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artemus24

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[Résolu]Utilisation d'Asterisk sous Debian 11
« le: 14 août 2023 à 12:54:25 »
Salut à tous.

J'ai installé Asterisk dans mon Debian 11 version Bullseye. J'ai pris la dernière version d'Asterisk en date, la 20.4.0.

J'ai suivi ce didacticiel pour installer Asterisk. Ainsi que celui-ci qui est en deux parties.

a) à la place de "sudo ./configure", j'ai fait cela "sudo ./configure --with-jansson-bundled".

b) dans le service "asterisk", un message d'anomalie concernant "radius" a été résolu avec ces explications là.

A priori, aucune grand difficulté pour l'installation d'Asterisk version 20.4.0. Le service Asterisk est opérationnel et j'ai pu entrer dans la console Asterisk. Donc tout est ok !

Là où j'ai eu plus de difficulté, c'est de paramétrer Asterisk avec les identifiants de ma ligne téléphonique SFR. Il faut savoir que les différentes versions d'Asterisk ne sont pas toujours compatibles entre elles. Ce qui m'a fait perdre beaucoup de temps, j'ai cru que je devais, comme dans les exemples trouvés dans ce forum, travaillé sur les fichiers "sip.conf", "extensions.conf" et "users.conf". Que nenni ! Il faut maintenant travailler sur "pjsip.conf" et "extensions.conf". Après l'installation, on trouve ces fichiers dans le répertoire "/ext/asterisk".

Voici mon "pjsip.conf" :
Code: [Sélectionner]
[registration]
auth_rejection_permanent=yes

[transport-udp-nat]
bind=0.0.0.0
external_media_address=xxx.xxx.xxx.xxx     ; c'est mon adresse IPv4 WAN ou Publique.
external_signaling_address=xxx.xxx.xxx.xxx ; c'est mon adresse IPv4 WAN ou Publique.
local_net=192.168.1.0/24                   ; c'est l'adresse de mon réseau local.
protocol=udp
type=transport

; --------- ;
; Trunk SFR ;
; --------- ;

[sfr]
contact=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
type=aor

[sfr]
password=XXXXXXXXXXXXXXXX              ; Password dans l'application de NextGens
username=NDIXXXXXXXXXX.XXX.XXX@sfr.fr  ; Username dans l'application de NextGens
type=auth

[sfr]
allow=!all,opus,speex,g722,alaw,ulaw,gsm
aors=sfr
context=incoming
from_domain=ims.mnc010.mcc208.3gppnetwork.org
from_user=+33XXXXXXXXX                 ; Display Name dans l'application de NextGens
outbound_auth=sfr
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
transport=transport-udp-nat
type=endpoint

[sfr]
endpoint=sfr
match=residential.p-cscf.sfr.net
type=identify

[sfr]
client_uri=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org  ; Display Name @ Domain dans l'application de NextGens
outbound_auth=sfr
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
server_uri=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org  ; Display Name @ Domain dans l'application de NextGens
transport=transport-udp-nat
type=registration

; --------- ;
; Templates ;
; --------- ;

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

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

[endpoint_internal](!)
allow=!all,alaw
context=outgoing
from_domain=ims.mnc010.mcc208.3gppnetwork.org  ; Domain dans l'application de NextGens
language=fr
type=endpoint

; ---------- ;
; Phone Line ;
; ---------- ;

[artemus](aor_dynamic)

[artemus](auth_userpass)
password=artemus
username=artemus

[artemus](endpoint_internal)
auth=artemus
aors=artemusEt voici le fichier "extensions.conf" :
Code: [Sélectionner]
[general]
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=DAHDI/G2
TRUNKMSD=1

[outgoing]
exten => _0[12345679]XXXXXXXX,1,Dial(PJSIP/${EXTEN}@sfr)
exten => _+33[12345679]XXXXXXXX,1,Dial(PJSIP/${EXTEN}@sfr)
exten => _1023,1,Dial(PJSIP/${EXTEN}@sfr)

[incoming]
exten => s,1,Dial(PJSIP/artemus)Ces deux fichiers de configurations sont opérationnels, mais ils ne sont pas définitifs. Je teste sur le SoftPhone Zoiper5 au travers du numéro de téléphone du Service Client SFR : 1023.

Après chaque modification des fichiers, il faut relancer le service Asterisk :
Code: [Sélectionner]
systemctl restart asterisk.service
systemctl status  asterisk.servicePour configurer Zoiper5, j'ai mis :
Première page :
--> artemus (c'est l'identifiant de mon poste téléphone)
--> artemus (c'est mon mot de passe)
Deuxième page
--> 192.168.1.11 (c'est l'adresse IPv4 dans mon réseau LAN où se trouve le serveur Asterisk).

Si tout ce passe bien, vous obtiendrez "UDP SIP : succès en vert".

Je vais dans la console asterisk :
Code: [Sélectionner]
asterisk -rvvvPuis sous Zoiper5, je tape le numéro 1023, celui du service client SFR. Voici le compte-rendu :
Code: [Sélectionner]
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 20.4.0 currently running on Debian (pid = 38311)
Debian*CLI>
    -- Executing [1023@outgoing:1] Dial("PJSIP/artemus-00000000", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
       > 0x7f3338052b70 -- Strict RTP learning after remote address set to: 92.91.230.104:5892
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/artemus-00000000
       > 0x7f333801bda0 -- Strict RTP learning after remote address set to: 192.168.1.11:40829
       > 0x7f333801bda0 -- Strict RTP switching to RTP target address 192.168.1.11:40829 as source
       > 0x7f3338052b70 -- Strict RTP switching to RTP target address 92.91.230.104:5892 as source
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/artemus-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/artemus-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/artemus-00000000
       > 0x7f3338052b70 -- Strict RTP learning complete - Locking on source address 92.91.230.104:5892
       > 0x7f333801bda0 -- Strict RTP learning complete - Locking on source address 192.168.1.11:40829
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/artemus-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/artemus-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/artemus-00000000
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/artemus-00000000' status is 'CHANUNAVAIL'
Debian*CLI> Pour ce test, mon Asus sous Debian est relié à la Box SFR. Je me retrouve déconnecté au bout des 32 secondes comme pour les tests précédents.
Nous avons une précision sur la nature de cette déconnexion : "CHANUNAVAIL".
J'en suis là pour l'instant. comme qui dirait, la suite au prochain épisode.

Cordialement.
Artemus24.
@+
« Modifié: 17 août 2023 à 17:24:12 par artemus24 »
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artemus24

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Utilisation d'Asterisk sous Debian 11
« Réponse #1 le: 16 août 2023 à 16:51:03 »
Comme j'ai modifié mes fichiers, je les remets :

pjsip.conf :

Code: [Sélectionner]
[transport-udp-nat]
bind=0.0.0.0
external_media_address=XXX.XXX.XXX.XXX
external_signaling_address=XXX.XXX.XXX.XXX
local_net=192.168.1.0/255.255.255.0
protocol=udp
type=transport

; --------- ;
; Templates ;
; --------- ;

[my_codecs](!)
disallow=all
allow=alaw
allow=ulaw
allow=gsm

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

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

[endpoint_internal](!,my_codecs)
context=outgoing
from_domain=ims.mnc010.mcc208.3gppnetwork.org
language=fr
type=endpoint

; --------- ;
; Trunk SFR ;
; --------- ;

[sfr]
contact=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
type=aor

[sfr]
auth_type=userpass
password=XXXXXXXXXXXXXXXX
;;realm=
username=NDIXXXXXXXXXX.XXX.XXX@sfr.fr
type=auth

[sfr](my_codecs)
100rel=required
aors=sfr
context=incoming
from_domain=ims.mnc010.mcc208.3gppnetwork.org
from_user=+33XXXXXXXXX
outbound_auth=sfr
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
transport=transport-udp-nat
type=endpoint

[sfr]
endpoint=sfr
match=residential.p-cscf.sfr.net
type=identify

[sfr]
client_uri=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
outbound_auth=sfr
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
server_uri=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
transport=transport-udp-nat
type=registration

; ------------------- ;
; Phone Line 'Zoiper' ;
; ------------------- ;

[zoiper](aor_dynamic)

[zoiper](auth_userpass)
password=zoiper
username=zoiper

[zoiper](endpoint_internal)
auth=zoiper
aors=zoiper
callerid=zoiper
extensions.conf :

Code: [Sélectionner]
[general]
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=DAHDI/G2
TRUNKMSD=1

[outgoing]
exten => _X.,1,Dial(PJSIP/${EXTEN}@sfr)
exten => _X.,n,Hangup()

[incoming]
exten => s,1,Dial(PJSIP/zoiper)
Ci-après une trace de la communication quand je fais le 1023 (Service Client SFR).
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artemus24

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Utilisation d'Asterisk sous Debian 11
« Réponse #2 le: 16 août 2023 à 16:53:26 »
Code: [Sélectionner]
<--- Received SIP request (1326 bytes) from UDP:192.168.1.11:45764 --->
INVITE sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45764;branch=z9hG4bK-524287-1---d7966bbe8cd328a8;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:45764;transport=UDP>
To: <sip:1023@192.168.1.11>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=09f12e30
Call-ID: gAgscy15debma_BfUc6IkA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 648

v=0
o=Z 0 21282916 IN IP4 192.168.1.11
s=Z
c=IN IP4 192.168.1.11
t=0 0
m=audio 55552 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux
m=audio 55552 RTP/AVPF 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir

<--- Transmitting SIP response (503 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---d7966bbe8cd328a8
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=z9hG4bK-524287-1---d7966bbe8cd328a8
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1692195895/bbd695058ecfe00e7ab5b81396661e4d",opaque="2ff404522100a4b3",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP request (351 bytes) from UDP:192.168.1.11:45764 --->
ACK sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45764;branch=z9hG4bK-524287-1---d7966bbe8cd328a8;rport
Max-Forwards: 70
To: <sip:1023@192.168.1.11>;tag=z9hG4bK-524287-1---d7966bbe8cd328a8
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=09f12e30
Call-ID: gAgscy15debma_BfUc6IkA..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1625 bytes) from UDP:192.168.1.11:45764 --->
INVITE sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45764;branch=z9hG4bK-524287-1---04203c58a61ade96;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:45764;transport=UDP>
To: <sip:1023@192.168.1.11>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=09f12e30
Call-ID: gAgscy15debma_BfUc6IkA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692195895/bbd695058ecfe00e7ab5b81396661e4d",uri="sip:1023@192.168.1.11;transport=UDP",response="9762d077578025d8447848e8f74b6156",cnonce="351f64212bb6fb16a306be607aef2657",nc=00000001,qop=auth,algorithm=MD5,opaque="2ff404522100a4b3"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 648

v=0
o=Z 0 21282916 IN IP4 192.168.1.11
s=Z
c=IN IP4 192.168.1.11
t=0 0
m=audio 55552 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux
m=audio 55552 RTP/AVPF 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir

<--- Transmitting SIP response (311 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Content-Length:  0


    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000012", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
<--- Transmitting SIP request (1069 bytes) to UDP:92.91.129.200:5062 --->
INVITE sip:1023@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>
Contact: <sip:+33XXXXXXXXX@XXX.XXX.XXX.XXX:5060>
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
CSeq: 23228 INVITE
Route: <sip:residential.p-cscf.sfr.net;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Require: 100rel
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Type: application/sdp
Content-Length:   280

v=0
o=- 383691559 383691559 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 17854 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (438 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 100 Trying
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Date: Wed, 16 Aug 2023 14:24:55 GMT
Content-Length: 0


<--- Received SIP response (942 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-110324@pcgw-0006.imsgroup-019.tng1asbc05.ims.sfr.net:5062;x-afi=105>
Content-Type: application/sdp
RSeq: 1
Content-Length: 235

v=0
o=LucentPCSF 1631970971 1631970971 IN IP4 imsgroup-019.tng1asbc05.ims.sfr.net
s=-
c=IN IP4 92.91.230.139
t=0 0
m=audio 35412 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -

    -- PJSIP/sfr-00000013 is making progress passing it to PJSIP/zoiper-00000012
<--- Transmitting SIP response (853 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Contact: <sip:192.168.1.11:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 0 21282918 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 10890 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=audio 0 RTP/AVPF 106 9 98 101 0 8 3

<--- Received SIP response (942 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-110324@pcgw-0006.imsgroup-019.tng1asbc05.ims.sfr.net:5062;x-afi=105>
Content-Type: application/sdp
RSeq: 1
Content-Length: 235

v=0
o=LucentPCSF 1631970971 1631970971 IN IP4 imsgroup-019.tng1asbc05.ims.sfr.net
s=-
c=IN IP4 92.91.230.139
t=0 0
m=audio 35412 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -

    -- PJSIP/sfr-00000013 is making progress passing it to PJSIP/zoiper-00000012
<--- Transmitting SIP response (853 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.11:5060>
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 0 21282918 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 10890 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=audio 0 RTP/AVPF 106 9 98 101 0 8 3

<--- Received SIP request (996 bytes) from UDP:192.168.1.11:45764 --->
REGISTER sip:192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45764;branch=z9hG4bK-524287-1---4f3e6e84ffcc40d2;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:45764;rinstance=f8e4393d973ab382;transport=UDP>
To: <sip:zoiper@192.168.1.11;transport=UDP>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=33da4d23
Call-ID: zH8FcrvsjHUcCYm3Sttw-Q..
CSeq: 257 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692195844/0ca0c31768bd2aa28614f20860933e24",uri="sip:192.168.1.11;transport=UDP",response="528a3430532bcbd39c7b9edc544a6dfe",cnonce="4e56f59af27e3276bf972f8c2dc36a9a",nc=00000002,qop=auth,algorithm=MD5,opaque="0b92ddfe333bcc7f"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---4f3e6e84ffcc40d2
Call-ID: zH8FcrvsjHUcCYm3Sttw-Q..
From: <sip:zoiper@192.168.1.11>;tag=33da4d23
To: <sip:zoiper@192.168.1.11>;tag=z9hG4bK-524287-1---4f3e6e84ffcc40d2
CSeq: 257 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1692195897/7a10e537cd5ed26aee104daea63071b1",opaque="37a560c9336e38b8",stale=true,algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP request (996 bytes) from UDP:192.168.1.11:45764 --->
REGISTER sip:192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45764;branch=z9hG4bK-524287-1---a1331db20e6fd74d;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:45764;rinstance=f8e4393d973ab382;transport=UDP>
To: <sip:zoiper@192.168.1.11;transport=UDP>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=33da4d23
Call-ID: zH8FcrvsjHUcCYm3Sttw-Q..
CSeq: 258 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692195897/7a10e537cd5ed26aee104daea63071b1",uri="sip:192.168.1.11;transport=UDP",response="3d5c2255078eed076b0c55931d63dfc1",cnonce="ff58e7a21bee38056d2a7a26faaab112",nc=00000001,qop=auth,algorithm=MD5,opaque="37a560c9336e38b8"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


<--- Transmitting SIP response (497 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---a1331db20e6fd74d
Call-ID: zH8FcrvsjHUcCYm3Sttw-Q..
From: <sip:zoiper@192.168.1.11>;tag=33da4d23
To: <sip:zoiper@192.168.1.11>;tag=z9hG4bK-524287-1---a1331db20e6fd74d
CSeq: 258 REGISTER
Date: Wed, 16 Aug 2023 14:24:57 GMT
Contact: <sip:zoiper@192.168.1.11:45764;transport=UDP;rinstance=f8e4393d973ab382>;expires=59
Expires: 60
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP response (942 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-110324@pcgw-0006.imsgroup-019.tng1asbc05.ims.sfr.net:5062;x-afi=105>
Content-Type: application/sdp
RSeq: 1
Content-Length: 235

v=0
o=LucentPCSF 1631970971 1631970971 IN IP4 imsgroup-019.tng1asbc05.ims.sfr.net
s=-
c=IN IP4 92.91.230.139
t=0 0
m=audio 35412 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -

    -- PJSIP/sfr-00000013 is making progress passing it to PJSIP/zoiper-00000012
IP archivée

artemus24

  • Abonné SFR fibre FttH
  • *
  • Messages: 782
  • Montignac Lascaux (24)
Utilisation d'Asterisk sous Debian 11
« Réponse #3 le: 16 août 2023 à 16:54:03 »
Comme cela dépasse les 20 000 caractères, voici la fin de ce debug :
Code: [Sélectionner]
<--- Transmitting SIP response (853 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.11:5060>
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 0 21282918 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 10890 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=audio 0 RTP/AVPF 106 9 98 101 0 8 3

<--- Received SIP response (942 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-110324@pcgw-0006.imsgroup-019.tng1asbc05.ims.sfr.net:5062;x-afi=105>
Content-Type: application/sdp
RSeq: 1
Content-Length: 235

v=0
o=LucentPCSF 1631970971 1631970971 IN IP4 imsgroup-019.tng1asbc05.ims.sfr.net
s=-
c=IN IP4 92.91.230.139
t=0 0
m=audio 35412 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -

    -- PJSIP/sfr-00000013 is making progress passing it to PJSIP/zoiper-00000012
<--- Transmitting SIP response (853 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.11:5060>
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 0 21282918 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 10890 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=audio 0 RTP/AVPF 106 9 98 101 0 8 3

<--- Received SIP response (942 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-110324@pcgw-0006.imsgroup-019.tng1asbc05.ims.sfr.net:5062;x-afi=105>
Content-Type: application/sdp
RSeq: 1
Content-Length: 235

v=0
o=LucentPCSF 1631970971 1631970971 IN IP4 imsgroup-019.tng1asbc05.ims.sfr.net
s=-
c=IN IP4 92.91.230.139
t=0 0
m=audio 35412 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -

    -- PJSIP/sfr-00000013 is making progress passing it to PJSIP/zoiper-00000012
<--- Transmitting SIP response (853 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.11:5060>
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 0 21282918 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 10890 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=audio 0 RTP/AVPF 106 9 98 101 0 8 3

<--- Received SIP response (942 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-110324@pcgw-0006.imsgroup-019.tng1asbc05.ims.sfr.net:5062;x-afi=105>
Content-Type: application/sdp
RSeq: 1
Content-Length: 235

v=0
o=LucentPCSF 1631970971 1631970971 IN IP4 imsgroup-019.tng1asbc05.ims.sfr.net
s=-
c=IN IP4 92.91.230.139
t=0 0
m=audio 35412 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -

    -- PJSIP/sfr-00000013 is making progress passing it to PJSIP/zoiper-00000012
<--- Transmitting SIP response (853 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.11:5060>
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 0 21282918 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 10890 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=audio 0 RTP/AVPF 106 9 98 101 0 8 3

<--- Received SIP response (942 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 183 Session Progress
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Require: 100rel
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-110324@pcgw-0006.imsgroup-019.tng1asbc05.ims.sfr.net:5062;x-afi=105>
Content-Type: application/sdp
RSeq: 1
Content-Length: 235

v=0
o=LucentPCSF 1631970971 1631970971 IN IP4 imsgroup-019.tng1asbc05.ims.sfr.net
s=-
c=IN IP4 92.91.230.139
t=0 0
m=audio 35412 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=ptime:20
a=silenceSupp:off - - - -

    -- PJSIP/sfr-00000013 is making progress passing it to PJSIP/zoiper-00000012
<--- Transmitting SIP response (853 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:192.168.1.11:5060>
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 0 21282918 IN IP4 192.168.1.11
s=Asterisk
c=IN IP4 192.168.1.11
t=0 0
m=audio 10890 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=audio 0 RTP/AVPF 106 9 98 101 0 8 3

<--- Received SIP request (996 bytes) from UDP:192.168.1.11:45764 --->
REGISTER sip:192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45764;branch=z9hG4bK-524287-1---daa603d1c87afe64;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:45764;rinstance=f8e4393d973ab382;transport=UDP>
To: <sip:zoiper@192.168.1.11;transport=UDP>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=33da4d23
Call-ID: zH8FcrvsjHUcCYm3Sttw-Q..
CSeq: 259 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692195897/7a10e537cd5ed26aee104daea63071b1",uri="sip:192.168.1.11;transport=UDP",response="60f88bf2676c226bf9060fe16caedb96",cnonce="74c7ab030d1b219f1954d0225e631369",nc=00000002,qop=auth,algorithm=MD5,opaque="37a560c9336e38b8"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


<--- Transmitting SIP response (520 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---daa603d1c87afe64
Call-ID: zH8FcrvsjHUcCYm3Sttw-Q..
From: <sip:zoiper@192.168.1.11>;tag=33da4d23
To: <sip:zoiper@192.168.1.11>;tag=z9hG4bK-524287-1---daa603d1c87afe64
CSeq: 259 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1692195950/6b3cf4806e6db0313a5b8090d332b0db",opaque="06af3be96e764dc7",stale=true,algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP request (996 bytes) from UDP:192.168.1.11:45764 --->
REGISTER sip:192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45764;branch=z9hG4bK-524287-1---f0a5c617e4e39de6;rport
Max-Forwards: 70
Contact: <sip:zoiper@192.168.1.11:45764;rinstance=f8e4393d973ab382;transport=UDP>
To: <sip:zoiper@192.168.1.11;transport=UDP>
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=33da4d23
Call-ID: zH8FcrvsjHUcCYm3Sttw-Q..
CSeq: 260 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="zoiper",realm="asterisk",nonce="1692195950/6b3cf4806e6db0313a5b8090d332b0db",uri="sip:192.168.1.11;transport=UDP",response="91465edc3410cf256308ff8c5748fe90",cnonce="b949d3427ee6322476338aaa28b1d990",nc=00000001,qop=auth,algorithm=MD5,opaque="06af3be96e764dc7"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


<--- Transmitting SIP response (497 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---f0a5c617e4e39de6
Call-ID: zH8FcrvsjHUcCYm3Sttw-Q..
From: <sip:zoiper@192.168.1.11>;tag=33da4d23
To: <sip:zoiper@192.168.1.11>;tag=z9hG4bK-524287-1---f0a5c617e4e39de6
CSeq: 260 REGISTER
Date: Wed, 16 Aug 2023 14:25:50 GMT
Contact: <sip:zoiper@192.168.1.11:45764;transport=UDP;rinstance=f8e4393d973ab382>;expires=59
Expires: 60
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP response (470 bytes) from UDP:92.91.129.200:5062 --->
SIP/2.0 500 Server Internal Error
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce;rport=5060
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org;user=phone>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
CSeq: 23228 INVITE
Content-Length: 0


<--- Transmitting SIP request (536 bytes) to UDP:92.91.129.200:5062 --->
ACK sip:1023@ims.mnc010.mcc208.3gppnetwork.org SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj857447e6-2a77-4c42-9111-4983a01df4ce
From: <sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org>;tag=d27f1e12-b985-4872-9327-ab9493c4a35d
To: <sip:1023@ims.mnc010.mcc208.3gppnetwork.org>;tag=64a4ab95-64dcdc3711acc271-gm-po-lucentPCSF-057234
Call-ID: 81d21fe3-eec7-4eaf-870c-ed8709c94c69
CSeq: 23228 ACK
Route: <sip:residential.p-cscf.sfr.net;lr>
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000012", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000012'
<--- Transmitting SIP response (502 bytes) to UDP:192.168.1.11:45764 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.1.11:45764;rport=45764;received=192.168.1.11;branch=z9hG4bK-524287-1---04203c58a61ade96
Call-ID: gAgscy15debma_BfUc6IkA..
From: <sip:zoiper@192.168.1.11>;tag=09f12e30
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
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 (352 bytes) from UDP:192.168.1.11:45764 --->
ACK sip:1023@192.168.1.11;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:45764;branch=z9hG4bK-524287-1---04203c58a61ade96;rport
Max-Forwards: 70
To: <sip:1023@192.168.1.11>;tag=62bd4cd3-00e9-4600-8abf-5973761bfe96
From: <sip:zoiper@192.168.1.11;transport=UDP>;tag=09f12e30
Call-ID: gAgscy15debma_BfUc6IkA..
CSeq: 2 ACK
Content-Length: 0


Debian*CLI>
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artemus24

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Utilisation d'Asterisk sous Debian 11
« Réponse #4 le: 16 août 2023 à 17:13:03 »
Et la même sortie dans la console Aserisk sans le trace du debut :
Code: [Sélectionner]
Asterisk 20.4.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 20.4.0 currently running on Debian (pid = 116941)
  == Contact zoiper/sip:zoiper@192.168.1.11:45764;transport=UDP;rinstance=f8e4393d973ab382 has been deleted
  == Endpoint zoiper is now Unreachable
    -- Added contact 'sip:zoiper@192.168.1.11:45764;transport=UDP;rinstance=07cff2a5eb20ee96' to AOR 'zoiper' with expiration of 60 seconds
  == Endpoint zoiper is now Reachable
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000000", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
[Aug 16 17:03:10] WARNING[116985]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'sfr': No auth objects matching realm(s) '' from challenge found.
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000000", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000000'
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000002", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
[Aug 16 17:03:14] WARNING[116985]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'sfr': No auth objects matching realm(s) '' from challenge found.
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000002", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000002'
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000004", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
[Aug 16 17:03:17] WARNING[116985]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'sfr': No auth objects matching realm(s) '' from challenge found.
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000004", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000004'
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000006", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
[Aug 16 17:03:21] WARNING[116985]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'sfr': No auth objects matching realm(s) '' from challenge found.
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000006", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000006'
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000008", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
    -- PJSIP/sfr-00000009 is making progress passing it to PJSIP/zoiper-00000008
    -- PJSIP/sfr-00000009 is making progress passing it to PJSIP/zoiper-00000008
    -- PJSIP/sfr-00000009 is making progress passing it to PJSIP/zoiper-00000008
    -- PJSIP/sfr-00000009 is making progress passing it to PJSIP/zoiper-00000008
    -- PJSIP/sfr-00000009 is making progress passing it to PJSIP/zoiper-00000008
    -- PJSIP/sfr-00000009 is making progress passing it to PJSIP/zoiper-00000008
    -- PJSIP/sfr-00000009 is making progress passing it to PJSIP/zoiper-00000008
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000008", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000008'
Debian*CLI> Ce n'est qu'à la cinquième tentative que j'ai pu obtenir la communication téléphonique au 1023.

Je ne comprends pas cette anomalie :
Code: [Sélectionner]
Aug 16 17:03:17] WARNING[116985]: res_pjsip_outbound_authenticator_digest.c:507 digest_create_request_with_auth: Endpoint: 'sfr': No auth objects matching realm(s) '' from challenge found.Il y a bien une option "realm" pour le type=auth du trunk SFR. J'ai essayé plusieurs combinaisons, sans résoudre le problèmes.
A vrai dire, je ne sais pas ce que je dois mettre pour éviter cet avertissement qui bloque la communication.

Comme on peut le voir, il y a eu sept "making progress passing" et sur le dernière, on atteint les 32 secondes et provoque l'arrêt de la communication.
« Modifié: 18 août 2023 à 11:48:30 par artemus24 »
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artemus24

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Utilisation d'Asterisk sous Debian 11
« Réponse #5 le: 17 août 2023 à 12:42:07 »
Voici deux communications consécutives réussies :
Code: [Sélectionner]
Asterisk 20.4.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 20.4.0 currently running on Debian (pid = 13170)
Debian*CLI> pjsip show registrations

 <Registration/ServerURI..............................>  <Auth....................>  <Status.......>
==========================================================================================

 sfr/sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org  sfr                         Registered        (exp. 3586s)

Objects found: 1

    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000000", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/zoiper-00000000
    -- PJSIP/sfr-00000001 requested media update control 26, passing it to PJSIP/zoiper-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/zoiper-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/zoiper-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/zoiper-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/zoiper-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/zoiper-00000000
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/zoiper-00000000
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000000", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000000'
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000002", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
    -- PJSIP/sfr-00000003 is making progress passing it to PJSIP/zoiper-00000002
    -- PJSIP/sfr-00000003 requested media update control 26, passing it to PJSIP/zoiper-00000002
    -- PJSIP/sfr-00000003 is making progress passing it to PJSIP/zoiper-00000002
    -- PJSIP/sfr-00000003 is making progress passing it to PJSIP/zoiper-00000002
    -- PJSIP/sfr-00000003 is making progress passing it to PJSIP/zoiper-00000002
    -- PJSIP/sfr-00000003 is making progress passing it to PJSIP/zoiper-00000002
    -- PJSIP/sfr-00000003 is making progress passing it to PJSIP/zoiper-00000002
    -- PJSIP/sfr-00000003 is making progress passing it to PJSIP/zoiper-00000002
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1023@outgoing:2] Hangup("PJSIP/zoiper-00000002", "") in new stack
  == Spawn extension (outgoing, 1023, 2) exited non-zero on 'PJSIP/zoiper-00000002'
Debian*CLI> Comme on peut le voir, il n'y a plus ce message d'avertissement mais je n'ai pas résolu le problème.
Il reste juste à résoudre le problème de la coupure de la communication au bout des 32 secondes.

Je n'ai pas trouvé grand chose dans le répertoire /var/log/asterisk/cdr-custom/Master.csv :
Code: [Sélectionner]
""""" <zoiper>","zoiper","1023","outgoing","PJSIP/zoiper-00000000","PJSIP/sfr-00000001","Dial","PJSIP/1023@sfr","2023-08-17 12:30:59","","2023-08-17 12:32:04","64","0","NO ANSWER","DOCUMENTATION","","1692268259.0","",0
""""" <zoiper>","zoiper","1023","outgoing","PJSIP/zoiper-00000002","PJSIP/sfr-00000003","Dial","PJSIP/1023@sfr","2023-08-17 12:32:12","","2023-08-17 12:33:17","64","0","NO ANSWER","DOCUMENTATION","","1692268332.3","",3Pour ce qui est des LOG, rien d'intéressant. Le mieux est encore d'utiliser la console Asterisk.
« Modifié: 18 août 2023 à 11:51:19 par artemus24 »
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artemus24

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Utilisation d'Asterisk sous Debian 11
« Réponse #6 le: 17 août 2023 à 17:20:11 »
J'ai résolu le problème des 32 secondes. :)

Dans le Trunc SFR, à la section "type=endpoint", il faut ajouter "rewrite_contact=yes" car par défaut, il est à "no".
J'ai fait le test de vérification à partir de mon Debian 11 derrière l'ONT de SFR et ça fonctionne !!!

Voici la solution définitive concernant le paramétrage de la configuration de "pjsip.conf".
Code: [Sélectionner]
[transport-udp-nat]
bind=0.0.0.0
external_media_address=XXX.XXX.XXX.XXX         ; c'est l'adresse IP WAN ou publique
external_signaling_address=XXX.XXX.XXX.XXX     ; c'est l'adresse IP WAN ou publique
local_net=XXX.XXX.XXX.XXX/255.255.255.255      ; c'est l'adresse IP WAN ou publique
protocol=udp
type=transport

; --------- ;
; Templates ;
; --------- ;

[my_codecs](!)
disallow=all
allow=alaw
allow=ulaw
allow=gsm

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

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

[endpoint_internal](!,my_codecs)
context=outgoing
from_domain=ims.mnc010.mcc208.3gppnetwork.org
language=fr
type=endpoint

; --------- ;
; Trunk SFR ;
; --------- ;

[sfr]
contact=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
type=aor

[sfr]
auth_type=userpass
password=XXXXXXXXXXXXXXXX
realm=                                                   ; pour résoudre le message d'anomalie qui me dérangeait
username=NDIXXXXXXXXXX.XXX.XXX@sfr.fr
type=auth

[sfr](my_codecs)
aors=sfr
context=incoming
from_domain=ims.mnc010.mcc208.3gppnetwork.org
from_user=+33XXXXXXXXX
outbound_auth=sfr
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
rewrite_contact=yes                                      ; pour ne plus avoir les messages "making progress passing", et la coupure au bout des 32 secondes.
transport=transport-udp-nat
type=endpoint

[sfr]
endpoint=sfr
match=residential.p-cscf.sfr.net
type=identify

[sfr]
client_uri=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
outbound_auth=sfr
outbound_proxy=sip:residential.p-cscf.sfr.net\;lr
server_uri=sip:+33XXXXXXXXX@ims.mnc010.mcc208.3gppnetwork.org
transport=transport-udp-nat
type=registration

; ------------------- ;
; Phone Line 'Zoiper' ;
; ------------------- ;

[zoiper](aor_dynamic)

[zoiper](auth_userpass)
password=zoiper
username=zoiper

[zoiper](endpoint_internal)
auth=zoiper
aors=zoiper
callerid=zoiperJe ne redonne pas le fichier "extension.conf" car il n'a pas changé.

Et voici un exemple du compte-rendu d'une communication réussie sans coupure au bout des 32 secondes :
Code: [Sélectionner]
Asterisk 20.4.0, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 20.4.0 currently running on Debian (pid = 31183)
Debian*CLI>
Debian*CLI>
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000008", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
    -- PJSIP/sfr-00000009 is making progress passing it to PJSIP/zoiper-00000008
    -- PJSIP/sfr-00000009 is ringing
[Aug 18 11:54:59] WARNING[31316][C-00000005]: channel.c:5765 set_format: Unable to find a codec translation path: (slin) -> (alaw)
[Aug 18 11:54:59] WARNING[31316][C-00000005]: indications.c:140 playtones_alloc: Unable to set 'PJSIP/zoiper-00000008' to signed linear format (write)
[Aug 18 11:54:59] WARNING[31316][C-00000005]: channel.c:4682 indicate_data_internal: Unable to handle indication 3 for 'PJSIP/zoiper-00000008'
    -- PJSIP/sfr-00000009 answered PJSIP/zoiper-00000008
    -- Channel PJSIP/sfr-00000009 joined 'simple_bridge' basic-bridge <8b25e4e8-6d86-4c95-8768-d82396fb8bc4>
    -- Channel PJSIP/zoiper-00000008 joined 'simple_bridge' basic-bridge <8b25e4e8-6d86-4c95-8768-d82396fb8bc4>
    -- Channel PJSIP/sfr-00000009 left 'native_rtp' basic-bridge <8b25e4e8-6d86-4c95-8768-d82396fb8bc4>
    -- Channel PJSIP/zoiper-00000008 left 'native_rtp' basic-bridge <8b25e4e8-6d86-4c95-8768-d82396fb8bc4>
  == Spawn extension (outgoing, 1023, 1) exited non-zero on 'PJSIP/zoiper-00000008'
Debian*CLI> Comme on peut le voir, les "making progress passing" ont été réduits à 1.
Maintenant, nous avons un "ringing" qui confirme que la communication a été reconnue par SFR.
Je suis allé au bout de la communication via le répondeur du Service Client SFR, soit un peu plus de 2 minutes.
Les trois Warning ne sont pas bloquants, sauf que je ne sais pas les résoudre.
Problème résolu en ce qui concerne les 32 secondes. :D

En l'état, cette configuration vous permet de passer des communications téléphoniques  depuis votre SoftPhone.
Ma configuration demande encore quelques ajustements et complémentarités en ce qui concerne le DialPlan, et l'IPv6.

Cordialement.
Artemus24.
@+
« Modifié: 18 août 2023 à 12:08:50 par artemus24 »
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rooot

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Utilisation d'Asterisk sous Debian 11
« Réponse #7 le: 18 août 2023 à 13:35:47 »
Citation de: artemus24 le 17 août 2023 à 17:20:11
Les trois Warning ne sont pas bloquants, sauf que je ne sais pas les résoudre.
c'est pas l'ordre dans lequel tu as mis tes codecs ?
Sur la box SFR l'ordre de préférence est celui-ci
Citer
<codec enable="true" id="1">G711_alaw</codec>
<codec enable="true" id="2">G711_mulaw</codec>
<codec enable="true" id="3">G729_ab</codec>
<codec enable="false" id="4">G726</codec>
<codec enable="false" id="5">G722</codec>
sur les péripheriques et asterisk, vérifie que tu as le meme ordre partout. Sinon je pense que ca va essayer des codecs qui ne conviennent peut etre pas et générer le warning.
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artemus24

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[Résolu]Utilisation d'Asterisk sous Debian 11
« Réponse #8 le: 18 août 2023 à 13:58:53 »
Dans Zoiper, j'ai mis dans cet ordre :
--> G.711 a-law
--> G.711 mu-law
--> GSM FR

Dans "sudo make menuselect", à l'installation, j'ai installé tous les sons français :
--> alaw
--> gsm
--> g729
--> g722
--> sln16
--> siren7
--> siren14

Dans Asterisk, j'ai mis dans cet ordre :
--> alaw
--> ulaw
--> gsm

Les codecs concernent le premier Warning. Mais les deux autres, je ne comprends pas.
Code: [Sélectionner]
Running as user 'asterisk'
Running under group 'asterisk'
Connected to Asterisk 20.4.0 currently running on Debian (pid = 3094)
    -- Executing [1023@outgoing:1] Dial("PJSIP/zoiper-00000000", "PJSIP/1023@sfr") in new stack
    -- Called PJSIP/1023@sfr
    -- PJSIP/sfr-00000001 is making progress passing it to PJSIP/zoiper-00000000
    -- PJSIP/sfr-00000001 is ringing
[Aug 18 14:26:40] WARNING[3219][C-00000001]: channel.c:5765 set_format: Unable to find a codec translation path: (slin) -> (alaw)
[Aug 18 14:26:40] WARNING[3219][C-00000001]: indications.c:140 playtones_alloc: Unable to set 'PJSIP/zoiper-00000000' to signed linear format (write)
[Aug 18 14:26:40] WARNING[3219][C-00000001]: channel.c:4682 indicate_data_internal: Unable to handle indication 3 for 'PJSIP/zoiper-00000000'
    -- PJSIP/sfr-00000001 answered PJSIP/zoiper-00000000
    -- Channel PJSIP/sfr-00000001 joined 'simple_bridge' basic-bridge <b172428c-b260-4bc5-97f4-3312a68d4eb2>
    -- Channel PJSIP/zoiper-00000000 joined 'simple_bridge' basic-bridge <b172428c-b260-4bc5-97f4-3312a68d4eb2>
    -- Channel PJSIP/zoiper-00000000 left 'native_rtp' basic-bridge <b172428c-b260-4bc5-97f4-3312a68d4eb2>
    -- Channel PJSIP/sfr-00000001 left 'native_rtp' basic-bridge <b172428c-b260-4bc5-97f4-3312a68d4eb2>
  == Spawn extension (outgoing, 1023, 1) exited non-zero on 'PJSIP/zoiper-00000000'
Debian*CLI> D'où sort ce SLIN ? De SFR, je suppose.
Mais je ne l'ai pas dans "sudo make menuselect", ni dans Asterisk et encore moins dans Zoiper.
A moins que "sln16" soit le "slin".

Edit: je n'ai pas de "sln16" ou de "slin" coté Zoiper.
Si je mets "sln16" dans Asterisk, je suis déconnecté de SFR.
« Modifié: 18 août 2023 à 14:36:28 par artemus24 »
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rooot

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[Résolu]Utilisation d'Asterisk sous Debian 11
« Réponse #9 le: 18 août 2023 à 16:59:09 »
A mon avis les 2 warnings suivants sont la conséquence du premier, mais cela reste que des warnings, c'est peut etre normal...en tout cas ça n'a pas d'incidence sur le fonctionnement final, non ?

EDIT:
une piste en tout cas une explication ici : https://community.freepbx.org/t/transcoding-from-slin-to-ulaw-and-back-why/58643
Citer
By default, Asterisk uses SLIN as internal codec so everything is transcoded to SLIN. Supposedly, you could disable that behaviour with parameter transcode_via_sln=no
à mettre dans asterisk.conf
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artemus24

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[Résolu]Utilisation d'Asterisk sous Debian 11
« Réponse #10 le: 18 août 2023 à 20:48:59 »
Je pense que "transcode_via_sln=no" ne correspond pas à ma version Asterisk 20.4.0.
Je l'ai mis dans le fichier "/etc/asterisk/asterisk.conf" et je l'ai testé. Ça ne fonctionne pas.
Merci de m'aider. :)

Pour les trois warnings de la codification slin -> alaw, cela ne pose aucun problème puisque la communication prend le codec "g.711 a-law".

Je viens de faire un test d'une communication entrante avec le mobile de mon voisin.
Ça ne fonctionne pas sous Asterisk alors qu'avec MicroSIP configuré, ça fonctionne.
Au mobile, SFR m'indique que mon numéro de téléphone n'existe pas.

Je soupçonne que le problème vient de ce Warning : "Endpoint: 'sfr': No auth objects matching realm(s) '' from challenge found.".
Je continue de chercher mais je ne trouve rien comme résolution de ce problème.

Il est difficile de s'y retrouver entre les différentes versions d'Asterisk et ce warning similaire au lien mais avec un contexte différent.
L'idée tourne autour du "domain" qu'il ne faudrait pas mettre ou de mettre la même chose entre "from_user" (endpoint) et "username" (auth) du Trunk SFR.
J'ai testé mas cela ne fonctionne pas.
C'est une anomalie qui ne se produit pas tout le temps et il est difficile d'en connaitre la cause.
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rooot

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[Résolu]Utilisation d'Asterisk sous Debian 11
« Réponse #11 le: 18 août 2023 à 21:02:51 »
je crois qu'a un moment tu combines des parametres et il ne faut pas. genre +33xxxxxxxxxx@domain
le +33xxxxxx doit aller quelque part et le domain est a renseigner. ensuite c'est asterisk qui se charge de combiner lorsque c'est necessaire je pense.
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