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Queue Recordings avoid delete

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@ncaridi wrote:

Dear Experts,

I would like to have our customer service queue automatically recorded but visible only to the manager of the CS.
right now each user can see the queue recordings and even delete them.
I would like the users to be able to delete only recordings which has been made on demand.
Is there a way to achieve this ?

Thank you !

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Forward Caller ID

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@Kawon wrote:

Hi there,
I'm having trouble to forward the caller ID from external . When closed, the call is going to an IVR which , when pressing one(1) forwards you to my cellphone. All I want is to see the caller ID from the caller and not my company number, can you help me out here ? I asked my provider to turn that option on, so now I have to configure it on Freepbx and I don`t know where to start.
Regards Harry

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DTLS enable=Yes with CSV file?

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@the_passerby wrote:

Hi to all,
I am new in freepbx. I am using freepbx 12.0.30 and i want to add some extensions via CSV file. But i want to add DTLS_Enable=Yes with CVS file. I can add DTLS enable=Yes with GUI but i could not add DTLS_Enable=Yes with CSV file. May i need to change some config files in freepbx?

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Call Parking Questions?

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@kwriley87 wrote:

I am using FreePBX Distro 12.0.76.2 with Grandstream GXP 2130v2 IP Phones.

My parking lot extension is 70 and I have 8 slots (71-78).

When I transfer a call to the parking lot by transferring to extension 70, the call transfers to the parking lot, but it does not announce to the transferring extension which parking slot it sent the call to so that they can tell the person which parking slot to dial to pick the call up.

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FreePBX 13 Wake up Calls

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@mcisar wrote:

Is there any chance that in the near future the wakeup calls module in FreePBX 13 will see any of the new functionalities that were present in the 2.11.4 beta 8 version (particularly recurring calls). Also I seem to recall there being a suggestion/request somewhere at one point to add an option to specify the alert-info to allow walkeup calls to use an alternate ring cadence and/or offhook automatically.

Cheers,
Mike

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How to enable DTMF tone?

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@dirweb wrote:

Hi
i have a problem because i can not press any number if the ivr on other line is active.
For example when the voice tell me: press 1 for, press 2 for, press 3 for, etc... if i press don't go.

What i have to set up on my freepbx?

If i add a new extension on "dtmf signaling" in my freepbx i have only 3 choice:
- rfc4733
- sip info (application dtmf/relay)
- in band audio (not raccomanded)

On x-lite or on my ip phone i saw i have RCF2833, maybe this is the problem?
How i can make for use DTMF?
Thank you

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Connecting two offices

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@ckattimeris wrote:

Hello,

i have 7 offices that i need to connect to my MAIN PBX for internal communication

my main PBX is FreePBX HA.
when i try to connect to my main asterisk server i am unable to connect to the remove offices as i am using the Virtual IP to connect to my main pbx from the countries. I ended configuring OPENVPN between countries but that is causing extreme delays on the communication.

I've tried IAX and SIP

any suggestion?

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Set Caller ID WITHOUT Changing Destination

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@SkydiverFL wrote:

Is there a way to set the Caller ID for an incoming number WITHOUT changing the destination? I have a client with over 1,000 cell phones in the field. They want the incoming displays to show the name of the employee to whom the cell phone is assigned. They DO NOT want to alter where the incoming call is directed to. That logic should be based off of the incoming route. The current "Set Caller ID" page forces a destination to be set which does not seem logical. Instead of simply setting the Caller ID, it is manipulating the routing of the incoming call.

What am I missing here?

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Network Time Server

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@bk6662 wrote:

Hello,

I'm a new FreePBX user so please forgive what is probably an elementary question. I've managed to get the system working with (so far) 1 Polycom IP phone, and a CounterPath X-Lite softphone. I can make and received external calls over a Digium SIP trunk. So far so good!

The question I have (one of many - but I'll start here!) :smile: - Is FreePBX able to act as an NTP Server? Or would I need to set the Host system as NTP server? Trying to find out the best place to push correct time & date out to my phones.

Thank you!
Brian

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Trying to create a "loopback trunk" but can't find "custom dial string"

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@prstech wrote:

I am trying to create a loopback trunk so that I can route calls to my DID's back into my PBX.

In this forum people give directions to add a custom dial string of "local/${OUTNUM}@from-trunk"

I can't seem to find the "custom dial string" field on the trunk setup screen.

I am using FreePBX verion 6.12.65-28. Did this field get renamed in this version? If so can someone tell me what the new field is called?

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Skype Connect

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@john_itworx wrote:

Hello,
I am having a problem a problem regarding Skype registration on FreePBX. The registration is fine on manager.skype.com but not on FreePBX. The string is as follows:
9905100027XXXX:password@sip.skype.com/9905100027XXXX

And this is what i get from the logs:
[2015-10-05 08:41:32] NOTICE[1730] chan_sip.c: -- Registration for '9905100027XXXX@sip.skype.com' timed out, trying again (Attempt #9)
[2015-10-05 08:41:52] NOTICE[1730] chan_sip.c: -- Registration for '9905100027XXXX@sip.skype.com' timed out, trying again (Attempt #10)
[2015-10-05 08:42:12] NOTICE[1730] chan_sip.c: -- Registration for '9905100027XXXX@sip.skype.com' timed out, trying again (Attempt #11)
[2015-10-05 08:42:32] NOTICE[1730] chan_sip.c: -- Registration for '9905100027XXXX@sip.skype.com' timed out, trying again (Attempt #12)

Please assist. Thanks in advance.

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PSTN One Way Audio through Pick (*8)

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@mohammed_kakuji wrote:

A little history about our system:

We had a major issue where every call timed out at around 30 seconds and most of the times resulted in one way audio.

Implementing a STUN server has fixed this issue..or has it?

When an external call comes in and the person who is calling inputs an extension and that extension picks up, the call is fine.

But when an external call comes in and they input an extension but another extension dials *8 to pick that call up, it results in one way audio. Is there an issue with the set-up of the dahdi card?

Hope this makes sense and let me know if you need any more details

Call log where it resulted in one way audio:

Deleted top part of log due to word count limit

[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@from-pstn:10] Set("PJSIP/Sipgate (01785 ******)-00000935", "CALLERPRES()=allowed_not_screened") in new stack
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@from-pstn:11] Goto("PJSIP/Sipgate (01785 ******)-00000935", "ivr-5,s,1") in new stack
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Goto (ivr-5,s,1)
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:1] Set("PJSIP/Sipgate (01785 ******)-00000935", "TIMEOUT_LOOPCOUNT=0") in new stack
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "INVALID_LOOPCOUNT=0") in new stack
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:3] Set("PJSIP/Sipgate (01785 ******)-00000935", "IVRCONTEXT_ivr-5=") in new stack
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:4] Set("PJSIP/Sipgate (01785 ******)-00000935", "IVRCONTEXT=ivr-5") in new stack
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:5] Set("PJSIP/Sipgate (01785 ******)-00000935", "_IVRRETVM=") in new stack
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:6] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?skip") in new stack
[2015-10-05 16:07:34] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:7] Answer("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:07:35] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:8] Wait("PJSIP/Sipgate (01785 ******)-00000935", "1") in new stack
[2015-10-05 16:07:35] WARNING[20067] .
[2015-10-05 16:07:36] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:9] Set("PJSIP/Sipgate (01785 ******)-00000935", "IVR_MSG=custom/TCB") in new stack
[2015-10-05 16:07:36] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:10] Set("PJSIP/Sipgate (01785 ******)-00000935", "TIMEOUT(digit)=3") in new stack
[2015-10-05 16:07:36] VERBOSE[14388][C-0000049d] func_timeout.c: Digit timeout set to 3.000
[2015-10-05 16:07:36] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@ivr-5:11] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "1?Background(custom/TCB)") in new stack
[2015-10-05 16:07:36] VERBOSE[14388][C-0000049d] file.c: Playing 'custom/TCB.slin' (language 'en')
[2015-10-05 16:07:41] WARNING[17461] .
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [243@ivr-5:1] Macro("PJSIP/Sipgate (01785 ******)-00000935", "blkvm-clr,") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-blkvm-clr:1] Set("PJSIP/Sipgate (01785 ******)-00000935", "SHARED(BLKVM,)=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-blkvm-clr:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "GOSUB_RETVAL=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-blkvm-clr:3] MacroExit("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [243@ivr-5:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "__NODEST=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [243@ivr-5:3] Goto("PJSIP/Sipgate (01785 ******)-00000935", "from-did-direct,243,1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (from-did-direct,243,1)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [243@from-did-direct:1] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?ext-local,243,1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (ext-local,243,1)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [243@ext-local:1] Set("PJSIP/Sipgate (01785 ******)-00000935", "__RINGTIMER=15") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [243@ext-local:2] Macro("PJSIP/Sipgate (01785 ******)-00000935", "exten-vm,novm,243,0,0,0") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:1] Macro("PJSIP/Sipgate (01785 ******)-00000935", "user-callerid,") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:1] Set("PJSIP/Sipgate (01785 ******)-00000935", "TOUCH_MONITOR=1444057654.17870") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "AMPUSER=07983497629") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?report") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "1?Set(REALCALLERIDNUM=07983497629)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:5] Set("PJSIP/Sipgate (01785 ******)-00000935", "AMPUSER=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?limit") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:7] Set("PJSIP/Sipgate (01785 ******)-00000935", "AMPUSERCIDNAME=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?report") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (macro-user-callerid,s,16)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:16] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?continue") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:17] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "1?Set(_CALLEEACCOUNCODE=)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:18] Set("PJSIP/Sipgate (01785 ******)-00000935", "__TTL=64") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:19] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?continue") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (macro-user-callerid,s,30)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:30] Set("PJSIP/Sipgate (01785 ******)-00000935", "CALLERID(number)=07983497629") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:31] Set("PJSIP/Sipgate (01785 ******)-00000935", "CALLERID(name)=07983497629") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:32] Set("PJSIP/Sipgate (01785 ******)-00000935", "CDR(cnum)=07983497629") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:33] Set("PJSIP/Sipgate (01785 ******)-00000935", "CDR(cnam)=07983497629") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-user-callerid:34] Set("PJSIP/Sipgate (01785 ******)-00000935", "CHANNEL(language)=en") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "RingGroupMethod=none") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:3] Set("PJSIP/Sipgate (01785 ******)-00000935", "__EXTTOCALL=243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:4] Set("PJSIP/Sipgate (01785 ******)-00000935", "__PICKUPMARK=243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:5] Set("PJSIP/Sipgate (01785 ******)-00000935", "RT=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:6] Gosub("PJSIP/Sipgate (01785 ******)-00000935", "sub-record-check,s,1(exten,243,dontcare)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@sub-record-check:1] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "11?initialized") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (sub-record-check,s,10)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@sub-record-check:10] NoOp("PJSIP/Sipgate (01785 ******)-00000935", "Recordings initialized") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@sub-record-check:11] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Set(ARG3=dontcare)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@sub-record-check:12] Set("PJSIP/Sipgate (01785 ******)-00000935", "REC_POLICY_MODE_SAVE=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@sub-record-check:13] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Set(REC_STATUS=NO)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@sub-record-check:14] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "5?checkaction") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (sub-record-check,s,17)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@sub-record-check:17] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?sub-record-check,exten,1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (sub-record-check,exten,1)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [exten@sub-record-check:1] NoOp("PJSIP/Sipgate (01785 ******)-00000935", "Exten Recording Check between 07983497629 and 243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [exten@sub-record-check:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "CALLTYPE=external") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [exten@sub-record-check:3] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Set(CALLTYPE=)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [exten@sub-record-check:4] Set("PJSIP/Sipgate (01785 ******)-00000935", "CALLEE=dontcare") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [exten@sub-record-check:5] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Set(CALLEE=dontcare)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [exten@sub-record-check:6] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?callee") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (sub-record-check,exten,11)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [exten@sub-record-check:11] Gosub("PJSIP/Sipgate (01785 ******)-00000935", "recordcheck,1(dontcare,external,243)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/Sipgate (01785 ******)-00000935", "Starting recording check against dontcare") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("PJSIP/Sipgate (01785 ******)-00000935", "dontcare") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (sub-record-check,recordcheck,3)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [recordcheck@sub-record-check:3] Return("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [exten@sub-record-check:12] Return("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:7] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?macrodial") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (macro-exten-vm,s,13)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:13] GosubIf("PJSIP/Sipgate (01785 ******)-00000935", "0?clrheader,1()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-exten-vm:14] Macro("PJSIP/Sipgate (01785 ******)-00000935", "dial-one,,Ttr,243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:1] Set("PJSIP/Sipgate (01785 ******)-00000935", "DEXTEN=243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "DIALSTATUS_CW=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:3] GosubIf("PJSIP/Sipgate (01785 ******)-00000935", "0?screen,1()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:4] GosubIf("PJSIP/Sipgate (01785 ******)-00000935", "0?cf,1()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:5] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?skip1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (macro-dial-one,s,8)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:8] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?nodial") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:9] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?continue") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:10] Set("PJSIP/Sipgate (01785 ******)-00000935", "EXTHASCW=ENABLED") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:11] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?next1:cwinusebusy") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (macro-dial-one,s,23)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:23] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?next3:continue") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (macro-dial-one,s,24)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:24] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:25] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?nodial") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:26] GosubIf("PJSIP/Sipgate (01785 ******)-00000935", "1?dstring,1():dlocal,1()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:1] Set("PJSIP/Sipgate (01785 ******)-00000935", "DSTRING=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "DEVICES=243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:3] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Return()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:4] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Set(DEVICES=43)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:5] Set("PJSIP/Sipgate (01785 ******)-00000935", "LOOPCNT=1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:6] Set("PJSIP/Sipgate (01785 ******)-00000935", "ITER=1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:7] Set("PJSIP/Sipgate (01785 ******)-00000935", "THISDIAL=PJSIP/243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:8] GosubIf("PJSIP/Sipgate (01785 ******)-00000935", "1?zap2dahdi,1()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:1] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Return()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:2] Set("PJSIP/Sipgate (01785 ******)-00000935", "NEWDIAL=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:3] Set("PJSIP/Sipgate (01785 ******)-00000935", "LOOPCNT2=1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:4] Set("PJSIP/Sipgate (01785 ******)-00000935", "ITER2=1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:5] Set("PJSIP/Sipgate (01785 ******)-00000935", "THISPART2=PJSIP/243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:6] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Set(THISPART2=DAHDIIP/243)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:7] Set("PJSIP/Sipgate (01785 ******)-00000935", "NEWDIAL=PJSIP/243&") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:8] Set("PJSIP/Sipgate (01785 ******)-00000935", "ITER2=2") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:9] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?begin2") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:10] Set("PJSIP/Sipgate (01785 ******)-00000935", "THISDIAL=PJSIP/243") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [zap2dahdi@macro-dial-one:11] Return("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:9] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?doset") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:10] NoOp("PJSIP/Sipgate (01785 ******)-00000935", "Debug: Found PJSIP Destination PJSIP/243, updating with PJSIP_DIAL_CONTACTS") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:11] Set("PJSIP/Sipgate (01785 ******)-00000935", "THISDIAL=PJSIP/243/sip:243@192.168.148.31:5062") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:12] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?skipset") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:13] Set("PJSIP/Sipgate (01785 ******)-00000935", "DSTRING=PJSIP/243/sip:243@192.168.148.31:5062&") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:14] Set("PJSIP/Sipgate (01785 ******)-00000935", "ITER=2") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:15] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?begin") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:16] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?Return()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:17] Set("PJSIP/Sipgate (01785 ******)-00000935", "DSTRING=PJSIP/243/sip:243@192.168.148.31:5062") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [dstring@macro-dial-one:18] Return("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:27] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?nodial") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:28] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?skiptrace") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:29] GosubIf("PJSIP/Sipgate (01785 ******)-00000935", "1?ctset,1():ctclear,1()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [ctset@macro-dial-one:1] Set("PJSIP/Sipgate (01785 ******)-00000935", "DB(CALLTRACE/243)=07983497629") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [ctset@macro-dial-one:2] Return("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:30] Set("PJSIP/Sipgate (01785 ******)-00000935", "D_OPTIONS=Ttr") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:31] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?SIPAddHeader(Alert-Info: )") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:32] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "0?SIPAddHeader()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:33] ExecIf("PJSIP/Sipgate (01785 ******)-00000935", "1?Set(CHANNEL(musicclass)=default)") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:34] GosubIf("PJSIP/Sipgate (01785 ******)-00000935", "0?qwait,1()") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:35] Set("PJSIP/Sipgate (01785 ******)-00000935", "__CWIGNORE=") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:36] Set("PJSIP/Sipgate (01785 ******)-00000935", "__KEEPCID=TRUE") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:37] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "0?usegoto,1") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:38] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?godial") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Goto (macro-dial-one,s,43)
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:43] Macro("PJSIP/Sipgate (01785 ******)-00000935", "dialout-one-predial-hook,") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dialout-one-predial-hook:1] MacroExit("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-dial-one:44] Dial("PJSIP/Sipgate (01785 ******)-00000935", "PJSIP/243/sip:243@192.168.148.31:5062,,Ttr") in new stack
[2015-10-05 16:07:44] VERBOSE[14388][C-0000049d] app_dial.c: Called PJSIP/243/sip:243@192.168.148.31:5062
[2015-10-05 16:07:45] VERBOSE[14388][C-0000049d] app_dial.c: PJSIP/243-00000936 is ringing
[2015-10-05 16:07:49] WARNING[17461] .
[2015-10-05 16:07:50] NOTICE[17461] pickup.c: Pickup PJSIP/243-00000936 attempt by PJSIP/242-00000937
[2015-10-05 16:07:50] VERBOSE[14388][C-0000049d] app_dial.c: PJSIP/242-00000937 answered PJSIP/Sipgate (01785 ******)-00000935
[2015-10-05 16:07:50] VERBOSE[14391][C-0000049d] bridge_channel.c: Channel PJSIP/242-00000937 joined 'simple_bridge' basic-bridge <06a98eb5-2262-4b0a-b92f-ed5b7cb706d7>
[2015-10-05 16:07:50] VERBOSE[14388][C-0000049d] bridge_channel.c: Channel PJSIP/Sipgate (01785 ******)-00000935 joined 'simple_bridge' basic-bridge <06a98eb5-2262-4b0a-b92f-ed5b7cb706d7>
[2015-10-05 16:07:55] WARNING[2226] .
[2015-10-05 16:07:56] WARNING[20067] .
[2015-10-05 16:08:02] WARNING[20067] .
[2015-10-05 16:08:10] WARNING[20067] .
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] bridge_channel.c: Channel PJSIP/Sipgate (01785 ******)-00000935 left 'simple_bridge' basic-bridge <06a98eb5-2262-4b0a-b92f-ed5b7cb706d7>
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] app_macro.c: Spawn extension (macro-dial-one, s, 44) exited non-zero on 'PJSIP/Sipgate (01785 ******)-00000935' in macro 'dial-one'
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] app_macro.c: Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'PJSIP/Sipgate (01785 ******)-00000935' in macro 'exten-vm'
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] pbx.c: Spawn extension (ext-local, 243, 2) exited non-zero on 'PJSIP/Sipgate (01785 ******)-00000935'
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] pbx.c: Executing [h@ext-local:1] Macro("PJSIP/Sipgate (01785 ******)-00000935", "hangupcall,") in new stack
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/Sipgate (01785 ******)-00000935", "1?theend") in new stack
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] pbx.c: Goto (macro-hangupcall,s,3)
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] pbx.c: Executing [s@macro-hangupcall:3] Hangup("PJSIP/Sipgate (01785 ******)-00000935", "") in new stack
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] app_macro.c: Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'PJSIP/Sipgate (01785 ******)-00000935' in macro 'hangupcall'
[2015-10-05 16:08:11] VERBOSE[14388][C-0000049d] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/Sipgate (01785 ******)-00000935'
[2015-10-05 16:08:11] VERBOSE[14391][C-0000049d] bridge_channel.c: Channel PJSIP/242-00000937 left 'simple_bridge' basic-bridge <06a98eb5-2262-4b0a-b92f-ed5b7cb706d7>

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Some RSS feeds do not load

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@Marbled wrote:

Hi!

I have been trying to add Slashdot's RSS feeds to my dashboard and it doesn't want to load, I only see the RSS feed title and nothing else.

According to the newly added hint about this the feed should be supported (I actually tried all three, the RSS 0.9, the RSS 1.0 and the Atom one) but it doesn't want to load.

Is there any way to figure out what's wrong?

I have checked into /var/log/asterisk/freepbx.log and /var/log/asterisk/full and I have not found anything interesting...

Thank you and have a nice day!

Nick

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How to connect freepbx-13/PJSIP+Sangoma Vega

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@james wrote:

Hello:
I tested with Sangoma Vega gateway and FreePBX-13 chan_pjsip. There are some steps as below:
1) Install FreePBX-13 distro version by the guild and using
chan_pjsip instead of using chan_SIP

2) Config the trunk, outbound and inbound from FreePBX-13



3) Config Sangoma Vega to support Tone( you have to upload the tone for your contry) and callerid (in china ,fsk) and SIP trunk account:



4) make test calls:

make sure all connections are Avail!
incoming calls

call records for outbound and inconing calls.


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Move file from freepbx to windows

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@xsirsys wrote:

Hello everyone,
I have FreePBX 2.11.0.38 with Asterisk Ver. 11.14.1

I'm trying to move or copy some of the recordings from /var/spool/asterisk/monitor to a windows system. Please help!! Thank you.

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Whishlist, fail2ban-function

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@sm0try wrote:

I want to see : the exension number (if any) that has been banned in the "fail2ban email", and not only the IP-adress.

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Voicemail to Email with justhost

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@raz23 wrote:

Hello everyone,

I've just started to use FreePbx recently and I was trying to get Voicemail to email to work with our justhost account. I'm receiving a 550 error which states: "status=bounced (host just58.justhost.com[173.254.28.58] said: 550 Unrouteable address (in reply to RCPT TO command))". I've tested the voicemail to email feature with my Google account and Yahoo account and was able to get it to work. Am I missing step somewhere or do I need to configure something to be able to receive my voicemail in my justhost account?

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DTMF issues 5% of the time calling from Cisco SPA to 3rd party conferencing providers on SIP trunk

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@alavelle wrote:

Trying to troubleshoot a DTMF issues where a small percentage of the time when people are calling to conferencing services like GoToMeeting (but not just GoToMeeting) it doesn't properly recognize the digits they type in.

FBPX 2.11.0.43
Phones: Cisco SPA504G running firmware 7.5.6a
Extension: DTMF on the extension is set to RFC 2833
SIP trunk: set to dtmf=auto on the PEER side

It happens infrequently so unfortunately I don't have logged examples of when the issue occurs. To test I called a conference # 50 times but couldn't recreate the issue

Here is what is in the logs:
[2015-10-09 15:25:57] DTMF[25340][C-0001480a]: channel.c:4214 _astread: DTMF begin '8' received on SIP/2281-00016531
[2015-10-09 15:25:57] DTMF[25340][C-0001480a]: channel.c:4225 _astread: DTMF begin passthrough '8' on SIP/2281-00016531
[2015-10-09 15:25:57] DTMF[25340][C-0001480a]: channel.c:4128 _astread: DTMF end '8' received on SIP/2281-00016531, duration 80 ms
[2015-10-09 15:25:57] DTMF[25340][C-0001480a]: channel.c:4169 _astread: DTMF end accepted with begin '8' on SIP/2281-00016531
[2015-10-09 15:25:57] DTMF[25340][C-0001480a]: channel.c:4184 _astread: DTMF end '8' detected to have actual duration 68 on the wire, emulation will be triggered on SIP/2281-00016531
[2015-10-09 15:25:57] DTMF[25340][C-0001480a]: channel.c:4191 _astread: DTMF end '8' has duration 68 but want minimum 80, emulating on SIP/2281-00016531
[2015-10-09 15:25:57] DTMF[25340][C-0001480a]: channel.c:4292 _astread: DTMF end emulation of '8' queued on SIP/2281-00016531

If I switch the extension DTMF setting to auto I see the same thing. Not sure if I need to restart the phone, rebuilt template or anything else of the change to happen?

The Cisco 504G is setup with the following settings:
G711u
DTMF Process AVT: Yes
DTMF Tx Method: Auto
DTMF Tx Volume for AVT Packet: 0
DTMF AVT Packet Interval: 0
application/dtmf-relay

Any suggestions?

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Update from 11 to 13

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@claloano wrote:

I latest version FreePBX, I run then upgrade to 13
The procedure started but then came into the loop ...
Thanks to the forum I found as take away from the problem, but ....
Now the module "Call Recordings" does not work from the following error:
Call to undefined function recording_report_get_from ()
At line:
/var/www/html/admin/modules/recording_report/views/page.recording_report.php20

I tried and tried what I know, someone has idea?

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No Audio IVR Playback

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@Marudi wrote:

Quite simply seen a few similar issues in the community, unfortunately there has been no definite solution. Dialing from an extension (7777) all works fine, once I dial into the SIP trunk (from an external line as should be) there's no audio on the IVR, DTMF works so i can select the options - timing it manually - (but no Audio), Sound files checked and several used including the inbuilt GSM files as a test (still no audio) seen where it was suggested to modify (Answer() ) because of the ringing possibility, changed permissions on the folder for asterisk - From the debug and verbose the files show they are playing so the log wouldnt show any issues.. Asterisk 11 + FreePBX + Centos 6.

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