Grandstream HT841 - Unable to make outgoing calls

FreePBX Endpoints
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1

Hello,
I’m unable to properly configure FreePBX/HT841 to be able to work for outgoing calls.
I tried tons of configs, and search the web, ask AI, I’m at the end of my legendary patience :slight_smile:
Incoming call no issue easy to have it working, but outgoing is a nightmare.
Anyone success to do it ? I have seen post on this forum, but no real solution or working config posted.

All are on the last version/firmware.
It is probably something very stupid that I’ve missed or a Codec or ports constraints…

I have configured a PJSIP Trunk, and let the defaults ports on the HT841 (6060,2,4,6) for each FXO. Here it is the FXO2 where I use to test the config, I didn’t activate yet the others.

On the FreePBX server, PJSip port is on 5060.

The HT841 register successfully to FreePBX, incomming calls are OK, but when I try to call out I receive the message that the numert requested is not in service…

I have this error in FreePBX log line 1768, not sure if it is the real issue:

1740 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@sub-flp-3:4] Set(“PJSIP/227-00000015”, “DIAL_NUMBER=0789663322”) in new stack
1741 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@sub-flp-3:5] Return(“PJSIP/227-00000015”, “”) in new stack
1742 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:18] Set(“PJSIP/227-00000015”, “OUTNUM=0789663322”) in new stack
1743 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:19] Set(“PJSIP/227-00000015”, “custom=PJSIP”) in new stack
1744 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:20] ExecIf(“PJSIP/227-00000015”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
1745 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:21] ExecIf(“PJSIP/227-00000015”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
1746 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:22] ExecIf(“PJSIP/227-00000015”, “0?AGI(allowlist-autoadd.agi,)”) in new stack
1747 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:23] Macro(“PJSIP/227-00000015”, “dialout-trunk-predial-hook,”) in new stack
1748 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/227-00000015”, “”) in new stack
1749 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:24] GotoIf(“PJSIP/227-00000015”, “0?bypass,1”) in new stack
1750 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:25] ExecIf(“PJSIP/227-00000015”, “1?Set(CONNECTEDLINE(num,i)=0789663322)”) in new stack
1751 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:26] ExecIf(“PJSIP/227-00000015”, “1?Set(CONNECTEDLINE(name,i)=CID:9988777300)”) in new stack
1752 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/227-00000015”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)9988777300)”) in new stack
1753 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:28] GotoIf(“PJSIP/227-00000015”, “0?customtrunk”) in new stack
1754 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/227-00000015”, “0?Set(DIAL_TRUNK_OPTIONS=)”) in new stack
1755 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:30] Set(“PJSIP/227-00000015”, “HASH(__SIPHEADERS,Alert-Info)=unset”) in new stack
1756 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:31] Dial(“PJSIP/227-00000015”, “PJSIP/0789663322@9988777300,300,Tb(func-apply-sipheaders^s^1,(3))U(sub-send-obroute-email^0789663322^0789663322^3^1739450821^^9988777300)”) in new stack
1757 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] app_stack.c: PJSIP/9988777300-00000016 Internal Gosub(func-apply-sipheaders,s,1(3)) start
1758 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/9988777300-00000016”, “Applying SIP Headers to channel PJSIP/9988777300-00000016”) in new stack
1759 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:2] Set(“PJSIP/9988777300-00000016”, “localchan=9988777300-00000016”) in new stack
1760 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“PJSIP/9988777300-00000016”, “DialMCEXT=9988777300”) in new stack
1761 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:4] Set(“PJSIP/9988777300-00000016”, “CHANNEL(hangup_handler_push)=app-missedcall-hangup,9988777300,1”) in new stack
1762 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:5] Set(“PJSIP/9988777300-00000016”, “TECH=PJSIP”) in new stack
1763 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:6] Set(“PJSIP/9988777300-00000016”, “SIPHEADERKEYS=Alert-Info”) in new stack
1764 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:7] While(“PJSIP/9988777300-00000016”, “1”) in new stack
1765 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:8] Set(“PJSIP/9988777300-00000016”, “sipheader=unset”) in new stack
1766 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:9] ExecIf(“PJSIP/9988777300-00000016”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
1767 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/9988777300-00000016”, “1?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
1768 [2025-02-13 12:47:01] ERROR[7580] res_pjsip_header_funcs.c: No headers had been previously added to this session.
1769 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:11] ExecIf(“PJSIP/9988777300-00000016”, “0?Set(sipheader=<127.0.0.1>;info=unset)”) in new stack
1770 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:12] ExecIf(“PJSIP/9988777300-00000016”, “0?Set(sipheader=<127.0.0.1>unset)”) in new stack
1771 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:13] ExecIf(“PJSIP/9988777300-00000016”, “0?SIPAddHeader(Alert-Info:unset)”) in new stack
1772 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:14] ExecIf(“PJSIP/9988777300-00000016”, “0?Set(PJSIP_HEADER(add,Alert-Info)=unset)”) in new stack
1773 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:15] EndWhile(“PJSIP/9988777300-00000016”, “”) in new stack
1774 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:7] While(“PJSIP/9988777300-00000016”, “0”) in new stack
1775 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@func-apply-sipheaders:16] Return(“PJSIP/9988777300-00000016”, “”) in new stack
1776 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] app_stack.c: Spawn extension (from-pstn, 0789663322, 1) exited non-zero on ‘PJSIP/9988777300-00000016’
1777 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] app_stack.c: PJSIP/9988777300-00000016 Internal Gosub(func-apply-sipheaders,s,1(3)) complete GOSUB_RETVAL=
1778 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] app_dial.c: Called PJSIP/0789663322@9988777300
1779 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] app_stack.c: PJSIP/9988777300-00000016 Internal Gosub(app-missedcall-hangup,9988777300,1) start
1780 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [9988777300@app-missedcall-hangup:1] NoOp(“PJSIP/9988777300-00000016”, “Dialed: 9988777300”) in new stack
1781 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [9988777300@app-missedcall-hangup:2] NoOp(“PJSIP/9988777300-00000016”, “Caller: 227”) in new stack
1782 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [9988777300@app-missedcall-hangup:3] GotoIf(“PJSIP/9988777300-00000016”, “0?exit”) in new stack
1783 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [9988777300@app-missedcall-hangup:4] Set(“PJSIP/9988777300-00000016”, “EXTENNUM=9988777300”) in new stack
1784 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [9988777300@app-missedcall-hangup:5] Set(“PJSIP/9988777300-00000016”, “FEXTENNUM=9988777300”) in new stack
1785 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [9988777300@app-missedcall-hangup:6] GotoIf(“PJSIP/9988777300-00000016”, “0?exit”) in new stack
1786 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [9988777300@app-missedcall-hangup:7] AGI(“PJSIP/9988777300-00000016”, “missedcallnotify.php,9988777300,9988777300,0,PJSIP/9988777300-00000016,”) in new stack
1787 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/missedcallnotify.php
1788 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] res_agi.c: <PJSIP/9988777300-00000016>AGI Script missedcallnotify.php completed, returning 0
1789 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [9988777300@app-missedcall-hangup:8] Return(“PJSIP/9988777300-00000016”, “”) in new stack
1790 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] app_stack.c: Spawn extension (from-pstn, 0789663322, 1) exited non-zero on ‘PJSIP/9988777300-00000016’
1791 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] app_stack.c: PJSIP/9988777300-00000016 Internal Gosub(app-missedcall-hangup,9988777300,1) complete GOSUB_RETVAL=
1792 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
1793 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:32] NoOp(“PJSIP/227-00000015”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1”) in new stack
1794 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s@macro-dialout-trunk:33] GotoIf(“PJSIP/227-00000015”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
1795 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
1796 [2025-02-13 12:47:01] VERBOSE[4486][C-0000000e] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“PJSIP/227-00000015”, “RC=1”) in new stack

Any help is more than WELCOME :slight_smile:

2

Did you take a look at your outbound routes?
If the call is reaching the fxo port, it can be an internal dialplan from the gateway.

3

The outbount route seems OK as the call select the right Trunk & transmit the right number.

4

The dial plan in the HT841 is:
{ x+ | +x+ | *x+ | xxx+ }

5

image
image3933×653 72.6 KB

Here the toNumber is N/A is it also a clue ?

6

And this is the sip log of the HT841: I have to add some space in the log ip to be able to publish

HT841 — 2025-02-13 15:29:29. 395 RECEIVING FROM 10. 6. 10. 100:5060
INVITE sip : xxxxxxx101@10. 6. 10. 20:47754 SIP/2. 0
Via: SIP/2. 0/UDP 10. 6. 10. 100:5060;rport;branch=z9hG4bKPj4a13f591-0c02-4a17-9f7f-e2ea14d8c017
From: <sip : xxxxxxx7300@10. 6. 10. 20>;tag=05b32886-2e7b-4fb2-9e8c-3c5802b3a9f7
To: <sip : xxxxxxx101@10. 6. 10. 20>
Contact: <sip : xxxxxxx7300@10. 6. 10. 100:5060>
Call-ID: cba4a9fc-35f6-48af-a495-68866f16c9f7
CSeq: 15713 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-16. 0. 40. 11(20. 5. 2)
Content-Type: application/sdp
Content-Length: 328

v=0
o=- 460351312 460351312 IN IP4 10. 6. 10. 100
s=Asterisk
c=IN IP4 10. 6. 10. 100
t=0 0
m=audio 16494 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

HT841 — 2025-02-13 15:29:29. 398 SENDING TO 10. 6. 10. 100:5060
SIP/2. 0 100 Trying
Via: SIP/2. 0/UDP 10. 6. 10. 100:5060;rport=5060;branch=z9hG4bKPj4a13f591-0c02-4a17-9f7f-e2ea14d8c017
From: <sip : xxxxxxx7300@10. 6. 10. 20>;tag=05b32886-2e7b-4fb2-9e8c-3c5802b3a9f7
To: <sip : xxxxxxx101@10. 6. 10. 20>
Call-ID: cba4a9fc-35f6-48af-a495-68866f16c9f7
CSeq: 15713 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT841 1. 0. 7. 3
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

HT841 — 2025-02-13 15:29:29. 405 SENDING TO 10. 6. 10. 100:5060
SIP/2. 0 404 Not Found
Via: SIP/2. 0/UDP 10. 6. 10. 100:5060;rport=5060;branch=z9hG4bKPj4a13f591-0c02-4a17-9f7f-e2ea14d8c017
From: <sip : xxxxxxx7300@10. 6. 10. 20>;tag=05b32886-2e7b-4fb2-9e8c-3c5802b3a9f7
To: <sip : xxxxxxx101@10. 6. 10. 20>;tag=1690782022
Call-ID: cba4a9fc-35f6-48af-a495-68866f16c9f7
CSeq: 15713 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT841 1. 0. 7. 3
Warning: 399 GS “Requested user ID is unknown”
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

HT841 — 2025-02-13 15:29:29. 407 RECEIVING FROM 10. 6. 10. 100:5060
ACK sip : xxxxxxx101@10. 6. 10. 20:47754 SIP/2. 0
Via: SIP/2. 0/UDP 10. 6. 10. 100:5060;rport;branch=z9hG4bKPj4a13f591-0c02-4a17-9f7f-e2ea14d8c017
From: <sip : xxxxxxx7300@10. 6. 10. 20>;tag=05b32886-2e7b-4fb2-9e8c-3c5802b3a9f7
To: <sip : xxxxxxx101@10. 6. 10. 20>;tag=1690782022
Call-ID: cba4a9fc-35f6-48af-a495-68866f16c9f7
CSeq: 15713 ACK
Max-Forwards: 70
User-Agent: FPBX-16. 0. 40. 11(20. 5. 2)
Content-Length: 0

HT841 — 2025-02-13 15:29:32. 139 SENDING TO 10. 6. 10. 100:5060
NOTIFY sip : 10. 6. 10. 100 SIP/2. 0
Via: SIP/2. 0/UDP 10. 6. 10. 20:47754;branch=z9hG4bK482821662;rport
From: <sip : xxxxxxx7300@10. 6. 10. 20>;tag=632529678
To: <sip : 10. 6. 10. 100>
Call-ID: 2132233037-47754-14@BA. G. BA. CA
CSeq: 130 NOTIFY
Contact: <sip : xxxxxxx7300@10. 6. 10. 20:47754>
Max-Forwards: 70
User-Agent: Grandstream HT841 1. 0. 7. 3
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

HT841 — 2025-02-13 15:29:32. 142 RECEIVING FROM 10. 6. 10. 100:5060
SIP/2. 0 401 Unauthorized
Via: SIP/2. 0/UDP 10. 6. 10. 20:47754;rport=47754;received=10. 6. 10. 20;branch=z9hG4bK482821662
Call-ID: 2132233037-47754-14@BA. G. BA. CA
From: <sip : xxxxxxx7300@10. 6. 10. 20>;tag=632529678
To: <sip : 10. 6. 10. 100>;tag=z9hG4bK482821662
CSeq: 130 NOTIFY
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1739456972/0de23efc391601c6f6eb627676861b29”,opaque=“4799553162c419e3”,algorithm=MD5,qop=“auth”
Server: FPBX-16. 0. 40. 11(20. 5. 2)
Content-Length: 0

HT841 — 2025-02-13 15:29:32. 144 SENDING TO 10. 6. 10. 100:5060
NOTIFY sip : 10. 6. 10. 100 SIP/2. 0
Via: SIP/2. 0/UDP 10. 6. 10. 20:47754;branch=z9hG4bK404702204;rport
From: <sip : xxxxxxx7300@10. 6. 10. 20>;tag=632529678
To: <sip : 10. 6. 10. 100>
Call-ID: 2132233037-47754-14@BA. G. BA. CA
CSeq: 131 NOTIFY
Contact: <sip : xxxxxxx7300@10. 6. 10. 20:47754>
Authorization: Digest username=“xxxxxxx7300”, realm=“asterisk”, nonce=“1739456972/0de23efc391601c6f6eb627676861b29”, uri=“sip : 10. 6. 10. 100”, response=“5152614e904f6d05102628a4a3cf3dc7”, algorithm=MD5, cnonce=“14669459”, opaque=“4799553162c419e3”, qop=auth, nc=00000001
Max-Forwards: 70
User-Agent: Grandstream HT841 1. 0. 7. 3
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

HT841 — 2025-02-13 15:29:32. 146 RECEIVING FROM 10. 6. 10. 100:5060
SIP/2. 0 501 Not Implemented
Via: SIP/2. 0/UDP 10. 6. 10. 20:47754;rport=47754;received=10. 6. 10. 20;branch=z9hG4bK404702204
Call-ID: 2132233037-47754-14@BA. G. BA. CA
From: <sip : xxxxxxx7300@10. 6. 10. 20>;tag=632529678
To: <sip : 10. 6. 10. 100>;tag=z9hG4bK404702204
CSeq: 131 NOTIFY
Server: FPBX-16. 0. 40. 11(20. 5. 2)
Content-Length: 0

7

image
image2626×924 146 KB

8

You couuld first try configuring it without authentication, to ensure that your issue is not related to that.

Double check that the UDP ports are correct too. On previous models, IIRC, the ports start at 5060 and go up by 2 for each port, so 5060, 5062, 5074 and so on.

9

On the HT841 UDP port by default are 6060 to 6066 (up by 2)
I have set on the trunk authentication to none, and by the way registration goes also to none.
No I have a different message saying that all line are busy, and the log say:

[2025-02-13 15:04:05] ERROR[2238] res_pjsip.c: Endpoint ‘xxxxxxx7300’: Could not create dialog to invalid URI ‘xxxxxxx7300’. Is endpoint registered and reachable?
9186 [2025-02-13 15:04:05] ERROR[2238] chan_pjsip.c: Failed to create outgoing session to endpoint ‘xxxxxxx7300’

And the asterisk info on pjsip section:

Endpoint: xxxxxxx7300 Unavailable 0 of inf
Aor: xxxxxxx7300 0
Contact: xxxxxxx7300/sip:[email protected]:477 10672c5432 Unavail nan
Contact: xxxxxxx7300/sip:10.6.10.20:6062 399fe382e8 Unavail nan
Transport: 0.0.0.0-udp udp 3 96 0.0.0.0:5060
Identify: xxxxxxx7300/xxxxxxx7300
Match: 10.6.10.20/32

10

That is probably related to some configuration left behind.
If you want, I can configure it for you step by step through AnyDesk.
If you find this convenient, just send me a private message with the AnyDesk number and I will connect and guide you so you can learn how to do it by yourself.

1 Like
11

This was before you made the latest changes to registration / authentication.
I suggest that you undo those and we troubleshoot the problem above.
Taking the Warning header at face value, it seems to be related to an obscure setting in the HT “Check SIP User ID for Incoming INVITE”. But this defaults to No and you would have had no reason to set it. Did you configure a used device without first doing a factory reset?

If this is not your issue (the setting is indeed No), try setting From User in the trunk to the same value you have for Trunk Name (which I assume is what you redacted to xxxxxxx7300).

12

Yes, this is before disabling authentification/registration
It is a new device, and I already reset it to factory few times…
I tryed so many things :slight_smile:
The “Check SIP User ID for Incoming INVITE” was Yes, now revert to No.
I also have set the from user to 999997300

I redo the trunk and config of the ht841 to 999997300 to simplify the anonymization :slight_smile:

This is the new log of the HT841

HT841 — 2025-02-13 17:48:13. 625 RECEIVING FROM 10. 6. 10. 100:5060
INVITE sip:0779999999@10. 6. 10. 20:6062 SIP/2. 0
Via: SIP/2. 0/UDP 10. 6. 10. 100:5060;rport;branch=z9hG4bKPjf94e2442-9331-4df4-aa64-b01b666c0a8b
From: <sip:999997300@10. 6. 10. 100>;tag=05cad87d-4509-4af6-a70f-5483f3eb7b71
To: <sip:0779999999@10. 6. 10. 20>
Contact: <sip:999997300@10. 6. 10. 100:5060>
Call-ID: d04d2c01-fc4f-463e-a039-f23dd2c1642d
CSeq: 3851 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: FPBX-16. 0. 40. 11(20. 5. 2)
Content-Type: application/sdp
Content-Length: 330

v=0
o=- 1243223130 1243223130 IN IP4 10. 6. 10. 100
s=Asterisk
c=IN IP4 10. 6. 10. 100
t=0 0
m=audio 18546 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

HT841 — 2025-02-13 17:48:13. 627 SENDING TO 10. 6. 10. 100:5060
SIP/2. 0 100 Trying
Via: SIP/2. 0/UDP 10. 6. 10. 100:5060;rport=5060;branch=z9hG4bKPjf94e2442-9331-4df4-aa64-b01b666c0a8b
From: <sip:999997300@10. 6. 10. 100>;tag=05cad87d-4509-4af6-a70f-5483f3eb7b71
To: <sip:0779999999@10. 6. 10. 20>
Call-ID: d04d2c01-fc4f-463e-a039-f23dd2c1642d
CSeq: 3851 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT841 1. 0. 7. 3
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

HT841 — 2025-02-13 17:48:13. 635 SENDING TO 10. 6. 10. 100:5060
SIP/2. 0 403 Forbidden
Via: SIP/2. 0/UDP 10. 6. 10. 100:5060;rport=5060;branch=z9hG4bKPjf94e2442-9331-4df4-aa64-b01b666c0a8b
From: <sip:999997300@10. 6. 10. 100>;tag=05cad87d-4509-4af6-a70f-5483f3eb7b71
To: <sip:0779999999@10. 6. 10. 20>;tag=584225122
Call-ID: d04d2c01-fc4f-463e-a039-f23dd2c1642d
CSeq: 3851 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT841 1. 0. 7. 3
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

HT841 — 2025-02-13 17:48:13. 637 RECEIVING FROM 10. 6. 10. 100:5060
ACK sip:0779999999@10. 6. 10. 20:6062 SIP/2. 0
Via: SIP/2. 0/UDP 10. 6. 10. 100:5060;rport;branch=z9hG4bKPjf94e2442-9331-4df4-aa64-b01b666c0a8b
From: <sip:999997300@10. 6. 10. 100>;tag=05cad87d-4509-4af6-a70f-5483f3eb7b71
To: <sip:0779999999@10. 6. 10. 20>;tag=584225122
Call-ID: d04d2c01-fc4f-463e-a039-f23dd2c1642d
CSeq: 3851 ACK
Max-Forwards: 70
User-Agent: FPBX-16. 0. 40. 11(20. 5. 2)
Content-Length: 0

HT841 — 2025-02-13 17:48:16. 697 SENDING TO 10. 6. 10. 100:5060
NOTIFY sip:10. 6. 10. 100 SIP/2. 0
Via: SIP/2. 0/UDP 10. 6. 10. 20:6062;branch=z9hG4bK34658190;rport
From: <sip:999997300@10. 6. 10. 20>;tag=257211330
To: <sip:10. 6. 10. 100>
Call-ID: 166114127-6062-6@BA. G. BA. CA
CSeq: 50 NOTIFY
Contact: <sip:999997300@10. 6. 10. 20:6062>
Max-Forwards: 70
User-Agent: Grandstream HT841 1. 0. 7. 3
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

HT841 — 2025-02-13 17:48:16. 699 RECEIVING FROM 10. 6. 10. 100:5060
SIP/2. 0 501 Not Implemented
Via: SIP/2. 0/UDP 10. 6. 10. 20:6062;rport=6062;received=10. 6. 10. 20;branch=z9hG4bK34658190
Call-ID: 166114127-6062-6@BA. G. BA. CA
From: <sip:999997300@10. 6. 10. 20>;tag=257211330
To: <sip:10. 6. 10. 100>;tag=z9hG4bK34658190
CSeq: 50 NOTIFY
Server: FPBX-16. 0. 40. 11(20. 5. 2)
Content-Length: 0

And FreePBX:

27070 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@sub-flp-3:4] Set(“PJSIP/227-00000006”, “DIAL_NUMBER=0779999999”) in new stack
27071 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@sub-flp-3:5] Return(“PJSIP/227-00000006”, “”) in new stack
27072 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:18] Set(“PJSIP/227-00000006”, “OUTNUM=0779999999”) in new stack
27073 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:19] Set(“PJSIP/227-00000006”, “custom=PJSIP”) in new stack
27074 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:20] ExecIf(“PJSIP/227-00000006”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
27075 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:21] ExecIf(“PJSIP/227-00000006”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
27076 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:22] ExecIf(“PJSIP/227-00000006”, “0?AGI(allowlist-autoadd.agi,)”) in new stack
27077 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:23] Macro(“PJSIP/227-00000006”, “dialout-trunk-predial-hook,”) in new stack
27078 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“PJSIP/227-00000006”, “”) in new stack
27079 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:24] GotoIf(“PJSIP/227-00000006”, “0?bypass,1”) in new stack
27080 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:25] ExecIf(“PJSIP/227-00000006”, “1?Set(CONNECTEDLINE(num,i)=0779999999)”) in new stack
27081 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:26] ExecIf(“PJSIP/227-00000006”, “1?Set(CONNECTEDLINE(name,i)=CID:999997300)”) in new stack
27082 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:27] ExecIf(“PJSIP/227-00000006”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)999997300)”) in new stack
27083 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:28] GotoIf(“PJSIP/227-00000006”, “0?customtrunk”) in new stack
27084 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf(“PJSIP/227-00000006”, “0?Set(DIAL_TRUNK_OPTIONS=)”) in new stack
27085 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:30] Set(“PJSIP/227-00000006”, “HASH(__SIPHEADERS,Alert-Info)=unset”) in new stack
27086 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:31] Dial(“PJSIP/227-00000006”, “PJSIP/0779999999@999997300,300,Tb(func-apply-sipheaders^s^1,(3))U(sub-send-obroute-email^0779999999^0779999999^3^1739465293^^999997300)”) in new stack
27087 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] app_stack.c: PJSIP/999997300-00000007 Internal Gosub(func-apply-sipheaders,s,1(3)) start
27088 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/999997300-00000007”, “Applying SIP Headers to channel PJSIP/999997300-00000007”) in new stack
27089 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:2] Set(“PJSIP/999997300-00000007”, “localchan=999997300-00000007”) in new stack
27090 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“PJSIP/999997300-00000007”, “DialMCEXT=999997300”) in new stack
27091 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:4] Set(“PJSIP/999997300-00000007”, “CHANNEL(hangup_handler_push)=app-missedcall-hangup,999997300,1”) in new stack
27092 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:5] Set(“PJSIP/999997300-00000007”, “TECH=PJSIP”) in new stack
27093 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:6] Set(“PJSIP/999997300-00000007”, “SIPHEADERKEYS=Alert-Info”) in new stack
27094 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:7] While(“PJSIP/999997300-00000007”, “1”) in new stack
27095 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:8] Set(“PJSIP/999997300-00000007”, “sipheader=unset”) in new stack
27096 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:9] ExecIf(“PJSIP/999997300-00000007”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
27097 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/999997300-00000007”, “1?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
27098 [2025-02-13 16:48:13] ERROR[2298] res_pjsip_header_funcs.c: No headers had been previously added to this session.
27099 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:11] ExecIf(“PJSIP/999997300-00000007”, “0?Set(sipheader=http://127.0.0.1;info=unset)”) in new stack
27100 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:12] ExecIf(“PJSIP/999997300-00000007”, “0?Set(sipheader=http://127.0.0.1unset)”) in new stack
27101 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:13] ExecIf(“PJSIP/999997300-00000007”, “0?SIPAddHeader(Alert-Info:unset)”) in new stack
27102 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:14] ExecIf(“PJSIP/999997300-00000007”, “0?Set(PJSIP_HEADER(add,Alert-Info)=unset)”) in new stack
27103 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:15] EndWhile(“PJSIP/999997300-00000007”, “”) in new stack
27104 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:7] While(“PJSIP/999997300-00000007”, “0”) in new stack
27105 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@func-apply-sipheaders:16] Return(“PJSIP/999997300-00000007”, “”) in new stack
27106 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] app_stack.c: Spawn extension (from-pstn, 0779999999, 1) exited non-zero on ‘PJSIP/999997300-00000007’
27107 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] app_stack.c: PJSIP/999997300-00000007 Internal Gosub(func-apply-sipheaders,s,1(3)) complete GOSUB_RETVAL=
27108 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] app_dial.c: Called PJSIP/0779999999@999997300
27109 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] app_stack.c: PJSIP/999997300-00000007 Internal Gosub(app-missedcall-hangup,999997300,1) start
27110 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [999997300@app-missedcall-hangup:1] NoOp(“PJSIP/999997300-00000007”, “Dialed: 999997300”) in new stack
27111 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [999997300@app-missedcall-hangup:2] NoOp(“PJSIP/999997300-00000007”, “Caller: 227”) in new stack
27112 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [999997300@app-missedcall-hangup:3] GotoIf(“PJSIP/999997300-00000007”, “0?exit”) in new stack
27113 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [999997300@app-missedcall-hangup:4] Set(“PJSIP/999997300-00000007”, “EXTENNUM=999997300”) in new stack
27114 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [999997300@app-missedcall-hangup:5] Set(“PJSIP/999997300-00000007”, “FEXTENNUM=999997300”) in new stack
27115 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [999997300@app-missedcall-hangup:6] GotoIf(“PJSIP/999997300-00000007”, “0?exit”) in new stack
27116 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [999997300@app-missedcall-hangup:7] AGI(“PJSIP/999997300-00000007”, “missedcallnotify.php,999997300,999997300,0,PJSIP/999997300-00000007,”) in new stack
27117 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/missedcallnotify.php
27118 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] res_agi.c: <PJSIP/999997300-00000007>AGI Script missedcallnotify.php completed, returning 0
27119 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [999997300@app-missedcall-hangup:8] Return(“PJSIP/999997300-00000007”, “”) in new stack
27120 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] app_stack.c: Spawn extension (from-pstn, 0779999999, 1) exited non-zero on ‘PJSIP/999997300-00000007’
27121 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] app_stack.c: PJSIP/999997300-00000007 Internal Gosub(app-missedcall-hangup,999997300,1) complete GOSUB_RETVAL=
27122 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
27123 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:32] NoOp(“PJSIP/227-00000006”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21”) in new stack
27124 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:33] GotoIf(“PJSIP/227-00000006”, “0?continue,1:s-CHANUNAVAIL,1”) in new stack
27125 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx_builtins.c: Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
27126 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set(“PJSIP/227-00000006”, “RC=21”) in new stack
27127 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto(“PJSIP/227-00000006”, “21,1”) in new stack
27128 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx_builtins.c: Goto (macro-dialout-trunk,21,1)
27129 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [21@macro-dialout-trunk:1] Goto(“PJSIP/227-00000006”, “continue,1”) in new stack
27130 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx_builtins.c: Goto (macro-dialout-trunk,continue,1)
27131 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [continue@macro-dialout-trunk:1] NoOp(“PJSIP/227-00000006”, “TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
27132 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [continue@macro-dialout-trunk:2] ExecIf(“PJSIP/227-00000006”, “1?Set(CALLERID(number)=227)”) in new stack
27133 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [0779999999@from-internal:13] Macro(“PJSIP/227-00000006”, “outisbusy,”) in new stack
27134 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-outisbusy:1] Progress(“PJSIP/227-00000006”, “”) in new stack
27135 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [s@macro-outisbusy:2] GotoIf(“PJSIP/227-00000006”, “1?emergency,1”) in new stack
27136 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx_builtins.c: Goto (macro-outisbusy,emergency,1)
27137 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] pbx.c: Executing [emergency@macro-outisbusy:1] Playback(“PJSIP/227-00000006”, “all-circuits-busy-now&please-try-call-later”) in new stack
27138 [2025-02-13 16:48:13] VERBOSE[5270][C-00000004] file.c: <PJSIP/227-00000006> Playing ‘all-circuits-busy-now.ulaw’ (language ‘fr’)
27139 [2025-02-13 16:48:15] VERBOSE[5270][C-00000004] file.c: <PJSIP/227-00000006> Playing ‘please-try-call-later.ulaw’ (language ‘fr’)
27140 [2025-02-13 16:48:15] VERBOSE[5270][C-00000004] app_macro.c: Spawn extension (macro-outisbusy, emergency, 1) exited non-zero on ‘PJSIP/227-00000006’ in macro ‘outisbusy’
27141 [2025-02-13 16:48:15] VERBOSE[5270][C-00000004] pbx.c: Spawn extension (from-internal, 0779999999, 13) exited non-zero on ‘PJSIP/227-00000006’
27142 [2025-02-13 16:48:15] VERBOSE[5270][C-00000004] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/227-00000006”, “hangupcall”) in new stack

13

The settings in the trunk are:

Authentication Outbound

Registration Receive

14

And in Asterisk info in the pjsip section I have for the trunk:
Endpoint: 999997300 Not in use 0 of inf
OutAuth: 999997300/999997300
Aor: 999997300 1
Contact: 999997300/sip:[email protected]:6062 e761324c82 Avail 4.923
Transport: 0.0.0.0-udp udp 3 96 0.0.0.0:5060

15

The INVITE sent to the HT looks ok to me, so I don’t know why HT responded with 403 Forbidden.
One thing that looks a little weird is the offered codecs, though if that were a problem I’d expect a different error. Try enabling only alaw in the trunk.
Otherwise, you could set up syslog on the HT (at extra debug level) and hope that it will show why the 403 was sent.
Or, try turning off restrictive settings such as Validate Incoming SIP Message, Allow Incoming SIP Messages from SIP Proxy Only, Wait for Dial Tone.
Or, post a screenshot of the Profile settings and maybe we can spot something.

16

I did a printscreen of all HT841 FXO settings and FreePBX trunk, hope the solution is in that :slight_smile:
The time the forum admin validate the pictures
Again thanks a lot for the support !!!

17

(deleted)

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