nonce="xxx", uri="sip:sip.alice-voip.de",
response="186x11yd22424424EDQb11133315b44ff1", algorithm=MD5,
cnonce="BasjdasKFHKbfhhfkjhfjkSGHF", qop=auth, nc=0001 Content-Length: 0


[Sep 1 17:32:06] VERBOSE[100188] res_pjsip_logger.c: <--- Received SIP response 
(589
bytes) from UDP:213.20.127.47:5060 ---> SIP/2.0 401 Unauthorized 1103003032F
Via: SIP/2.0/UDP
XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=xxx
To: 
;tag=xxx
From: ;tag=y 
Call-ID:
yxyxyxyxyxyxyxyxyxyxyxyxyxyxy CSeq: 15095 REGISTER
Service-Route: 
WWW-Authenticate: Digest
realm="ims.telefonica.de",nonce="xx",algorithm=MD5,qop="auth"
Content-Length: 0

[Sep 1 17:32:06] WARNING[100189] res_pjsip_outbound_registration.c: Temporal 
response
'401' received from 'sip:sip.alice-voip.de' on registration attempt to
'sip:491234567...@sip.alice-voip.de', retrying in '30'

[...] 



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Re: [asterisk-users] Received REGISTER response 401(Unauthorized 1103003032F)

[O. Hartmann]

Am Sat, 2 Sep 2017 09:58:09 +0200
"O. Hartmann" <ohartm...@walstatt.org> schrieb:

Is this question to "blunt" for this forum? The background is, that I have two 
ITSP
providing VoIP. One works with Asterisk 13 like a charme, but the other one 
not. This
specific ITSP claims that they've provided me with all the necessary 
informations -
comprised from registrar, username, password and SIP server. nothing more. The 
working
one did the same, and it worked.

Now I need to figure out what is wrong. I suspect the password, but before 
pressing
charges, I need to know some more proof ...

So far thanks.

oh

> It might sound stupid and a kind of "noobish", but I have serious trouble with
> registering one of my ITSP to Asterisk 13, running on a FreeBSD 12-CURRENT 
> box.
> 
> The following is seen in the log and anything seems somehow "normal", my PBX 
> tries to
> REGISTER, receives 401, and then  nothing more!
> 
> I can't see why the REGISTER attempt dies that early (reason?). The only hint 
> is:
> 
> SIP/2.0 401 Unauthorized 1103003032F
> 
> Can someone shed some light/help onto this?
> 
> Thanks in advance,
> 
> Oliver
> 
> [...]
> [Sep 1 17:32:06] VERBOSE[100189] res_pjsip_logger.c: <--- Transmitting SIP 
> request (829
> bytes) to UDP:213.20.127.47:5060 ---> REGISTER sip:sip.alice-voip.de SIP/2.0
> Via: SIP/2.0/UDP
> XXX.XXX.XXX.XXX:5060;rport;branch= 
> From:
> <sip:491234567...@sip.alice-voip.de>;tag= 
> To:
> <sip:491234567...@sip.alice-voip.de> Call-ID: yxyxyxyxyxyxyxyxyxyxyxyxyxyxy
> CSeq: 15095 REGISTER
> Contact: <sip:491234567...@xxx.xxx.xxx.xxx:5060>
> Expires: 1800
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
> PRACK,
> REGISTER, REFER, MESSAGE Max-Forwards: 70
> User-Agent: Asterisk13
> Authorization: Digest username="491234567890", realm="ims.telefonica.de",
> nonce="xxx", uri="sip:sip.alice-voip.de",
> response="186x11yd22424424EDQb11133315b44ff1", algorithm=MD5,
> cnonce="BasjdasKFHKbfhhfkjhfjkSGHF", qop=auth, nc=0001 Content-Length: 0
> 
> 
> [Sep 1 17:32:06] VERBOSE[100188] res_pjsip_logger.c: <--- Received SIP 
> response (589
> bytes) from UDP:213.20.127.47:5060 ---> SIP/2.0 401 Unauthorized 1103003032F
> Via: SIP/2.0/UDP
> XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=xxx
> To: 
> <sip:491234567...@sip.alice-voip.de>;tag=xxx
> From: <sip:491234567...@sip.alice-voip.de>;tag=y 
> Call-ID:
> yxyxyxyxyxyxyxyxyxyxyxyxyxyxy CSeq: 15095 REGISTER
> Service-Route: <sip:213.20.127.47:5060;transport=udp;lr>
> WWW-Authenticate: Digest
> realm="ims.telefonica.de",nonce="xx",algorithm=MD5,qop="auth"
> Content-Length: 0
> 
> [Sep 1 17:32:06] WARNING[100189] res_pjsip_outbound_registration.c: Temporal 
> response
> '401' received from 'sip:sip.alice-voip.de' on registration attempt to
> 'sip:491234567...@sip.alice-voip.de', retrying in '30'
> 
> [...] 
> 



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[asterisk-users] PJSIP, NAT and STUN/ICE

[O. Hartmann]

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is 
behind a
NAT router, the physical setup is very much a trivial one. The Asterisk PBX is 
supposed
to act as the telephone gateway for several VoIP/SIP phones.

I'm using throughout pjsip as configuration, I have no experience with chan_sip 
since I
started recently using Asterisk for several SoHo and lab's projects. So be 
alarmed, there
may come some noobish questions.

When planning and setting up the Asterisk, I had to deal with NAT. The Asterisk 
config
object of type=transport knows about essential entries:

local_net=  192.168.254.1/24
bind=   192.168.254.1:5060
external_media_address= dyndns FQDN
external_signaling_address= dyndns FQDN
direct_media=   no
rtp_symmetric=  yes
force_rport=yes

dyndns FQDN is the FQDN of my broadband access point provided by some dynamical 
DNS
provider.

This setup is not working properly with when external_media_address= and
external_signaling_address= are set that way, but commenting out both makes all 
of the
ITSP which provides me with service happy.

I think, at this point I have no idea of the concept or, there is simply a 
missing link,
even a (dangerously) misconfigured router - on which the Asterisk runs. When
external__ are not set, I suppose they're set to 0.0.0.0/0.0.0.0, 
aren't they,
implying that they listen on all configured IPs?

If so, is there a way to show the setting of external_media_address= and
external_signaling_address= on the CLI?

When using PJSIP with the setting excluding attributes external_xxx_xxx, does 
pjsip do
some magic to traverse the NAT in a proper way? At this very moment, I do not 
understand
how things would work avoiding setting external_media_address= and
external_signaling_address=. The Asterisk 13 I'm running is supposed to be 
bound to IP
192.168.254.1, which is routed to the gateway. 

My problem is, I don not understand why the communication is working fluently 
with both
external_xxx_xxx atrributes commeneted out which I suppose to be crucial - 
following the
Asterisk 13 documentation!

Maybe someone can sched some light onto this. I think, my view on the matter is
completely confused.

Thanks in advance,

oh

Another point


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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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