@bflinte wrote:

Newbie question: Is there a way for an Administrator to change Call Forwarding or DND for an end user? I can see if these features are active through User Management, but the only way I have found to modify them is through the UCP. The only 2 ways I have found is to log into the UCP as that end user or as another user that has rights to modify another person's settings. If I choose the latter I have to specify each user has rights to do this. I'm sure there is an easy way, but it is evading me.

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@ipserviceskenya wrote:

Dear All,
Still newbie on FreePBX. This might have been discussed before with no specific answer.
Am using FreePBX 13.0.124 I want to allow ONLY CIDs with 11 digits

Thanks
IPservices

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@Vyspa wrote:

I have a system where we want to direct each persons DDI to them for 5 seconds, if they dont answer it goes to the main sales queue for 10 seconds. if it is not answered by an agent within 10 seconds it goes to the original users voicemail. How can i do this without having to create a whole load of queues. Is this a feature of vqplus?

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@hiddenvision wrote:

Hi All,
I wonder if this is a feature or perhaps an oversight.

With voicemail blasting I notice that if a group member has a temporary message recorded then the caller hears that before they get to leave a message.
This is great for a single member voicemail group.! but pops up an annoying situation if this is the general afterhours message group, where as it happens one of the staff members is away and has set a message on her personal voicemail box.

Would hate to think what would happen if more members of the group also had temp messages recorded.
It may take a while to finally leave that message.!!

Is there a setting I have missed to turn of this feature.?
I would prefer it to ignore any temp message.
Actually it would be nice if it was switchable, per extension / group perhaps.!

I also notice that directing the call to voicemail (no-msg) still plays the temp message.

or could it be described as funky and need surgery.?

On the same-ish subject
I noticed that the Press"0" for operator was not working via voicemail blasting yet it does on calls to individual extension voicemails.

EDIT:-
Oops.. Versions OLD
Voicemail 12.0.44
VmBlasting 12.0.1
FreePBX 6.12.65-30
Asterisk 1.8.32.3

Hv.

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@kwriley87 wrote:

Hello

This should be pretty basic but I am running into an issue here.
I have two PBX's both running FreePBX 12.0.76.3.
I have created IAX trunks on both PBX's and the status on both PBX's is showing to be okay.

I have System1 and System2
System1 has a DID that I am sending over the IAX trunk to System2
System2 has that same DID listed under inbound routes that is pointing to a voicemail box.
When I dial the DID, I see it hit System2 via Putty logs but the call does not complete.. here is the output where the call fails:

-- Called SIP/VOIP_INNOV_PRIMARY/16822027111
[2016-06-08 22:35:38] WARNING[26964]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 190481e9074b21566c4fde9d55f827a1@71.164.156.230:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2016-06-08 22:35:38] WARNING[26964]: chan_sip.c:4053 retrans_pkt: Hanging up call 190481e9074b21566c4fde9d55f827a1@71.164.156.230:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:23] NoOp("IAX2/iaxtest-1179", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 18") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("IAX2/iaxtest-1179", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("IAX2/iaxtest-1179", "RC=18") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("IAX2/iaxtest-1179", "18,1") in new stack
-- Goto (macro-dialout-trunk,18,1)
-- Executing [18@macro-dialout-trunk:1] Goto("IAX2/iaxtest-1179", "s-NOANSWER,1") in new stack
-- Goto (macro-dialout-trunk,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-dialout-trunk:1] NoOp("IAX2/iaxtest-1179", "Dial failed due to trunk reporting NOANSWER - giving up") in new stack
-- Executing [s-NOANSWER@macro-dialout-trunk:2] Progress("IAX2/iaxtest-1179", "") in new stack
-- Executing [s-NOANSWER@macro-dialout-trunk:3] Playback("IAX2/iaxtest-1179", "number-not-answering,noanswer") in new stack
-- Playing 'number-not-answering.ulaw' (language 'en')

Can anyone provide any advice as to what the issue is?

Thank you!

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@ganiherawan wrote:

help me.

How to route server PBX non line to server PBX any line to call phone number ??

thnks

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@rco wrote:

I have a ring group that calls four internal extensions and two external numbers in 'ringall' configuration. When I call the ring group extension, all internal extensions ring, but the external lines do not. When I look at the CDR reports, the external numbers are reported as 'busy'. The numbers are written with a # at the end of the last number in the extension list.

This has been working perfectly from the time I set it up, but have just noticed this issue now. I haven't made any changes to the system since the time that they were functioning correctly that I know of. When I call the external numbers directly, the call goes through without issue.

I attached a screenshot of the ring group settings (numbers removed) and my outgoing dialing rules (these haven't been changed at all).

Would anyone have any advice? its just a home setup, so don't have a great deal of experience with some of the finer points of freepbx (:

Thanks

2016-06-10 02_37_46-Greenshot.png947x949 50.8 KB

Dialing rules are 'XX.', for the match pattern, couldn't upload a second screenshot

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@amztransit wrote:

Followed these instructions: http://charlesauer.net/tutorials/centos/postfix-as-gmail-relay-centos.php

And still having trouble. Trying to set this up so that I can send emails out from FreePBX obviously. Everytime I send a test message, /var/log/maillog still returns:

postfix/smtp[31970]: 7880343D513C: SASL authentication failed; server smtp.gmail.com[74.125.22.108] said: 534-5.7.14 <https://accounts.google.com/signin/continue?sarp=1&scc=1&plt=AKgnsbsS?534-5.7.14 SVSFq7E-OTTONQlngmEb_fm4fRMazei637OGMU4Kq2MXKcF0YWxX0rtwtXGsGjCywTNu_a?534-5.7.14 1y4zA0I0b3XAMM6zPiTef2Ob4Pk1KpBXT0NvqW2vVV5Y3-BJliImFGtngnAI0d2Dcyuua9?534-5.7.14 Y7Myd-hNV5QxUWrLytnHI4FQmQuU91MYI-AkVNqTF_JZGBuVX236dDoRt_i42DQPnFdlyB?534-5.7.14 WWXVYStViTgACtoFs_7sWtmhI168s> Please log in via your web browser and?534-5.7.14 then try again.?534-5.7.14 Learn more at?534 5.7.14 https://support.google.com/mail/answer/78754 q190sm2001023qhb.9 - gsmtp

I'm stuck. What else can I try to get SASL to work?

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@orkwizard wrote:

Hi Everybody,

This seems a really active and helpful community. Im pretty new to FreePBX so im sorry if i make a newbie question. I work for an hotel which has finally accepted that Avaya is not the "future". We need to integrate the PMS (Property Managment System) with Asterisk to perform some hotelier stuff (voice mail, caller_id etc). We pretty much figure out how to do some stuff but one is bugging our mind if how we can dynamically when the PMS perfoms a guest check in let know Asterisk to change the DIALPLAN to allow calls, and the when a checkout is flagged modify the extension to close the line. In our PMS we can generate a script or file to be read by freepbx. Does anyone can point out what files or tables in mysql (in case we have to) modify. My logic says that i can create a config file and have amportal reload the configuration...

Again thanks for your help!

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@ganiherawan wrote:

how to secure a server with a sip trunks of the user or an outsider trying to break into the server . to my server using linux cent os.

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@dsgii wrote:

Using the latest FreePBX distro (10.13.66-12) and I have been noticing something I just don't get.

Commercial Modules that require a license to function: Do these do ANYTHING other than provide useless GUI menus and confusion. System Administration module would be an exception as the Commercial License enables additional features. I saw a couple other Commercial modules that do not state they require anything at all to function and do not show the "Buy"
button (Zulu would be one).

I am only using softphones with my installation so some of the other modules like OSS and EndPoint manager are fairly useless to me (unless I am missing something). I use the Bulk Phone Restart module, but every single update script barks about this or that not updating because of this or that module.

For me I am removing all commercial modules that require a license to run, or when you go to their UI within FreePBX you get a message indicating - no license, no workie.

There's more and these dependencies would be nice to know in advance. I am not afraid of doing a little RTFM, but I am not finding what I am looking for on the documentation WIKI. If anyone can point me in a direction this is what I am looking for:

Module Administration Best Practices - Disable, Enable, Remove. When and why per module. I'm not worried about disk space or running a lean system, I just don't/can't use certain default modules, I get it all setup and the update script puts it all back

Module Dependency Mapping - Which Modules Require Which other Modules. If a "free" module depends on a commercial module and the commercial module requires a license, is the free module actually working?

I am not necessarily looking for specific answers to my thoughts here, I am hoping someone out there understands the general idea of what I mean here.

Thank you for your help in advance.

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@andrew_miller wrote:

Hello,

I have FreePBX 13.0.124 installed and working with a number of Yealink and soft phones.

I've purchased a new Cisco SPA112 and upgraded the firmware to 1.4.1(002) as the previous one did not work either.

According to the SPA112 it fails to register with freePBX but on the FreePBX dashboard it says the SPA112 is online but you can't call the SPA112 extension.

I used the Quick setup screen on the SPA112 and set Proxy, Display Name, User ID and Password. And I have changed the standard 5060 port to another for security reasons.

The SPA112 extension is configured for pjsip. I've tried channel_sip but the FreePBX never shows it online.

Has anyone got the SPA112 to successfully register with FreePBX ? Are there any other settings that need to be change for it to register properly ?

Regards,

Andy

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@gatozgz wrote:

Hello good morning,

I have a doubt about that: How I can use different Outbound CIDs for different outbound routes.
It is not as easy as looks like to be, im going to show you an example related with I am triying to do through the FreePBX GUI.
From extensions.conf I know how can I do it but im not sure if I can do it from FreePBX GUI directly.

Example:
I have two groups:
- Systems: It belongs the extensions 1001,1002 and 1003
- Purchases: It belongs the extensions 2001,2002 and 2003

I have two PRI gateways:
- CISCO 2811, which connects to a PRI line (fixed) --> PRINCIPAL NUMBER 976000000
- CISCO 2801, which connects to a PRI line (mobile) --> PRINCIPAL NUMBER 600000000

I have two DID (fixed PRI line) associated to both groups:
- Systems DID --> 976100000
- Purchases DID --> 976200000

I have two ring groups associated to the groups and to the DID:
- Systems Ring Group --> 1000
- Purchases Ring Group --> 2000

What I want to do?
Assuming that I am extension 1001:
- If I do a call through PRI fixed I want to show this CID: 976100000
- If I do a call through PRI mobile I want to show this CID: 600000000 + 1000

I only can use one mask for outbound CID and one mask for CID Num Alias. Anyone know if I can do what I want using only FreePBX GUI? Or I will need to modify extensions_custom.conf?

Regards and thanks by all

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@al2lo wrote:

All calls timeout with cause code 18 gave by DAHI.

The debug info is below:

Have replaced card after lightning and still no luck.

Wanrouter Status:

Device name | Protocol | Station | Status |
wanpipe2 | AFT ISDN | N/A | Disconnected |
wanpipe1 | AFT ISDN | N/A | Disconnected |

pri status is:
PRI span 1/0: Up, Active
PRI span 2/0: Up, Active

any help is appreciated. we have forwarding on line at present as well to SIP(working).

[2016-06-10 13:42:37] VERBOSE[4415] chan_dahdi.c: PRI Span: 1 > Calling Number (len=17) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
[2016-06-10 13:42:37] VERBOSE[4415] chan_dahdi.c: PRI Span: 1 > Presentation: Presentation permitted, user number not screened (0) '+3534995XXXXX' ]
[2016-06-10 13:42:37] VERBOSE[4415] chan_dahdi.c: PRI Span: 1 > [70 0d a1 39 30 37 39 37 30 30 37 33 38 37 37]
[2016-06-10 13:42:37] VERBOSE[4415] chan_dahdi.c: PRI Span: 1 > Called Number (len=15) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '907970073877' ]
[2016-06-10 13:42:37] VERBOSE[4415] chan_dahdi.c: PRI Span: 1 > [a1]
[2016-06-10 13:42:37] VERBOSE[4415] chan_dahdi.c: PRI Span: 1 > Sending Complete (len= 1)
[2016-06-10 13:42:41] VERBOSE[4415] chan_dahdi.c: PRI Span: 1 T303 timed out. cref:32770

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@garethqct wrote:

Hi All,

I'm just in the process of setting up a new FreePBX 13 box installed on a MS HyperV box from the FreePBX-64bit-10.13.66 iso. It's to replace an old install that we have been using for a few years. I've got my trunk, incoming and outgoing routes, extensions, ring groups and voicemail setup fine and all working, however...

I have been trying to get some announcements up and running and seem to be struggling.

I recorded them using admin/system recordings, record over extension.
The recordings play back fine from the system recording page
Enabling the feature code allows me to listen to the recording by dialing the feature code from an extension.
I have created an announcement from the applications menu and picked the recording I made
I have also set the destination after playback to be my main ring group.

I then change my inbound routes from ring group to announcement.
The call is answered, no sound is played the call is eventually transfered to the ring group but picking up the receiver does not answer the call.

The following from the logs seems to indicate that the audio is being played and when the playback finishes it forwards on to ring group 600 (which is correct)

[2016-06-10 20:15:41] VERBOSE[47743][C-0000006c] pbx_builtins.c: Goto (app-announcement-1,s,1)
[2016-06-10 20:15:41] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:1] GotoIf("SIP/orbtalk_in-00000041", "0?begin") in new stack
[2016-06-10 20:15:41] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:2] Answer("SIP/orbtalk_in-00000041", "") in new stack
[2016-06-10 20:15:42] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:3] Wait("SIP/orbtalk_in-00000041", "1") in new stack
[2016-06-10 20:15:43] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:4] NoOp("SIP/orbtalk_in-00000041", "Playing announcement Daven Not Here") in new stack
[2016-06-10 20:15:43] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:5] Playback("SIP/orbtalk_in-00000041", "custom/davien_not_here,noanswer") in new stack
[2016-06-10 20:15:43] VERBOSE[47743][C-0000006c] file.c: <SIP/orbtalk_in-00000041> Playing 'custom/davien_not_here.slin' (language 'en')
[2016-06-10 20:15:57] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@app-announcement-1:6] Goto("SIP/orbtalk_in-00000041", "ext-group,600,1") in new stack
[2016-06-10 20:15:57] VERBOSE[47743][C-0000006c] pbx_builtins.c: Goto (ext-group,600,1)

Then after some more normal looking ring group entries I get the following;

[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] app_dial.c: PJSIP/127-00000130 answered SIP/orbtalk_in-00000041
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:1] Set("PJSIP/127-00000130", "__MACRO_RESULT=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:2] Set("PJSIP/127-00000130", "CFIGNORE=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:3] Set("PJSIP/127-00000130", "MASTER_CHANNEL(CFIGNORE)=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:4] Set("PJSIP/127-00000130", "FORWARD_CONTEXT=from-internal") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:5] Set("PJSIP/127-00000130", "MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:6] Macro("PJSIP/127-00000130", "blkvm-clr,") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-blkvm-clr:1] Set("PJSIP/127-00000130", "SHARED(BLKVM,SIP/orbtalk_in-00000041)=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-blkvm-clr:2] Set("PJSIP/127-00000130", "GOSUB_RETVAL=") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-blkvm-clr:3] MacroExit("PJSIP/127-00000130", "") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:7] ExecIf("PJSIP/127-00000130", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=127/sip:127@172.17.0.156:5060)") in new stack
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-auto-blkvm:8] ExecIf("PJSIP/127-00000130", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)") in new stack
[2016-06-10 20:16:02] VERBOSE[47822][C-0000006c] bridge_channel.c: Channel PJSIP/127-00000130 joined 'simple_bridge' basic-bridge <a94b380a-01e4-4d38-8a3e-bd7363543782>
[2016-06-10 20:16:02] VERBOSE[47743][C-0000006c] bridge_channel.c: Channel SIP/orbtalk_in-00000041 joined 'simple_bridge' basic-bridge <a94b380a-01e4-4d38-8a3e-bd7363543782>
[2016-06-10 20:16:02] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:03] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:03] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:04] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:04] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:05] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:05] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:06] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:06] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:07] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:07] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:08] WARNING[47822][C-0000006c] chan_pjsip.c: Can't send 10 type frames with PJSIP
[2016-06-10 20:16:08] VERBOSE[47822][C-0000006c] bridge_channel.c: Channel PJSIP/127-00000130 left 'simple_bridge' basic-bridge <a94b380a-01e4-4d38-8a3e-bd7363543782>
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] bridge_channel.c: Channel SIP/orbtalk_in-00000041 left 'simple_bridge' basic-bridge <a94b380a-01e4-4d38-8a3e-bd7363543782>
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] app_macro.c: Spawn extension (macro-dial, s, 17) exited non-zero on 'SIP/orbtalk_in-00000041' in macro 'dial'
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Spawn extension (ext-group, 600, 14) exited non-zero on 'SIP/orbtalk_in-00000041'
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Executing [h@ext-group:1] Macro("SIP/orbtalk_in-00000041", "hangupcall,") in new stack
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/orbtalk_in-00000041", "1?theend") in new stack
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/orbtalk_in-00000041", "0?Set(CDR(recordingfile)=)") in new stack
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/orbtalk_in-00000041", "") in new stack
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/orbtalk_in-00000041' in macro 'hangupcall'
[2016-06-10 20:16:08] VERBOSE[47743][C-0000006c] pbx.c: Spawn extension (ext-group, h, 1) exited non-zero on 'SIP/orbtalk_in-00000041'

I am behind a NAT router with 8 static address, I have port 5060 and 10000 - 20000 forwarding in on one of those address and have all outbound traffic from my Freepbx box going through the same IP.

I have tried setting up an IVR instead of the announcement and get the same issue.

Voicemail seems to be working fine. I can hear the recording and record a message with no problem.

Any help, pointers or additional reading suggestions for getting Announcements / IVR working would be greatly appreciated.

Thanks in advance

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@translux wrote:

I'm a bit over my skis on this one.
After updating several modules followed by an orderly shutdown we initially lost inbound/outbound service. I was so panicked trying to restore I'm not totally sure what I did that resolved connectivity, I believe it was editing DAHDI trunk channel name. We can make and receive calls but caller id is not working.

Any help would be greatly appreciated.

tail -f /var/log/asterisk/full

lay:1] Goto("DAHDI/i1/*phone_number*-4ca", "state-,1") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (sub-presencestate-display,state-,1)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [state-@sub-presencestate-display:1] Set("DAHDI/i1/*phone_number*-4ca", "PRESENCESTATE_DISPLAY=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [state-@sub-presencestate-display:2] Return("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:6] Set("DAHDI/i1/*phone_number*-4ca", "CONNECTEDLINE(name)=Eric Schmidt") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:7] Set("DAHDI/i1/*phone_number*-4ca", "FM_DIALSTATUS=NOT_INUSE") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:8] Set("DAHDI/i1/*phone_number*-4ca", "__EXTTOCALL=105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:9] Set("DAHDI/i1/*phone_number*-4ca", "__PICKUPMARK=105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:10] Macro("DAHDI/i1/*phone_number*-4ca", "blkvm-setifempty,") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-blkvm-setifempty:1] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?init") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (macro-blkvm-setifempty,s,4)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-blkvm-setifempty:4] Set("DAHDI/i1/*phone_number*-4ca", "_BLKVMCHANNEL=DAHDI/i1/*phone_number*-4ca") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-blkvm-setifempty:5] Set("DAHDI/i1/*phone_number*-4ca", "SHARED(BLKVM,DAHDI/i1/*phone_number*-4ca)=TRUE") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-blkvm-setifempty:6] Set("DAHDI/i1/*phone_number*-4ca", "GOSUB_RETVAL=TRUE") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-blkvm-setifempty:7] MacroExit("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:11] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?skipov") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (from-did-direct,105,14)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:14] Set("DAHDI/i1/*phone_number*-4ca", "RRNODEST=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:15] Set("DAHDI/i1/*phone_number*-4ca", "__NODEST=105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:16] GosubIf("DAHDI/i1/*phone_number*-4ca", "0?sub-fmsetcid,s,1()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:17] Set("DAHDI/i1/*phone_number*-4ca", "RecordMethod=Group") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:18] Gosub("DAHDI/i1/*phone_number*-4ca", "sub-record-check,s,1(exten,105,)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@sub-record-check:1] GotoIf("DAHDI/i1/*phone_number*-4ca", "10?initialized") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (sub-record-check,s,10)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@sub-record-check:10] NoOp("DAHDI/i1/*phone_number*-4ca", "Recordings initialized") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@sub-record-check:11] ExecIf("DAHDI/i1/*phone_number*-4ca", "1?Set(ARG3=dontcare)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@sub-record-check:12] Set("DAHDI/i1/*phone_number*-4ca", "REC_POLICY_MODE_SAVE=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@sub-record-check:13] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Set(REC_STATUS=NO)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@sub-record-check:14] GotoIf("DAHDI/i1/*phone_number*-4ca", "5?checkaction") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (sub-record-check,s,17)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@sub-record-check:17] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?sub-record-check,exten,1") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (sub-record-check,exten,1)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [exten@sub-record-check:1] NoOp("DAHDI/i1/*phone_number*-4ca", "Exten Recording Check between phone_number and 105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [exten@sub-record-check:2] Set("DAHDI/i1/*phone_number*-4ca", "CALLTYPE=external") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [exten@sub-record-check:3] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Set(CALLTYPE=)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [exten@sub-record-check:4] Set("DAHDI/i1/*phone_number*-4ca", "CALLEE=dontcare") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [exten@sub-record-check:5] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Set(CALLEE=dontcare)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [exten@sub-record-check:6] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?callee") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (sub-record-check,exten,11)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [exten@sub-record-check:11] Gosub("DAHDI/i1/*phone_number*-4ca", "recordcheck,1(dontcare,external,105)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [recordcheck@sub-record-check:1] NoOp("DAHDI/i1/*phone_number*-4ca", "Starting recording check against dontcare") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [recordcheck@sub-record-check:2] Goto("DAHDI/i1/*phone_number*-4ca", "dontcare") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (sub-record-check,recordcheck,3)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [recordcheck@sub-record-check:3] Return("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [exten@sub-record-check:12] Return("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:19] GotoIf("DAHDI/i1/*phone_number*-4ca", "0 ?skipsimple") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [105@from-did-direct:20] Macro("DAHDI/i1/*phone_number*-4ca", "simple-dial,105,19") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-simple-dial:1] Set("DAHDI/i1/*phone_number*-4ca", "__EXTTOCALL=105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-simple-dial:2] Set("DAHDI/i1/*phone_number*-4ca", "RT=19") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-simple-dial:3] Set("DAHDI/i1/*phone_number*-4ca", "CFUEXT=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-simple-dial:4] Set("DAHDI/i1/*phone_number*-4ca", "CFBEXT=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-simple-dial:5] Set("DAHDI/i1/*phone_number*-4ca", "CWI_TMP=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-simple-dial:6] Macro("DAHDI/i1/*phone_number*-4ca", "dial-one,19,TtrI,105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:1] Set("DAHDI/i1/*phone_number*-4ca", "DEXTEN=105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:2] Set("DAHDI/i1/*phone_number*-4ca", "DIALSTATUS_CW=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:3] GosubIf("DAHDI/i1/*phone_number*-4ca", "0?screen,1()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:4] GosubIf("DAHDI/i1/*phone_number*-4ca", "0?cf,1()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:5] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?skip1") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (macro-dial-one,s,8)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:8] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?nodial") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:9] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?continue") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:10] Set("DAHDI/i1/*phone_number*-4ca", "EXTHASCW=ENABLED") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:11] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?next1:cwinusebusy") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (macro-dial-one,s,23)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:23] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?next3:continue") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (macro-dial-one,s,24)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:24] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:25] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?nodial") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:26] GosubIf("DAHDI/i1/*phone_number*-4ca", "1?dstring,1():dlocal,1()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:1] Set("DAHDI/i1/*phone_number*-4ca", "DSTRING=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:2] Set("DAHDI/i1/*phone_number*-4ca", "DEVICES=105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:3] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Return()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:4] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Set(DEVICES=05)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:5] Set("DAHDI/i1/*phone_number*-4ca", "LOOPCNT=1") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:6] Set("DAHDI/i1/*phone_number*-4ca", "ITER=1") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:7] Set("DAHDI/i1/*phone_number*-4ca", "THISDIAL=SIP/105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:8] GosubIf("DAHDI/i1/*phone_number*-4ca", "1?zap2dahdi,1()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Return()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:2] Set("DAHDI/i1/*phone_number*-4ca", "NEWDIAL=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:3] Set("DAHDI/i1/*phone_number*-4ca", "LOOPCNT2=1") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:4] Set("DAHDI/i1/*phone_number*-4ca", "ITER2=1") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:5] Set("DAHDI/i1/*phone_number*-4ca", "THISPART2=SIP/105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Set(THISPART2=DAHDI/105)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:7] Set("DAHDI/i1/*phone_number*-4ca", "NEWDIAL=SIP/105&") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:8] Set("DAHDI/i1/*phone_number*-4ca", "ITER2=2") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?begin2") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:10] Set("DAHDI/i1/*phone_number*-4ca", "THISDIAL=SIP/105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [zap2dahdi@macro-dial-one:11] Return("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:9] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?doset") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (macro-dial-one,dstring,13)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:13] Set("DAHDI/i1/*phone_number*-4ca", "DSTRING=SIP/105&") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:14] Set("DAHDI/i1/*phone_number*-4ca", "ITER=2") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:15] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?begin") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:16] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Return()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:17] Set("DAHDI/i1/*phone_number*-4ca", "DSTRING=SIP/105") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [dstring@macro-dial-one:18] Return("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:27] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?nodial") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:28] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?skiptrace") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:29] GosubIf("DAHDI/i1/*phone_number*-4ca", "1?ctset,1():ctclear,1()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [ctset@macro-dial-one:1] Set("DAHDI/i1/*phone_number*-4ca", "DB(CALLTRACE/105)=phone_number") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [ctset@macro-dial-one:2] Return("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:30] Set("DAHDI/i1/*phone_number*-4ca", "D_OPTIONS=TtrIM(auto-blkvm)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:31] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?SIPAddHeader(Alert-Info: )") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:32] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?SIPAddHeader()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:33] ExecIf("DAHDI/i1/*phone_number*-4ca", "1?Set(CHANNEL(musicclass)=Movie-Themes)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:34] GosubIf("DAHDI/i1/*phone_number*-4ca", "0?qwait,1()") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:35] Set("DAHDI/i1/*phone_number*-4ca", "__CWIGNORE=") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:36] Set("DAHDI/i1/*phone_number*-4ca", "__KEEPCID=TRUE") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:37] GotoIf("DAHDI/i1/*phone_number*-4ca", "0?usegoto,1") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:38] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?godial") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (macro-dial-one,s,43)
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:43] Macro("DAHDI/i1/*phone_number*-4ca", "dialout-one-predial-hook,") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:44] ExecIf("DAHDI/i1/*phone_number*-4ca", "1?Set(D_OPTIONS=trIM(auto-blkvm)I)") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-dial-one:45] Dial("DAHDI/i1/*phone_number*-4ca", "SIP/105,19,trIM(auto-blkvm)I") in new stack
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] netsock2.c: == Using SIP RTP TOS bits 184
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] netsock2.c: == Using SIP RTP CoS mark 5
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] app_dial.c: -- Called SIP/105
[2016-06-10 14:44:12] VERBOSE[2546][C-0000ce74] app_dial.c: -- Connected line update to DAHDI/i1/*phone_number*-4ca prevented.
[2016-06-10 14:44:13] VERBOSE[2546][C-0000ce74] app_dial.c: -- SIP/105-000007d4 is ringing
[2016-06-10 14:44:21] VERBOSE[5918][C-0000ce74] sig_pri.c: -- Span 1: Channel 0/1 got hangup request, cause 16
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] app_macro.c: == Spawn extension (macro-dial-one, s, 45) exited non-zero on 'DAHDI/i1/*phone_number*-4ca' in macro 'dial-one'
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] app_macro.c: == Spawn extension (macro-simple-dial, s, 6) exited non-zero on 'DAHDI/i1/*phone_number*-4ca' in macro 'simple-dial'
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] pbx.c: == Spawn extension (from-did-direct, 105, 20) exited non-zero on 'DAHDI/i1/*phone_number*-4ca'
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [h@from-did-direct:1] Macro("DAHDI/i1/*phone_number*-4ca", "hangupcall,") in new stack
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-hangupcall:1] ExecIf("DAHDI/i1/*phone_number*-4ca", "0?Set(CDR(recordingfile)=.wav)") in new stack
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-hangupcall:2] GotoIf("DAHDI/i1/*phone_number*-4ca", "1?theend") in new stack
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] pbx.c: -- Goto (macro-hangupcall,s,4)
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("DAHDI/i1/*phone_number*-4ca", "") in new stack
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'DAHDI/i1/*phone_number*-4ca' in macro 'hangupcall'
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] pbx.c: == Spawn extension (from-did-direct, h, 1) exited non-zero on 'DAHDI/i1/*phone_number*-4ca'
[2016-06-10 14:44:21] VERBOSE[2546][C-0000ce74] chan_dahdi.c: -- Hungup 'DAHDI/i1/*phone_number*-4ca'
tail -f /var/log/asterisk/full

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@jfinstrom wrote:

There has been much discussion over the FCC robocall list and why it’s the greatest thing since canned bread or why drinking bleach would be a better choice. Current implementations all leave something to be desired.
So currently folks are downloading and parsing a massive csv into astdb to take advantage of the blacklist.

Here are my issues with this implementation:

• This puts over 97,000 entries in your astdb. To begin with this is madness. There is no reason you should push that much of any kind of data into astdb.

• Every week you have to insert a new list, which takes about 5-10 minutes. This is likely not an issue for most as they likely use a cron job.

• This process as discovered by one user can render GUI based management difficult if not impossible.

• The numbers on this list have had a complaint by someone. The complaints do not have to be valid. This list is not vetted and filtered. Anyone could potentially report anyone and suddenly you are not getting calls from them. One company on the list is Fedex. Using the import method, fedex can’t call you. It seems like a terrible idea to block almost a million callers without a second thought.

• Data normalization. Caller ID is not one size fits all. Even between your trunks you may get 2 or 3 caller ID formats. One carrier may send +14805551212 another 4805551212, and finally 14805551212. The way blacklisting works is on an exact match. You would have to put 3 entries for each of the 97K numbers to satisfy a proper block.

These reasons are all why I have resisted any promotion of using this data. There has to be a better, perhaps sane way to use this data without falling into these pits. My knee jerk reaction was to put up an API. I could keep a sync’d copy of this list and superfecta could make calls to the web service. This clears up 1, 2, 3 and 5. We still fall into pit number 4. Well 3 out of 4 ain’t bad.

So I went to the FCC site and noticed a button. They have an API. I clicked on the API and discovered I can pass a phone number to them and get all the complaints back for a given number. This was kind of awesome!

So I put together a Superfecta source that I think solves 1, 2, 3, 5 and some of 4.

The source normalizes the number, calls the FCC API and counts the complaints. The more complaints, the more likely the number is a bad guy. By default the threshold is 2. For reference, Fedex mentioned above is at 2, so for their call to not be spam you would have to bump it to 3. Now you can use the data on any trunk without need for multiple formats. You don’t have to pollute your blacklist with 970K entries. No need to sync the data or wait a week for an update as the information is pulled in real time.

In the edge version of Superfecta, there is a new source called FCC Complaints. Give it a look and reclaim your blacklist!

If you are using a framework version of 13.0.96+ you can get Edge modules as follows.

To enable the edge track, go to “Advanced settings and set
“Set Module Admin to Edge mode” to “Yes”
Then go to module admin and click "Check Online". Note this will show updates
for ALL modules in the edge track. Update the blacklist module.
Once finished go to “Advanced settings and set “Set Module Admin to Edge mode” to “No”
You may also upgrade from the command line with "fwconsole ma --edge upgrade blacklist”

For more information on what we are doing please follow @Sangoma and @FreePBX on twitter.

If you find bugs please file a bug report at http://issues.freepbx.org
FreePBX documentation on our wiki http://wiki.freepbx.org
Professional support services at http://ussupport.sangoma.com

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@bksales wrote:

There are two offices, both connecting to the same hosted FreePBX virtual instance. Once a month or so a couple of the phones will go offline and when we look at it the phone appears to have been reset to factory defaults. There are 12 phones, 6 in each location. There does not seem to be a pattern as to which ones are getting reset and when. Both locations are using Comcast who has been having lots of issues in that area. They also occasionally have power issues too.

Any ideas what could be causing this?

Some background info
64 bit FreePBX system. It's been happening since the 6 track, so I don't think the specific version is relevant.
Aastra 57i, 6731i, and 6739i (they've all experienced this issue at least once at different times)
Comcast internet
Routers are freeware. Entangle.

Right now my main suspect is the Entangle router, which does not have Option 66 set in it FYI.

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@TugBoat wrote:

Following installation of the FreepBX 13.0 distro I am getting some emails delivered regularly to the asterisk user. I don't exactly know if these are 'normal' or not, so I thought I should report them anyway.

1: Every 5 minutes the following email:

PHP Fatal error:  Class 'Whoops\Util\Misc' not found in /var/www/html/admin/modules/qxact_reports/functions.inc/import_queue_data.php on line 0
Whoops\Exception\ErrorException: Class 'Whoops\Util\Misc' not found in file /var/www/html/admin/modules/qxact_reports/functions.inc/import_queue_data.php on line 0
Stack trace:
  1. () /var/www/html/admin/modules/qxact_reports/functions.inc/import_queue_data.php:0

1. Not sure how often this one is occuring, I think once a day:

   Exception: No deployment name, this is unpossible in file /var/www/html/admin/modules/sysadmin/Sysadmin.class.php on line 1270
   Stack trace:
     1. Exception->() /var/www/html/admin/modules/sysadmin/Sysadmin.class.php:1270
     2. FreePBX\modules\Sysadmin->sendDDNSUpdate() /var/lib/asterisk/agi-bin/ddns_client.php:11

I hope this helps someone.
Tim

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@TugBoat wrote:

Firstly, I should say that this is not a complaint - it is more that I would like to understand something that appears unusual to me.

I am running FreePBX (13.0.124 from distro) in a virtualised ESXi 6.0 environment on a system using SSD. (Yes, I am aware of all the caveats etc about virtualised/SSD etc.) As part of most installations in this environment I do the normal modifications to reduce the write rate based on the underlying SSD storage. This includes:
- moving /tmp to RAM
- moving /var/log to RAM (with save/restore on shutdown/start etc.)
- update the mount options to include: noatime,data=writeback,commit=60
- run without swap

On the whole this normally works well and prior to running FreePBX in this environment the daily write rate was about 0.07GB for two VMs (a FreeBSD router, and a Linux server).

When I install FreePBX in this environment my daily write rate goes up be a factor of 10 to 0.70 GB. Now this is probably acceptable, however, I am wondering exactly what all this write traffic is?

Even on a physical disk based system the disks would be active almost 100% of the time. The ESXi performance information shows continues write rates of 15 KByte/Sec (avg) for the FreePBX VM.

My installation has 2 x VOIP Phone, 1 x ATA (FXO+FXS) Trunk, 1 x IAX2 Trunk, 1 x SIP Trunk to external provider (60sec registration rate).

Another interesting thing is that if I run a copy my FreePBX system connected to an 'empty' network (ie. disconnected from the world) then the write rate drops a little, but not that much.

So what I am trying to understand is where exactly this write traffic is occurring and what it is related to?

I should add that altering the commit rate on the ext4 file systems doesn't appear to have much effect on the write traffic.

Thanks,
Tim

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