@Lutiana wrote:

I have a Cisco 7975G setup with SIP firmware and registered with Asterisk. So now I am just tweaking the XML setup for my use using this site: http://usecallmanager.nz/sepmac-cnf-xml.html

So far so good, but I am having an odd thing happen with the screen timeouts you can set. I have them configured like this:

 <daysDisplayNotActive></daysDisplayNotActive>
 <displayOnTime>06:00</displayOnTime>
 <displayOnDuration>17:00</displayOnDuration>
 <displayIdleTimeout>00:10</displayIdleTimeout>
 <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
 <backlightOnTime></backlightOnTime>
 <backlightOnDuration></backlightOnDuration>
 <backlightIdleTimeout>00:10</backlightIdleTimeout>
 <backlightOnWhenIncomingCall>1</backlightOnWhenIncomingCall>

The active display settings seem to work fine, however when they kick in any calls to the extension are immediately sent to voicemail. You can, however, pickup the phone and make a call with no problem.

And then there is the backlight settings here, they don’t seem to do anything.

My goal here is to basically have the screen on for most of the day, but have the backlight turn off after being idle for 20 minutes, but otherwise the phone should work as it usually does and not kick calls to voicemail after the screen is turned off.

Is this the expected behaviour? If so, then why is there an option to turn on the screen when a call comes in?

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@psdk wrote:

Hi folks,

Is it possible to create inbound and outbound through API module? if yes, is there any sample?
Wiki page for this module is not good.

BR

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@chris_unit wrote:

I’m about to roll out softphones for our staff in the office.

these will work over 3G and wifi.

they’re going to quickly realise they can use the softphone for personal calls any time.

i’d like to limit outbound calls from the softphone outside of working hours and weekends.

how could i go about this?

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@Prodna wrote:

Hi;

We dynamically add the extensions to the queue with “AMI”. We accept one call for each extension.
However, in a way that we cannot understand, sometimes the call drops as soon as it transfers to the call extension. Asteriks hangup cause comes as “16-Normal clearing”.
(Asterisk Version: 16.6.2)

What could be the reason for you?

queue.conf

[100001]
announce_frequency=0
announce_holdtime=no
announce_position=no
autofill=yes
autopause=no
autopausebusy=no
autopausedelay=0
autopauseunavail=no
joinempty=no
leavewhenempty=penalty,paused,invalid,unavailable,inuse,ringing
maxlen=0
memberdelay=0
min_announce_frequency=15
monitor_join=yes
musicclass=calmasesi
penaltymemberslimit=0
periodic_announce_frequency=0
queue_callswaiting=silence/1
queue_thereare=silence/1
queue_youarenext=silence/1
reportholdtime=no
retry=1
ringinuse=no
servicelevel=60
strategy=rrmemory
timeout=15
timeoutpriority=conf
timeoutrestart=no
weight=0
wrapuptime=5
context=

pjsip_endpoint.conf

[147] ;extension
type=endpoint
aors=147
auth=147-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,g723,g729,g726,g722,gsm,ilbc,opus
context=RCOPOUT_A
callerid=7@admin2 <147>
dtmf_mode=rfc4733
aggregate_mwi=yes
use_avpf=yes
rtcp_mux=yes
bundle=no
ice_support=yes
media_use_received_transport=yes
trust_id_inbound=yes
send_connected_line=yes
media_encryption=dtls
timers=yes
media_encryption_optimistic=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
dtls_rekey=0
dtls_cert_file=/etc/asterisk/keys/default.crt
dtls_private_key=/etc/asterisk/keys/default.key

cdr_logs

Time Event CNAM CNUM ANI DID AMA exten context App channel UserDefType EventExtra CEL Table
Sat, 23 May 2020 18:07 CHAN_START DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_START DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 CHAN_START DEFAULT s from-pstn PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 ANSWER DIAL DEFAULT DIAL from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 ANSWER 49345***** DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 ANSWER DIAL 49345***** DEFAULT DIAL RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_ENTER 49345***** DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 APP_START DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_START 1@master3 4 DEFAULT s RCOPOUT_A PJSIP/4-0000006f
Sat, 23 May 2020 18:07 ANSWER 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 APP_START 1@master3 4 4 DEFAULT s RCOUT_A_KUYRUK_UYE MixMonitor PJSIP/4-0000006f
Sat, 23 May 2020 18:07 APP_END 1@master3 4 4 DEFAULT s RCOUT_A_KUYRUK_UYE MixMonitor PJSIP/4-0000006f
Sat, 23 May 2020 18:07 BRIDGE_ENTER 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_ENTER DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 LOCAL_OPTIMIZE DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT 1@master3 4 DEFAULT DIAL RCOUT_A Dial Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 HANGUP 1@master3 4 DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 CHAN_END 1@master3 4 DEFAULT DIAL RCOUT_A Local/DIAL@RCOUT_A-0000005c;2
Sat, 23 May 2020 18:07 APP_END DIAL 49345***** DEFAULT ANSWERED RCOUT_A Queue Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 HANGUP DIAL 49345***** DEFAULT ANSWERED RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 CHAN_END DIAL 49345***** DEFAULT ANSWERED RCOUT_A AppDial2 Local/DIAL@RCOUT_A-0000005c;1
Sat, 23 May 2020 18:07 BRIDGE_EXIT DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 BRIDGE_EXIT 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 HANGUP 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 CHAN_END 1@master3 4 4 DEFAULT ANSWERED RCOPOUT_A AppQueue PJSIP/4-0000006f
Sat, 23 May 2020 18:07 HANGUP DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Time Event CNAM CNUM ANI DID AMA exten context App channel UserDefType EventExtra CEL Table
Sat, 23 May 2020 18:07 CHAN_END DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Sat, 23 May 2020 18:07 LINKEDID_END DIAL DEFAULT from-pstn AppDial PJSIP/t_1_14-0000006d
Related Call Detail Records
Call Date Recording System CallerID Outbound CallerID DID App Destination Disposition Duration Userfield Account CDR Table CDR Graph
Sat, 23 May 2020 18:07 1.590.250.051.295 DIAL Return ANSWERED 00:00 89098
Sat, 23 May 2020 18:07 o DIAL Queue ANSWERED ANSWERED 00:11 89098
Sat, 23 May 2020 18:07 o 4 Dial DIAL ANSWERED 00:11 89098

asterisk_logs

<— Received SIP response (956 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;branch=z9hG4bKPj39a93181-2955-47ac-85bb-76d174751ea2;rport=8000
Record-Route: sip:87.238.XXX.XX;lr;ep
Contact: sip:87.238.XXX.XX:5074
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13652 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
Content-Length: 392

v=0
o=PortaSIP 2130776891631118155 1 IN IP4 87.238.XXX.XX
s=Phone Call via hiQ9200 SIPCA
t=0 0
m=audio 41492 RTP/AVP 8 0 18 101
c=IN IP4 87.238.XXX.XX
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20

-- PJSIP/t_1_14-0000006d answered Local/DIAL@RCOUT_A-0000005c;2
-- Local/DIAL@RCOUT_A-0000005c;1 answered
-- Executing [ANSWERED@RCOUT_A:1] Answer("Local/DIAL@RCOUT_A-0000005c;1", "") in new stack

<— Transmitting SIP request (450 bytes) to UDP:87.238.XXX.XX:5060 —>
ACK sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj866061f5-366f-415d-9c15-6181c45325f3
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13652 ACK
Route: sip:87.238.XXX.XX;lr;ep
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

-- Executing [ANSWERED@RCOUT_A:2] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCUID=o-100016-41-182520") in new stack
-- Executing [ANSWERED@RCOUT_A:3] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCTIP=o") in new stack
-- Executing [ANSWERED@RCOUT_A:4] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCKID=100016") in new stack
-- Executing [ANSWERED@RCOUT_A:5] Set("Local/DIAL@RCOUT_A-0000005c;1", "RCAID=182520") in new stack
-- Channel PJSIP/t_1_14-0000006d joined 'simple_bridge' basic-bridge <e1d70cb4-56b5-46b4-8cdf-03f2f6d5e42d>
-- Channel Local/DIAL@RCOUT_A-0000005c;2 joined 'simple_bridge' basic-bridge <e1d70cb4-56b5-46b4-8cdf-03f2f6d5e42d>
-- Executing [ANSWERED@RCOUT_A:6] Verbose("Local/DIAL@RCOUT_A-0000005c;1", "1, RCOUT 3 PJSIP/0001498454*****@t_1_14 - o-100016-41-182520 - Local/DIAL@RCOUT_A-0000005c;1 - 2020-05-23 18:07:42") in new stack

RCOUT 3 PJSIP/0001498454*****@t_1_14 - o-100016-41-182520 - Local/DIAL@RCOUT_A-0000005c;1 - 2020-05-23 18:07:42
– Executing [ANSWERED@RCOUT_A:7] Gosub(“Local/DIAL@RCOUT_A-0000005c;1”, “RCAMD,s,1(100016,182520,0)”) in new stack
– Executing [s@RCAMD:1] Answer(“Local/DIAL@RCOUT_A-0000005c;1”, “”) in new stack
– Executing [s@RCAMD:2] Verbose(“Local/DIAL@RCOUT_A-0000005c;1”, “1,AMD BASLADI 100016-182520-0”) in new stack
AMD BASLADI 100016-182520-0
– Executing [s@RCAMD:3] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “DURUM=ANSWER”) in new stack
– Executing [s@RCAMD:4] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “1?son”) in new stack
– Goto (RCAMD,s,14)
– Executing [s@RCAMD:14] Return(“Local/DIAL@RCOUT_A-0000005c;1”, “ANSWER”) in new stack
– Executing [ANSWERED@RCOUT_A:8] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “RCSTATU=ANSWER”) in new stack
– Executing [ANSWERED@RCOUT_A:9] Set(“Local/DIAL@RCOUT_A-0000005c;1”, “GELEN=1”) in new stack
– Executing [ANSWERED@RCOUT_A:10] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “1?atla”) in new stack
– Goto (RCOUT_A,ANSWERED,14)
– Executing [ANSWERED@RCOUT_A:14] NoOp(“Local/DIAL@RCOUT_A-0000005c;1”, “”) in new stack
– Executing [ANSWERED@RCOUT_A:15] GotoIf(“Local/DIAL@RCOUT_A-0000005c;1”, “0?kapat”) in new stack
– Executing [ANSWERED@RCOUT_A:16] Queue(“Local/DIAL@RCOUT_A-0000005c;1”, “100016,tn,10,RCOUT_A_KUYRUK_UYE”) in new stack
– Started music on hold, class ‘calmasesi’, on channel ‘Local/DIAL@RCOUT_A-0000005c;1’
– Called PJSIP/4
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
== DTLS ECDH initialized (automatic), faster PFS enabled
<— Transmitting SIP request (1692 bytes) to WSS:178.246.XXX.XX:18439 —>
INVITE sip:vujiu9og@178.246.XXX.XX:18439;transport=ws SIP/2.0
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX
Contact: sip:asterisk@pbxtest.xxxxx.com:5060;transport=ws
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
CSeq: 1327 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: sip:DIAL@pbxtest.xxxxx.com
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Type: application/sdp
Content-Length: 927

v=0
o=- 1690486252 1690486252 IN IP4 85.95.XXX.XX
s=Asterisk
c=IN IP4 85.95.XXX.XX
t=0 0
m=audio 24360 UDP/TLS/RTP/SAVPF 0 8 4 18 111 9 3 97 107 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 49:42:49:29:BE:E1:65:AD:1B:80:7E:E1:75:5D:5F:E3:63:FC:8D:D7:32:D6:94:98:BC:4C:33:D0:7B:1D:36:0B
a=ice-ufrag:310c33e03aa30526508942d35514483e
a=ice-pwd:292f7726449b35512adb649f416e5aa0
a=candidate:Hbfbf9d57 1 UDP 2130706431 fe80::e050:1636:aa00:252e 24360 typ host
a=candidate:H555ff211 1 UDP 2130706431 85.95.XXX.XX 24360 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux

<— Received SIP response (370 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Content-Length: 0

<— Received SIP response (436 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Contact: sip:vujiu9og@192.0.X.XXX;transport=ws
Content-Length: 0

-- PJSIP/4-0000006f is ringing
-- PJSIP/4-0000006f is ringing

<— Received SIP response (1901 bytes) from WSS:178.246.XXX.XX:18439 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPjfdfe26c6-e26f-4198-a7ee-2b40b65a9d13;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
CSeq: 1327 INVITE
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
Supported: outbound
User-Agent: SIP.js/0.14.4
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,SUBSCRIBE
Contact: sip:vujiu9og@192.0.X.XXX;transport=ws
Content-Type: application/sdp
Content-Length: 1356

v=0
o=- 4567787090042083750 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS 1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW
m=audio 18512 UDP/TLS/RTP/SAVPF 0 8 9 107 101
c=IN IP4 178.246.XXX.XX
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1798856119 1 udp 2122260223 192.168.43.236 59130 typ host generation 0 network-id 1 network-cost 10
a=candidate:2813682212 1 udp 1686052607 178.246.XXX.XX 18512 typ srflx raddr 192.168.43.236 rport 59130 generation 0 network-id 1 network-cost 10
a=candidate:633053511 1 tcp 1518280447 192.168.43.236 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:weD2
a=ice-pwd:AJqFH6qGz+P6Yrdo45YWg6GB
a=ice-options:trickle
a=fingerprint:sha-256 3B:8D:ED:F8:C7:56:E8:95:F9:3D:78:5B:15:F2:40:19:37:9F:21:B7:7D:2E:6C:99:34:6D:9D:52:D2:B9:5B:84
a=setup:active
a=mid:0
a=sendrecv
a=msid:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW dd127927-5900-4267-aef5-b85b92d1d8cd
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 minptime=10;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=ssrc:397357961 cname:eq6+5jnRwp9EfZ6s
a=ssrc:397357961 msid:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW dd127927-5900-4267-aef5-b85b92d1d8cd
a=ssrc:397357961 mslabel:1gQ0FLUOf9GslA86Y7uBOZZ3ecACM0dkMbNW
a=ssrc:397357961 label:dd127927-5900-4267-aef5-b85b92d1d8cd

<— Transmitting SIP request (430 bytes) to WSS:178.246.XXX.XX:18439 —>
ACK sip:vujiu9og@178.246.XXX.XX:18439;transport=ws SIP/2.0
Via: SIP/2.0/WSS 85.95.XXX.XX:8002;rport;branch=z9hG4bKPj531dc48c-82c9-4c77-9e3b-a4a8c5bcb073;alias
From: sip:DIAL@pbxtest.xxxxx.com;tag=e2f7c171-1f8f-4a93-a8c2-628292ac491c
To: sip:vujiu9og@178.246.XXX.XX;tag=mlglunc7ma
Call-ID: e87b6de1-05dc-45ee-8d4d-1d7da862a6d5
CSeq: 1327 ACK
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

-- PJSIP/4-0000006f answered Local/DIAL@RCOUT_A-0000005c;1
-- Stopped music on hold on Local/DIAL@RCOUT_A-0000005c;1
-- PJSIP/4-0000006f Internal Gosub(RCOUT_A_KUYRUK_UYE,s,1) start
-- Executing [s@RCOUT_A_KUYRUK_UYE:1] NoOp("PJSIP/4-0000006f", "") in new stack
-- Executing [s@RCOUT_A_KUYRUK_UYE:2] Verbose("PJSIP/4-0000006f", "1, RCOUT 6  - o-100016-41-182520 - PJSIP/4-0000006f - RCOUT_A_KUYRUK_UYE - PJSIP/4-0000006f - s") in new stack

RCOUT 6 - o-100016-41-182520 - PJSIP/4-0000006f - RCOUT_A_KUYRUK_UYE - PJSIP/4-0000006f - s
– Executing [s@RCOUT_A_KUYRUK_UYE:3] Set(“PJSIP/4-0000006f”, “RCUID=o-100016-41-182520”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:4] Set(“PJSIP/4-0000006f”, “RCKID=100016”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:5] Set(“PJSIP/4-0000006f”, “RCDTID=41”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:6] Set(“PJSIP/4-0000006f”, “RCAID=182520”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:7] Set(“PJSIP/4-0000006f”, “RECDOSYA=/var/spool/asterisk/recording/182520-”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:8] Set(“PJSIP/4-0000006f”, “GELEN=1”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:9] GotoIf(“PJSIP/4-0000006f”, “1?atla”) in new stack
– Goto (RCOUT_A_KUYRUK_UYE,s,13)
– Executing [s@RCOUT_A_KUYRUK_UYE:13] NoOp(“PJSIP/4-0000006f”, “”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:14] UserEvent(“PJSIP/4-0000006f”, “RCDialBegin_Op,RCAID:182520,RCDTID:41,RCUYE:4,RCDNID:89098”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:15] MixMonitor(“PJSIP/4-0000006f”, “/var/spool/asterisk/recording/182520-sistem.wav,a”) in new stack
– Executing [s@RCOUT_A_KUYRUK_UYE:16] Return(“PJSIP/4-0000006f”, “”) in new stack
== Begin MixMonitor Recording PJSIP/4-0000006f
== Spawn extension (RCOPOUT_A, ANSWERED, 1) exited non-zero on ‘PJSIP/4-0000006f’
– PJSIP/4-0000006f Internal Gosub(RCOUT_A_KUYRUK_UYE,s,1) complete GOSUB_RETVAL=
– Channel PJSIP/4-0000006f joined ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel Local/DIAL@RCOUT_A-0000005c;1 joined ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel PJSIP/t_1_14-0000006d left ‘simple_bridge’ basic-bridge
– Channel Local/DIAL@RCOUT_A-0000005c;1 left ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel PJSIP/t_1_14-0000006d swapped with Local/DIAL@RCOUT_A-0000005c;1 into ‘simple_bridge’ basic-bridge <0c7ae476-3a83-4f55-8d77-d7271181acca>
– Channel Local/DIAL@RCOUT_A-0000005c;2 left ‘simple_bridge’ basic-bridge
== Spawn extension (RCOUT_A, DIAL, 8) exited non-zero on ‘Local/DIAL@RCOUT_A-0000005c;2’
<— Transmitting SIP request (1035 bytes) to UDP:87.238.XXX.XX:5060 —>
INVITE sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Contact: sip:asterisk@85.95.XXX.XX:8000
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Route: sip:87.238.XXX.XX;lr;ep
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Type: application/sdp
Content-Length: 308

v=0
o=- 1908267525 1908267526 IN IP4 85.95.XXX.XX
s=Asterisk
c=IN IP4 85.95.XXX.XX
t=0 0
m=audio 23890 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
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:150
a=sendrecv

== Spawn extension (RCOUT_A, ANSWERED, 16) exited non-zero on ‘Local/DIAL@RCOUT_A-0000005c;1’
<— Received SIP response (352 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport=8000;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Content-Length: 0

<— Received SIP response (915 bytes) from UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;branch=z9hG4bKPj17aae872-b13f-4b1e-ab0c-9f04657c2d91;rport=8000
Contact: sip:87.238.XXX.XX:5074
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 INVITE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
Content-Type: application/sdp
Server: PortaSIP
Content-Length: 392

v=0
o=PortaSIP 2130776891631118155 2 IN IP4 87.238.XXX.XX
s=Phone Call via hiQ9200 SIPCA
t=0 0
m=audio 41492 RTP/AVP 0 8 18 101
c=IN IP4 87.238.XXX.XX
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20

<— Transmitting SIP request (450 bytes) to UDP:87.238.XXX.XX:5060 —>
ACK sip:87.238.XXX.XX:5074 SIP/2.0
Via: SIP/2.0/UDP 85.95.XXX.XX:8000;rport;branch=z9hG4bKPj12e9a84e-da11-4215-87e6-8ab565bdf41b
From: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
To: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 13653 ACK
Route: sip:87.238.XXX.XX;lr;ep
Max-Forwards: 70
User-Agent: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

<— Received SIP request (731 bytes) from UDP:87.238.XXX.XX:5060 —>
BYE sip:asterisk@85.95.XXX.XX:8000 SIP/2.0
Via: SIP/2.0/UDP 87.238.XXX.XX:5060;branch=z9hG4bK-524287-1—683e14e478b76ff31034c2a7e9ee6db3;rport
Via: SIP/2.0/UDP 87.238.XXX.XX:5074;branch=z9hG4bK-ga4exivxulbugknp;rport=5074
Max-Forwards: 69
Contact: sip:87.238.XXX.XX:5074
To: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
From: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
CSeq: 555 BYE
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS, UPDATE
User-Agent: PortaSIP
h323-conf-id: 1503134234-742797846-1530232655-1293871401
cisco-GUID: 1503134234-742797846-1530232655-1293871401
Content-Length: 0

<— Transmitting SIP response (487 bytes) to UDP:87.238.XXX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.238.XXX.XX:5060;rport=5060;received=87.238.XXX.XX;branch=z9hG4bK-524287-1—683e14e478b76ff31034c2a7e9ee6db3
Via: SIP/2.0/UDP 87.238.XXX.XX:5074;rport=5074;branch=z9hG4bK-ga4exivxulbugknp
Call-ID: 14ea6875-d488-4e5c-8109-9b12f1214bc6
From: sip:0001498454*****@87.238.XXX.XX;tag=W+21UxHZA73meptD.i
To: sip:49345*****@85.95.XXX.XX;tag=5ec9d5a7-1ec0-4bea-957e-90b2f50a6bf1
CSeq: 555 BYE
Server: FPBX-14.0.13.26(16.6.2)
Content-Length: 0

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@WB3FFV wrote:

I am running a current PBXact server with about 500 active extensions, which most are connected using PJsip. It’s running asterisk 16.9.0 for the release, and all modules are current. I am seeing the following errors in the logging, and I am wondering if something needs to be adjusted up for the extension count we have. It’s on a pretty rocking server, so not a problem at all to allow it to use more of the resources on the server.

Here are the errors I am seeing:

[2020-05-26 16:48:57] WARNING[26365][C-00000569] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 16:48:57] WARNING[26367][C-00000569] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 16:52:20] WARNING[29631][C-00000572] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:02:05] WARNING[4350][C-00000580] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:08:53] WARNING[9307][C-0000058e] taskprocessor.c: The ‘stasis/p:channel:all-000028ef’ task processor queue reached 500 scheduled tasks.
[2020-05-26 17:15:20] WARNING[13284][C-00000598] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:23:52] WARNING[15986][C-000005a3] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:32:49] WARNING[17784][C-000005a9] taskprocessor.c: The ‘stasis/p:channel:all-00002923’ task processor queue reached 500 scheduled tasks.
[2020-05-26 17:32:49] WARNING[17784][C-000005a9] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:39:21] WARNING[20522][C-000005af] taskprocessor.c: The ‘stasis/m:channel:all-00000f53’ task processor queue reached 500 scheduled tasks again.
[2020-05-26 17:40:21] WARNING[18975][C-000005af] taskprocessor.c: The ‘stasis/m:cache_pattern:1/channel:all-00000f52’ task processor queue reached 500 scheduled tasks again.

If anyone can offer any clues, or point me in the right direction it would be great…

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@douggoens wrote:

I recently deployed a Cisco SPA8000 which is an 8 channel trunking box.
Not specific to anything other than I am receiving inbound calls by the caller bypassing the freepbx and probably calling the phone directly. I have changed the ports from 5060 to other port numbers randomly.
Caller ID is 8888 or 1111 or 888 or other crap and when they answer, no caller.
Any advice? Is there somewhere I can tell the SPA8000 only to accept calls from one source server?
If so, where do I do this on the admin page?

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@Eric_THM wrote:

I’ve been running FreePBX on an EC2 instance since early April of this year and it has not been a smooth transition.

Ever since switching over we have calls cut out randomly on all the extensions. It can happen anywhere between 2-5x a day.
In light of current events we don’t even have many people on the phone at once. Only about 10 max (very small company). Before this was deployed my test calls didn’t seem to have any issues. However, please note I didn’t test the calls for very long, a few minutes at a time at different times of the day.

Complete list of symptoms:

• Calls cutting out (can be anywhere between less than a half a second, to 5 seconds)
• It can happen on all the calls that are live, or just on a couple of therm
• The times it cuts out are at random times. some days it’s heavier in the morning, other days it’s in the afternoon.
• The length of call doesn’t matter, the call itself can be only a few minutes in, or starting at a half hour or even hour in if a call goes that long.
• The extensions affected are all of them, and it seems to surge with all active calls before dying down and performing as expected for another couple of hours.
• Some surges cause very little interference (less than half a second), to a very noticeable disturbance (agents/clients both saying “You’re cutting out” to calls flat out dropping)
• The surges can be a couple of cuts before being fine, to being 5 or 6 times before finally straitening out
• Happens for both IB and OB calls.

Setup of our network:

• The FreePBX server is on an EC2 instance, connected to an S3 bucket for easy access/storage to our recordings.
• FreePBX version is 14.0.13.33.
• All extensions are using PJSIP
• We have an SD-WAN setup before it goes to our internal network, there’s no firewall rules setup to block VOIP calls from our specific network EC2 instance.
• We have two routers going to our locations each with QoS setup. Please note, these are home/small business routers and not commercial routers and our ‘locations’ is in the same place, only different suites in the building.

Here’s what has been tried so far:

• Setup QoS on the routers and the SD-WAN we have running to prioritize VOIP calls.
• Put data on one channel and voice on the other for the SD-WAN so VOIP calls have a dedicated channel.
• Each of our ISP’s have 500MB and the monitoring I have setup shows we don’t use more than 20MB at a time (we only have a few calls and people checking different map apps on stations for our clients)
• Double checked all connections (I know someone would’ve asked that at one point)
• Double-checked that the audio codecs on our VOIP Server and softphone are the same (currently using the softphone using MicroSIP).
• The onboard FreePBX firewall is setup to allow our ISP’s

Other notes:

• Whenever a surge happens I bring up one of the recordings to see if the recording is breaking up as well. The recording has no such issue. The recording catches everything.
• I’ve checked the EC2 instance via putty to see if it’s running high for some reason and the server is running at 10% or lower with plenty of VRAM to spare during the surges, so I figure it’s not the server or installation itself
• The logs on the FreePBX during one of these surges I haven’t seen anything pop out at me (although, I’ll be the first to admit that I’m no expert at these FreePBX logs). I don’t see any uptick or different errors on the FreePBX logs when a surge happens versus when there’s no issues at all.
• The changes I’ve been doing have made small improvements to the VOIP calls. Prioritizing VOIP traffic throughout the network, matching the audio codecs as well as switching over to less bandwidth heavy codecs, have made the surges less disruptive than before but hasn’t completely solved the problem. This tells me I might be on the right track.

The last idea I have is to change our network landscape so that our routers are no longer issuing DHCP and only having our SD-WAN do the job which in turn will have ALL of our internet traffic not be filtered by any router/switch and going right to the SD-WAN, with our routers only being WiFi hubs in a sense. However, I was told by the SD-WAN administrator it may only solve the issue for a couple of days. I’m currently still planning to make the change however.

At this point, I’m at a complete lost and am looking for ANY ideas of what might be happening to the FreePBX installation

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@truongtv wrote:

Hi guys,
I am working with FreePBX 15.0.16.49 and I want to monitor FXO/FXS port status as Busy, Idle…
How can I monitor it?

Thank you so much!

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@Odi wrote:

Hello,

i read the post from Eric and have the same issues sometimes. I will not enter his thread so i opened a new one for my case.

I have a TCPdump running and have actually one cut off. What is needed for informations from this dump?

Cutoff 1:
SIP Error 480 Request: BYE

I’m happy about every helping hand and I just want to get an idea whether it’s at freepbx or our ISP or any other ISP in Germany.

Regards
Nico

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@gathlamia wrote:

Hello, first time poster.

I have set up a small installation for a client with 1 trunk and 6 extensions on a raspberryPi. Due to the nature of the device, every SD card fails after a 2-3 months period, in which case I have to always keep an sd card backed up and ready with any changes I might make so that he has as minimal downtime as possible.

Is there a way I can migrate the installation from the pi to an sff - low consumption PC? I would like to avoid the hassle of unregistering and reregistering the product if that’s possible.

Best regards and thanks in advance,
George

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@PoDuck wrote:

I have had issues with my server becoming deactivated apparently at random. It happened again today. My server suffered a power loss, and when I restarted it, it was deactivated. I am now out of resets on the account, and I have purchased modules I can’t use. I assume I can contact someone and they can reset things for me, but if all it takes is a reboot to lose my activation, how am I supposed to keep this from happening again?

What is it that Sangoma uses to determine that the hardware has changed? My hardware is exactly the same, and it thinks it is different hardware.

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@milanolarry wrote:

I have Nextcloud (mariadb + Caddy + php7.3) in my Raspberry Pi. After installing Freepbx (php7.3 + Apache2 + mariadb + Asterisk), I can no longer log in to my Nextcloud. Any ideas what FreePBX changed during installation?

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@busster8 wrote:

All of the sudden have no access to some GUI menu options. in/out routes, Administrators, and some others.
Tried to restore from two previous backups, but getting an error. Error references core_usersastdb().

I assume the system has been hacked. Is there a way to recover? Get to user database and see what is there?

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@mobeus wrote:

It is possible to have a virtual extension (voicemail only) assigned as a user in UCP for easy voicemail retrieval?

Example would be after hours general voicemail box setup as ext 2201.

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@amardeeprana wrote:

Hi

I am facing high CPU usage spikes on asterisk service. Please help.

FreePBX 14.0.13.24

Thanks
Amardeep

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@byrnejb wrote:

We have a FreePBX system that has been running for two years. Last week we shut it down to clone the HDD as we suspected that the drive was about to fail from the smartctl reports. This turned out to be an error in interpretation due to the way Seagate report raw error rates.

However, after restarting the system we began to experience intermittent cut outs to audio reception on the extensions. We have made no direct changes to any of the configuration files, everything has been handled through the web interface. In any case there were no changes made there either since the last restart of Asterisk before the reboot.

After the reboot we also had problems with the fax lines not picking up. This was traced back to a corrupted astdb.sqlite3 database that was dealt with through the web interface by simply resubmitting the device and applying the configuration.

This morning in an attempt to clear up any residual sqlite3 problems we shut down asterisk and renamed astdb.sqlite3, and restarted asterisk to recreate the database from scratch. This asterisk did do. However, it also brought on the symptom where none of the internal extensions can accept calls, either internal or external. Outgoing calls are not affected.

This has killed our pbx. We have run fwconsol stop and fwconsole start but nothing has changed insofar as we can see.

This is the output fromfwconsole start:
fwconsole start
Running FreePBX startup…
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions…
Setting base permissions…Done
Setting specific permissions…
27381 [============================]
Finished setting permissions
Running Asterisk pre from Dahdiconfig module
Wanrouter: No valid Sangoma Hardware found, if you have no Sangoma cards this is OK
Starting DAHDi for Digium Cards
DAHDi Started
Running Asterisk pre from Firewall module
Running Asterisk pre from Sysadmin module
Running Sysadmin Hooks
Restarting fail2ban
fail2ban Restarted
Updating License Information for 15423705
Checking Vpn server
Starting Asterisk…
[============================] 34 secs
Asterisk Started
Running Asterisk post from Dahdiconfig module
Running Asterisk post from Endpoint module
Running Asterisk post from Pagingpro module
Running Asterisk post from Restapps module
Starting RestApps Server…
[>---------------------------] 4 secs
Started RestApps Server. PID is 9940
Running Asterisk post from Ucpnode module
Starting UCP Node Server…
[>---------------------------] 9 secs
Started UCP Node Server. PID is 10017
Running Asterisk post from Vqplus module
RestApps is not licensed.
Running Asterisk post from Xmpp module
[>---------------------------] < 1 secStarting Chat Server…
[>---------------------------] 7 secs
Started Chat Server. PID is 10272
Running Asterisk post from Zulu module
This product is not licensed

Any help is appreciated:

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@byrnejb wrote:

I am trying to debug a PBX failure. In the process I have to regenerate the device entries in astdb.sqlite3. At the moment, for want of knowledge of a better way, I open the device application in the web gui and individually select and submit every extension before applying the configuration changes.

Is there a script that accomplishes the same thing?

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@byrnejb wrote:

Assuming that I cannot recover the current FreeBPX system

# cat /etc/issue
SHMZ release 6.6 (Final)
Kernel \r on an \m

I have regular backups of the database:

l> l /var/www/avantfax/dbdump/ # mysql backups

total 5219784
. . .
avantfax.mysql.dump.20200112071001.log
-rw-r–r-- 1 root root 17396371 Jan 12 19:10 avantfax.mysql.dump.20200112191001.gz
-rw-r–r-- 1 root root 0 Jan 12 19:10 avantfax.mysql.dump.20200112191001.log

Can I provision a new host with the latest FreeBSD and reload the database backup into it? If not, is there a procedure to accomplish the same effect?

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@thetoad30 wrote:

I had one of my FreePBX machines crash and have disk errors. I repaired them, but to be safe, I migrated to a new FreePBX install.

That install will accept no incoming calls. They are always anonymous, even if the IP is in the subnet range of the pjsip list identifies command.

I resurrected my old FreePBX box, and everything worked as expected. As soon as I upgraded the modules, incoming calls stopped working.

I noticed that the global variable SIPDOMAIN changed from my local machine’s IP to my external IP after the modules upgraded.

Is this a bug?

EDIT:

So I restored to a snapshot I had of my old box, and the calls are working again. Something is not working with whatever updates I installed.

FreePBX Upgrades that Crashed Incoming Calls.png716×602 84.3 KB

Working Version:

Not working version:

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@arroinie wrote:

Hi everyone,

Can we put a new FreePBX logo on our website which is for a commercial product?

Update: I know it depends on how it would be actually used. In our case for the purpose of mentioning that our product works with FreePBX.

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