Solved issue

Turns out the person with cell/bria had my extension blocked in their bria. And of course they say they didn’t put the block there.

Your log shows BUSY, not UNAVAILABLE!

Hi David55, thanks for replying. I saw CHANUAVAIL but looking again, it does say busy/congested. The FACT is there is no one using the extension. How do I know? The person with the softphone was sitting beside me, I called their extension 726 from my A30 phone using ext. 471 and it goes to vmail after 1 ring. Do exact same with my Bria ext. 471 and it does exactly same… 1 ring and vmail. The person with 726 was as I mentioned above, sitting right here and their phone did not ring.

Then with my A30 phone, dial 726 with a different line (different ext #) and their softphone rings. Do same from my Bria again with a different line (ext) and again their ext. 726 rings.

All lines/ext’s on my A30 & Bria all go to the same fpbx.

All the lines on the A30 and my Bria are configured the same… PJSIP/TLS/SRTP and only IPv4.

It’s specifically busy, not congested. The three numbers separated by /s are the number of the contacts that returned busy, congested, and unavailable, respectively (the number before the : is the total number included i the dial call). Also the BUSY in s-BUSY is the result of expanding the DIALSTATUS variable.

Asterisk is actually quite fussy about what it will report as busy, and the CoS and ToS lines indicate that it actually tried to set up the call, rather than making a local decision that the device was busy, so I think one can confidently say that the phone, itself, thought it was busy.

A SIP trace would be useful here, to see if an INVITE actually went out, and if it was rejected for some reason.

Hi David,
I think you are missing the part when I said the person was sitting right beside me with their cell/bria and they were defintely not on any call. So when I call from just ONE extension, their phone magically thinks it’s on a call but calling from another ext. on same devices the other phone rings and can answer the call?? Makes no sense.

Which sip debug? For PJSIP it’s “pjsip set logger on”, and I’d expect at least SIP traffic for the incoming call leg.

Hi JColp,

It does show busy here below but the endpoint ext. 726 is NOT using the phone. You have to trust me on that.

<--- Received SIP request (864 bytes) from TLS:77.137.66.143:19973 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.14:1290;branch=z9hG4bK40141394042663511988
From: PHIL <sip:[email protected]>;tag=6546621740
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:1290;transport=tls>
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Asterisk A30 Phone 1.6.2.2 000fd3cf185a
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 285

v=0
o=471 1145951630 2530154369 IN IP4 192.168.1.14
s=A conversation
c=IN IP4 192.168.1.14
t=0 0
m=audio 10064 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:koOxSfgodwGf8+NhJFeiFEwe9RJUmltDiYGOTeDM
a=sendrecv

<--- Transmitting SIP response (502 bytes) to TLS:77.137.66.143:19973 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 192.168.1.14:1290;rport=19973;received=77.137.66.143;branch=z9hG4bK40141394042663511988
Call-ID: [email protected]
From: "PHIL" <sip:[email protected]>;tag=6546621740
To: <sip:[email protected]>;tag=z9hG4bK40141394042663511988
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1644339404/caf2d25dbbe0ddee0dbafbea6ea7397e",opaque="4669d14452620b0f",algorithm=md5,qop="auth"
Server: FPBX-15.0.17.68(18.6.0)
Content-Length:  0


<--- Received SIP request (408 bytes) from TLS:77.137.66.143:19973 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.14:1290;branch=z9hG4bK40141394042663511988
From: "PHIL" <sip:[email protected]>;tag=6546621740
To: <sip:[email protected]>;tag=z9hG4bK40141394042663511988
Call-ID: [email protected]
CSeq: 1 ACK
Contact: <sip:[email protected]:1290;transport=tls>
User-Agent: Asterisk A30 Phone 1.6.2.2 000fd3cf185a
Max-Forwards: 70
Content-Length: 0


<--- Received SIP request (1123 bytes) from TLS:77.137.66.143:19973 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.14:1290;branch=z9hG4bK12828467423522821204
From: PHIL <sip:[email protected]>;tag=6546621740
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:1290;transport=tls>
Authorization: Digest username="471", realm="asterisk", nonce="1644339404/caf2d25dbbe0ddee0dbafbea6ea7397e", uri="sip:[email protected]", response="d14b602cff59ab4f001aaa97d74dcc0f", algorithm=MD5, cnonce="ebcb82d2", opaque="4669d14452620b0f", qop=auth, nc=00030963
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Asterisk A30 Phone 1.6.2.2 000fd3cf185a
P-Early-Media: supported
Allow-Events: talk
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 285

v=0
o=471 1145951630 2530154369 IN IP4 192.168.1.14
s=A conversation
c=IN IP4 192.168.1.14
t=0 0
m=audio 10064 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:koOxSfgodwGf8+NhJFeiFEwe9RJUmltDiYGOTeDM
a=sendrecv

<--- Transmitting SIP response (318 bytes) to TLS:77.137.66.143:19973 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.1.14:1290;rport=19973;received=77.137.66.143;branch=z9hG4bK12828467423522821204
Call-ID: [email protected]
From: "PHIL" <sip:[email protected]>;tag=6546621740
To: <sip:[email protected]>
CSeq: 2 INVITE
Server: FPBX-15.0.17.68(18.6.0)
Content-Length:  0


[2022-02-08 11:56:45] WARNING[19630][C-00000054]: func_strings.c:1442 function_eval: EVAL requires an argument: EVAL(<string>)
<--- Transmitting SIP request (1175 bytes) to TLS:77.139.237.105:45478 --->
INVITE sip:[email protected]:45478;transport=TLS;rinstance=da1d7e9c04860999 SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.37.86:5161;rport;branch=z9hG4bKPj0c2532bc-1808-447a-9e6f-031edaadd277;alias
From: "Phil" <sip:[email protected]>;tag=4e685fb8-f5fa-4243-8f1e-7c4bad071775
To: <sip:[email protected];rinstance=da1d7e9c04860999>
Contact: <sip:[email protected]:5161;transport=TLS>
Call-ID: 1f7bd151-968a-4462-a675-288e72b6b70e
CSeq: 18107 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: "Phil" <sip:[email protected]>;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: FPBX-15.0.17.68(18.6.0)
Content-Type: application/sdp
Content-Length:   324

v=0
o=- 1783384131 1783384131 IN IP4 xxx.xxx.37.86
s=Asterisk
c=IN IP4 xxx.xxx.37.86
t=0 0
m=audio 12866 RTP/SAVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RCPdkGcO5s9H6pE6TUdMG0vYn8uiUHx9uckufHX3
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (596 bytes) to TLS:77.137.66.143:19973 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.1.14:1290;rport=19973;received=77.137.66.143;branch=z9hG4bK12828467423522821204
Call-ID: [email protected]
From: "PHIL" <sip:[email protected]>;tag=6546621740
To: <sip:[email protected]>;tag=5a596d2d-311e-4178-8e7c-1223f4fca162
CSeq: 2 INVITE
Server: FPBX-15.0.17.68(18.6.0)
Contact: <sip:xxx.xxx.37.86:5161;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Remote-Party-ID: "Niri" <sip:[email protected]>;party=called;privacy=off;screen=no
Content-Length:  0


<--- Received SIP response (352 bytes) from TLS:77.139.237.105:45478 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS xxx.xxx.37.86:5161;rport=5161;branch=z9hG4bKPj0c2532bc-1808-447a-9e6f-031edaadd277;alias
To: <sip:[email protected];rinstance=da1d7e9c04860999>
From: "Phil" <sip:[email protected]>;tag=4e685fb8-f5fa-4243-8f1e-7c4bad071775
Call-ID: 1f7bd151-968a-4462-a675-288e72b6b70e
CSeq: 18107 INVITE
Content-Length: 0


<--- Received SIP response (420 bytes) from TLS:77.139.237.105:45478 --->
**SIP/2.0 486 Busy Here**
Via: SIP/2.0/TLS xxx.xxx.37.86:5161;rport=5161;branch=z9hG4bKPj0c2532bc-1808-447a-9e6f-031edaadd277;alias
To: <sip:[email protected];rinstance=da1d7e9c04860999>;tag=555094bb
From: "Phil" <sip:[email protected]>;tag=4e685fb8-f5fa-4243-8f1e-7c4bad071775
Call-ID: 1f7bd151-968a-4462-a675-288e72b6b70e
CSeq: 18107 INVITE
User-Agent: Bria Mobile Android 6.6.3 build 129673
Content-Length: 0


<--- Transmitting SIP request (476 bytes) to TLS:77.139.237.105:45478 --->
ACK sip:[email protected]:45478;transport=TLS;rinstance=da1d7e9c04860999 SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.37.86:5161;rport;branch=z9hG4bKPj0c2532bc-1808-447a-9e6f-031edaadd277;alias
From: "Phil" <sip:[email protected]>;tag=4e685fb8-f5fa-4243-8f1e-7c4bad071775
To: <sip:[email protected];rinstance=da1d7e9c04860999>;tag=555094bb
Call-ID: 1f7bd151-968a-4462-a675-288e72b6b70e
CSeq: 18107 ACK
Max-Forwards: 70
User-Agent: FPBX-15.0.17.68(18.6.0)
Content-Length:  0


<--- Transmitting SIP response (997 bytes) to TLS:77.137.66.143:19973 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.14:1290;rport=19973;received=77.137.66.143;branch=z9hG4bK12828467423522821204
Call-ID: [email protected]
From: "PHIL" <sip:[email protected]>;tag=6546621740
To: <sip:[email protected]>;tag=5a596d2d-311e-4178-8e7c-1223f4fca162
CSeq: 2 INVITE
Server: FPBX-15.0.17.68(18.6.0)
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:xxx.xxx.37.86:5161;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Remote-Party-ID: "Niri" <sip:[email protected]>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   324

v=0
o=- 1145951630 2530154371 IN IP4 xxx.xxx.37.86
s=Asterisk
c=IN IP4 xxx.xxx.37.86
t=0 0
m=audio 17768 RTP/SAVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:FaKlgR3rh8nmRF13WAhFZPITNGxDa6fxv5RSvmDg
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (438 bytes) from TLS:77.137.66.143:19973 --->
ACK sip:xxx.xxx.37.86:5161;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.1.14:1290;branch=z9hG4bK55091580365618155022
From: PHIL <sip:[email protected]>;tag=6546621740
To: <sip:[email protected]>;tag=5a596d2d-311e-4178-8e7c-1223f4fca162
Call-ID: [email protected]
CSeq: 2 ACK
Contact: <sip:[email protected]:1290;transport=tls>
Max-Forwards: 70
User-Agent: Asterisk A30 Phone 1.6.2.2 000fd3cf185a
Content-Length: 0

Well, that’s what the endpoint sent back. The log message is exactly what was received back from it.

1 Like

Could the user have put it into a local Do Not Disturb state?

The problem is with the phone.

David, you were dead on. There is a place to block callers. Found it on my bria and told the person to go look and yup, there it was.
Thanks for your help.

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