Problems setting up extension

I’m used to Asterisk, more so the flat file configuration. I have done some with MySQL and Asterisk 1.4. I am using PBX In A Flash.

I setup an extension. It seemed fairly straight forward. However, my soft phone doesn’t register. It will register to another box.

| 8159814211 | account | 8159814211 | 25 | | 8159814211 | permit | 0.0.0.0/0.0.0.0 | 24 | | 8159814211 | deny | 0.0.0.0/0.0.0.0 | 23 | | 8159814211 | mailbox | [email protected] | 22 | | 8159814211 | accountcode | | 21 | | 8159814211 | dial | SIP/8159814211 | 20 | | 8159814211 | allow | | 19 | | 8159814211 | disallow | | 18 | | 8159814211 | pickupgroup | | 17 | | 8159814211 | callgroup | | 16 | | 8159814211 | encryption | no | 15 | | 8159814211 | transport | udp | 14 | | 8159814211 | qualifyfreq | 60 | 13 | | 8159814211 | qualify | yes | 12 | | 8159814211 | port | 5060 | 11 | | 8159814211 | type | friend | 9 | | 8159814211 | nat | no | 10 | | 8159814211 | sendrpid | no | 8 | | 8159814211 | trustrpid | yes | 7 | | 8159814211 | host | dynamic | 6 | | 8159814211 | context | from-internal | 5 | | 8159814211 | canreinvite | no | 4 | | 8159814211 | dtmfmode | rfc2833 | 3 | | 8159814211 | secret | Abc123 | 2 | | 8159814211 | callerid | device <8159814211> | 26 | | 8159814211 | record_in | Adhoc | 27 | | 8159814211 | record_out | Adhoc | 28 | +------------+-------------+-----------------------------+-------+

However, this is what Asterisk is reporting:

[code]<— SIP read from UDP:10.1.1.38:5060 —>
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-947afbb4506ae49d-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:5060;transport=UDP
To: sip:[email protected];transport=UDP
From: sip:[email protected];transport=UDP;tag=bf679828
Call-ID: YzYxNmQxODRmY2YzOWNjY2RhODAyNGY1NmMxYjhlMGM.
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to 10.1.1.38:5060 (NAT)
list_route: hop: sip:[email protected]:5060;transport=UDP
Found peer ‘8159814211’ for ‘8159814211’ from 10.1.1.38:5060

<— Transmitting (NAT) to 10.1.1.38:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-947afbb4506ae49d-1—d8754z-;received=10.1.1.38;rport=5060
From: sip:[email protected];transport=UDP;tag=bf679828
To: sip:[email protected];transport=UDP;tag=as3f048477
Call-ID: YzYxNmQxODRmY2YzOWNjY2RhODAyNGY1NmMxYjhlMGM.
CSeq: 1 SUBSCRIBE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1f57715b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘YzYxNmQxODRmY2YzOWNjY2RhODAyNGY1NmMxYjhlMGM.’ in 32000 ms (Method: SUBSCRIBE)

<— SIP read from UDP:10.1.1.38:5060 —>
REGISTER sip:10.1.8.16;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-ea1aaea390f03ab6-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:5060;rinstance=08770de9833d2f0f;transport=UDP
To: sip:[email protected];transport=UDP
From: sip:[email protected];transport=UDP;tag=6f633572
Call-ID: ZmQ0NmRmYzAyMzhkNzY5ZjFhN2IzODU5YzljY2U4ZTc.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 10.1.1.38:5060 (NAT)

<— Transmitting (NAT) to 10.1.1.38:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-ea1aaea390f03ab6-1—d8754z-;received=10.1.1.38;rport=5060
From: sip:[email protected];transport=UDP;tag=6f633572
To: sip:[email protected];transport=UDP;tag=as29cf2e74
Call-ID: ZmQ0NmRmYzAyMzhkNzY5ZjFhN2IzODU5YzljY2U4ZTc.
CSeq: 1 REGISTER
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="441cc5ce"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZmQ0NmRmYzAyMzhkNzY5ZjFhN2IzODU5YzljY2U4ZTc.’ in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog ‘ZmQ0NmRmYzAyMzhkNzY5ZjFhN2IzODU5YzljY2U4ZTc.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:10.1.1.38:5060 —>
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-a6f07f3c94b90c73-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:5060;transport=UDP
To: sip:[email protected];transport=UDP
From: sip:[email protected];transport=UDP;tag=bf679828
Call-ID: YzYxNmQxODRmY2YzOWNjY2RhODAyNGY1NmMxYjhlMGM.
CSeq: 2 SUBSCRIBE
Expires: 3600
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Authorization: Digest username=“8159814211”,realm=“asterisk”,nonce=“1f57715b”,uri="sip:[email protected];transport=UDP",response=“37ec71f14bffaf49c2f1f4f74976d914”,algorithm=MD5
Event: message-summary
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 10.1.1.38:5060 (NAT)
Found peer ‘8159814211’ for ‘8159814211’ from 10.1.1.38:5060

<— Reliably Transmitting (NAT) to 10.1.1.38:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-a6f07f3c94b90c73-1—d8754z-;received=10.1.1.38;rport=5060
From: sip:[email protected];transport=UDP;tag=bf679828
To: sip:[email protected];transport=UDP;tag=as3f048477
Call-ID: YzYxNmQxODRmY2YzOWNjY2RhODAyNGY1NmMxYjhlMGM.
CSeq: 2 SUBSCRIBE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>

<— SIP read from UDP:10.1.1.38:5060 —>
REGISTER sip:10.1.8.16;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-2e35cb6433915eb4-1—d8754z-
Max-Forwards: 70
Contact: sip:[email protected]:5060;rinstance=08770de9833d2f0f;transport=UDP
To: sip:[email protected];transport=UDP
From: sip:[email protected];transport=UDP;tag=6f633572
Call-ID: ZmQ0NmRmYzAyMzhkNzY5ZjFhN2IzODU5YzljY2U4ZTc.
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Authorization: Digest username=“8159814211”,realm=“asterisk”,nonce=“441cc5ce”,uri=“sip:10.1.8.16;transport=UDP”,response=“7c1011f1ec18b800d28cfa0abb2e7ea5”,algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 10.1.1.38:5060 (NAT)

<— Transmitting (NAT) to 10.1.1.38:5060 —>
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-2e35cb6433915eb4-1—d8754z-;received=10.1.1.38;rport=5060
From: sip:[email protected];transport=UDP;tag=6f633572
To: sip:[email protected];transport=UDP;tag=as29cf2e74
Call-ID: ZmQ0NmRmYzAyMzhkNzY5ZjFhN2IzODU5YzljY2U4ZTc.
CSeq: 2 REGISTER
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZmQ0NmRmYzAyMzhkNzY5ZjFhN2IzODU5YzljY2U4ZTc.’ in 32000 ms (Method: REGISTER)
Retransmitting #1 (NAT) to 10.1.1.38:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-a6f07f3c94b90c73-1—d8754z-;received=10.1.1.38;rport=5060
From: sip:[email protected];transport=UDP;tag=bf679828
To: sip:[email protected];transport=UDP;tag=as3f048477
Call-ID: YzYxNmQxODRmY2YzOWNjY2RhODAyNGY1NmMxYjhlMGM.
CSeq: 2 SUBSCRIBE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #2 (NAT) to 10.1.1.38:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-a6f07f3c94b90c73-1—d8754z-;received=10.1.1.38;rport=5060
From: sip:[email protected];transport=UDP;tag=bf679828
To: sip:[email protected];transport=UDP;tag=as3f048477
Call-ID: YzYxNmQxODRmY2YzOWNjY2RhODAyNGY1NmMxYjhlMGM.
CSeq: 2 SUBSCRIBE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #3 (NAT) to 10.1.1.38:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.1.1.38:5060;branch=z9hG4bK-d8754z-a6f07f3c94b90c73-1—d8754z-;received=10.1.1.38;rport=5060
From: sip:[email protected];transport=UDP;tag=bf679828
To: sip:[email protected];transport=UDP;tag=as3f048477
Call-ID: YzYxNmQxODRmY2YzOWNjY2RhODAyNGY1NmMxYjhlMGM.
CSeq: 2 SUBSCRIBE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

—[/code]

More log information I found:[2012-07-17 15:11:16] VERBOSE[6486] asterisk.c: -- Remote UNIX connection disconnected [2012-07-17 15:11:58] NOTICE[1883] chan_sip.c: Registration from '<sip:[email protected];transport=UDP>' failed for '10.1.1.38:5060' - No matching peer found [2012-07-17 15:11:58] NOTICE[1883] chan_sip.c: Failed to authenticate device <sip:[email protected];transport=UDP>;tag=926fe803 for SUBSCRIBE [2012-07-17 15:11:58] NOTICE[1883] chan_sip.c: Registration from '<sip:[email protected];transport=UDP>' failed for '10.1.1.38:5060' - No matching peer found [2012-07-17 15:12:02] VERBOSE[1869] asterisk.c: -- Remote UNIX connection [2012-07-17 15:12:06] NOTICE[1883] chan_sip.c: Registration from '<sip:[email protected];transport=UDP>' failed for '10.1.1.38:5060' - No matching peer found [2012-07-17 15:12:06] NOTICE[1883] chan_sip.c: Failed to authenticate device <sip:[email protected];transport=UDP>;tag=b502435f for SUBSCRIBE [2012-07-17 15:12:06] NOTICE[1883] chan_sip.c: Registration from '<sip:[email protected];transport=UDP>' failed for '10.1.1.38:5060' - No matching peer found [2012-07-17 15:12:07] NOTICE[1883] chan_sip.c: Registration from '<sip:[email protected];transport=UDP>' failed for '10.1.1.38:5060' - No matching peer found [2012-07-17 15:12:07] NOTICE[1883] chan_sip.c: Failed to authenticate device <sip:[email protected];transport=UDP>;tag=2e756a5d for SUBSCRIBE [2012-07-17 15:12:07] NOTICE[1883] chan_sip.c: Registration from '<sip:[email protected];transport=UDP>' failed for '10.1.1.38:5060' - No matching peer found

Ideas?

If using PBX in a flash you should try their forums for help.

I did, but I posted here since Free PBX is the interface providing all of this.

The device at 10.1.8.16 is not sending the correct credentials to Asterisk.

secret is case sensitive.

UserID/PeerID is the extension number in FreePBX speak.

10.1.8.16 is the Asterisk server itself. 10.1.1.38 is the PC I’m launching the soft phone on.

I copied and pasted the account and secret fields into the soft phones.

Regardless of copying and pasting it is still wrong. GIGO. This is clear by the log messages you are receiving.

From the message below the phone is trying to register as 8159814211

[2012-07-17 15:12:07] NOTICE[1883] chan_sip.c: Registration from '' failed for '10.1.1.38:5060' - No matching peer found

Do a “sip show peers”. Do you see a peer for 8159814211?

We can’t support you if you are changing code, accessing the DB directly etc.

If you are using non-standard interfaces you should state that up front.

I’m not changing them. All changes are on the GUI, which makes it more difficult for me to use.

I’m pulling the information from the text files to show what they are set to. Easier and more clean than screen shots from the GUI.

A trunk had the same name, but wasn’t making it to the conf file. When I was working with conf files or MySQL directly, I could see on the screen any conflicts. In PIAF I have to make sure there are no conflicts across different screens. I resolved that now.

I do have a new problem now. DOH!

The PIAF install is registered to my main Asterisk install. The PIAF install shows it has registered and both installs show the other as peers. Calls aren’t accepted by PIAF.

On the Asterisk server -- Executing Macro("IAX2/aiur-3", "stdexten|[email protected]|SIP/FreePBX|15") -- Executing [[email protected]:1] Dial("IAX2/aiur-3", "SIP/FreePBX|15") in new stack -- Called FreePBX [Jul 19 07:39:54] WARNING[3533]: chan_sip.c:12192 handle_response_invite: Received response: "Forbidden" from '"8157395582" <sip:[email protected]>;tag=as6598236d' -- SIP/FreePBX-01d5a550 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [[email protected]:2] Goto("IAX2/aiur-3", "s-CONGESTION|1") in new stack -- Goto (macro-stdexten,s-CONGESTION,1) -- Executing [[email protected]:1] Goto("IAX2/aiur-3", "s-NOANSWER|1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing [[email protected]:1] VoiceMail("IAX2/aiur-3", "[email protected]|u") in new stack -- <IAX2/aiur-3> Playing 'vm-theperson' (language 'en') -- <IAX2/aiur-3> Playing 'digits/8' (language 'en') -- <IAX2/aiur-3> Playing 'digits/1' (language 'en') -- <IAX2/aiur-3> Playing 'digits/5' (language 'en') -- <IAX2/aiur-3> Playing 'digits/9' (language 'en') -- <IAX2/aiur-3> Playing 'digits/8' (language 'en') == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'IAX2/aiur-3' in macro 'stdexten' == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'IAX2/aiur-3' -- Hungup 'IAX2/aiur-3'

SIP debugging on the PIAF install[code]<— SIP read from UDP:10.1.5.5:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK6bc76e48;rport
From: “8157395582” sip:[email protected];tag=as4ff0f096
To: sip:[email protected]:5060
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 19 Jul 2012 12:21:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 3172 3172 IN IP4 10.1.5.5
s=session
c=IN IP4 10.1.5.5
t=0 0
m=audio 14532 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (14 headers 13 lines) —
Sending to 10.1.5.5:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘Aiur’ for ‘8157395582’ from 10.1.5.5:5060

<— Reliably Transmitting (NAT) to 10.1.5.5:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK6bc76e48;received=10.1.5.5;rport=5060
From: “8157395582” sip:[email protected];tag=as4ff0f096
To: sip:[email protected]:5060;tag=as759dfb16
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3fb790c1"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:10.1.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK6bc76e48;rport
From: “8157395582” sip:[email protected];tag=as4ff0f096
To: sip:[email protected]:5060;tag=as759dfb16
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:10.1.5.5:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK422508b6;rport
From: “8157395582” sip:[email protected];tag=as4ff0f096
To: sip:[email protected]:5060
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“FreePBX”, realm=“asterisk”, algorithm=MD5, uri=“sip:[email protected]:5060”, nonce=“3fb790c1”, response=“74da51dabdfc1b9667b0a14603ab3f0b”, opaque=""
Date: Thu, 19 Jul 2012 12:21:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 3172 3173 IN IP4 10.1.5.5
s=session
c=IN IP4 10.1.5.5
t=0 0
m=audio 14532 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (15 headers 13 lines) —
Sending to 10.1.5.5:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘Aiur’ for ‘8157395582’ from 10.1.5.5:5060

<— Reliably Transmitting (NAT) to 10.1.5.5:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK422508b6;received=10.1.5.5;rport=5060
From: “8157395582” sip:[email protected];tag=as4ff0f096
To: sip:[email protected]:5060;tag=as759dfb16
Call-ID: [email protected]
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:10.1.5.5:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK422508b6;rport
From: “8157395582” sip:[email protected];tag=as4ff0f096
To: sip:[email protected]:5060;tag=as759dfb16
Contact: sip:[email protected]
Call-ID: 7[email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS[/code]

Here is my sip_additional.conf [code][Aiur]
host=10.1.5.5
username=FreePBX
secret=Abc.123
type=peer
context=from-trunk-sip-Aiur

[FreePBX]
secret=Abc.123
type=user
context=from-trunk[/code]

Parts of extensions_additional.conf [code][from-trunk-sip-Aiur]
include => from-trunk-sip-Aiur-custom
exten => _.,1,Set(GROUP()=OUT_2)
exten => _.,n,Goto(from-trunk,${EXTEN},1)

; end of [from-trunk-sip-Aiur][/code]

I suppose another fat finger somewhere, but being as it injects a whole pile of stuff I’m not used to, I’m not even sure where to begin. Peer and user vs. friend is one area I have a problem getting used to. I’m required to have different names and different types when traditionally, one name as a friend is all that was required.

I had to add insecure=very to my FreePBX PEER details for the specific trunk. I guess now I need to figure out why I had to make it insecure to work. At least it works.