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Hide the delete action

April 23, 2020, 8:09 am
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@VoipCool wrote:

An user has rights in the Freepbx administration to edit some extensions. Is it possible to hide the delete action so that the user cannot delete an extension?

Posts: 5

Participants: 4

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Endpoint manager Template has no phone buttons

May 12, 2020, 7:00 am
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Sip client registration ok , debug shows correct action based on dial plan, RTP ports open but no sound

May 12, 2020, 12:17 pm
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@lmon wrote:

Hi
I have my sip client which register, but when I try for example to get extension name via *65, I get no sound.
I’m using a manual install of FreePBX ( even I don’t think it is changing anything on this config point ).

here is some CLI debug

no client connected, so I can not set debug to extension :

adc1*CLI> SIP SET DEBUG PEER 5555
Unable to get IP address of peer '5555'

Then I register my client

 -- Registered SIP '5555' at 37.170.156.93:52566
[2020-05-12 20:35:27] NOTICE[21959]: chan_sip.c:25008 handle_response_peerpoke: Peer '5555' is now Reachable. (119ms / 2000ms)

Then I call the *65 ( to get my extension ) :

adc1*CLI> SIP SET DEBUG PEER 5555
SIP Debugging Enabled for IP: [My Client IP]
adc1*CLI> *******
No such command '*******' (type 'core show help *******' for other possible commands)
adc1*CLI> ****
No such command '****' (type 'core show help ****' for other possible commands)
adc1*CLI> *****
No such command '*****' (type 'core show help *****' for other possible commands)
adc1*CLI> *******
No such command '*******' (type 'core show help *******' for other possible commands)
adc1*CLI> *****
No such command '*****' (type 'core show help *****' for other possible commands)

<--- SIP read from UDP:[My Client IP]:52566 --->
INVITE sip:*65@[FQDN of my Server ]:6116 SIP/2.0
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---42f0001b95815d6e;rport
Max-Forwards: 70
Contact: <sip:5555@10.115.236.181:36571>
To: <sip:*65@[FQDN of my Server ]:6116>
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 1 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Content-Type: application/sdp
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 6.0
Content-Length: 398

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 10.115.236.181
t=0 0
m=audio 4086 RTP/AVP 3 102 0 8 9 120 101
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 17 lines) ---
Sending to [My Client IP]:52566 (NAT)
Sending to [My Client IP]:52566 (NAT)
Using INVITE request as basis request - UBKs1nTIc_v_ckIarWFcqw..
Found peer '5555' for '5555' from [My Client IP]:52566

<--- Reliably Transmitting (NAT) to [My Client IP]:52566 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---42f0001b95815d6e;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as650fda78
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 1 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5dbbdc85"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'UBKs1nTIc_v_ckIarWFcqw..' in 7616 ms (Method: INVITE)

<--- SIP read from UDP:[My Client IP]:52566 --->
ACK sip:*65@[FQDN of my Server ]:6116 SIP/2.0
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---42f0001b95815d6e;rport
Max-Forwards: 70
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as650fda78
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:[My Client IP]:52566 --->
INVITE sip:*65@[FQDN of my Server ]:6116 SIP/2.0
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;rport
Max-Forwards: 70
Contact: <sip:5555@10.115.236.181:36571>
To: <sip:*65@[FQDN of my Server ]:6116>
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Content-Type: application/sdp
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 6.0
Authorization: Digest username="5555",realm="asterisk",nonce="5dbbdc85",uri="sip:*65@[FQDN of my Server ]:6116",response="0bda3cdf17b7a72f627228325b12bacf",algorithm=MD5
Content-Length: 398

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 10.115.236.181
t=0 0
m=audio 4086 RTP/AVP 3 102 0 8 9 120 101
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (16 headers 17 lines) ---
Sending to [My Client IP]:52566 (NAT)
Using INVITE request as basis request - UBKs1nTIc_v_ckIarWFcqw..
Found peer '5555' for '5555' from [My Client IP]:52566
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Got SDP version 1 and unique parts [- 0 IN IP4 192.168.0.250]
Found RTP audio format 3
Found RTP audio format 102
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 120
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 102
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format opus for ID 120
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw|g722|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x559b9dbdfe40 -- Strict RTP learning after remote address set to: 10.115.236.181:4086
Peer audio RTP is at port 10.115.236.181:4086
Looking for *65 in from-internal (domain [FQDN of my Server ])
sip_route_dump: route/path hop: <sip:5555@10.115.236.181:36571>

<--- Transmitting (NAT) to [My Client IP]:52566 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
Content-Length: 0


<------------>
    -- Executing [*65@from-internal:1] Set("SIP/5555-00000027", "CONNECTEDLINE(name-charset,i)=utf8") in new stack
    -- Executing [*65@from-internal:2] Set("SIP/5555-00000027", "CONNECTEDLINE(name,i)=Speak Extension") in new stack
    -- Executing [*65@from-internal:3] Set("SIP/5555-00000027", "CONNECTEDLINE(num,i)=*65") in new stack
    -- Executing [*65@from-internal:4] Answer("SIP/5555-00000027", "") in new stack
Audio is at 16602
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to [My Client IP]:52566 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
Retransmitting #1 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
Retransmitting #2 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [*65@from-internal:5] Wait("SIP/5555-00000027", "1") in new stack
Retransmitting #3 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:[My Client IP]:52566 --->


<------------->
    -- Executing [*65@from-internal:6] Macro("SIP/5555-00000027", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/5555-00000027", "TOUCH_MONITOR=1589308555.39") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/5555-00000027", "AMPUSER=5555") in new stack
    -- Executing [s@macro-user-callerid:3] Set("SIP/5555-00000027", "HOTDESCKCHAN=5555-00000027") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/5555-00000027", "HOTDESKEXTEN=5555") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/5555-00000027", "HOTDESKCALL=0") in new stack
    -- Executing [s@macro-user-callerid:6] ExecIf("SIP/5555-00000027", "0?Set(HOTDESKCALL=1)") in new stack
    -- Executing [s@macro-user-callerid:7] ExecIf("SIP/5555-00000027", "0?Set(CALLERID(name)=)") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/5555-00000027", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/5555-00000027", "1?Set(REALCALLERIDNUM=5555)") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/5555-00000027", "AMPUSER=5555") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/5555-00000027", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/5555-00000027", "AMPUSERCIDNAME=5555") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("SIP/5555-00000027", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/5555-00000027", "0?report") in new stack
    -- Executing [s@macro-user-callerid:15] Set("SIP/5555-00000027", "AMPUSERCID=5555") in new stack
    -- Executing [s@macro-user-callerid:16] Set("SIP/5555-00000027", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-user-callerid:17] Set("SIP/5555-00000027", "CALLERID(all)="5555" <5555>") in new stack
    -- Executing [s@macro-user-callerid:18] ExecIf("SIP/5555-00000027", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-user-callerid:19] GotoIf("SIP/5555-00000027", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:20] ExecIf("SIP/5555-00000027", "0?Set(GROUP(concurrency_limit)=5555)") in new stack
    -- Executing [s@macro-user-callerid:21] ExecIf("SIP/5555-00000027", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:22] NoOp("SIP/5555-00000027", "Macro Depth is 1") in new stack
    -- Executing [s@macro-user-callerid:23] GotoIf("SIP/5555-00000027", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,24)
    -- Executing [s@macro-user-callerid:24] GotoIf("SIP/5555-00000027", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:25] ExecIf("SIP/5555-00000027", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s@macro-user-callerid:26] Set("SIP/5555-00000027", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:27] GotoIf("SIP/5555-00000027", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,43)
    -- Executing [s@macro-user-callerid:43] Set("SIP/5555-00000027", "CALLERID(number)=5555") in new stack
    -- Executing [s@macro-user-callerid:44] Set("SIP/5555-00000027", "CALLERID(name)=5555") in new stack
    -- Executing [s@macro-user-callerid:45] GotoIf("SIP/5555-00000027", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:46] Set("SIP/5555-00000027", "CDR(cnam)=5555") in new stack
    -- Executing [s@macro-user-callerid:47] Set("SIP/5555-00000027", "CDR(cnum)=5555") in new stack
    -- Executing [s@macro-user-callerid:48] Set("SIP/5555-00000027", "CHANNEL(language)=en") in new stack
    -- Executing [*65@from-internal:7] GotoIf("SIP/5555-00000027", "1?app-speakextennum,en,1:app-speakextennum,en,1") in new stack
    -- Goto (app-speakextennum,en,1)
    -- Executing [en@app-speakextennum:1] Playback("SIP/5555-00000027", "your") in new stack
    -- <SIP/5555-00000027> Playing 'your.ulaw' (language 'en')
Retransmitting #4 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [en@app-speakextennum:2] Playback("SIP/5555-00000027", "extension") in new stack
    -- <SIP/5555-00000027> Playing 'extension.ulaw' (language 'en')
Really destroying SIP dialog 'vjmd4jeo7BIjdBqOtwYtuw..' Method: REGISTER
    -- Executing [en@app-speakextennum:3] Playback("SIP/5555-00000027", "number") in new stack
    -- <SIP/5555-00000027> Playing 'number.ulaw' (language 'en')
Retransmitting #5 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [en@app-speakextennum:4] Playback("SIP/5555-00000027", "is") in new stack
    -- <SIP/5555-00000027> Playing 'is.ulaw' (language 'en')
    -- Executing [en@app-speakextennum:5] SayDigits("SIP/5555-00000027", "5555") in new stack
    -- <SIP/5555-00000027> Playing 'digits/5.ulaw' (language 'en')
    -- <SIP/5555-00000027> Playing 'digits/5.ulaw' (language 'en')
    -- <SIP/5555-00000027> Playing 'digits/5.ulaw' (language 'en')
    -- <SIP/5555-00000027> Playing 'digits/5.ulaw' (language 'en')
Retransmitting #6 (NAT) to [My Client IP]:52566:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.115.236.181:36571;branch=z9hG4bK-524287-1---1bf5ec52b4995164;received=[My Client IP];rport=52566
From: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
To: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 2 INVITE
Server: FPBX-15.0.16.52(16.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:*65@[My Server IP]:6116>
P-Asserted-Identity: "Speak Extension" <sip:*65@[FQDN of my Server ]>
Content-Type: application/sdp
Require: timer
Content-Length: 327

v=0
o=root 1173292838 1173292838 IN IP4 [My Server IP]
s=Asterisk PBX 16.10.0
c=IN IP4 [My Server IP]
t=0 0
m=audio 16602 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
[2020-05-12 20:36:03] WARNING[21959]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission UBKs1nTIc_v_ckIarWFcqw.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 7617ms with no response
[2020-05-12 20:36:03] WARNING[21959]: chan_sip.c:4166 retrans_pkt: Hanging up call UBKs1nTIc_v_ckIarWFcqw.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (app-speakextennum, en, 5) exited non-zero on 'SIP/5555-00000027'
Scheduling destruction of SIP dialog 'UBKs1nTIc_v_ckIarWFcqw..' in 7616 ms (Method: INVITE)
Reliably Transmitting (NAT) to [My Client IP]:52566:
BYE sip:5555@10.115.236.181:36571 SIP/2.0
Via: SIP/2.0/UDP [My Server IP]:6116;branch=z9hG4bK3e40705e;rport
Max-Forwards: 70
From: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
To: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 102 BYE
User-Agent: FPBX-15.0.16.52(16.10.0)
Proxy-Authorization: Digest username="5555", realm="asterisk", algorithm=MD5, uri="sip:[FQDN of my Server ]", nonce="5dbbdc85", response="fdb70a489f709d119201a63c4d7530ee"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---

<--- SIP read from UDP:[My Client IP]:52566 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [My Server IP]:6116;branch=z9hG4bK3e40705e;rport=6116;received=[Second IP of my Server ]
Contact: <sip:5555@10.115.236.181:36571>
To: "5555"<sip:5555@[FQDN of my Server ]:6116>;tag=10a3d249
From: <sip:*65@[FQDN of my Server ]:6116>;tag=as6b4c13cf
Call-ID: UBKs1nTIc_v_ckIarWFcqw..
CSeq: 102 BYE
User-Agent: SessionTalk 6.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'UBKs1nTIc_v_ckIarWFcqw..' Method: INVITE

No sound at all.
as we are behind an infrasctruture we are note managing , please note that :slight_smile:
The SIP port is 6116 ( instead of the standard 5060)
I restricted for test for now the RTP ports range to 16600 to 16605 ( so enough to have 2 calls at the same time for now, I will ask later to a bigger range when I will get my FreePBX working )

Any insights about these lines ? it relates to RTP ports to be open for ex ? any method to troubleshoot ?
[2020-05-12 20:36:03] WARNING[21959]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission UBKs1nTIc_v_ckIarWFcqw… for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 7617ms with no response
[2020-05-12 20:36:03] WARNING[21959]: chan_sip.c:4166 retrans_pkt: Hanging up call UBKs1nTIc_v_ckIarWFcqw… - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (app-speakextennum, en, 5) exited non-zero on ‘SIP/5555-00000027’

Thanks for your insights

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Directing network traffic to different FreePBX instances using one WAN

May 12, 2020, 12:20 pm
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@kfeen wrote:

Ok the title might not be the best but let me explain. Right now I have one server at my office that’s running one instance of FreePBX that only one remote client needs to connect to. All I do is have ports 5060 and 10000-20000 forwarded to the LAN IP that is assigned to that server, the clients phones point to my WAN and it works great. All of my other clients have the servers at their offices so no real networking config is required.

In the very near future I plan on moving all of my clients FreePBX instances into my office for various reasons. I will be using Proxmox to run VMs that contain each FreePBX instance. My question is how would I direct each one of my clients to the LAN IP address that is assigned to their FreePBX instance if all I have is one WAN? I’m using a network bridge to connect Proxmox’s internal LAN to the rest of my network. I’m not that great at networking so the help is much appreciated.

I know that this is more of a networking question rather than a FreePBX question but I thought this would be the best place to get a direct answer.

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Help! Can not connect to asterisk

May 12, 2020, 4:12 pm
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@dsubs wrote:

I have inherited a system that was upgraded from freepbx 12 to 14.
It only had chan-sip running. To install zulu, I enabled pjsip today and did fwconsole restart.
Since then, I am getting “Cannot connect to asterisk” in the gui.

I have checked /etc/asterisk/manager.conf and the AMI password is matching with the one in the GUI.

fwconsole restart -verbose

In ArgvInput.php line 125:

[Symfony\Component\Console\Exception\RuntimeException]
The “-e” option does not exist.

Exception trace:
() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Input/ArgvInput.php:125
Symfony\Component\Console\Input\ArgvInput->parseShortOptionSet() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Input/ArgvInput.php:105
Symfony\Component\Console\Input\ArgvInput->parseShortOption() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Input/ArgvInput.php:84
Symfony\Component\Console\Input\ArgvInput->parse() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Input/Input.php:55
Symfony\Component\Console\Input\Input->bind() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Command/Command.php:214
Symfony\Component\Console\Command\Command->run() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:960
Symfony\Component\Console\Application->doRunCommand() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:255
Symfony\Component\Console\Application->doRun() at /var/www/html/admin/libraries/Composer/vendor/symfony/console/Application.php:148
Symfony\Component\Console\Application->run() at /var/lib/asterisk/bin/fwconsole:164

restart [-i|–immediate] [–] []…

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AMI > Abandoned Call Report

May 13, 2020, 8:23 pm
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@vaibhav wrote:

Whenever a call is abandoned, I need to capture the following information:

  1. Queue number
  2. Extensions logged into the queue
  3. Current state of each extension (busy, available, not logged in etc)
  4. Caller id of the abandoned call
  5. Wait time before the call was abandoned

What is the best way to get this data? Can an AMI script do this?

Looking for some directions to get started.

Thanks!!

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New user email SIP details

April 24, 2020, 9:44 pm
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@nickzed wrote:

OK, so, I must be having a blonde moment, in the add new extension , in particular quick add extension, when submitting email, it notifies new user, but strangely only for UCP details, I’m thinking to myself WTF good is that, no, its 2020, I must have missed something as to why it does not send the new users SIP details as well, but only UCP, but nope, can’t find it, this is FPBX 14 latest up to date, what have I missed?

Whats the point emailing them their UCP details if we have to verbally give their SIP login to them…

Secondly, kind of related, UCP well dont get me started on the blank UCP dashboard, that Andrew and Tony keep telling people to code it up themselves, but I also dont see where the users can change their SIP password, only their UCP password, kind of makes the entire UCP pointless.

But again, it is 2020 this project is over 15yrs old, so I’m sure I just need someone to tell me where the enabling settings are that I seem to not be able to find.

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Error in loading FreePBX dashboard

May 14, 2020, 7:51 am
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@Pentium5 wrote:

Hi all,
I have a fresh Asterisk 16 + FreePBX 15 install on a CentOS 8.1 (not a distro).
The issue is that dashboard is loading too long, getting stuck at 75%:

image
image.png1525×276 17.2 KB

When timeout expires (50 s), it finally loads, but with empty section “System overview”:

image
image.png1748×341 37.4 KB

Apache logs contains the following lines:

[Thu May 14 16:41:52.110192 2020] [proxy_fcgi:error] [pid 8643:tid 140221907703552] (70007)The timeout specified has expired: [client 10.33.3.248:54675] AH01075: Error dispatching request to : (polling), referer: http://10.10.1.149/admin/config.php
[Thu May 14 17:04:12.601101 2020] [proxy_fcgi:error] [pid 8643:tid 140222033770240] (70007)The timeout specified has expired: [client 10.33.3.248:56184] AH01075: Error dispatching request to : (polling), referer: http://10.10.1.149/admin/config.php?display=index

PHP version:

[root@hostname ~]# php -v
PHP 7.2.11 (cli) (built: Oct  9 2018 15:09:36) ( NTS )
Copyright (c) 1997-2018 The PHP Group
Zend Engine v3.2.0, Copyright (c) 1998-2018 Zend Technologies

Based on similar topics and my observations, I tried the following actions, but none of them helped:

  • Disable RSS feeds
  • Upgrade dashboard module (its current version is 15.0.5)
  • Provide correct hostname in /etc/hosts

Apart from this issue, the system seem to be operating properly; provisioning to Asterisk also works.
Could you please guide me what else should I check / correct?

Thanks.

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Enable verbose ssh in backups?

May 14, 2020, 8:33 am
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@wayne_pbx wrote:

I have created an ssh server in filestore.
I have set keys correctly. This command works correctly:
$ sudo -u asterisk <username>@<backupserver> pwd
(both with IP address and hostname)

But backups always fail with
Could not login with username: <username>, host: <hostname>
(username and IP address are correct)

tcpdump on the remote ssh server shows ssh traffic but, of course it’s encrypted. I think that to get to the bottom of this, I need to enable verbose mode (-vvv) on the ssh command that backup is using.

Is there an easy way for me to do that?

Furthermore, while the backup is running, I can see it in
/var/spool/asterisk/backup/<backupname>
but as soon as the backup completes, it is removed. From reading the wiki, my understanding is that failed backups are supposed to remain there?

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Responsive Firewall

May 14, 2020, 11:55 am
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@AsherN wrote:

I’m trying to get the responsive firewall to work.

My edge firewall is allowing ports 5060-5061 and 10000-20000 through to my PBX. There is a single interface in the PBX.

I can register my extension, but it does not show as registered in the firewall. I can also see other traffic to 5060 from unknown address, and they do not appear to be blocked.

Am I doing something wrong?

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GUI stopped opening

May 14, 2020, 12:34 pm
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@penbrock wrote:

I have a FreePBX that has been rining for a while.
This week it have stopped allowing login. It does not show the login windows. It flashes an error to check the console. The console is showing a referenceError modules not defined. Can anyone help with where to start getting access again?

ReferenceError: modules is not defined

jsphpg_ae2b33db35c56d3d87a2b8828c33b54c.js:2446:1038
autoload http://10.0.3.101:8080/ucp/assets/js/compiled/main/jsphpg_ae2b33db35c56d3d87a2b8828c33b54c.js?load_version=v15.0.6.14:2446
setupDashboard http://10.0.3.101:8080/ucp/assets/js/compiled/main/jsphpg_ae2b33db35c56d3d87a2b8828c33b54c.js?load_version=v15.0.6.14:2440
ready http://10.0.3.101:8080/ucp/assets/js/compiled/main/jsphpg_ae2b33db35c56d3d87a2b8828c33b54c.js?load_version=v15.0.6.14:2431
http://10.0.3.101:8080/ucp/assets/js/compiled/main/jsphpg_ae2b33db35c56d3d87a2b8828c33b54c.js?load_version=v15.0.6.14:2462
jQuery 2
Source map error: Error: request failed with status 403
Resource URL: http://10.0.3.101:8080/ucp/assets/js/compiled/main/jsphpg_ae2b33db35c56d3d87a2b8828c33b54c.js?load_version=v15.0.6.14

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VTech VCS754 unable to register

May 14, 2020, 3:56 pm
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@RBIT wrote:

I’m using FreePBX 12.0.76.6.
I’ll start by saying i’m new to Free PBX so please forgive the ignorance.

I normally use Cisco SPA5114g Phones but i have a Vtech VCS754 that i’m trying to register with my phone system. the phone keeps state Lines unregistered.
under endpoint manager I’m creating a new template and i get a blank screen for the VTech VCS754, normally i’m able to select account speed dial and so on then save that phones template.
my questions are,

  1. is there another way to register the phone?
  2. Do i need an update to my software that will give me what is needed to register this phone?
  3. is there something else i’m missing?

Any help would be much appreciated.

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MariaDB hourly high IO usage

May 15, 2020, 8:21 am
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@oliviermarchand wrote:

Hi to all,

We’ve depolyed about 20 FreePBX instances into Digital Ocean droplets. Each of them are using the same FreePBX ISO disk for initial installation : SNG7-PBX-64bit-1805 (FreePBX 14.0.13.26 , Current Asterisk Version: 13.32.0)

Server is using 2vCPU and 4GIGs of Ram.

Now, a single instance had a strange behaviour. At every hours, the IO usage is increasing up for one or two minutes, than returns to its normal “almost idle” state. It is the only instance experiencing that trouble.

All this started few days back, used to had no issue with that server.
Call volume is really low, in fact, almost no calls are being processed as our customer is still not ready to migrate its SIP phone lines to that server.

Using iotop command, I was able to find out that mariadb logs is the issue here:

09:00:55 18419 be/4 mysql 0.00 K/s 0.00 K/s 0.00 % 99.99 % mysqld --basedir=/usr --datadir=/var/lib/mysql --plugin-dir=/usr/lib64/m
ysql/plugin --log-error=/var/log/mariadb/mariadb.log --pid-file=/var/run/mariadb/mariadb.pid --socket=/var/lib/mysql/mysql.sock
09:00:55 1597 be/4 mysql 0.00 K/s 7.89 K/s 0.00 % 0.00 % mysqld --basedir=/usr --datadir=/var/lib/mysql --plugin-dir=/usr/lib64/m
ysql/plugin --log-error=/var/log/mariadb/mariadb.log --pid-file=/var/run/mariadb/mariadb.pid --socket=/var/lib/mysql/mysql.sock
09:00:56 1605 be/4 root 0.00 K/s 0.00 K/s 0.00 % 0.00 % rsyslogd -n [in:imjournal]
09:00:56 720 be/3 root 0.00 K/s 3.94 K/s 0.00 % 0.00 % auditd
09:00:56 13222 be/4 asterisk 0.00 K/s 3.94 K/s 0.00 % 0.00 % asterisk -f -U asterisk -G asterisk -vvvg -c
09:00:57 721 be/3 root 0.00 K/s 0.00 K/s 0.00 % 99.99 % auditd
09:00:57 1605 be/4 root 0.00 K/s 0.00 K/s 0.00 % 0.00 % rsyslogd -n [in:imjournal]
09:00:58 1597 be/4 mysql 0.00 K/s 142.02 K/s 0.00 % 99.99 % mysqld --basedir=/usr --datadir=/var/lib/mysql --plugin-dir=/usr/lib64/m
ysql/plugin --log-error=/var/log/mariadb/mariadb.log --pid-file=/var/run/mariadb/mariadb.pid --socket=/var/lib/mysql/mysql.sock

As all our deployments are using same CPU size, same amount of RAM , and so on, did I missed something in the Advance Settings of FreePBX that might start up a debug log of some sort ?

I alredy went into all the advance settings and disable all FreePBX debug loggings and try all the usual reboots , module upgrades, yum upgrade, no luck.

Any advise what can be my next steps ?

Thanks.

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Terrible audio on fresh install

May 15, 2020, 7:51 pm
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@bigunk wrote:

FreePBX 15.0.16.49 / Asterisk Version:16.6.2. Fresh install because my old one got a bit scrambled. The audio is not that good. Lots of clicks and pops and very short dropouts. No pattern to it. Sounds quite random. Everything else is unchanged. The PC. phones, network, and carriers are all the same. Anyone got a thought or three?

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Fixed Public IP needed for freePBX?

May 15, 2020, 11:09 pm
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@Wallyrt wrote:

Hi there,

I want to install or buy a Freepbx System. My ISP doo not provide me with an fixed WAN IP adress. Can I use freepbx with an dynamic pubilc IP oder must I have a static IP adress like in an 3CX Installation ?

Greetings
WALLY

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Filtering pjsip show channels

May 18, 2020, 1:20 am
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@wassy83 wrote:

Hi to all,
I’m using

asterisk -rx "pjsip show channels"

to show ringing and active calls, but since I have a lot of extensions ringing at same time, I got something like this

    [root@freepbx ~]# asterisk -rx "pjsip show channels"

  Channel:  <ChannelId........................................>  <State.....>  <Time.....>
      Exten: <DialedExten.............>  CLCID: <ConnectedLineCID.......>
==========================================================================================

  Channel: PJSIP/214-0001190d/AppDial                            Ringing       00:00:19
      Exten: 214                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/215-0001190e/AppDial                            Ringing       00:00:19
      Exten: 215                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/221-00011909/AppDial                            Ringing       00:00:19
      Exten: 221                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/222-00011910/AppDial                            Ringing       00:00:19
      Exten: 222                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/223-0001190b/AppDial                            Ringing       00:00:19
      Exten: 223                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/225-00011912/AppDial                            Ringing       00:00:19
      Exten: 225                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/226-00011914/AppDial                            Ringing       00:00:19
      Exten: 226                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/227-0001190a/AppDial                            Ringing       00:00:19
      Exten: 227                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/228-00011915/AppDial                            Ringing       00:00:19
      Exten: 228                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/230-00011913/AppDial                            Ringing       00:00:19
      Exten: 230                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/231-000118d7/AppDial                            Up            00:01:41
      Exten: s                           CLCID: "0566437XX" <0566437XX>

  Channel: PJSIP/232-0001190f/AppDial                            Ringing       00:00:19
      Exten: 232                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/234-00011911/AppDial                            Ringing       00:00:19
      Exten: 234                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/235-00011908/AppDial                            Ringing       00:00:19
      Exten: 235                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/236-00011907/AppDial                            Ringing       00:00:19
      Exten: 236                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/237-000118ed/AppDial                            Up            00:01:01
      Exten: s                           CLCID: "0415346979" <0415346979>

  Channel: PJSIP/238-0001190c/AppDial                            Ringing       00:00:19
      Exten: 238                         CLCID: "+336506736XX" <+336506736XX>

  Channel: PJSIP/272-00011919/AppDial                            Ringing       00:00:09
      Exten: 2030                        CLCID: "Venezia:33339180XX" <33339180XX>

  Channel: PJSIP/276-0001191a/AppDial                            Ringing       00:00:09
      Exten: 2030                        CLCID: "Venezia:33339180XX" <33339180XX>

  Channel: PJSIP/ANCONA_UFFICIO-0001184b/Queue                   Up            00:05:21
      Exten: 2000                        CLCID: "" <>

  Channel: PJSIP/ANCONA_UFFICIO-0001189c/Queue                   Up            00:03:20
      Exten: 2000                        CLCID: "" <>

  Channel: PJSIP/ANCONA_UFFICIO-000118e9/Queue                   Up            00:01:14
      Exten: 2000                        CLCID: "" <>

  Channel: PJSIP/VENEZIA_PASSEGGERI-000118e6/Queue               Up            00:01:20
      Exten: 2000                        CLCID: "" <>

  Channel: PJSIP/VENEZIA_PASSEGGERI-00011918/Dial                Ring          00:00:09
      Exten: s                           CLCID: "" <>


Objects found: 24

so…is there a way to filter this in a way that will show only one entries for each incoming call ringing and up? many thanks

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Bulk upload content into the Directory - via MySQL?

May 18, 2020, 8:16 am
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@cfapress wrote:

We are going to begin using the Directory feature to route incoming calls from our main site to satellite offices which also run FreePBX. We have over 150 extensions which people could direct-dial and I understand the only way to manage this is using the Directory.

I cringe at the amount of clicking, copying, and pasting involved in hand-building such a large directory.

I prefer to take an Excel spreadsheet and upload the content.

Since that doesn’t seem possible I’ve looked at the MySQL database for Asterisk. It looks like I could run some SQL statements and it will update info properly. Based upon my very short tests it looks like these two statements could achieve a desired result:

select max(e_id)+1 into @newID from directory_entries; 
insert into directory_entries (id, e_id, name, type, audio, dial) values (1, @newID, "test person3", "custom", "tts", "9991");

NOTE: my single Directory has ID=1 to match the “id” column above

But I’d like opinions of more senior FreePBX people before I head too far down this road.

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Call from one site to another site thru inta-company trunk and then call forward

May 18, 2020, 2:11 pm
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@rdkerns wrote:

So here’s my issue. I have two PBX’s SITE A and SITE B.

When one employee in SITE A Calls another Employee in Site B it works fine. But when Employee in Site B has their find me follow me turned on the call to the Find Me/Follow Me destination fails. In this case the destination is a cellphone. If an employee in site b calls the same employee in site B the find me follow me works fine.

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Transfer with abnormal behavior

May 19, 2020, 4:15 am
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@claloano wrote:

Transfer with abnormal behavior

I have a system with freepbx that does a strange thing about transfers: “Call Forward All Activate * 72”

In practice, if an extension calls the extension that is being transferred, everything works but if the call comes from an external trunk, the transfer takes place but the communication is not heard.

The switchboard is in the cloud and between the cloud and the customer network there is a vpn, NAT problems I never had …
But above all, according to the client, the problem is not present on all extension but only on some

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IAX2 trunk for 2 x FreePBX 14

May 19, 2020, 8:51 am
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@cmstechnical wrote:

Hi All,

I am hoping to get some help. i had this working on an older version of Freepbx in the past and have followed multiple guides and youtube videos with no joy this time.

I need the most basic IAX2 trunk to transfer calls between two sites.

I have created a trunk on both sites (only outgoing as the guide suggested and ignored the incoming)

with the following

trunk name:
pbx1_to_pbx2

peer details:
type=friend
qualify=yes
host= (pbx2ip)
context=from-internal
disallow=-all
allow=ulaw

and the reverse on the other system only in outgoing.

IAX2 Info shows a peer connection on both

I have setup an outgoing route with dial plans on both and selected the trunks i created.

I am not very comfortable using the dial plans.

One system has extensions 213 - 223 and 801 to 809
The other has extensions 201 - 210

I have tried X. and just . on them both and both says all circuits are busy

Any suggestions welcome :slight_smile:

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