Repa Muchas gracias por tu respuesta
Esta configurado un SIP trunk, configuro el issabel troncal, ruta entrante, ruta saliente, extension
Hago una llamada desde la central (la primera luego de configurar el troncal) la llamada se escucha bien, conecta rapido, al cortar y luego querer hacer una llamada desde la extensión que corto la llamda reciente da el error de todas las lineas estan ocupadas
Audio is at 13432
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to XXXXXXXXXX6:5060:
INVITE sip:00110116542322@XXXXXXXXXX6 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX6:5060;branch=z9hG4bK2ded2fd0;rport
Max-Forwards: 70
From: <sip:011XXXX31X8002@XXXXXXXXXX6>;tag=as0bd1bc0e
To: <sip:00110116542322@XXXXXXXXXX6>
Contact: <sip:011XXXX31X8002@XXXXXXXXXX6:5060>
Call-ID: 05c291a27cc67cdb430aec314a21e4cf@XXXXXXXXXX6:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0(16.7.0)
Date: Thu, 11 Jun 2020 17:38:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "011XXXX31X8002" <sip:011XXXX31X8002@XXXXXXXXXX6>;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 2081438984 2081438984 IN IP4 XXXXXXXXXX6
s=Asterisk PBX 16.7.0
c=IN IP4 XXXXXXXXXX6
t=0 0
m=audio 13432 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:230
a=sendrecv
<--- SIP read from UDP:XXXXXXXXXX6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XXXXXXXXXX6:5060;received=XXXXXXXXXX6;branch=z9hG4bK2ded2fd0;rport=5060
From: <sip:011XXXX31X8002@XXXXXXXXXX6>;tag=as0bd1bc0e
To: <sip:00110116542322@XXXXXXXXXX6>
Call-ID: 05c291a27cc67cdb430aec314a21e4cf@XXXXXXXXXX6:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:XXXXXXXXXX6:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP XXXXXXXXXX6:5060;received=XXXXXXXXXX6;branch=z9hG4bK2ded2fd0;rport=5060
From: <sip:011XXXX31X8002@XXXXXXXXXX6>;tag=as0bd1bc0e
To: <sip:00110116542322@XXXXXXXXXX6>;tag=SDeo2d799-43deb25d6df7e008a8cc39a4e03655bf-a607
Call-ID: 05c291a27cc67cdb430aec314a21e4cf@XXXXXXXXXX6:5060
CSeq: 102 INVITE
Server: Darwin Proxy Sip
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to XXXXXXXXXX6:5060:
ACK sip:00110116542322@XXXXXXXXXX6 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX6:5060;branch=z9hG4bK2ded2fd0;rport
Max-Forwards: 70
From: <sip:011XXXX31X8002@XXXXXXXXXX6>;tag=as0bd1bc0e
To: <sip:00110116542322@XXXXXXXXXX6>;tag=SDeo2d799-43deb25d6df7e008a8cc39a4e03655bf-a607
Contact: <sip:011XXXX31X8002@XXXXXXXXXX6:5060>
Call-ID: 05c291a27cc67cdb430aec314a21e4cf@XXXXXXXXXX6:5060
CSeq: 102 ACK
User-Agent: IPBX-2.11.0(16.7.0)
Content-Length: 0
Really destroying SIP dialog '05c291a27cc67cdb430aec314a21e4cf@XXXXXXXXXX6:5060' Method: INVITE
[2020-06-11 13:38:36] WARNING[4025][C-0000000c]: file.c:782 ast_openstream_full: File pls-try-call-later does not exist in any format
[2020-06-11 13:38:36] WARNING[4025][C-0000000c]: file.c:1255 ast_streamfile: Unable to open pls-try-call-later (format (alaw)): No such file or directory
[2020-06-11 13:38:36] WARNING[4025][C-0000000c]: app_playback.c:497 playback_exec: Playback failed on IAX2/200-10589 for all-circuits-busy-now&pls-try-call-later, noanswer
Really destroying SIP dialog '4122621267' Method: REGISTER
Reliably Transmitting (NAT) to XXXXXXXXXX6:5060:
OPTIONS sip:XXXXXXXXXX6 SIP/2.0
Via: SIP/2.0/UDP XXXXXXXXXX6:5060;branch=z9hG4bK04b5eca9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@XXXXXXXXXX6>;tag=as00b721f1
To: <sip:XXXXXXXXXX6>
Contact: <sip:Unknown@XXXXXXXXXX6:5060>
Call-ID: 0c5a43d613516e7d438ee740702d4b41@XXXXXXXXXX6:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(16.7.0)
Date: Thu, 11 Jun 2020 17:39:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<--- SIP read from UDP:XXXXXXXXXX6:5060 --->
SIP/2.0 200 Keepalive
Via: SIP/2.0/UDP XXXXXXXXXX6:5060;received=XXXXXXXXXX6;branch=z9hG4bK04b5eca9;rport=5060
From: "Unknown" <sip:Unknown@XXXXXXXXXX6>;tag=as00b721f1
To: <sip:XXXXXXXXXX6>;tag=SDmmup599-940e20d5808f5fe8982a5ade50ea88f2.2df2
Call-ID: 0c5a43d613516e7d438ee740702d4b41@XXXXXXXXXX6:5060
CSeq: 102 OPTIONS
Server: Darwin Proxy Sip
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '0c5a43d613516e7d438ee740702d4b41@XXXXXXXXXX6:5060' Method: OPTIONS
