Can you connect a phone to the wall and place a call on each of the 8-phone lines you have coming into your house?

Here’s where the problem is. Without more information about your configuration, it’s hard to say exactly what is happening, but a couple of things that will help you troubleshoot:

• FXS and FXO ports are not interchangeable. The port type on your card is the opposite of what DAHDI identifies them as (from the book “Did you forget that FXS interfaces are configured with FXO signaling and that FXO interfaces use FXS signaling?”), so the assumption is that you are using an 8-port FXS card as your outbound connection. The outbound call was trying to go through port 1 of your FXS card. Is that what you were expecting? Asterisk Book Reference.
• When you call a number through Asterisk, you aren’t calling anything directly. You dial the number and the PBX places the call. The ringing you are hearing is the ringing of the PBX working on your call.

I’ve been quietly watching this thread. I only have a couple of quickies:

• 1900 extensions is a lot of extensions for a single instance of anything. I have the magic number “800” in my head as the most that will work reliably. It could be from discussions here, or I might have dreamed it, but almost 2000 extensions on a single instance is a LOT.
• We’ve heard lots of reports in the past with lots smaller systems that hint processing takes a considerable amount of CPU horsepower. Even if all that’s happening is all of the 1900 extensions setting up their hints, that’s still something like 5500 hints to process.

From experience:

Doing the code is the easy part. Building it into a module is no small undertaking. I found the new object model daunting, and I’m not new to this programming thing. What I did that with the SCCP Manager on the old FreePBX (1.8 and before) was still a LOT of work. All of the new supporting code, packaging, BMO-interfacing, and GUI tricks make it a big effort and time investment for someone that only does this once. Even then, the project ends up in the repository as a “here you go, knock yourself out” piece of software.

Don’t get me wrong, I’d do that work again if I wasn’t working full time and running my own business. Having some teaming from the commercial folks for an “open” module would be really cool. Of course, how does one fund something like that? Sangoma’s people are paid to do this, so asking us to ask them isn’t really appropriate, and if Sangoma asks them to do it for free, there are labor rules about that…

A compromise solution might be to take James’ “basic” hello world starter kit and put together a webinar explaining how the pieces go together for us “wanna-be” module developers. Perhaps a clear path for “promotion” and (of course) the obligatory NDA/Release that Sangoma has all
developers sign might be a good first step.

I have a personalized version of this module (there is a copy on Github) which adds repeating alarms functionality, the lack of which is holding me back from moving to V14.

I want to port this to v14. I have now made two attempts to do this. The first was partially working when I realized that it was actually based on V13. I then tried using ‘Ring-groups’ as a base but am really at a loss to complete this. There seems to be so much ‘hidden’ stuff. For example, I cannot work out how the ‘Submit’ button works so can’t modify it to save the details to a different table. There are also loads of functions defined which are not explicitly referenced from anywhere in the module code.

The main functionality I need is to input several fields including time and date and to be able to select an extension, and then save them to a table.

Also the ability to amend and delete entries.

Any advice on what existing module to base it on and pointers to V14 methodology would be greatly welcome.

I have managed to work out how to display existing data from the database table but that is about it.

I’m not intending to become a FreePBX developer - just to gain the basic knowledge to achieve the above.

<OCD Mode>32 BYTE password - 256 bit password.</OCD Mode>

Hi Greg,

Thanks for your response. You are correct about #1. Auto-Intercom is enabled and works very well. On #2, I have set the Grandstream phone to default transfer mode to Attended Transfer Only (Advanced settings - Call Features - Default Transfer Mode) This takes away the blind/attended option and forces it to be attended.

I can hit transfer, extension, send and announce the call. When I hit END to drop off, the transferred caller is then on the speakerphone. I would like for the phone to ring and the person have to answer it, or let it ring thru to voice mail. Could I be missing a setting in FreePBX on transfered call behavior? There are so many setting…

Thanks,
Trey

thats mean not to port forward right?

MySQL buf info has some interesting discussion about this error. Make sure your queries are not exceeding the “in application” timeout (which is usually about 30 seconds).

I vaguely remember this error from about a year (maybe 18 months) ago. It was early in the Firewall implementation, but I don’t remember what the cure was. Perhaps a quick scan through the archives would benefit you?

If that’s the Sangoma USB GSM dongle - you’re out of luck. The card is End of Life and is no longer supported.

If not, then never mind…

You might try cross-posting this to one of the Asterisk forums, since it appears to be the underlying Asterisk (version 13) that’s giving you problems. While there’s every possibility someone here might be able to help you, going to the source (as a backup) can’t hurt.

Short of that, I think there is a way to set the system to use a shorter RTP packet size in one of the config files in /etc/asterisk.

Not the reload but the GUI slowness that others were experiencing with a large number of extensions is what I read somewhere will be much improved with PHP 7, or did I dream that?

Hello!

Yes im still using elastix 2.5 with a freepbx 2.10

I just purchased a Yealink T54S with EXP50 Sidecar. I know the EPM does not show EXP50 as an option to auto provision the sidecar. Does anyone know if by selecting EXP40 if that will work? Once I get to the site I can test and experiment. I am too far away at the moment.

Sorry that’s not how SIP transfers work. Once the phone auto answer it starts the transfer process. It can’t than hangup the auto answer and send the transfer after

Long past end of life. Upgrade to a supported system:
https://wiki.freepbx.org/display/PPS/Elastix+and+PBXinaFlash+to+FreePBX+Distro+Conversion+Tool

I was afraid of that.

Is there a best practice for users that want their calls screened and announced before taking the call?

Thanks again for your help - I am not 100% sure what I’m looking for here so have some more detailed log info:

<— SIP read from UDP:XXX.XXX.XXX.XXX:48892 —>
REGISTER sip:SIPSERVER:5060 SIP/2.0
Call-ID: 8773a18d@192.168.0.152
Content-Length: 0
CSeq: 36166 REGISTER
From: sip:PSTNLINE@SIPSERVER;tag=SP22cc12aea5b6b12a3
Max-Forwards: 70
To: sip:PSTNLINE@SIPSERVER
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:48892;branch=z9hG4bK-5ae61fd8;rport
Authorization: DIGEST algorithm=MD5,nonce=“7ff3c6c4”,realm=“asterisk”,response=“bb4ffe07e598d966371a185cd5c746d6”,uri=“sip:SIPSERVER:5060”,username=“PSTNLINE”
User-Agent: OBIHAI/OBi110-1.3.0.2860
Contact: sip:PSTNLINE@XXX.XXX.XXX.XXX:48892;expires=60;+sip.instance=“urn:uuid:00000000-0000-0000-0000-9cadef01531a”
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE
Supported: replaces

<------------->
— (13 headers 0 lines) —
Sending to XXX.XXX.XXX.XXX:48892 (NAT)
[2018-04-02 16:19:01] NOTICE[1686]: chan_sip.c:17278 check_auth: Correct auth, but based on stale nonce received from ‘sip:PSTNLINE@SIPSERVER;tag=SP22cc12aea5b6b12a3’

<— Transmitting (NAT) to XXX.XXX.XXX.XXX:48892 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:48892;branch=z9hG4bK-5ae61fd8;received=XXX.XXX.XXX.XXX;rport=48892
From: sip:PSTNLINE@SIPSERVER;tag=SP22cc12aea5b6b12a3
To: sip:PSTNLINE@SIPSERVER;tag=as2c7f3bab
Call-ID: 8773a18d@192.168.0.152
CSeq: 36166 REGISTER
Server: FPBX-14.0.1.19(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“661e9dc4”, stale=true
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘8773a18d@192.168.0.152’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:XXX.XXX.XXX.XXX:48892 —>
REGISTER sip:SIPSERVER:5060 SIP/2.0
Call-ID: 8773a18d@192.168.0.152
Content-Length: 0
CSeq: 36167 REGISTER
From: sip:PSTNLINE@SIPSERVER;tag=SP22cc12aea5b6b12a3
Max-Forwards: 70
To: sip:PSTNLINE@SIPSERVER
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:48892;branch=z9hG4bK-88969d8;rport
Authorization: DIGEST algorithm=MD5,nonce=“661e9dc4”,realm=“asterisk”,response=“a2848b869056b41d2bba406d96217916”,uri=“sip:SIPSERVER:5060”,username=“PSTNLINE”
User-Agent: OBIHAI/OBi110-1.3.0.2860
Contact: sip:PSTNLINE@XXX.XXX.XXX.XXX:48892;expires=60;+sip.instance=“urn:uuid:00000000-0000-0000-0000-9cadef01531a”
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE
Supported: replaces

<------------->
— (13 headers 0 lines) —
Sending to XXX.XXX.XXX.XXX:48892 (NAT)
Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:48892:
OPTIONS sip:PSTNLINE@XXX.XXX.XXX.XXX:48892 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK02d24a2f;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@YYY.YYY.YYY.YYY;tag=as534d40de
To: sip:PSTNLINE@XXX.XXX.XXX.XXX:48892
Contact: sip:Unknown@YYY.YYY.YYY.YYY:5060
Call-ID: 57a660aa4cf92cae482c6b7e5884ce2f@YYY.YYY.YYY.YYY:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.1.19(13.17.2)
Date: Mon, 02 Apr 2018 15:19:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<— Transmitting (NAT) to XXX.XXX.XXX.XXX:48892 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:48892;branch=z9hG4bK-88969d8;received=XXX.XXX.XXX.XXX;rport=48892
From: sip:PSTNLINE@SIPSERVER;tag=SP22cc12aea5b6b12a3
To: sip:PSTNLINE@SIPSERVER;tag=as2c7f3bab
Call-ID: 8773a18d@192.168.0.152
CSeq: 36167 REGISTER
Server: FPBX-14.0.1.19(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: sip:PSTNLINE@XXX.XXX.XXX.XXX:48892;expires=60
Date: Mon, 02 Apr 2018 15:19:01 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘8773a18d@192.168.0.152’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:XXX.XXX.XXX.XXX:48892 —>
SIP/2.0 200 OK
Call-ID: 57a660aa4cf92cae482c6b7e5884ce2f@YYY.YYY.YYY.YYY:5060
CSeq: 102 OPTIONS
Content-Length: 0
From: “Unknown” sip:Unknown@YYY.YYY.YYY.YYY;tag=as534d40de
To: sip:PSTNLINE@XXX.XXX.XXX.XXX:48892
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK02d24a2f;received=YYY.YYY.YYY.YYY;rport=5060

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘57a660aa4cf92cae482c6b7e5884ce2f@YYY.YYY.YYY.YYY:5060’ Method: OPTIONS
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
– Remote UNIX connection disconnected

<— SIP read from UDP:ZZZ.ZZZ.ZZZ.ZZZ:2925 —>
INVITE sip:01234567890@SIPSERVER SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bKfGu5BBiXwSRdyaMI;rport
Contact: sip:205@ZZZ.ZZZ.ZZZ.ZZZ:2925
Max-Forwards: 70
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
To: sip:01234567890@SIPSERVER
Content-Type: application/sdp
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
CSeq: 1 INVITE
User-Agent: Acrobits Softphone/5.9
Content-Length: 362

v=0
o=- 6064877934 42815 IN IP4 172.26.170.170
s=hmnnshv
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 43930 RTP/AVP 0 8 9 103 102 3 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:102 iLBC/8000
a=rtpmap:103 opus/48000/2
a=fmtp:101 0-15
a=fmtp:102 mode=20
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=ptime:20
a=sendrecv
<------------->
— (13 headers 14 lines) —
Sending to ZZZ.ZZZ.ZZZ.ZZZ:2925 (NAT)
Sending to ZZZ.ZZZ.ZZZ.ZZZ:2925 (NAT)
Using INVITE request as basis request - F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
Found peer ‘205’ for ‘205’ from ZZZ.ZZZ.ZZZ.ZZZ:2925

<— Reliably Transmitting (NAT) to ZZZ.ZZZ.ZZZ.ZZZ:2925 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bKfGu5BBiXwSRdyaMI;received=ZZZ.ZZZ.ZZZ.ZZZ;rport=2925
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
To: sip:01234567890@SIPSERVER;tag=as5f384dde
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
CSeq: 1 INVITE
Server: FPBX-14.0.1.19(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“14435003”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B’ in 13760 ms (Method: INVITE)
Retransmitting #1 (NAT) to ZZZ.ZZZ.ZZZ.ZZZ:2925:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bKfGu5BBiXwSRdyaMI;received=ZZZ.ZZZ.ZZZ.ZZZ;rport=2925
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
To: sip:01234567890@SIPSERVER;tag=as5f384dde
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
CSeq: 1 INVITE
Server: FPBX-14.0.1.19(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“14435003”
Content-Length: 0

---

Retransmitting #2 (NAT) to ZZZ.ZZZ.ZZZ.ZZZ:2925:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bKfGu5BBiXwSRdyaMI;received=ZZZ.ZZZ.ZZZ.ZZZ;rport=2925
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
To: sip:01234567890@SIPSERVER;tag=as5f384dde
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
CSeq: 1 INVITE
Server: FPBX-14.0.1.19(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“14435003”
Content-Length: 0

---

<— SIP read from UDP:ZZZ.ZZZ.ZZZ.ZZZ:2925 —>
ACK sip:01234567890@SIPSERVER SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bKfGu5BBiXwSRdyaMI;rport
Max-Forwards: 70
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
To: sip:01234567890@SIPSERVER;tag=as5f384dde
CSeq: 1 ACK
User-Agent: Acrobits Softphone/5.9
Content-Length: 0

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

<— SIP read from UDP:ZZZ.ZZZ.ZZZ.ZZZ:2925 —>
ACK sip:01234567890@SIPSERVER SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bKfGu5BBiXwSRdyaMI;rport
Max-Forwards: 70
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
To: sip:01234567890@SIPSERVER;tag=as5f384dde
CSeq: 1 ACK
User-Agent: Acrobits Softphone/5.9
Content-Length: 0

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

<— SIP read from UDP:ZZZ.ZZZ.ZZZ.ZZZ:2925 —>
INVITE sip:01234567890@SIPSERVER SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bK3R14bbSuUFTneRrP;rport
Contact: sip:205@ZZZ.ZZZ.ZZZ.ZZZ:2925
Max-Forwards: 70
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
To: sip:01234567890@SIPSERVER
Content-Type: application/sdp
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
CSeq: 2 INVITE
Authorization: Digest username=“205”,realm=“asterisk”,algorithm=MD5,uri=“sip:01234567890@SIPSERVER”,nonce=“14435003”,response=“59db2506905aa8d91c574920f8e32326”
User-Agent: Acrobits Softphone/5.9
Content-Length: 362

v=0
o=- 6064877934 42815 IN IP4 172.26.170.170
s=hmnnshv
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 43930 RTP/AVP 0 8 9 103 102 3 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:102 iLBC/8000
a=rtpmap:103 opus/48000/2
a=fmtp:101 0-15
a=fmtp:102 mode=20
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=ptime:20
a=sendrecv
<------------->
— (14 headers 14 lines) —
Sending to ZZZ.ZZZ.ZZZ.ZZZ:2925 (NAT)
Using INVITE request as basis request - F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
Found peer ‘205’ for ‘205’ from ZZZ.ZZZ.ZZZ.ZZZ:2925
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found audio description format iLBC for ID 102
Found audio description format opus for ID 103
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|)
Peer audio RTP is at port ZZZ.ZZZ.ZZZ.ZZZ:43930
Looking for 01234567890 in from-internal (domain SIPSERVER)
sip_route_dump: route/path hop: sip:205@ZZZ.ZZZ.ZZZ.ZZZ:2925

<— Transmitting (NAT) to ZZZ.ZZZ.ZZZ.ZZZ:2925 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bK3R14bbSuUFTneRrP;received=ZZZ.ZZZ.ZZZ.ZZZ;rport=2925
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
To: sip:01234567890@SIPSERVER
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
CSeq: 2 INVITE
Server: FPBX-14.0.1.19(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:01234567890@YYY.YYY.YYY.YYY:5060
Content-Length: 0

<------------>
– Executing [01234567890@from-internal:1] Macro(“SIP/205-00000189”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/205-00000189”, “TOUCH_MONITOR=1522682344.800”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/205-00000189”, “AMPUSER=205”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/205-00000189”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/205-00000189”, “1?Set(__REALCALLERIDNUM=205)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/205-00000189”, “AMPUSER=205”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/205-00000189”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/205-00000189”, “AMPUSERCIDNAME=Anthony C”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/205-00000189”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/205-00000189”, “AMPUSERCID=205”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/205-00000189”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/205-00000189”, “CALLERID(all)=“Anthony C” <205>”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/205-00000189”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“SIP/205-00000189”, “1?Set(GROUP(concurrency_limit)=205)”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/205-00000189”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:15] GotoIf(“SIP/205-00000189”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,29)
– Executing [s@macro-user-callerid:29] Set(“SIP/205-00000189”, “CALLERID(number)=205”) in new stack
– Executing [s@macro-user-callerid:30] Set(“SIP/205-00000189”, “CALLERID(name)=Anthony C”) in new stack
– Executing [s@macro-user-callerid:31] GotoIf(“SIP/205-00000189”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:32] Set(“SIP/205-00000189”, “CDR(cnam)=Anthony C”) in new stack
– Executing [s@macro-user-callerid:33] Set(“SIP/205-00000189”, “CDR(cnum)=205”) in new stack
– Executing [s@macro-user-callerid:34] Set(“SIP/205-00000189”, “CHANNEL(language)=en_GB”) in new stack
– Executing [01234567890@from-internal:2] Gosub(“SIP/205-00000189”, “sub-record-check,s,1(out,01234567890,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/205-00000189”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/205-00000189”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/205-00000189”, “NOW=1522682344”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/205-00000189”, “__DAY=02”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/205-00000189”, “__MONTH=04”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/205-00000189”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/205-00000189”, “__TIMESTR=20180402-161904”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/205-00000189”, “__FROMEXTEN=205”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/205-00000189”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/205-00000189”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/205-00000189”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/205-00000189”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/205-00000189”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/205-00000189”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/205-00000189”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“SIP/205-00000189”, “Outbound Recording Check from 205 to 01234567890”) in new stack
– Executing [out@sub-record-check:2] Set(“SIP/205-00000189”, “RECMODE=force”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“SIP/205-00000189”, “0?Goto(routewins)”) in new stack
– Executing [out@sub-record-check:4] ExecIf(“SIP/205-00000189”, “0?Goto(routewins)”) in new stack
– Executing [out@sub-record-check:5] Gosub(“SIP/205-00000189”, “recordcheck,1(force,out,01234567890)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/205-00000189”, “Starting recording check against force”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/205-00000189”, “force”) in new stack
– Goto (sub-record-check,recordcheck,5)
– Executing [recordcheck@sub-record-check:5] Set(“SIP/205-00000189”, “__REC_POLICY_MODE=FORCE”) in new stack
– Executing [recordcheck@sub-record-check:6] GotoIf(“SIP/205-00000189”, “1?startrec”) in new stack
– Goto (sub-record-check,recordcheck,16)
– Executing [recordcheck@sub-record-check:16] NoOp(“SIP/205-00000189”, “Starting recording: out, 01234567890”) in new stack
– Executing [recordcheck@sub-record-check:17] Set(“SIP/205-00000189”, “AUDIOHOOK_INHERIT(MixMonitor)=yes”) in new stack
– Executing [recordcheck@sub-record-check:18] Set(“SIP/205-00000189”, “__CALLFILENAME=out-01234567890-205-20180402-161904-1522682344.800”) in new stack
– Executing [recordcheck@sub-record-check:19] MixMonitor(“SIP/205-00000189”, “2018/04/02/out-01234567890-205-20180402-161904-1522682344.800.wav,abi(LOCAL_MIXMON_ID),”) in new stack
– Executing [recordcheck@sub-record-check:20] Set(“SIP/205-00000189”, “__MIXMON_ID=0x7f9f08074ab0”) in new stack
– Executing [recordcheck@sub-record-check:21] Set(“SIP/205-00000189”, “__RECORD_ID=SIP/205-00000189”) in new stack
– Executing [recordcheck@sub-record-check:22] Set(“SIP/205-00000189”, “__REC_STATUS=RECORDING”) in new stack
– Executing [recordcheck@sub-record-check:23] Set(“SIP/205-00000189”, “CDR(recordingfile)=out-01234567890-205-20180402-161904-1522682344.800.wav”) in new stack
– Executing [recordcheck@sub-record-check:24] Return(“SIP/205-00000189”, “”) in new stack
– Executing [out@sub-record-check:6] Return(“SIP/205-00000189”, “”) in new stack
– Executing [01234567890@from-internal:3] ExecIf(“SIP/205-00000189”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [01234567890@from-internal:4] Set(“SIP/205-00000189”, “MOHCLASS=default”) in new stack
– Executing [01234567890@from-internal:5] Set(“SIP/205-00000189”, “_NODEST=”) in new stack
– Executing [01234567890@from-internal:6] Macro(“SIP/205-00000189”, “dialout-trunk,2,01234567890,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/205-00000189”, “DIAL_TRUNK=2”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/205-00000189”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/205-00000189”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/205-00000189”, “DIAL_NUMBER=01234567890”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/205-00000189”, “DIAL_TRUNK_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/205-00000189”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/205-00000189”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/205-00000189”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/205-00000189”, “DIAL_TRUNK_OPTIONS=T”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/205-00000189”, “outbound-callerid,2”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/205-00000189”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/205-00000189”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:3] ExecIf(“SIP/205-00000189”, “0?Set(REALCALLERIDNUM=205)”) in new stack
– Executing [s@macro-outbound-callerid:4] GotoIf(“SIP/205-00000189”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,7)
– Executing [s@macro-outbound-callerid:7] Set(“SIP/205-00000189”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/205-00000189”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:9] Set(“SIP/205-00000189”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:10] GotoIf(“SIP/205-00000189”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,15)
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/205-00000189”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:16] ExecIf(“SIP/205-00000189”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:17] ExecIf(“SIP/205-00000189”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:18] ExecIf(“SIP/205-00000189”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:19] ExecIf(“SIP/205-00000189”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:20] Set(“SIP/205-00000189”, “CDR(outbound_cnum)=205”) in new stack

<— SIP read from UDP:ZZZ.ZZZ.ZZZ.ZZZ:2925 —>
ACK sip:01234567890@SIPSERVER SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bKfGu5BBiXwSRdyaMI;rport
Max-Forwards: 70
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
To: sip:01234567890@SIPSERVER;tag=as5f384dde
CSeq: 1 ACK
User-Agent: Acrobits Softphone/5.9
Content-Length: 0

<------------->
— (9 headers 0 lines) —
– Executing [s@macro-outbound-callerid:21] Set(“SIP/205-00000189”, “CDR(outbound_cnam)=Anthony C”) in new stack
– Executing [s@macro-dialout-trunk:12] GosubIf(“SIP/205-00000189”, “0?sub-flp-2,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/205-00000189”, “OUTNUM=01234567890”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/205-00000189”, “custom=SIP/PSTNLINE”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/205-00000189”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“SIP/205-00000189”, “0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:17] Macro(“SIP/205-00000189”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/205-00000189”, “”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/205-00000189”, “0?skipcrm”) in new stack
– Executing [s@macro-dialout-trunk:19] Set(“SIP/205-00000189”, “__CRM_DIRECTION=OUTBOUND”) in new stack
– Executing [s@macro-dialout-trunk:20] Set(“SIP/205-00000189”, “__CRM_DESTINATION=01234567890”) in new stack
– Executing [s@macro-dialout-trunk:21] Set(“SIP/205-00000189”, “__CRM_SOURCE=205”) in new stack
– Executing [s@macro-dialout-trunk:22] AGI(“SIP/205-00000189”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
== Begin MixMonitor Recording SIP/205-00000189
– <SIP/205-00000189>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@macro-dialout-trunk:23] Set(“SIP/205-00000189”, “CHANNEL(hangup_handler_push)=crm-hangup,s,1”) in new stack
– Executing [s@macro-dialout-trunk:24] NoOp(“SIP/205-00000189”, “CRM Finished”) in new stack
– Executing [s@macro-dialout-trunk:25] GotoIf(“SIP/205-00000189”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:26] ExecIf(“SIP/205-00000189”, “1?Set(CONNECTEDLINE(num,i)=01234567890)”) in new stack
– Executing [s@macro-dialout-trunk:27] GotoIf(“SIP/205-00000189”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:28] Dial(“SIP/205-00000189”, “SIP/PSTNLINE/01234567890,300,T”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 16614
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:48892:
INVITE sip:01234567890@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK0ce2ac51;rport
Max-Forwards: 70
From: “Anthony C” sip:205@YYY.YYY.YYY.YYY;tag=as70ec5382
To: sip:01234567890@XXX.XXX.XXX.XXX
Contact: sip:205@YYY.YYY.YYY.YYY:5060
Call-ID: 159b94f11e34287a08b813332adf1bad@YYY.YYY.YYY.YYY:5060
CSeq: 102 INVITE
User-Agent: FPBX-14.0.1.19(13.17.2)
Date: Mon, 02 Apr 2018 15:19:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 192411319 192411319 IN IP4 YYY.YYY.YYY.YYY
s=Asterisk PBX 13.17.2
c=IN IP4 YYY.YYY.YYY.YYY
t=0 0
m=audio 16614 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

-- Called SIP/PSTNLINE/01234567890

<— SIP read from UDP:XXX.XXX.XXX.XXX:48892 —>
SIP/2.0 100 Trying
Call-ID: 159b94f11e34287a08b813332adf1bad@YYY.YYY.YYY.YYY:5060
CSeq: 102 INVITE
Content-Length: 0
From: “Anthony C” sip:205@YYY.YYY.YYY.YYY;tag=as70ec5382
To: sip:01234567890@XXX.XXX.XXX.XXX;tag=SP22cc12aea5b6b12a3
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK0ce2ac51;received=YYY.YYY.YYY.YYY;rport=5060
Server: OBIHAI/OBi110-1.3.0.2860

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

<— SIP read from UDP:XXX.XXX.XXX.XXX:48892 —>
SIP/2.0 486 Busy Here
Call-ID: 159b94f11e34287a08b813332adf1bad@YYY.YYY.YYY.YYY:5060
CSeq: 102 INVITE
Content-Length: 0
From: “Anthony C” sip:205@YYY.YYY.YYY.YYY;tag=as70ec5382
To: sip:01234567890@XXX.XXX.XXX.XXX;tag=SP22cc12aea5b6b12a3
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK0ce2ac51;received=YYY.YYY.YYY.YYY;rport=5060
Server: OBIHAI/OBi110-1.3.0.2860

<------------->
— (8 headers 0 lines) —
– Got SIP response 486 “Busy Here” back from XXX.XXX.XXX.XXX:48892
Transmitting (NAT) to XXX.XXX.XXX.XXX:48892:
ACK sip:01234567890@XXX.XXX.XXX.XXX SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK0ce2ac51;rport
Max-Forwards: 70
From: “Anthony C” sip:205@YYY.YYY.YYY.YYY;tag=as70ec5382
To: sip:01234567890@XXX.XXX.XXX.XXX;tag=SP22cc12aea5b6b12a3
Contact: sip:205@YYY.YYY.YYY.YYY:5060
Call-ID: 159b94f11e34287a08b813332adf1bad@YYY.YYY.YYY.YYY:5060
CSeq: 102 ACK
User-Agent: FPBX-14.0.1.19(13.17.2)
Content-Length: 0

---

-- SIP/PSTNLINE-0000018a is busy

== Everyone is busy/congested at this time (1:1/0/0)
– Executing [s@macro-dialout-trunk:29] NoOp(“SIP/205-00000189”, “Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17”) in new stack
– Executing [s@macro-dialout-trunk:30] GotoIf(“SIP/205-00000189”, “0?continue,1:s-BUSY,1”) in new stack
– Goto (macro-dialout-trunk,s-BUSY,1)
– Executing [s-BUSY@macro-dialout-trunk:1] NoOp(“SIP/205-00000189”, “Dial failed due to trunk reporting BUSY - giving up”) in new stack
– Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(“SIP/205-00000189”, “busy”) in new stack
[2018-04-02 16:19:05] WARNING[26088][C-00000633]: translate.c:407 framein: no samples for ulawtolin
– Executing [s-BUSY@macro-dialout-trunk:3] Busy(“SIP/205-00000189”, “20”) in new stack

<— Reliably Transmitting (NAT) to ZZZ.ZZZ.ZZZ.ZZZ:2925 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:2925;branch=z9hG4bK3R14bbSuUFTneRrP;received=ZZZ.ZZZ.ZZZ.ZZZ;rport=2925
From: “Anthony C” sip:205@SIPSERVER;tag=20ACEB9E71BCCD97699AAE4FD9398D7B
To: sip:01234567890@SIPSERVER;tag=as6514128f
Call-ID: F434F1AFF89E090B8CBF3FD1D9212ADF5949EA3B
CSeq: 2 INVITE
Server: FPBX-14.0.1.19(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: User busy
X-Asterisk-HangupCauseCode: 17
Content-Length: 0

<------------>
== Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on ‘SIP/205-00000189’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 01234567890, 6) exited non-zero on ‘SIP/205-00000189’
– Executing [h@from-internal:1] Macro(“SIP/205-00000189”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/205-00000189”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
Really destroying SIP dialog ‘159b94f11e34287a08b813332adf1bad@YYY.YYY.YYY.YYY:5060’ Method: INVITE
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/205-00000189”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“SIP/205-00000189”, " monior file= /var/spool/asterisk/monitor/2018/04/02/out-01234567890-205-20180402-161904-1522682344.800.wav") in new stack
– Executing [s@macro-hangupcall:5] AGI(“SIP/205-00000189”, “attendedtransfer-rec-restart.php,/var/spool/asterisk/monitor/2018/04/02/out-01234567890-205-20180402-161904-1522682344.800.wav”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php

good day dear alan,

i have such trouble.

i want to call some numbers and have their self numbers in their phones.

i need something like set outbound CLI = Dilaled number.

please advise,

I have a number of Fail2Ban reports from several of our FreePBX servers where the server’s IP itself is banned. So far, there’s one ban for most of the servers. One particular server, however, has recurring bans for the last 12 hours.

Is this some sort of spoof attack?