Telecare not working after update

Hi,

After updating the freePBX to the 14.0 version useres using telecare stop working. We have check everything and it is the exact same configuration we had on the 13.0 version.

If we debug the call to telecare center the negotiation is well done but for some reasson it end up in a bucle witch the telecare center didnt receive the call.

I attach the Sip debug of one call, everything seems find to me, lets see if you see anything strange

< <--- SIP read from UDP:10.0.5.137:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.137;branch=z9hG4bKb7d38b6e868471c66;rport
Max-Forwards: 70
From: 938810914 <sip:[email protected]>;tag=9531a2705f
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 983478748 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, SUBSCRIBE, UPDATE
Contact: <sip:[email protected]>
Supported: 100rel, answermode, early-session, eventlist, histinfo, join, path, replaces, tdialog, timer
User-Agent: PTInS GR2402GA Build 3RGW030500m139
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 457
v=0
o=MxSIP 889 889 IN IP4 10.0.5.137
s=CMGR_20_DSL_CLIENT
c=IN IP4 10.0.5.137
t=0 0
m=audio 28318 RTP/AVP 8 18 4 108 109 34 0 13 117
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:108 G726-24/8000
a=rtpmap:109 G726-32/8000
a=rtpmap:34 L16/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:117 CN/16000
a=fmtp:18 annexb=yes
a=fmtp:4 annexa=yes
a=ptime:20
a=rtcp:24498
a=sendrecv
a=silenceSupp:on - - - -
<------------->
--- (14 headers 21 lines) ---
Sending to 10.0.5.137:5060 (NAT)
Sending to 10.0.5.137:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '938810914' for '938810914' from 10.0.5.137:5060
<--- Reliably Transmitting (no NAT) to 10.0.5.137:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.5.137;branch=z9hG4bKb7d38b6e868471c66;received=10.0.5.137;rport=5060
From: 938810914 <sip:[email protected]>;tag=9531a2705f
To: <sip:[email protected]:5060>;tag=as3c9d28e4
Call-ID: [email protected]
CSeq: 983478748 INVITE
Server: FPBX-14.0.13.12(13.27.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17ec9ec6"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.0.5.137:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.137;branch=z9hG4bKb7d38b6e868471c66;rport
Max-Forwards: 70
From: 938810914 <sip:[email protected]>;tag=9531a2705f
To: <sip:[email protected]:5060>;tag=as3c9d28e4
Call-ID: [email protected]
CSeq: 983478748 ACK
User-Agent: PTInS GR2402GA Build 3RGW030500m139
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:10.0.5.137:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.137;branch=z9hG4bK844a8330fa25fcb5e;rport
Max-Forwards: 70
From: 938810914 <sip:[email protected]>;tag=9531a2705f
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 983478749 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, SUBSCRIBE, UPDATE
Authorization: Digest username="938810914",realm="asterisk",nonce="17ec9ec6",uri="sip:[email protected]:5060",response="551d70131be86df176b32068796532bc",algorithm=MD5
Contact: <sip:[email protected]>
Supported: 100rel, answermode, early-session, eventlist, histinfo, join, path, replaces, tdialog, timer
User-Agent: PTInS GR2402GA Build 3RGW030500m139
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 457
v=0
o=MxSIP 889 889 IN IP4 10.0.5.137
s=CMGR_20_DSL_CLIENT
c=IN IP4 10.0.5.137
t=0 0
m=audio 28318 RTP/AVP 8 18 4 108 109 34 0 13 117
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:108 G726-24/8000
a=rtpmap:109 G726-32/8000
a=rtpmap:34 L16/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:117 CN/16000
a=fmtp:18 annexb=yes
a=fmtp:4 annexa=yes
a=ptime:20
a=rtcp:24498
a=sendrecv
a=silenceSupp:on - - - -
<------------->
--- (15 headers 21 lines) ---
Sending to 10.0.5.137:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '938810914' for '938810914' from 10.0.5.137:5060
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 108
Found RTP audio format 109
Found RTP audio format 34
Found RTP audio format 0
Found RTP audio format 13
Found RTP audio format 117
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found unknown media description format G726-24 for ID 108
Found audio description format G726-32 for ID 109
Found audio description format L16 for ID 34
Found audio description format PCMU for ID 0
Found audio description format CN for ID 13
Found unknown media description format CN for ID 117
Capabilities: us - (ulaw|alaw|g729|g722|gsm|g726), peer - audio=(ulaw|g723|alaw|g729|slin16|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729|g726)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x2 (CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 10.0.5.137:28318
Looking for 901501004 in from-internal (domain 10.0.2.1)
sip_route_dump: route/path hop: <sip:[email protected]>
<--- Transmitting (no NAT) to 10.0.5.137:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.5.137;branch=z9hG4bK844a8330fa25fcb5e;received=10.0.5.137;rport=5060
From: 938810914 <sip:[email protected]>;tag=9531a2705f
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 983478749 INVITE
Server: FPBX-14.0.13.12(13.27.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
[2019-11-26 15:01:02] WARNING[25153][C-00084d88]: ast_expr2.fl:470 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input:
 = 1 & 0 = 0
 ^
[2019-11-26 15:01:02] WARNING[25153][C-00084d88]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
[2019-11-26 15:01:02] WARNING[25153][C-00084d88]: ast_expr2.fl:470 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input:
 = 1 & 0 = 0
 ^
[2019-11-26 15:01:02] WARNING[25153][C-00084d88]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
Audio is at 14264
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding codec g726 to SDP
<--- Transmitting (no NAT) to 10.0.5.137:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.5.137;branch=z9hG4bK844a8330fa25fcb5e;received=10.0.5.137;rport=5060
From: 938810914 <sip:[email protected]>;tag=9531a2705f
To: <sip:[email protected]:5060>;tag=as4b36f3e9
Call-ID: [email protected]
CSeq: 983478749 INVITE
Server: FPBX-14.0.13.12(13.27.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 290
v=0
o=root 1252813700 1252813700 IN IP4 10.0.2.1
s=Asterisk PBX 13.27.1
c=IN IP4 10.0.2.1
t=0 0
m=audio 14264 RTP/AVP 0 8 18 109
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:109 G726-32/8000
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
Audio is at 14264
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g729 to SDP
Adding codec g726 to SDP
<--- Reliably Transmitting (no NAT) to 10.0.5.137:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.5.137;branch=z9hG4bK844a8330fa25fcb5e;received=10.0.5.137;rport=5060
From: 938810914 <sip:[email protected]>;tag=9531a2705f
To: <sip:[email protected]:5060>;tag=as4b36f3e9
Call-ID: [email protected]
CSeq: 983478749 INVITE
Server: FPBX-14.0.13.12(13.27.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
P-Asserted-Identity: "CID:938810914" <sip:[email protected]>
Content-Type: application/sdp
Require: timer
Content-Length: 290
v=0
o=root 1252813700 1252813700 IN IP4 10.0.2.1
s=Asterisk PBX 13.27.1
c=IN IP4 10.0.2.1
t=0 0
m=audio 14264 RTP/AVP 0 8 18 109
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:109 G726-32/8000
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
<--- SIP read from UDP:10.0.5.137:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.137;branch=z9hG4bK4a02624b73424e359;rport
Max-Forwards: 70
From: 938810914 <sip:[email protected]>;tag=9531a2705f
To: <sip:[email protected]:5060>;tag=as4b36f3e9
Call-ID: [email protected]
CSeq: 983478749 ACK
Authorization: Digest username="938810914",realm="asterisk",nonce="17ec9ec6",uri="sip:[email protected]:5060",response="551d70131be86df176b32068796532bc",algorithm=MD5
User-Agent: PTInS GR2402GA Build 3RGW030500m139
Content-Length: 0
<------------->
--- (10 headers 0 lines) --->`Texto preformateado`

Kind regards

Looks clean to me too. What is the provider saying?

<— SIP read from UDP:10.0.5.137:5060 —> ACK sip:[email protected]:5060 SIP/2.0

Answer was acknowledged so you should check logs at the other end, 10.0.5.137

Need to see a verbose output of the call so we can see where those errors are being generated. Based on a report from yesterday by someone it could be in the outbound-callerid macro. That could be causing the issue.

Those are warnings and you would get those when value of one of the variables in comparison is blank.

I know exactly what they are and wanted to see where they were being generated. Because while they are just warnings it still could impact other things because formatting was messed up. Also it’s not that they a blank, I’ve set vars with no value and compared them in If and other statements, that doesn’t happen. That more points to the fact that a var that is being check isn’t set either will a null/empty value or a value. So why isn’t that var being set? What is the problem in the dialplan that is causing this issue?

That’s what I wanted to see.

satish-dev*CLI> dialplan show testc

[ Context ‘testc’ created by ‘pbx_config’ ]

‘s’ => 1. Set(v=) [pbx_config]

  1. ExecIf($[${v} = 1]?NoOp(true):NoOp(false)) [pbx_config]

-= 1 extension (2 priorities) in 1 context. =-

satish-dev*CLI>

satish-dev*CLI> channel originate local/[email protected] application wait 5

– Executing [[email protected]:1] Set (" Local/[email protected];2 ", " v= ") in new stack

[2019-11-27 09:54:09] WARNING [3612][C-00000003]: ast_expr2.fl : 470 ast_yyerror : ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:

= 1

^

[2019-11-27 09:54:09] WARNING [3612][C-00000003]: ast_expr2.fl : 474 ast_yyerror : If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables

– Executing [[email protected]:2] ExecIf (" Local/[email protected];2 ", " ?NoOp(true):NoOp(false) ") in new stack

How can I extract the information you are requesting?

In the Asterisk console do: core set verbose 10 then make the call again.

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