Estou instalando um telefone SIP em um ambiente VoIP. Existem 2 telefones do sistema que funcionam perfeitamente (mesmo fabricante do PBX), e o terceiro telefone pode ser chamado, mas não pode ligar para os outros 2 telefones.
O PBX mostra um erro: "Nenhum codec correspondente! Chamada rejeitada" Esta é a conversa da perspectiva do terceiro telefone:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport
From: <sip:192.168.0.250>;tag=1540961770
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 160 INVITE
Contact: <sip:192.168.0.14:5060>
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2140 1.0.5.18
Privacy: none
P-Preferred-Identity: <sip:192.168.0.250>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 306
v=0
o=- 8000 8000 IN IP4 192.168.0.14
s=SIP Call
c=IN IP4 192.168.0.14
t=0 0
m=audio 5004 RTP/AVP 9 8 18 2 101
a=sendrecv
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
SIP/2.0 480 Temporarily not available
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport=5060
From: <sip:192.168.0.250>;tag=1540961770
To: <sip:[email protected]>;tag=74C1BCB3775433109F0E49A014240025
Call-ID: [email protected]
CSeq: 160 INVITE
Content-Length: 0
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK1040318360;rport
From: <sip:192.168.0.250>;tag=1540961770
To: <sip:[email protected]>;tag=74C1BCB3775433109F0E49A014240025
Call-ID: [email protected]
CSeq: 160 ACK
Content-Length: 0
Mas de uma chamada recebida:
INVITE sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060;transport=udp>
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, MESSAGE, SUBSCRIBE, UPDATE, PRACK, REFER
Supported: 100rel, replaces, timer
User-Agent: hybird_130j V.9.1 Rev. 10 (Patch 4) IPSec
Alert-Info: <http://127.0.0.1>;info=alert-internal
Allow-Events: refer, message-summary, dialog
P-Asserted-Identity: "Sys Tel 20" <sip:[email protected];user=phone>
Session-Expires: 1800
Content-Type: application/sdp
Content-Length: 328
v=0
o=- 71 1 IN IP4 192.168.0.250
s=SIP call
c=IN IP4 192.168.0.250
t=0 0
m=audio 10848 RTP/AVP 0 8 18 2 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.5.18
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>;tag=471383942
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Contact: <sip:192.168.0.14:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.5.18
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.250:5060;branch=z9hG4bKE85E5CC7F25533109F4949A014240025;rport=5060
From: "Sys Tel 20" <sip:[email protected];user=phone>;tag=1E2DB6AE775433109F0C49A014240025
To: <sip:[email protected];user=phone>;tag=471383942
Call-ID: 303F5CC7F25533109F4849A014240025
CSeq: 1 INVITE
Contact: <sip:192.168.0.14:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2140 1.0.5.18
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 306
Observe que ambos oferecem PCMU/PCMA e alguns outros codecs. Por que a chamada falha?
Os IPs dos telefones são 192.168.0.12e 192.168.0.14, o PBX tem 192.168.0.250.
Responder1
A triste verdade é que esses telefones que não são do sistema Elmeg não foram registrados corretamente. Tive que atribuir PINs para os usuários na interface Hybird 130j, que é então usada para autenticação no sistema.
Elmeg não se preocupou em declarar em lugar nenhum que esse PIN deve ser definido e é usado como senha SIP para telefones não-elmeg.
Responder2
Remova o g729, a menos que você tenha a licença.
Use 711a, 711u, 722, GSM.
Verifique se todos os telefones estão registrados e se estão usando o mesmo transporte (udp) e se as configurações de nat são todas iguais.