Bonjour à tous !
J’ai un soucis d’accès à la boite vocale lorsque le combiné d’une base Gigaset est éteint.
J’utilise Wazo en Version : 18.03 avec un trunk OVH.
J’ai des bases Gigaset C530IP et N510IP.
Même si le (ou tous les) combiné(s) sont éteint(s) la base, elle, est toujours enregistrée dans Wazo
Name/username Host Dyn Forcerport Comedia ACL Port Status Description 25dirjyv/25dirjyv 92.184.x.x D Yes Yes 5360 OK (95 ms)
Lors d’un appel de l’extérieur (de mon portable), je me fais directement raccroché au nez. Je suppose que c’est le CANCEL à 20:57:02
[Apr 23 20:56:58] – Executing [s @ user:39] Dial(“SIP/trunk-000002eb”, “SIP/25dirjyv,30,”) in new stack
[Apr 23 20:56:58] == Using SIP RTP TOS bits 184
[Apr 23 20:56:58] == Using SIP RTP CoS mark 6
[Apr 23 20:56:58] Audio is at 10474
[Apr 23 20:56:58] Adding codec alaw to SDP
[Apr 23 20:56:58] Adding codec g722 to SDP
[Apr 23 20:56:58] Adding codec ulaw to SDP
[Apr 23 20:56:58] Adding non-codec 0x1 (telephone-event) to SDP
[Apr 23 20:56:58] Reliably Transmitting (NAT) to 92.184.x.x:5360:
[Apr 23 20:56:58] INVITE sip:25dirjyv @ 172.19.1.100:5360 SIP/2.0
[Apr 23 20:56:58] Via: SIP/2.0/UDP 172.16.0.5:5060;branch=z9hG4bK02841e9c;rport
[Apr 23 20:56:58] Max-Forwards: 70
[Apr 23 20:56:58] From: “+336xxxxxxxx” <sip:06xxxxxxxx @ 172.16.0.5>;tag=as6fb0b697
[Apr 23 20:56:58] To: <sip:25dirjyv @ 172.19.1.100:5360>
[Apr 23 20:56:58] Contact: <sip:06xxxxxxxx @ 172.16.0.5:5060>
[Apr 23 20:56:58] Call-ID: 42892ff042e7c2707328d78e17108b32 @ 172.16.0.5:5060
[Apr 23 20:56:58] CSeq: 102 INVITE
[Apr 23 20:56:58] User-Agent: Wazo PBX
[Apr 23 20:56:58] Date: Fri, 23 Apr 2021 18:56:58 GMT
[Apr 23 20:56:58] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Apr 23 20:56:58] Supported: replaces, timer
[Apr 23 20:56:58] Content-Type: application/sdp
[Apr 23 20:56:58] Content-Length: 293
[Apr 23 20:56:58]
[Apr 23 20:56:58] v=0
[Apr 23 20:56:58] o=root 452654938 452654938 IN IP4 172.16.0.5
[Apr 23 20:56:58] s=Asterisk PBX 15.2.2
[Apr 23 20:56:58] c=IN IP4 172.16.0.5
[Apr 23 20:56:58] t=0 0
[Apr 23 20:56:58] m=audio 10474 RTP/AVP 8 9 0 101
[Apr 23 20:56:58] a=rtpmap:8 PCMA/8000
[Apr 23 20:56:58] a=rtpmap:9 G722/8000
[Apr 23 20:56:58] a=rtpmap:0 PCMU/8000
[Apr 23 20:56:58] a=rtpmap:101 telephone-event/8000
[Apr 23 20:56:58] a=fmtp:101 0-16
[Apr 23 20:56:58] a=ptime:20
[Apr 23 20:56:58] a=maxptime:150
[Apr 23 20:56:58] a=sendrecv
[Apr 23 20:56:58]
[Apr 23 20:56:58] —
[Apr 23 20:56:58] – Called SIP/25dirjyv
[Apr 23 20:56:58]
[Apr 23 20:56:58] <— SIP read from UDP:92.184.x.x:5360 —>
[Apr 23 20:56:58] SIP/2.0 100 Trying
[Apr 23 20:56:58] Via: SIP/2.0/UDP 172.16.0.5:5060;branch=z9hG4bK02841e9c;rport=5060
[Apr 23 20:56:58] From: “+336xxxxxxxx” <sip:06xxxxxxxx @ 172.16.0.5:5060>;tag=as6fb0b697
[Apr 23 20:56:58] To: <sip:25dirjyv @ 172.19.1.100:5360>;tag=198435648
[Apr 23 20:56:58] Call-ID: 42892ff042e7c2707328d78e17108b32 @ 172.16.0.5:5060
[Apr 23 20:56:58] CSeq: 102 INVITE
[Apr 23 20:56:58] Contact: <sip:25dirjyv @ 172.19.1.100:5360>
[Apr 23 20:56:58] User-Agent: N510 IP PRO/42.258.00.000.000
[Apr 23 20:56:58] Content-Length: 0
[Apr 23 20:56:58]
[Apr 23 20:56:58] <------------->
[Apr 23 20:56:58] — (9 headers 0 lines) —
[Apr 23 20:57:02] Scheduling destruction of SIP dialog ‘42892ff042e7c2707328d78e17108b32 @ 172.16.0.5:5060’ in 32000 ms (Method: INVITE)
[Apr 23 20:57:02] Reliably Transmitting (NAT) to 92.184.x.x:5360:
[Apr 23 20:57:02] CANCEL sip:25dirjyv @ 172.19.1.100:5360 SIP/2.0
[Apr 23 20:57:02] Via: SIP/2.0/UDP 172.16.0.5:5060;branch=z9hG4bK02841e9c;rport
[Apr 23 20:57:02] Max-Forwards: 70
[Apr 23 20:57:02] From: “+336xxxxxxxx” <sip:06xxxxxxxx @ 172.16.0.5>;tag=as6fb0b697
[Apr 23 20:57:02] To: <sip:25dirjyv @ 172.19.1.100:5360>
[Apr 23 20:57:02] Call-ID: 42892ff042e7c2707328d78e17108b32 @ 172.16.0.5:5060
[Apr 23 20:57:02] CSeq: 102 CANCEL
[Apr 23 20:57:02] User-Agent: Wazo PBX
[Apr 23 20:57:02] Content-Length: 0
Alors que si j’appel d’un poste interne je tombe bien sur la boite vocale, grâce au 480 à 21:06:09
[Apr 23 21:06:00] – Executing [s @ user:39] Dial(“SIP/labbjuli-000002ed”, “SIP/25dirjyv,30,”) in new stack
[Apr 23 21:06:00] == Using SIP RTP TOS bits 184
[Apr 23 21:06:00] == Using SIP RTP CoS mark 6
[Apr 23 21:06:00] Audio is at 15066
[Apr 23 21:06:00] Adding codec g722 to SDP
[Apr 23 21:06:00] Adding codec alaw to SDP
[Apr 23 21:06:00] Adding codec ulaw to SDP
[Apr 23 21:06:00] Adding non-codec 0x1 (telephone-event) to SDP
[Apr 23 21:06:00] Reliably Transmitting (NAT) to 92.184.x.x:5360:
[Apr 23 21:06:00] INVITE sip:25dirjyv @ 172.19.1.100:5360 SIP/2.0
[Apr 23 21:06:00] Via: SIP/2.0/UDP 172.16.0.5:5060;branch=z9hG4bK48116c90;rport
[Apr 23 21:06:00] Max-Forwards: 70
[Apr 23 21:06:00] From: “toto” <sip:04xxxxxxxx @ 172.16.0.5>;tag=as3fa4617c
[Apr 23 21:06:00] To: <sip:25dirjyv @ 172.19.1.100:5360>
[Apr 23 21:06:00] Contact: <sip:04xxxxxxxx @ 172.16.0.5:5060>
[Apr 23 21:06:00] Call-ID: 62f7201f28fb72451c84373641efe911 @ 172.16.0.5:5060
[Apr 23 21:06:00] CSeq: 102 INVITE
[Apr 23 21:06:00] User-Agent: Wazo PBX
[Apr 23 21:06:00] Date: Fri, 23 Apr 2021 19:06:00 GMT
[Apr 23 21:06:00] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Apr 23 21:06:00] Supported: replaces, timer
[Apr 23 21:06:00] Content-Type: application/sdp
[Apr 23 21:06:00] Content-Length: 293
[Apr 23 21:06:00]
[Apr 23 21:06:00] v=0
[Apr 23 21:06:00] o=root 211750862 211750862 IN IP4 172.16.0.5
[Apr 23 21:06:00] s=Asterisk PBX 15.2.2
[Apr 23 21:06:00] c=IN IP4 172.16.0.5
[Apr 23 21:06:00] t=0 0
[Apr 23 21:06:00] m=audio 15066 RTP/AVP 9 8 0 101
[Apr 23 21:06:00] a=rtpmap:9 G722/8000
[Apr 23 21:06:00] a=rtpmap:8 PCMA/8000
[Apr 23 21:06:00] a=rtpmap:0 PCMU/8000
[Apr 23 21:06:00] a=rtpmap:101 telephone-event/8000
[Apr 23 21:06:00] a=fmtp:101 0-16
[Apr 23 21:06:00] a=ptime:20
[Apr 23 21:06:00] a=maxptime:150
[Apr 23 21:06:00] a=sendrecv
[Apr 23 21:06:00]
[Apr 23 21:06:00] —
[Apr 23 21:06:00] – Called SIP/25dirjyv
[Apr 23 21:06:00]
[Apr 23 21:06:00] <— SIP read from UDP:92.184.x.x:5360 —>
[Apr 23 21:06:00] SIP/2.0 100 Trying
[Apr 23 21:06:00] Via: SIP/2.0/UDP 172.16.0.5:5060;branch=z9hG4bK48116c90;rport=5060
[Apr 23 21:06:00] From: “toto” <sip:04xxxxxxxx @ 172.16.0.5:5060>;tag=as3fa4617c
[Apr 23 21:06:00] To: <sip:25dirjyv @ 172.19.1.100:5360>;tag=257995669
[Apr 23 21:06:00] Call-ID: 62f7201f28fb72451c84373641efe911 @ 172.16.0.5:5060
[Apr 23 21:06:00] CSeq: 102 INVITE
[Apr 23 21:06:00] Contact: <sip:25dirjyv @ 172.19.1.100:5360>
[Apr 23 21:06:00] User-Agent: N510 IP PRO/42.258.00.000.000
[Apr 23 21:06:00] Content-Length: 0
[Apr 23 21:06:00]
[Apr 23 21:06:00]
[Apr 23 21:06:00] <------------->
[Apr 23 21:06:00] — (9 headers 0 lines) —
[Apr 23 21:06:09]
[Apr 23 21:06:09] <— SIP read from UDP:92.184.x.x:5360 —>
[Apr 23 21:06:09] SIP/2.0 480 Temporarily not available
[Apr 23 21:06:09] Via: SIP/2.0/UDP 172.16.0.5:5060;branch=z9hG4bK48116c90;rport=5060
[Apr 23 21:06:09] From: “toto” <sip:04xxxxxxxx @ 172.16.0.5:5060>;tag=as3fa4617c
[Apr 23 21:06:09] To: <sip:25dirjyv @ 172.19.1.100:5360>;tag=257995669
[Apr 23 21:06:09] Call-ID: 62f7201f28fb72451c84373641efe911 @ 172.16.0.5:5060
[Apr 23 21:06:09] CSeq: 102 INVITE
[Apr 23 21:06:09] Contact: <sip:25dirjyv @ 172.19.1.100:5360>
[Apr 23 21:06:09] User-Agent: N510 IP PRO/42.258.00.000.000
[Apr 23 21:06:09] Content-Length: 0
Quelqu’un a t’il eu le même problème ?
J’ai aussi une base Yealink, et là, pas de soucis de boite vocale.