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Problems with trunks after updating all the modules


(Franklin8686) #1

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 <markster@digium.com>

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:081@130.16.50.130 SIP/2.0

Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—468a4d6f76e9ff21;rport

Max-Forwards: 70

Contact: <sip:10300@130.16.50.131:61031;rinstance=bb23a1cf71d09619>

To: <sip:081@130.16.50.130>

From: "PC TEC COMM"<sip:10300@130.16.50.130>;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:10300@130.16.50.130>;tag=d1c0450a

To: <sip:081@130.16.50.130>;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:081@130.16.50.130 SIP/2.0

Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—468a4d6f76e9ff21;rport

Max-Forwards: 70

To: <sip:081@130.16.50.130>;tag=as78820e77

From: "PC TEC COMM"<sip:10300@130.16.50.130>;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:081@130.16.50.130 SIP/2.0

Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;rport

Max-Forwards: 70

Contact: <sip:10300@130.16.50.131:61031;rinstance=bb23a1cf71d09619>

To: <sip:081@130.16.50.130>

From: "PC TEC COMM"<sip:10300@130.16.50.130>;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:081@130.16.50.130",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:10300@130.16.50.131:61031;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:10300@130.16.50.130>;tag=d1c0450a

To: <sip:081@130.16.50.130>

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:081@130.16.50.130:5060>

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 ‘6495796c74dbd39e1090eaef2cf71819@130.16.50.130:5060’ 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:10300@130.16.50.130>;tag=d1c0450a

To: <sip:081@130.16.50.130>;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:081@130.16.50.130:5060>

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:10300@130.16.50.130>;tag=d1c0450a

To: <sip:081@130.16.50.130>;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:10300@130.16.50.130>;tag=d1c0450a

To: <sip:081@130.16.50.130>;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:10300@130.16.50.130>;tag=d1c0450a

To: <sip:081@130.16.50.130>;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:081@130.16.50.130 SIP/2.0

Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;rport

Max-Forwards: 70

To: <sip:081@130.16.50.130>;tag=as37bb394f

From: "PC TEC COMM"<sip:10300@130.16.50.130>;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:081@130.16.50.130 SIP/2.0

Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;rport

Max-Forwards: 70

To: <sip:081@130.16.50.130>;tag=as37bb394f

From: "PC TEC COMM"<sip:10300@130.16.50.130>;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:081@130.16.50.130 SIP/2.0

Via: SIP/2.0/UDP 130.16.50.131:61031;branch=z9hG4bK-524287-1—6da6662e086f6c3b;rport

Max-Forwards: 70

To: <sip:081@130.16.50.130>;tag=as37bb394f

From: "PC TEC COMM"<sip:10300@130.16.50.130>;tag=d1c0450a

Call-ID: 97566Y2M2ODEzNzBmYmY1Y2VjY2NkZDJlNmJhMjcyMzVjNWM

CSeq: 2 ACK

Content-Length: 0

<------------->

— (8 headers 0 lines) —

Really destroying SIP dialog ‘6495796c74dbd39e1090eaef2cf71819@130.16.50.130:5060’ Method: INVITE

localhost*CLI> sip set debug off

SIP Debugging Disabled


(Dave Burgess) #2

I’m gonna guess that your upstream provider doesn’t support GSM as a primary codec.


(system) closed #3

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