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Estoy configurando un HT503 (ver 1.0.6.13 "ultima version") con el 3CX (ver 10 SP 6) (ALGUIEN PUEDE AYUDARME CON ESTO?)
Llamadas entrantes funcionan bien.
Llamadas Salientes se escucha tono de ocupado.
Configuracion en 3CX
PSTN Device (grandstream -> ht488)
Name: HT503
Gateway Hostname or ip: 192.168.1.2 (IP del HT503)
Port Identification: 80712
Authentication ID: 80712
Password: 1234
OUTBOUND RULES
Prefix: 9
calls from extension group: 9 (8 digitos para la ciudad Monterrey, Mexico para llamadas locales)
Route 1: HT503 (nombre del Pstn device)
Strip Digits: 1
STATUS
Registered (idle)
En HT503 esta configurado lo siguiente (FXO PORT)
SIP Server: 192.168.1.1 (Servidor 3CX)
User ID: 80712
Auth ID: 80712
Password: 1234
Local sip port: 5062
Local RTP port: 5012
Preferred DTMF method: RF2833
PSTN Ring Thru FXS: NO
Number of Rings: 1
Wait for Dial-Tone: NO
Stage Method (1/2): 1
BASIC SETTINGS
Unconditional Call Forward to VOIP: 555 @ 192.168.1.1 : 5060
ERROR EN 3CX: (Anexo archivo)
10:41:14.616|.\CallCtrl.cpp(346)|Log2||CallCtrl:nIncomingCall:[CM503001]: Call(2): Incoming call from Ext.100 to <sip:[email protected];user=phone><br>
10:41:14.620|.\Extension.cpp(1407)|Log3||Extension:rintEndpointInfo:[CM505001]: Ext.100: Device info: Device Identified: [Man: GrandStream;Mod: GXP series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP2124 1.0.3.19] PBX contact: [sip:[email protected]:5060]<br>
10:41:14.626|.\CallCtrl.cpp(529)|Log3||CallCtrl:nSelectRouteReq:[CM503010]: Making route(s) to <sip:[email protected];user=phone><br>
10:41:14.627|.\CallCtrl.cpp(708)|Log2||CallCtrl:nSelectRouteReq:[CM503004]: Call(2): Route 1: PSTNline:83834700@(Ln.10000@HT-503)@[Dev:sip:[email protected]:5062]<br>
10:41:14.664|.\Target.cpp(441)|Log2||Target::makeOneInvite:[CM503025]: Call(2): Calling PSTNline:83834700@(Ln.10000@HT-503)@[Dev:sip:[email protected]:5062]<br>
10:41:15.191|.\Call.cpp(42)|Log3||??:Currently active calls - 1: [2]<br>
10:41:16.643|.\CallLeg.cpp(315)|Log3||CallLeg:nAnswer:[CM503002]: Call(2): Alerting sip:[email protected]:5062<br>
10:41:16.643|.\Line.cpp(1452)|Log2||Line:rintEndpointInfo:[CM505002]: Gateway:[HT-503] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream HT-503 V1.1B 1.0.6.13 chip V2.2] PBX contact: [sip:[email protected]:5060]<br>
10:41:16.670|.\CallCtrl.cpp(885)|Log2||CallCtrl:nLegConnected:[CM503007]: Call(2): Device joined: sip:[email protected]:5060;user=phone<br>
10:41:16.674|.\CallCtrl.cpp(885)|Log2||CallCtrl:nLegConnected:[CM503007]: Call(2): Device joined: sip:[email protected]:5062<br>
10:41:20.203|.\Call.cpp(1396)|Log2||Call::Terminate:[CM503008]: Call(2): Call is terminated<br>
Llamadas entrantes funcionan bien.
Llamadas Salientes se escucha tono de ocupado.
Configuracion en 3CX
PSTN Device (grandstream -> ht488)
Name: HT503
Gateway Hostname or ip: 192.168.1.2 (IP del HT503)
Port Identification: 80712
Authentication ID: 80712
Password: 1234
OUTBOUND RULES
Prefix: 9
calls from extension group: 9 (8 digitos para la ciudad Monterrey, Mexico para llamadas locales)
Route 1: HT503 (nombre del Pstn device)
Strip Digits: 1
STATUS
Registered (idle)
En HT503 esta configurado lo siguiente (FXO PORT)
SIP Server: 192.168.1.1 (Servidor 3CX)
User ID: 80712
Auth ID: 80712
Password: 1234
Local sip port: 5062
Local RTP port: 5012
Preferred DTMF method: RF2833
PSTN Ring Thru FXS: NO
Number of Rings: 1
Wait for Dial-Tone: NO
Stage Method (1/2): 1
BASIC SETTINGS
Unconditional Call Forward to VOIP: 555 @ 192.168.1.1 : 5060
ERROR EN 3CX: (Anexo archivo)
10:41:14.616|.\CallCtrl.cpp(346)|Log2||CallCtrl:nIncomingCall:[CM503001]: Call(2): Incoming call from Ext.100 to <sip:[email protected];user=phone><br>
10:41:14.620|.\Extension.cpp(1407)|Log3||Extension:rintEndpointInfo:[CM505001]: Ext.100: Device info: Device Identified: [Man: GrandStream;Mod: GXP series;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXP2124 1.0.3.19] PBX contact: [sip:[email protected]:5060]<br>
10:41:14.626|.\CallCtrl.cpp(529)|Log3||CallCtrl:nSelectRouteReq:[CM503010]: Making route(s) to <sip:[email protected];user=phone><br>
10:41:14.627|.\CallCtrl.cpp(708)|Log2||CallCtrl:nSelectRouteReq:[CM503004]: Call(2): Route 1: PSTNline:83834700@(Ln.10000@HT-503)@[Dev:sip:[email protected]:5062]<br>
10:41:14.664|.\Target.cpp(441)|Log2||Target::makeOneInvite:[CM503025]: Call(2): Calling PSTNline:83834700@(Ln.10000@HT-503)@[Dev:sip:[email protected]:5062]<br>
10:41:15.191|.\Call.cpp(42)|Log3||??:Currently active calls - 1: [2]<br>
10:41:16.643|.\CallLeg.cpp(315)|Log3||CallLeg:nAnswer:[CM503002]: Call(2): Alerting sip:[email protected]:5062<br>
10:41:16.643|.\Line.cpp(1452)|Log2||Line:rintEndpointInfo:[CM505002]: Gateway:[HT-503] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream HT-503 V1.1B 1.0.6.13 chip V2.2] PBX contact: [sip:[email protected]:5060]<br>
10:41:16.670|.\CallCtrl.cpp(885)|Log2||CallCtrl:nLegConnected:[CM503007]: Call(2): Device joined: sip:[email protected]:5060;user=phone<br>
10:41:16.674|.\CallCtrl.cpp(885)|Log2||CallCtrl:nLegConnected:[CM503007]: Call(2): Device joined: sip:[email protected]:5062<br>
10:41:20.203|.\Call.cpp(1396)|Log2||Call::Terminate:[CM503008]: Call(2): Call is terminated<br>