hello … in advance thanks for the help. I’m new to freepbx I downloaded and installed the freepbx version of the page for 32 bits. Configure some extensions, and configure a trunk with a grandstream gwx4104. Everything worked perfect. But then connect the server to the internet to update modules and download a language pack in Spanish and the trunk stopped working.
I paste down what I get from sip debug when I want to use the trunk.
[root@localhost ~]# asterisk -r
Asterisk 11.25.1, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
=========================================================================
Connected to Asterisk 11.25.1 currently running on localhost (pid = 1923)
localhost*CLI> sip set debug on
SIP Debugging enabled
<— SIP read from UDP:130.16.50.131:61031 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—468a4d6f76e9ff21;rport
Max-Forwards: 70
Contact: <sip:[email protected]:61031;rinstance=bb23a1cf71d09619>
To: <sip:[email protected]>
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 1 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.5.0 stamp 97566
Content-Length: 211
v=0
o=- 13199836383296716 1 IN IP4 130.16.50.131
s=X-Lite release 5.5.0 stamp 97566
c=IN IP4 130.16.50.131
t=0 0
m=audio 50666 RTP/AVP 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 9 lines) —
Sending to 130.16.50.131:61031 (NAT)
Sending to 130.16.50.131:61031 (NAT)
Using INVITE request as basis request - 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
Found peer ‘10300’ for ‘10300’ from 130.16.50.131:61031
<— Reliably Transmitting (no NAT) to 130.16.50.131:61031 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—468a4d6f76e9ff21;received=130.16.50.131;rport=61031
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
To: <sip:[email protected]>;tag=as78820e77
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 1 INVITE
Server: FPBX-13.0.196.2(11.25.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fba3d6a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:130.16.50.131:61031 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—468a4d6f76e9ff21;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as78820e77
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 1 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from UDP:130.16.50.131:61031 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;rport
Max-Forwards: 70
Contact: <sip:[email protected]:61031;rinstance=bb23a1cf71d09619>
To: <sip:[email protected]>
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.5.0 stamp 97566
Authorization: Digest username="10300",realm="asterisk",nonce="0fba3d6a",uri="sip:[email protected]",response="e92fcadf81da74907e4de97d5e291f84",algorithm=MD5
Content-Length: 211
v=0
o=- 13199836383296716 1 IN IP4 130.16.50.131
s=X-Lite release 5.5.0 stamp 97566
c=IN IP4 130.16.50.131
t=0 0
m=audio 50666 RTP/AVP 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 9 lines) —
Sending to 130.16.50.131:61031 (no NAT)
Using INVITE request as basis request - 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
Found peer ‘10300’ for ‘10300’ from 130.16.50.131:61031
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726|g722), peer - audio=(gsm)/video=(nothing)/text=(nothing), combined - (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 130.16.50.131:50666
Looking for 081 in from-internal (domain 130.16.50.130)
list_route: hop: <sip:[email protected]:61031;rinstance=bb23a1cf71d09619>
<— Transmitting (no NAT) to 130.16.50.131:61031 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;received=130.16.50.131;rport=61031
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
To: <sip:[email protected]>
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 INVITE
Server: FPBX-13.0.196.2(11.25.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
[2019-04-15 18:12:38] WARNING[29506][C-00000032]: chan_sip.c:6274 sip_call: No audio format found to offer. Cancelling call to 81
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: INVITE)
Audio is at 18570
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (no NAT) to 130.16.50.131:61031 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;received=130.16.50.131;rport=61031
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
To: <sip:[email protected]>;tag=as37bb394f
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 INVITE
Server: FPBX-13.0.196.2(11.25.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 233
v=0
o=root 56541431 56541431 IN IP4 130.16.50.130
s=Asterisk PBX 11.25.1
c=IN IP4 130.16.50.130
t=0 0
m=audio 18570 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:39] NOTICE[29506][C-00000032]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 126 received from ‘130.16.50.131:50666’
[2019-04-15 18:12:40] WARNING[29506][C-00000032]: file.c:701 ast_openstream_full: File please-try-call-later does not exist in any format
[2019-04-15 18:12:40] WARNING[29506][C-00000032]: file.c:1017 ast_streamfile: Unable to open please-try-call-later (format (gsm)): No such file or directory
[2019-04-15 18:12:40] WARNING[29506][C-00000032]: app_playback.c:484 playback_exec: ast_streamfile failed on SIP/10300-00000051 for all-circuits-busy-now&please-try-call-later, noanswer
<— Reliably Transmitting (no NAT) to 130.16.50.131:61031 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;received=130.16.50.131;rport=61031
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
To: <sip:[email protected]>;tag=as37bb394f
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 INVITE
Server: FPBX-13.0.196.2(11.25.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[2019-04-15 18:12:40] WARNING[29506][C-00000032]: channel.c:4861 ast_prod: Prodding channel ‘SIP/10300-00000051’ failed
Retransmitting #1 (no NAT) to 130.16.50.131:61031:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;received=130.16.50.131;rport=61031
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
To: <sip:[email protected]>;tag=as37bb394f
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 INVITE
Server: FPBX-13.0.196.2(11.25.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
—
Retransmitting #2 (no NAT) to 130.16.50.131:61031:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;received=130.16.50.131;rport=61031
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
To: <sip:[email protected]>;tag=as37bb394f
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 INVITE
Server: FPBX-13.0.196.2(11.25.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
—
<— SIP read from UDP:130.16.50.131:61031 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as37bb394f
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM’ Method: ACK
<— SIP read from UDP:130.16.50.131:61031 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as37bb394f
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from UDP:130.16.50.131:61031 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as37bb394f
From: "PC TEC COMM"<sip:[email protected]>;tag=d1c0450a
Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM
CSeq: 2 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
localhost*CLI> sip set debug off
SIP Debugging Disabled