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Trasferimento nr. esterno

F

fractals87

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Registrato
11 Marzo 2015
Messaggi
11
Ciao a tutti,

Ho un problema con il trasferimento di chiamata verso numero estero.
Questo è il log che ottengo, cosa può essere?

23/03/2020 20:07:41 - Currently active calls [none]
23/03/2020 20:07:41 - Leg L:832.1[Line:10001<<anonymous] is terminated: Cause: BYE from PBX
23/03/2020 20:07:41 - Leg L:832.2[Line:10001>>340xxxxxxx] is terminated: Cause: 503 Service Unavailable/INVITE from 192.168.1.8:5060
23/03/2020 20:07:41 - [CM503020]: Call(C:832): Normal call termination. Call originator: Line:10001<<anonymous. Reason: Server Failure
23/03/2020 20:07:41 - L:832.1[Line:10001<<anonymous] failed to reach Line:10001>>340xxxxxxx, reason Server Failure
23/03/2020 20:07:41 - Call to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060] from L:832.1[Line:10001<<anonymous] failed, cause: Cause: 503 Service Unavailable/INVITE from 192.168.1.8:5060
23/03/2020 20:07:41 - [CM503003]: Call(C:832): Call to <sip:[email protected]:5060> has failed; Cause: 503 Service Unavailable/INVITE from 192.168.1.8:5060
23/03/2020 20:07:40 - [CM503025]: Call(C:832): Calling T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060] for L:832.1[Line:10001<<anonymous]
23/03/2020 20:07:40 - [CM503027]: Call(C:832): From: Line:10001<<anonymous (<sip:[email protected]:5060>) to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060]
23/03/2020 20:07:40 - [CM503004]: Call(C:832): Route 1: from L:832.1[Line:10001<<anonymous] to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060]
23/03/2020 20:07:40 - Line limit check: Current # of calls for line Lc:10001(@Patton Analog FXO[<sip:[email protected]:5060>]) is 1; limit is 2
23/03/2020 20:07:40 - Call(C:832): Call from Line:10001<<anonymous to 340xxxxxxx matches outbound rule 'Generale'
23/03/2020 20:07:40 - [Flow] Call(C:832): has built target endpoint: Out#:>>Rule{Generale}>>340xxxxxxx for call from L:832.1[Line:10001<<anonymous]
23/03/2020 20:07:40 - [Flow] Target endpoint for 340xxxxxxx is Out#:>>Rule{Generale}>>340xxxxxxx
23/03/2020 20:07:40 - [CM503010]: Call(C:832): Making route(s) from Line:10001<<anonymous to <sip:[email protected]:5060>
23/03/2020 20:07:40 - [CM505002]: Gateway:[Patton Analog FXO] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4112 JO EUI 00A0BA063824 R6.5 2014-01-09 H323 SIP FXS FXO M5T SIP Stack/4.2.8.10] PBX contact: [sip:[email protected]:5060]
23/03/2020 20:07:40 - [CM500002]: Call(C:832): Info on incoming INVITE from Line:10001<<anonymous:
Invite-IN Recv Req INVITE from 192.168.1.8:5060 tid=e32ee85ccba9a0aee Call-ID=bca2fd0e3e031ab7:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKe32ee85ccba9a0aee
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp>
To: <sip:[email protected]:5060>
From: <sip:[email protected]:5060>;tag=8a68181b69
Call-ID: bca2fd0e3e031ab7
CSeq: 28422 INVITE
Content-Type: application/sdp
Proxy-Authorization: Digest username="10001",realm="3CXPhoneSystem",nonce="414d535c148999fc78:869245ee44c0b28543ed596b6504631c",uri="sip:[email protected]:5060",response="1e0e24bd373f267a55cb14c0e083cfd3",algorithm=MD5
Supported: replaces
User-Agent: Patton SN4112 JO EUI 00A0BA063824 R6.5 2014-01-09 H323 SIP FXS FXO M5T SIP Stack/4.2.8.10
Content-Length: 246

v=0
o=MxSIP 0 28517 IN IP4 192.168.1.8
s=SIP Call
c=IN IP4 192.168.1.8
t=0 0
m=audio 5344 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
23/03/2020 20:07:40 - [CM503001]: Call(C:832): Incoming call from Line:10001<<anonymous to <sip:[email protected]:5060>
23/03/2020 20:07:40 - Line limit check: Current # of calls for line Lc:10001(@Patton Analog FXO[<sip:[email protected]:5060>]) is 1; limit is 2
23/03/2020 20:07:40 - [CM503012]: Inbound office hours rule (unnamed) for 10001 forwards to DN:340xxxxxxx
23/03/2020 20:07:40 - [Flow] No office hours set, office hours assumed
23/03/2020 20:07:40 - [Flow] Looking for inbound target: called=000001; caller=<sip:anonymous@:5060>
 
Ciao,
due cose:
il firmware del Patton è obsoleto (è del 2014), aggiornalo al 6.11
alla voce Dial Delay nelle opzioni del Patton su 3CX, al posto di "Wait for dial tone" imposta "2 secondi" poi rigenera e ricarica il file sul Patton
 
Effettuato upgrade, ed impostato il wait for dial (prima lo avevo impostato a 1 sec)
Purtroppo non è cambiato nulla.

25/03/2020 09:38:47 - [CM503025]: Call(C:893): Calling T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060] for L:893.1[Line:10001<<anonymous]
25/03/2020 09:38:47 - [CM503027]: Call(C:893): From: Line:10001<<anonymous (<sip:[email protected]:5060>) to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060]
25/03/2020 09:38:47 - [CM503004]: Call(C:893): Route 1: from L:893.1[Line:10001<<anonymous] to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060]
25/03/2020 09:38:47 - Line limit check: Current # of calls for line Lc:10001(@Patton Analog FXO[<sip:[email protected]:5060>]) is 1; limit is 2
25/03/2020 09:38:47 - Call(C:893): Call from Line:10001<<anonymous to 340xxxxxxx matches outbound rule 'Generale'
25/03/2020 09:38:47 - [Flow] Call(C:893): has built target endpoint: Out#:>>Rule{Generale}>>340xxxxxxx for call from L:893.1[Line:10001<<anonymous]
25/03/2020 09:38:47 - [Flow] Target endpoint for 340xxxxxxx is Out#:>>Rule{Generale}>>340xxxxxxx
25/03/2020 09:38:47 - [CM503010]: Call(C:893): Making route(s) from Line:10001<<anonymous to <sip:[email protected]:5060>
25/03/2020 09:38:47 - [CM505002]: Gateway:[Patton Analog FXO] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4112 JO EUI 00A0BA063824 R6.11 2019-07-02 H323 SIP FXS FXO M5T SIP Stack/4.2.26.112] PBX contact: [sip:[email protected]:5060]
25/03/2020 09:38:47 - [CM500002]: Call(C:893): Info on incoming INVITE from Line:10001<<anonymous: Invite-IN Recv Req INVITE from 192.168.1.8:5060 tid=6a69559d58e96b118 Call-ID=37e6826225ecd422: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a69559d58e96b118 Max-Forwards: 70 Contact: <sip:[email protected]:5060;transport=udp> To: <sip:[email protected]:5060> From: <sip:[email protected]:5060>;tag=da7a20fefa Call-ID: 37e6826225ecd422 CSeq: 17544 INVITE Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, UPDATE, REFER, REGISTER Content-Type: application/sdp Proxy-Authorization: Digest username="10001",realm="3CXPhoneSystem",nonce="414d535c148ba99721:2f371870198e382ccef5c060e182ad0d",uri="sip:[email protected]:5060",response="dc68592d0820467dc572a88809d43b4b",algorithm=MD5 Supported: replaces User-Agent: Patton SN4112 JO EUI 00A0BA063824 R6.11 2019-07-02 H323 SIP FXS FXO M5T SIP Stack/4.2.26.112 Content-Length: 243 v=0 o=MxSIP 0 13 IN IP4 192.168.1.8 s=SIP Call c=IN IP4 192.168.1.8 t=0 0 m=audio 4878 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv
25/03/2020 09:38:47 - [CM503001]: Call(C:893): Incoming call from Line:10001<<anonymous to <sip:[email protected]:5060>
25/03/2020 09:38:47 - Line limit check: Current # of calls for line Lc:10001(@Patton Analog FXO[<sip:[email protected]:5060>]) is 1; limit is 2
25/03/2020 09:38:47 - [CM503012]: Inbound office hours rule (unnamed) for 10001 forwards to DN:340xxxxxxx
25/03/2020 09:38:47 - [Flow] No office hours set, office hours assumed
25/03/2020 09:38:47 - [Flow] Looking for inbound target: called=000000; caller=<sip:anonymous@:5060>
25/03/2020 09:38:44 - Leg L:892.1[Line:10001<<anonymous] is terminated: Cause: BYE from PBX
25/03/2020 09:38:44 - [CM503020]: Call(C:892): Normal call termination. Call originator: Line:10001<<anonymous. Reason: Server Failure
25/03/2020 09:38:44 - L:892.1[Line:10001<<anonymous] failed to reach Line:10001>>340xxxxxxx, reason Server Failure
25/03/2020 09:38:44 - Leg L:892.2[Line:10001>>340xxxxxxx] is terminated: Cause: 503 Service Unavailable/INVITE from 192.168.1.8:5060
25/03/2020 09:38:44 - Call to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060] from L:892.1[Line:10001<<anonymous] failed, cause: Cause: 503 Service Unavailable/INVITE from 192.168.1.8:5060
25/03/2020 09:38:44 - [CM503003]: Call(C:892): Call to <sip:[email protected]:5060> has failed; Cause: 503 Service Unavailable/INVITE from 192.168.1.8:5060
25/03/2020 09:38:44 - [CM503025]: Call(C:892): Calling T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060] for L:892.1[Line:10001<<anonymous]
25/03/2020 09:38:44 - [CM503027]: Call(C:892): From: Line:10001<<anonymous (<sip:[email protected]:5060>) to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060]
25/03/2020 09:38:44 - [CM503004]: Call(C:892): Route 1: from L:892.1[Line:10001<<anonymous] to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060]
25/03/2020 09:38:44 - Line limit check: Current # of calls for line Lc:10001(@Patton Analog FXO[<sip:[email protected]:5060>]) is 1; limit is 2
25/03/2020 09:38:44 - Call(C:892): Call from Line:10001<<anonymous to 340xxxxxxx matches outbound rule 'Generale'
25/03/2020 09:38:44 - [Flow] Call(C:892): has built target endpoint: Out#:>>Rule{Generale}>>340xxxxxxx for call from L:892.1[Line:10001<<anonymous]
25/03/2020 09:38:44 - [Flow] Target endpoint for 340xxxxxxx is Out#:>>Rule{Generale}>>340xxxxxxx
25/03/2020 09:38:44 - [CM503010]: Call(C:892): Making route(s) from Line:10001<<anonymous to <sip:[email protected]:5060>
25/03/2020 09:38:44 - [CM505002]: Gateway:[Patton Analog FXO] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4112 JO EUI 00A0BA063824 R6.11 2019-07-02 H323 SIP FXS FXO M5T SIP Stack/4.2.26.112] PBX contact: [sip:[email protected]:5060]
25/03/2020 09:38:44 - [CM500002]: Call(C:892): Info on incoming INVITE from Line:10001<<anonymous: Invite-IN Recv Req INVITE from 192.168.1.8:5060 tid=671e467b3cf53ccc5 Call-ID=b5daf68effcbcb99: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK671e467b3cf53ccc5 Max-Forwards: 70 Contact: <sip:[email protected]:5060;transport=udp> To: <sip:[email protected]:5060> From: <sip:[email protected]:5060>;tag=c98fb662bd Call-ID: b5daf68effcbcb99 CSeq: 9085 INVITE Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, INFO, UPDATE, REFER, REGISTER Content-Type: application/sdp Proxy-Authorization: Digest username="10001",realm="3CXPhoneSystem",nonce="414d535c148ba99412:4aadd3a32c5a323f7d65ab3f03d41c6a",uri="sip:[email protected]:5060",response="108f60dad220300abe21567c5066b67e",algorithm=MD5 Supported: replaces User-Agent: Patton SN4112 JO EUI 00A0BA063824 R6.11 2019-07-02 H323 SIP FXS FXO M5T SIP Stack/4.2.26.112 Content-Length: 243 v=0 o=MxSIP 0 10 IN IP4 192.168.1.8 s=SIP Call c=IN IP4 192.168.1.8 t=0 0 m=audio 4874 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv
25/03/2020 09:38:44 - [CM503001]: Call(C:892): Incoming call from Line:10001<<anonymous to <sip:[email protected]:5060>
25/03/2020 09:38:44 - Line limit check: Current # of calls for line Lc:10001(@Patton Analog FXO[<sip:[email protected]:5060>]) is 1; limit is 2
25/03/2020 09:38:44 - [CM503012]: Inbound office hours rule (unnamed) for 10001 forwards to DN:340xxxxxxx
25/03/2020 09:38:44 - [Flow] No office hours set, office hours assumed
25/03/2020 09:38:44 - [Flow] Looking for inbound target: called=000001; caller=<sip:anonymous@:5060>
 
Ultima modifica:
il Patton continua a dare errore:
25/03/2020 09:38:44 - Call to T:Line:10001>>340xxxxxxx@[Dev:sip:[email protected]:5060] from L:892.1[Line:10001<<anonymous] failed, cause: Cause: 503 Service Unavailable/INVITE from 192.168.1.8:5060

il problema avviene solo per le chiamate inoltrate? se chiami il numero esterno direttamente, la chiamata esce?
 
Si le chiamate escono con successo e funzionano anche due chiamate contemporanee (il patton dispone di due linee fxo configurate).
Ho scoperto questo problema proprio in questo periodo perchè avrei dovuto configurare il trasferimento di chiamata direttamente nel trunk (prima non ne ho mai avuto necessità).
Per ora l'ho aggiustata attaccando telefono sip alla borchia telecom e configurando direttamente il trasferimento.
Ma proprio non riesco ad immaginare da cosa possa dipendere.
Ho altre installazioni però con trunk voip e non ho mai avuto problemi.
 
sinceramente non so dirti...l'unica cosa che noto è che il chiamante è anonimo...ma non vedo come questo possa influire.
Non ti dico neanche di aprire un ticket con 3CX, perchè dovrei rimandarti al supporto Patton, in quanto l'errore arriva da lì
 
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