[PJSIP] CHANUNAVAIL - Endpoint: 'trunk-sfr': No auth objects matching realm(s) '' from challenge found

Hello,

I cannot make an outgoing call with the error CHANUNAVAIL :

    -- Executing [+336xxxxxxxx@work:1] Dial("PJSIP/1002-00000054", "PJSIP/+336xxxxxxxx@trunk-sfr") in new stack
    -- Called PJSIP/+336xxxxxxxx@trunk-sfr
[Jan 21 21:00:34] WARNING[538]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'trunk-sfr': No auth objects matching realm(s) '' from challenge found.
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/1002-00000054' status is 'CHANUNAVAIL'

Anybody has an idea?
Thanks!

My extensions.conf file:

[general] 
static=yes 
writeprotect=no 
autofallthrough=yes 
clearglobalvars=no
priorityjumping=no 

[general] 
static=yes 
writeprotect=no 
autofallthrough=yes 
clearglobalvars=no
priorityjumping=no 

[globals] 
CONSOLE=Console/dsp 
IAXINFO=guest 
TRUNK=Zap/g2 
TRUNKMSD=1 

[work]
exten => _0[12345679]XXXXXXXX,1,Dial(PJSIP/${EXTEN}@trunk-sfr)
exten => _+33[12345679]XXXXXXXX,1,Dial(PJSIP/${EXTEN}@trunk-sfr)
exten => _XXXX,1,Dial(PJSIP/${EXTEN})

; VoiceMail
; exten => 123,1,VoiceMailMain(${CALLERID(num)}) ; l'activation de cette ligne requiert l'activation du service de message via voicemail.conf

[from-sfr]
; exten => s,1,Dial(PJSIP/1001,60)
exten => s,1,Dial(PJSIP/1001)

My pjsip.conf file:

[global]

[registration]
auth_rejection_permanent=yes

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

[trunk-sfr]
type=registration
transport=transport-udp-nat
outbound_auth=trunk-sfr
server_uri=sip:+33123456789@ims.mnc010.mcc208.3gppnetwork.org
client_uri=sip:+33123456789@ims.mnc010.mcc208.3gppnetwork.org
outbound_proxy=sip:corbas.p-cscf.sfr.net:5062\;lr

[trunk-sfr]
type=auth
password=XXXXXXXXXXX
username=XXXXXXXXXXXX@sfr.fr

[trunk-sfr]
type=aor
contact=sip:+33123456789@ims.mnc010.mcc208.3gppnetwork.org
outbound_proxy=sip:corbas.p-cscf.sfr.net:5062\;lr

[trunk-sfr]
type=identify
endpoint=trunk-sfr
match=corbas.p-cscf.sfr.net:5062

[trunk-sfr]
type=endpoint
transport=transport-udp-nat
context=from-sfr
outbound_auth=trunk-sfr
from_domain=ims.mnc010.mcc208.3gppnetwork.org
from_user=+33123456789 
allow=!all,opus,speex,g722,alaw,ulaw,gsm
outbound_proxy = sip:corbas.p-cscf.sfr.net:5062\;lr
aors = trunk-sfr

[endpoint_internal](!)
type=endpoint
context=work
disallow=all
allow=ulaw
language=fr

[auth_userpass](!)
type=auth
auth_type=userpass

[aor_dynamic](!)
type=aor
max_contacts=1
remove_existing=yes

[1001](endpoint_internal)
auth=1001
aors=1001
[1001](auth_userpass)
password=XXXXXXXXXXX
username=1001
[1001](aor_dynamic)

One of the parameters in the 401 response can be a realm. It appears to have been set to a null string, rather than left out. I suspect they are not the same.

You will need to get the actual contents of the 401, using “pjsip set logger on”, and check and if there is a realm= parameters, with no value, check the RFCs for how this should be handled.

Specifying the realm in the type=auth section, as an empty string, might work.

The detailed logs with pjsip set logger on:

[Jan 21 20:26:46] ERROR[459]: res_pjsip_session.c:937 handle_incoming_sdp:  1002: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
    -- Executing [0612345678@work:1] Dial("PJSIP/1002-00000048", "PJSIP/0612345678@trunk-sfr") in new stack
    -- Called PJSIP/0612345678@trunk-sfr
[Jan 21 20:26:47] WARNING[459]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'trunk-sfr': No auth objects matching realm(s) '' from challenge found.
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/1002-00000048' status is 'CHANUNAVAIL'
router3*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (1165 bytes) from UDP:192.168.3.173:5060 --->
INVITE sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.ZUxlhrt0v;rport
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: sip:0612345678@domain.local
CSeq: 20 INVITE
Call-ID: 4TLbNc67YF
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 495
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
User-Agent: Linphone Desktop/ (Ubuntu 22.04.1 LTS, Qt 5.15.3) LinphoneCore/4.4.21

v=0
o=1002 3348 599 IN IP4 192.168.3.173
s=Talk
c=IN IP4 192.168.3.173
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVPF 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (474 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.ZUxlhrt0v
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>;tag=z9hG4bK.ZUxlhrt0v
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1674332930/df807965872c3191c1685c3a733f96d8",opaque="5c8b7332194275dc",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.15.0
Content-Length:  0


<--- Received SIP request (418 bytes) from UDP:192.168.3.173:5060 --->
ACK sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.ZUxlhrt0v;rport
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>;tag=z9hG4bK.ZUxlhrt0v
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
Max-Forwards: 70
CSeq: 20 ACK


<--- Received SIP request (1461 bytes) from UDP:192.168.3.173:5060 --->
INVITE sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.aknG4Xhac;rport
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: sip:0612345678@domain.local
CSeq: 21 INVITE
Call-ID: 4TLbNc67YF
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 495
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
User-Agent: Linphone Desktop/ (Ubuntu 22.04.1 LTS, Qt 5.15.3) LinphoneCore/4.4.21
Authorization:  Digest realm="asterisk", nonce="1674332930/df807965872c3191c1685c3a733f96d8", algorithm=MD5, opaque="5c8b7332194275dc", username="1002",  uri="sip:0612345678@domain.local:5060", response="40e82d2720974dc417430f3f582b52b2", cnonce="V3qiXuvH7f2skqiR", nc=00000001, qop=auth

v=0
o=1002 3348 599 IN IP4 192.168.3.173
s=Talk
c=IN IP4 192.168.3.173
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVPF 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (300 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.aknG4Xhac
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Content-Length:  0


[Jan 21 20:28:50] ERROR[470]: res_pjsip_session.c:937 handle_incoming_sdp:  1002: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
<--- Transmitting SIP response (350 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.aknG4Xhac
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>;tag=6X0ZD4BQOU.RQH4OCMm6GSb70gsekUi.
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Content-Length:  0


<--- Received SIP request (433 bytes) from UDP:192.168.3.173:5060 --->
ACK sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.aknG4Xhac;rport
Call-ID: 4TLbNc67YF
From: <sip:1001@domain.local>;tag=kHQwvy0k5
To: <sip:0612345678@domain.local>;tag=6X0ZD4BQOU.RQH4OCMm6GSb70gsekUi.
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
Max-Forwards: 70
CSeq: 21 ACK


<--- Received SIP request (1164 bytes) from UDP:192.168.3.173:5060 --->
INVITE sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.LTKWgc1dQ;rport
From: <sip:1001@domain.local>;tag=tGlzd585N
To: sip:0612345678@domain.local
CSeq: 20 INVITE
Call-ID: w~ARmS6~3G
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 494
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
User-Agent: Linphone Desktop/ (Ubuntu 22.04.1 LTS, Qt 5.15.3) LinphoneCore/4.4.21

v=0
o=1002 3348 600 IN IP4 192.168.3.173
s=Talk
c=IN IP4 192.168.3.173
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (474 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.LTKWgc1dQ
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=z9hG4bK.LTKWgc1dQ
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1674332930/df807965872c3191c1685c3a733f96d8",opaque="0687bf9f261c9f3d",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.15.0
Content-Length:  0


<--- Received SIP request (418 bytes) from UDP:192.168.3.173:5060 --->
ACK sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.LTKWgc1dQ;rport
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=z9hG4bK.LTKWgc1dQ
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
Max-Forwards: 70
CSeq: 20 ACK


<--- Received SIP request (1460 bytes) from UDP:192.168.3.173:5060 --->
INVITE sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.uUBBaJGQ1;rport
From: <sip:1001@domain.local>;tag=tGlzd585N
To: sip:0612345678@domain.local
CSeq: 21 INVITE
Call-ID: w~ARmS6~3G
Max-Forwards: 70
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 494
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
User-Agent: Linphone Desktop/ (Ubuntu 22.04.1 LTS, Qt 5.15.3) LinphoneCore/4.4.21
Authorization:  Digest realm="asterisk", nonce="1674332930/df807965872c3191c1685c3a733f96d8", algorithm=MD5, opaque="0687bf9f261c9f3d", username="1002",  uri="sip:0612345678@domain.local:5060", response="bf4421cff9b28229f9b928c9a7a06528", cnonce="5JnwzOzN2e790UHz", nc=00000001, qop=auth

v=0
o=1002 3348 600 IN IP4 192.168.3.173
s=Talk
c=IN IP4 192.168.3.173
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 0 8 101 99 100
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:99 telephone-event/16000
a=rtpmap:100 telephone-event/8000
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (300 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Content-Length:  0


    -- Executing [0612345678@work:1] Dial("PJSIP/1002-0000004a", "PJSIP/0612345678@trunk-sfr") in new stack
    -- Called PJSIP/0612345678@trunk-sfr
<--- Transmitting SIP request (1132 bytes) to UDP:92.91.179.40:5062 --->
INVITE sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 93.4.216.9:5060;rport;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org>
Contact: <sip:+33287654321@93.4.216.9:5060>
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
CSeq: 18726 INVITE
Route: <sip:corbas.p-cscf.sfr.net:5062;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Type: application/sdp
Content-Length:   361

v=0
o=- 906875053 906875053 IN IP4 93.4.216.9
s=Asterisk
c=IN IP4 93.4.216.9
t=0 0
m=audio 10044 RTP/AVP 0 107 110 9 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

<--- Received SIP response (430 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 100 Trying
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Date: Sat, 21 Jan 2023 20:28:37 GMT
Content-Length: 0


<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256

v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20

       > 0x7f9f46497400 -- Strict RTP learning after remote address set to: 92.91.224.78:17964
    -- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
       > 0x7f9f464b93b0 -- Strict RTP learning after remote address set to: 192.168.3.173:7078
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Contact: <sip:192.168.3.254:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   226

v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

       > 0x7f9f464b93b0 -- Strict RTP switching to RTP target address 192.168.3.173:7078 as source
<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256

v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20

    -- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length:   226

v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256

v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20

    -- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length:   226

v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256

v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20

    -- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length:   226

v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

       > 0x7f9f464b93b0 -- Strict RTP learning complete - Locking on source address 192.168.3.173:7078
<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256

v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20

    -- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length:   226

v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (1104 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-118977@pcgw-0006.imsgroup-003.cor1asbc03.ims.sfr.net:5062;x-afi=8>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel";audio
Content-Type: application/sdp
P-Early-Media: inactive
RSeq: 1
Server: Alcatel-Lucent-HPSS/3.0.3
Supported: precondition
Content-Length: 256

v=0
o=LucentPCSF 830991649 830991649 IN IP4 imsgroup-003.cor1asbc03.ims.sfr.net
s=-
c=IN IP4 92.91.224.78
t=0 0
m=audio 17964 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20

    -- PJSIP/trunk-sfr-0000004b is making progress passing it to PJSIP/1002-0000004a
<--- Transmitting SIP response (753 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.3.254:5060>
Content-Type: application/sdp
Content-Length:   226

v=0
o=- 3348 602 IN IP4 192.168.3.254
s=Asterisk
c=IN IP4 192.168.3.254
t=0 0
m=audio 10034 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (496 bytes) from UDP:92.91.179.40:5062 --->
SIP/2.0 500 Server Internal Error
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
CSeq: 18726 INVITE
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0


<--- Transmitting SIP request (535 bytes) to UDP:92.91.179.40:5062 --->
ACK sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 93.4.216.9:5060;rport;branch=z9hG4bKPjrbz92zKKeWKGLuj2-X4WF0vPlpecPpo2
From: <sip:+33287654321@ims.mnc010.mcc208.3gppnetwork.org>;tag=hCHqGKT2rc0KomF9-.lBvKeaT6oTKtWR
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org>;tag=62ba2c57-63cc4af5183a903-gm-po-lucentPCSF-012866
Call-ID: 5UrSOYtvmclD2JBK3SFYEadnlj6WSRZZ
CSeq: 18726 ACK
Route: <sip:corbas.p-cscf.sfr.net:5062;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/1002-0000004a' status is 'CHANUNAVAIL'
<--- Transmitting SIP response (485 bytes) to UDP:192.168.3.173:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.3.173:5060;rport=5060;received=192.168.3.173;branch=z9hG4bK.uUBBaJGQ1
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
CSeq: 21 INVITE
Server: Asterisk PBX 18.15.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Reason: Q.850;cause=38
Content-Length:  0


<--- Received SIP request (433 bytes) from UDP:192.168.3.173:5060 --->
ACK sip:0612345678@domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.173:5060;branch=z9hG4bK.uUBBaJGQ1;rport
Call-ID: w~ARmS6~3G
From: <sip:1001@domain.local>;tag=tGlzd585N
To: <sip:0612345678@domain.local>;tag=SgToqSznO0Eeq00C5ISb861.oeq7skfU
Contact: <sip:1001@192.168.3.173;transport=udp>;expires=3599;+sip.instance="<urn:uuid:af16bccc-f148-00fe-91eb-78c65b9d3d31>"
Max-Forwards: 70
CSeq: 21 ACK

I had updated the auth config like this:

[trunk-sfr]
type=auth
password=XXXXXXXXXXXXXX
username=XXXXXXXXXXXXXXXXXXX@sfr.fr
realm=
CLI> pjsip show auth trunk-sfr

  I/OAuth:  <AuthId/UserName.............................................................>
==========================================================================================

     Auth:  trunk-sfr/NDI0247672022.CTR.THD@sfr.fr

 ParameterName  : ParameterValue
 =============================================
 auth_type      : userpass
 md5_cred       : 
 nonce_lifetime : 32
 oauth_clientid : 
 oauth_secret   : 
 password       : XXXXXXXXXXXXXX
 realm          : 
 refresh_token  : 
 username       : XXXXXXXXXXXXXXXXXXX@sfr.fr

but unfortunately, same error… :frowning:

There is no inbound authentication request in that log!

The interesting bit has already happened before the log starts.

As a general principle, do not screen scrape logs, enable the full log and take your logs from /var/log/asterisk/full

with :

full => notice,warning,error,debug,verbose,dtmf,fax

in logger.conf configuration file
/var/log/asterisk/full contains the lines:

[Jan 22 13:50:47] VERBOSE[832][C-00000031] pbx.c: Executing [+33612345678@work:1] Dial("PJSIP/1001-0000005e", "PJSIP/+33612345678@trunk-sfr") in new stack
[Jan 22 13:50:47] VERBOSE[832][C-00000031] app_dial.c: Called PJSIP/+33612345678@trunk-sfr
[Jan 22 13:50:47] WARNING[823] res_pjsip_outbound_authenticator_digest.c: Endpoint: 'trunk-sfr': No auth objects matching realm(s) '' from challenge found.
[Jan 22 13:50:47] VERBOSE[832][C-00000031] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[Jan 22 13:50:47] VERBOSE[832][C-00000031] pbx.c: Auto fallthrough, channel 'PJSIP/1001-0000005e' status is 'CHANUNAVAIL'

You’ve turned off the pjsip set logger on (it is not a permanent change)

[Jan 22 14:00:26] VERBOSE[1] asterisk.c: Asterisk Ready.
[Jan 22 14:00:36] VERBOSE[18] asterisk.c: Remote UNIX connection
[Jan 22 14:02:27] VERBOSE[104][C-00000001] pbx.c: Executing [0612345678@work:1] Dial("PJSIP/1001-00000000", "PJSIP/0612345678@trunk-sfr") in new stack
[Jan 22 14:02:27] VERBOSE[104][C-00000001] app_dial.c: Called PJSIP/0612345678@trunk-sfr
[Jan 22 14:02:28] WARNING[99] res_pjsip_outbound_authenticator_digest.c: Endpoint: 'trunk-sfr': No auth objects matching realm(s) '' from challenge found.
[Jan 22 14:02:28] VERBOSE[104][C-00000001] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[Jan 22 14:02:28] VERBOSE[104][C-00000001] pbx.c: Auto fallthrough, channel 'PJSIP/1001-00000000' status is 'CHANUNAVAIL'
[Jan 22 14:03:56] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP request (975 bytes) from UDP:192.168.3.62:5061 --->
INVITE sip:0612345678@sip.domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.62:5061;branch=z9hG4bK-8089b011
From: "DECT" <sip:1001@sip.domain.local:5060>;tag=1c33cbb71e7f0c03o1
To: <sip:0612345678@sip.domain.local:5060>
Remote-Party-ID: "DECT" <sip:1001@sip.domain.local:5060>;screen=yes;party=calling
Call-ID: ad5c7cc7-5e3f8e33@192.168.3.62
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "DECT" <sip:1001@192.168.3.62:5061;ref=1001>
Expires: 240
User-Agent: Cisco/SPA112-1.4.1_SR5
Content-Length: 333
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 6916559 6916559 IN IP4 192.168.3.62
s=-
c=IN IP4 192.168.3.62
t=0 0
m=audio 16478 RTP/AVP 8 0 2 18 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

[Jan 22 14:03:56] VERBOSE[107] res_pjsip_logger.c: <--- Transmitting SIP response (507 bytes) to UDP:192.168.3.62:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.62:5061;rport=5061;received=192.168.3.62;branch=z9hG4bK-8089b011
Call-ID: ad5c7cc7-5e3f8e33@192.168.3.62
From: "DECT" <sip:1001@sip.domain.local>;tag=1c33cbb71e7f0c03o1
To: <sip:0612345678@sip.domain.local>;tag=z9hG4bK-8089b011
CSeq: 101 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1674396236/3b4390ed53909c40431cd911fc9d2ba1",opaque="1dd583361b87d555",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.15.0
Content-Length:  0


[Jan 22 14:03:56] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP request (440 bytes) from UDP:192.168.3.62:5061 --->
ACK sip:0612345678@sip.domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.62:5061;branch=z9hG4bK-8089b011
From: "DECT" <sip:1001@sip.domain.local:5060>;tag=1c33cbb71e7f0c03o1
To: <sip:0612345678@sip.domain.local:5060>;tag=z9hG4bK-8089b011
Call-ID: ad5c7cc7-5e3f8e33@192.168.3.62
CSeq: 101 ACK
Max-Forwards: 70
Contact: "DECT" <sip:1001@192.168.3.62:5061;ref=1001>
User-Agent: Cisco/SPA112-1.4.1_SR5
Content-Length: 0


[Jan 22 14:03:56] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP request (1252 bytes) from UDP:192.168.3.62:5061 --->
INVITE sip:0612345678@sip.domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.62:5061;branch=z9hG4bK-ad6e37cd
From: "DECT" <sip:1001@sip.domain.local:5060>;tag=1c33cbb71e7f0c03o1
To: <sip:0612345678@sip.domain.local:5060>
Remote-Party-ID: "DECT" <sip:1001@sip.domain.local:5060>;screen=yes;party=calling
Call-ID: ad5c7cc7-5e3f8e33@192.168.3.62
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="1001",realm="asterisk",nonce="1674396236/3b4390ed53909c40431cd911fc9d2ba1",uri="sip:0612345678@sip.domain.local:5060",algorithm=MD5,response="aa81119a9260360bbe245b863884fae2",opaque="1dd583361b87d555",qop=auth,nc=00000001,cnonce="9bd5e56c"
Contact: "DECT" <sip:1001@192.168.3.62:5061;ref=1001>
Expires: 240
User-Agent: Cisco/SPA112-1.4.1_SR5
Content-Length: 333
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 6916559 6916559 IN IP4 192.168.3.62
s=-
c=IN IP4 192.168.3.62
t=0 0
m=audio 16478 RTP/AVP 8 0 2 18 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

[Jan 22 14:03:56] VERBOSE[107] res_pjsip_logger.c: <--- Transmitting SIP response (334 bytes) to UDP:192.168.3.62:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.62:5061;rport=5061;received=192.168.3.62;branch=z9hG4bK-ad6e37cd
Call-ID: ad5c7cc7-5e3f8e33@192.168.3.62
From: "DECT" <sip:1001@sip.domain.local>;tag=1c33cbb71e7f0c03o1
To: <sip:0612345678@sip.domain.local>
CSeq: 102 INVITE
Server: Asterisk PBX 18.15.0
Content-Length:  0


[Jan 22 14:03:56] VERBOSE[113][C-00000002] pbx.c: Executing [0612345678@work:1] Dial("PJSIP/1001-00000002", "PJSIP/0612345678@trunk-sfr") in new stack
[Jan 22 14:03:56] VERBOSE[113][C-00000002] app_dial.c: Called PJSIP/0612345678@trunk-sfr
[Jan 22 14:03:56] VERBOSE[107] res_pjsip_logger.c: <--- Transmitting SIP request (1133 bytes) to UDP:92.91.179.24:5062 --->
INVITE sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 93.4.216.9:5060;rport;branch=z9hG4bKPjV6ePSo02lq-NqKXAAfpaW3la1XklgR4g
From: <sip:+33247672022@ims.mnc010.mcc208.3gppnetwork.org>;tag=Vs7eMs3wvEDEJAloXgIlamF5zL6KkgOJ
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org>
Contact: <sip:+33247672022@93.4.216.9:5060>
Call-ID: ZEpiDooMTBLWA7zoc58j8KSd6slgCOdt
CSeq: 8444 INVITE
Route: <sip:corbas.p-cscf.sfr.net:5062;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Type: application/sdp
Content-Length:   363

v=0
o=- 1702257044 1702257044 IN IP4 93.4.216.9
s=Asterisk
c=IN IP4 93.4.216.9
t=0 0
m=audio 10016 RTP/AVP 0 107 110 9 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

[Jan 22 14:03:56] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP response (429 bytes) from UDP:92.91.179.24:5062 --->
SIP/2.0 100 Trying
Call-ID: ZEpiDooMTBLWA7zoc58j8KSd6slgCOdt
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjV6ePSo02lq-NqKXAAfpaW3la1XklgR4g;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
From: <sip:+33247672022@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=Vs7eMs3wvEDEJAloXgIlamF5zL6KkgOJ
CSeq: 8444 INVITE
Date: Sun, 22 Jan 2023 14:03:41 GMT
Content-Length: 0


[Jan 22 14:03:56] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP response (482 bytes) from UDP:92.91.179.24:5062 --->
SIP/2.0 407 Proxy Authentication Required
Call-ID: ZEpiDooMTBLWA7zoc58j8KSd6slgCOdt
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjV6ePSo02lq-NqKXAAfpaW3la1XklgR4g;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=62fd62fc-63cd423d1ffbe614
From: <sip:+33247672022@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=Vs7eMs3wvEDEJAloXgIlamF5zL6KkgOJ
CSeq: 8444 INVITE
Date: Sun, 22 Jan 2023 14:03:41 GMT
Content-Length: 0


[Jan 22 14:03:56] VERBOSE[107] res_pjsip_logger.c: <--- Transmitting SIP request (511 bytes) to UDP:92.91.179.24:5062 --->
ACK sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 93.4.216.9:5060;rport;branch=z9hG4bKPjV6ePSo02lq-NqKXAAfpaW3la1XklgR4g
From: <sip:+33247672022@ims.mnc010.mcc208.3gppnetwork.org>;tag=Vs7eMs3wvEDEJAloXgIlamF5zL6KkgOJ
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org>;tag=62fd62fc-63cd423d1ffbe614
Call-ID: ZEpiDooMTBLWA7zoc58j8KSd6slgCOdt
CSeq: 8444 ACK
Route: <sip:corbas.p-cscf.sfr.net:5062;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Length:  0


[Jan 22 14:03:56] WARNING[107] res_pjsip_outbound_authenticator_digest.c: Endpoint: 'trunk-sfr': No auth objects matching realm(s) '' from challenge found.
[Jan 22 14:03:56] VERBOSE[113][C-00000002] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[Jan 22 14:03:56] VERBOSE[113][C-00000002] pbx.c: Auto fallthrough, channel 'PJSIP/1001-00000002' status is 'CHANUNAVAIL'
[Jan 22 14:03:56] VERBOSE[108] res_pjsip_logger.c: <--- Transmitting SIP response (408 bytes) to UDP:192.168.3.62:5061 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.3.62:5061;rport=5061;received=192.168.3.62;branch=z9hG4bK-ad6e37cd
Call-ID: ad5c7cc7-5e3f8e33@192.168.3.62
From: "DECT" <sip:1001@sip.domain.local>;tag=1c33cbb71e7f0c03o1
To: <sip:0612345678@sip.domain.local>;tag=yY4BRHToaYcLRllMLkml8j.Rrz1JS0qz
CSeq: 102 INVITE
Server: Asterisk PBX 18.15.0
Reason: Q.850;cause=34
Content-Length:  0


[Jan 22 14:03:56] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP request (733 bytes) from UDP:192.168.3.62:5061 --->
ACK sip:0612345678@sip.domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.62:5061;branch=z9hG4bK-ad6e37cd
From: "DECT" <sip:1001@sip.domain.local:5060>;tag=1c33cbb71e7f0c03o1
To: <sip:0612345678@sip.domain.local:5060>;tag=yY4BRHToaYcLRllMLkml8j.Rrz1JS0qz
Call-ID: ad5c7cc7-5e3f8e33@192.168.3.62
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="1001",realm="asterisk",nonce="1674396236/3b4390ed53909c40431cd911fc9d2ba1",uri="sip:0612345678@sip.domain.local:5060",algorithm=MD5,response="aa81119a9260360bbe245b863884fae2",opaque="1dd583361b87d555",qop=auth,nc=00000001,cnonce="9bd5e56c"
Contact: "DECT" <sip:1001@192.168.3.62:5061;ref=1001>
User-Agent: Cisco/SPA112-1.4.1_SR5
Content-Length: 0

I will retry after an asterisk restart

For additional information, the registration is OK and incoming call also OK…

The complete full logs from Asterisk start with pjsip set logger on: PsiTransfer
The call part:

[Jan 22 14:10:43] VERBOSE[1] asterisk.c: Asterisk Ready.
[Jan 22 14:10:47] VERBOSE[13] asterisk.c: Remote UNIX connection
[Jan 22 14:11:02] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP request (975 bytes) from UDP:192.168.3.62:5061 --->
INVITE sip:0612345678@sip.domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.62:5061;branch=z9hG4bK-f544a247
From: "DECT" <sip:1001@sip.domain.local:5060>;tag=2bfb9269be3a4d64o1
To: <sip:0612345678@sip.domain.local:5060>
Remote-Party-ID: "DECT" <sip:1001@sip.domain.local:5060>;screen=yes;party=calling
Call-ID: fbd9568d-4c78b790@192.168.3.62
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "DECT" <sip:1001@192.168.3.62:5061;ref=1001>
Expires: 240
User-Agent: Cisco/SPA112-1.4.1_SR5
Content-Length: 333
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 6959674 6959674 IN IP4 192.168.3.62
s=-
c=IN IP4 192.168.3.62
t=0 0
m=audio 16480 RTP/AVP 8 0 2 18 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

[Jan 22 14:11:02] VERBOSE[36] res_pjsip_logger.c: <--- Transmitting SIP response (507 bytes) to UDP:192.168.3.62:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.62:5061;rport=5061;received=192.168.3.62;branch=z9hG4bK-f544a247
Call-ID: fbd9568d-4c78b790@192.168.3.62
From: "DECT" <sip:1001@sip.domain.local>;tag=2bfb9269be3a4d64o1
To: <sip:0612345678@sip.domain.local>;tag=z9hG4bK-f544a247
CSeq: 101 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1674396662/6ae31de8973cfbd29f7d7fe9bc48cec6",opaque="41615da110adf537",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.15.0
Content-Length:  0


[Jan 22 14:11:02] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP request (440 bytes) from UDP:192.168.3.62:5061 --->
ACK sip:0612345678@sip.domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.62:5061;branch=z9hG4bK-f544a247
From: "DECT" <sip:1001@sip.domain.local:5060>;tag=2bfb9269be3a4d64o1
To: <sip:0612345678@sip.domain.local:5060>;tag=z9hG4bK-f544a247
Call-ID: fbd9568d-4c78b790@192.168.3.62
CSeq: 101 ACK
Max-Forwards: 70
Contact: "DECT" <sip:1001@192.168.3.62:5061;ref=1001>
User-Agent: Cisco/SPA112-1.4.1_SR5
Content-Length: 0


[Jan 22 14:11:02] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP request (1252 bytes) from UDP:192.168.3.62:5061 --->
INVITE sip:0612345678@sip.domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.62:5061;branch=z9hG4bK-ae59efd3
From: "DECT" <sip:1001@sip.domain.local:5060>;tag=2bfb9269be3a4d64o1
To: <sip:0612345678@sip.domain.local:5060>
Remote-Party-ID: "DECT" <sip:1001@sip.domain.local:5060>;screen=yes;party=calling
Call-ID: fbd9568d-4c78b790@192.168.3.62
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="1001",realm="asterisk",nonce="1674396662/6ae31de8973cfbd29f7d7fe9bc48cec6",uri="sip:0612345678@sip.domain.local:5060",algorithm=MD5,response="c9a25aad329c5b6ef41156e4b4129658",opaque="41615da110adf537",qop=auth,nc=00000001,cnonce="1a3a4f26"
Contact: "DECT" <sip:1001@192.168.3.62:5061;ref=1001>
Expires: 240
User-Agent: Cisco/SPA112-1.4.1_SR5
Content-Length: 333
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 6959674 6959674 IN IP4 192.168.3.62
s=-
c=IN IP4 192.168.3.62
t=0 0
m=audio 16480 RTP/AVP 8 0 2 18 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

[Jan 22 14:11:02] VERBOSE[36] res_pjsip_logger.c: <--- Transmitting SIP response (334 bytes) to UDP:192.168.3.62:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.62:5061;rport=5061;received=192.168.3.62;branch=z9hG4bK-ae59efd3
Call-ID: fbd9568d-4c78b790@192.168.3.62
From: "DECT" <sip:1001@sip.domain.local>;tag=2bfb9269be3a4d64o1
To: <sip:0612345678@sip.domain.local>
CSeq: 102 INVITE
Server: Asterisk PBX 18.15.0
Content-Length:  0


[Jan 22 14:11:02] VERBOSE[97][C-00000001] pbx.c: Executing [0612345678@work:1] Dial("PJSIP/1001-00000000", "PJSIP/0612345678@trunk-sfr") in new stack
[Jan 22 14:11:02] VERBOSE[97][C-00000001] app_dial.c: Called PJSIP/0612345678@trunk-sfr
[Jan 22 14:11:02] VERBOSE[37] res_pjsip_logger.c: <--- Transmitting SIP request (1132 bytes) to UDP:77.136.7.5:5062 --->
INVITE sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 93.4.216.9:5060;rport;branch=z9hG4bKPjdz.6Sx3pj-kEtJte4fXkamdtCo-9-aRN
From: <sip:+33247672022@ims.mnc010.mcc208.3gppnetwork.org>;tag=pQAWhkato2ecEAyjOFfPDFIT2qEYx6da
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org>
Contact: <sip:+33247672022@93.4.216.9:5060>
Call-ID: zWsGk6hCICBAhdqVnvRhsKwiJj1ZRO0d
CSeq: 13318 INVITE
Route: <sip:corbas.p-cscf.sfr.net:5062;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Type: application/sdp
Content-Length:   361

v=0
o=- 300950253 300950253 IN IP4 93.4.216.9
s=Asterisk
c=IN IP4 93.4.216.9
t=0 0
m=audio 10046 RTP/AVP 0 107 110 9 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:110 speex/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

[Jan 22 14:11:02] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP response (465 bytes) from UDP:77.136.7.5:5062 --->
SIP/2.0 100 Trying
Call-ID: zWsGk6hCICBAhdqVnvRhsKwiJj1ZRO0d
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjdz.6Sx3pj-kEtJte4fXkamdtCo-9-aRN;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
From: <sip:+33247672022@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=pQAWhkato2ecEAyjOFfPDFIT2qEYx6da
CSeq: 13318 INVITE
Date: Sun, 22 Jan 2023 14:10:47 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0


[Jan 22 14:11:02] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP response (518 bytes) from UDP:77.136.7.5:5062 --->
SIP/2.0 407 Proxy Authentication Required
Call-ID: zWsGk6hCICBAhdqVnvRhsKwiJj1ZRO0d
Via: SIP/2.0/UDP 93.4.216.9:5060;received=93.4.216.9;branch=z9hG4bKPjdz.6Sx3pj-kEtJte4fXkamdtCo-9-aRN;rport=5060
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=6210272a-63cd43e72903d2c2
From: <sip:+33247672022@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=pQAWhkato2ecEAyjOFfPDFIT2qEYx6da
CSeq: 13318 INVITE
Date: Sun, 22 Jan 2023 14:10:47 GMT
Server: Alcatel-Lucent-HPSS/3.0.3
Content-Length: 0


[Jan 22 14:11:02] VERBOSE[36] res_pjsip_logger.c: <--- Transmitting SIP request (512 bytes) to UDP:77.136.7.5:5062 --->
ACK sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP 93.4.216.9:5060;rport;branch=z9hG4bKPjdz.6Sx3pj-kEtJte4fXkamdtCo-9-aRN
From: <sip:+33247672022@ims.mnc010.mcc208.3gppnetwork.org>;tag=pQAWhkato2ecEAyjOFfPDFIT2qEYx6da
To: <sip:0612345678@ims.mnc010.mcc208.3gppnetwork.org>;tag=6210272a-63cd43e72903d2c2
Call-ID: zWsGk6hCICBAhdqVnvRhsKwiJj1ZRO0d
CSeq: 13318 ACK
Route: <sip:corbas.p-cscf.sfr.net:5062;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 18.15.0
Content-Length:  0


[Jan 22 14:11:02] WARNING[36] res_pjsip_outbound_authenticator_digest.c: Endpoint: 'trunk-sfr': No auth objects matching realm(s) '' from challenge found.
[Jan 22 14:11:02] VERBOSE[97][C-00000001] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[Jan 22 14:11:02] VERBOSE[97][C-00000001] pbx.c: Auto fallthrough, channel 'PJSIP/1001-00000000' status is 'CHANUNAVAIL'
[Jan 22 14:11:02] VERBOSE[37] res_pjsip_logger.c: <--- Transmitting SIP response (408 bytes) to UDP:192.168.3.62:5061 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.3.62:5061;rport=5061;received=192.168.3.62;branch=z9hG4bK-ae59efd3
Call-ID: fbd9568d-4c78b790@192.168.3.62
From: "DECT" <sip:1001@sip.domain.local>;tag=2bfb9269be3a4d64o1
To: <sip:0612345678@sip.domain.local>;tag=xvFCqGDcHYOKpLEcidMYBhrMViRtUN2k
CSeq: 102 INVITE
Server: Asterisk PBX 18.15.0
Reason: Q.850;cause=34
Content-Length:  0


[Jan 22 14:11:02] VERBOSE[35] res_pjsip_logger.c: <--- Received SIP request (733 bytes) from UDP:192.168.3.62:5061 --->
ACK sip:0612345678@sip.domain.local:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.62:5061;branch=z9hG4bK-ae59efd3
From: "DECT" <sip:1001@sip.domain.local:5060>;tag=2bfb9269be3a4d64o1
To: <sip:0612345678@sip.domain.local:5060>;tag=xvFCqGDcHYOKpLEcidMYBhrMViRtUN2k
Call-ID: fbd9568d-4c78b790@192.168.3.62
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="1001",realm="asterisk",nonce="1674396662/6ae31de8973cfbd29f7d7fe9bc48cec6",uri="sip:0612345678@sip.domain.local:5060",algorithm=MD5,response="c9a25aad329c5b6ef41156e4b4129658",opaque="41615da110adf537",qop=auth,nc=00000001,cnonce="1a3a4f26"
Contact: "DECT" <sip:1001@192.168.3.62:5061;ref=1001>
User-Agent: Cisco/SPA112-1.4.1_SR5
Content-Length: 0

This is a protocol violation by the other party. It violates this mandatory requirement in the SIP specification:

The proxy MUST populate the 407 (Proxy Authentication Required) message with a Proxy-Authenticate header field value applicable to the proxy for the requested resource.

I guess it could be the proxy’s way of telling you it doesn’t want to talk to you, but it is not the right way of doing that.

There is nothing you can sensibly do from your side as you cannot construct the challenge response without the nonce value from the missing header.

2 Likes

YEAAAAAAAAAH, just a first message to say that I solve my issue (thanks to @david551 for it’s tips for the debug) :grin:

I will post a second bigger message with my analysis and the way to bypass the bad SIP server behavior.

In fact, following the response of @david551 that said that the proxy wont to talk me and the outgoing calls work some infrequent time without any explanation, I think that my ISP (SFR) infrastructure only accept outgoing call on the same server where the registration is performed.
In addition, the outgoing call with my Cisco SPA112 works without any problems on the same ISP SIP server.

To bypass this issue, I had fixed the DNS entry to one server, and now all my outgoing calls work perfectly.

Question: There is a way to ask pjsip module to perform outgoing call on the same host (ip) where the registration is performed?

Don’t specify a proxy. Do specify the correct name in the contact.