@susfreepbx wrote:

Just like with the cudatel box we had before it seems call numbers are schewed by using ques and so on. I have a polycom phone that shows 400 calls in a day went to it but that can’t be right. It has to be bouncing around a que or something.
The problem: We need to be able to know what is happening on the phone system, if our employees are answering the phone properly, and what exactly is happening to calls as they go through the system so that we can tweak the setting and make adjustments and document when employees aren’t answering the phone.
How are we set up: We use IVR’s and a single QUEUE. There are 4 or 5 static queue agents and a couple of dynamic agents. For failover it goes to ext 106 which is setup to not ring but go directly to voicemail in which all agents receive the voicemail through email and return the call. Pretty simple. We use polycom phones from our last cudatel system.

I need a reporting mechanism that will give me accurate numbers for:
Total Incoming Calls for the Day
How many of those were answered?
Which user/extension answered each one?
How many were not answered?
How long did it take before it was answered?
We use an automated attendant - How many of each selection was made that day?
And it would be good to be able to do a date range as well.
We would also like to easily be able to “follow” a call and see everything that happened to it as it came it and went through the system until it was answered, transferred to voicemail, or they hung up.

We don’t mind buying a module if it gives us what we need.

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@mannebk wrote:

Hi Folks,

I’m just curios how big company FreePBX admins (as the rent out admin guys at Sangoma for sure do) handle 5000 extensions 35 locations 250 trunks or what ever?

If I want to compare 2 extensions its an awfully work (clicking around in the GUI and waiting for the page to be loaded many times over) to do. I only run 40 extensions, 22 trunks and 4 locations on 1 server.

For sure they don’t export it by bulk handler, convert the “;” into “,” for excel to edit the CSV file and convert it back, then re import. I’m sure because this just asks for screwed up settings. Cant count on how many times the re import failed due to data row count mismatch due to excel fooling with the CSV format.

so how do you pros do administrate a FreePBX machine?

command line?
I have a feeling that’s the only way to comfortably do it as a professional PBX admin, administrating what ever asterisk based PBX.

unfortunately, that wont work if you don’t work that system 7hoursday / 5daysweek / 40 weeksyear as a PBX admin, but are just a normal guy that isn’t be able to remember all this command line syntax.

for my feeling, while the FreePBX is awesome, the GUI is a total miss in question for streamlined setup work and admin tasks.

that starts as short as the submit button in extensions reloads the extension list while in trunks and routes it goes back to the edited trunk or route and then you need to move the mouse to the side menu, open it, move the mouse again to click list, as the supplied side menu trunk list is not the same order as the main list. so you cant work your way down the list. and since I don’t remember which of my 12 trunks I have already changed this is very important to just work down the list.

also the gui does not remember how many entries it should show. it always drops back (in a new session) to the lowest number and I could not find any post or wiki or setting relating to this.

also, many options are good explained, there are some stuff, like the backup and restore module, that is not intuitively operated, as you have to set some settings at places where in no other part of the FreePBX gui there are settings to be found to set (like you have to use the side menu to switch to restore, servers and templates) in all other places you have it below the top menu like tabs. took me about 3 month’s to figure that one out, and since an other guy that has been asking here was f***ed over by some pros how he could ask such a stupid question, I kept quiet and just didn’t use the backup module until I lost my first server due to me accidentally delete the VM… That made me almost dump FreePBX but then, I, by accident, found the required settings for SSH backups.

Or the bulk handler, who only offers csv files, and only a few things like extensions, but no routes, or trunks could be exported.

what I as a user really would like is this:

first of all, a consistent layout. buttons behave the same, are at the same locations, settings and options are being reached the same way, and list have always the same (preferably system wide adjustable) order.

I also would like a table that shows horizontally all settings for e.g. extensions while it shows all extensions vertically. ideally I could change settings in this table, preferably on several extensions before the need of clicking “submit” that goes for every thing that’s not there once. like time conditions, routes, trunks, what ever. this would make administer the server so much easier.

I also would love, that if I work on several tabs, and I do change some settings (like for e.g. the name of a trunk in incoming sip settings) including clicking submit and the red “apply” button, that the next browser tab, in case it finds some “mismatch” like the same (but already changed) trunk name before informing me about this problem, rather looks up the database to discovers, the matching trunk has been renamed, an my input is perfectly valid, rather than make me reload the tab and enter every thing again.

I also would love a log show option in the gui, so that I would not need to open the cli via putty, enter the asterisk console and set it to debug. rather open the log view in gui and see the same output with drop down verbose level.

Also it would be great if the gui would be smart phone capable.

I also have a big problem with the language in FreePBX. so now, my system is set to de_de and my menu is partly translated. I would be glad to help with that translation, but, well, I could not find a way to do so. what irritates me much more is, that in English its sorted by name, in German, it once was, so new name and new location, that was so messy I got back to US_en. Back a few days I did built a new server, and now its not, its at the same location, but named in German, and I cant change that in the settings. its very irritating. how about you stop showing a partly translated gui until the column is fully translated. and offer a right or long click option to supply a better translation.

The language is also a mess if you want your voice prompts in German. there is some old de asterisk files out there, difficult to locate, I have them, so if some other German speaker needs them, send PM; but the current FreePBX version does use different prompts, so its 50/50 what prompt you get. so you start rerecording with different voice, well that sounds very professional to the caller and I cant order a professional voice to do the recordings, as I cant find a list of those recordings required by FreePBX. Since Sangoma offer them in English (only as far as I have seen, so I cant buy the voice prompts from Sangoma in German), I don’t see where the problem is, to somewhere in the wiki post all required voice prompts with filename and the English text for the current version as for sure you do have this list. so we not English user’s could develop a community based local language file.

so that’s it from my side, much longer than I intended to, but not jet with everything that’s annoying in the gui.

I just thought, the developer may want to know, what a user may think

Cheers
Manne

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@MacsOffice wrote:

I found out the hard way that activation uses the ETH ports config as part of the activation key process when it deactivated my machine when I re-ordered the ports. I think I have those assignments worked out but I need to know if the activation process uses IP assigned to those ports as well?
I bought a package and I want to install it but not if I need to use another activation.

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@mkaye wrote:

image.png1202x103 24.3 KB

trying to use my Exchange Calendar
it autodetects my info

mark

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@jcroy wrote:

So we moved our PBXACT 40 from one network to an entirely separate network. Before I could get to the GUI but now I cannot. I’m guessing its a simple problem, I’ve tried setting a static ip but I am messing up somewhere in that process it appears. Ideas?

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@matthewljensen wrote:

I’m not able to add a custom basefile edit for a grandstream GXP-2170. The “add entry” button is there, but when I click it, add the description, parameter, and value, and hit change basefile, nothing seems to have changed.

The way I understand it, is that after i go through that process, this new parameter should appear in red at the bottom. But not only does it not appear at the bottom, it doesn’t appear anywhere.

I’m thinking that this must be a bug, because I can’t see what I might be doing wrong.

Thanks for the help.

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@Nksahu wrote:

Dear Team,

Now-a-days we are facing strange and serious issue with DAHDI calling. We are using FreePBX with this card. We are getting message “All Circuits are busy now”. I am novice to handle this issue so please suggest me some steps to get rid off this issue.

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@Robbert wrote:

Hello all,

We are using freepbx with one incoming trunk and one ring group with a couple of phones.

When we are receiving a call and we are not answering the call in 20 seconds or when all extensions are busy or not available or on do not disturb the call is forwarded to a mobile phone.
This is by design and works fine.

But when we are receiving a call and all the phones are ringing and at the same time another call comes in we are unreachable for the second call and this call transfers to the mobile phone.
This is because Asterisk says all the extensions are busy because of ExtensionState: 8

Has anybody any idea how to prevent this behaviour ?

Below a piece of logfile.
This happends with every extension.
The phones we are using are Yealink phones

[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: Methodology of ring is ‘ringall’
[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: Added extension 101 to extension map
[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: Extension 101 cf is disabled
[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: Extension 101 do not disturb is disabled

[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: EXTENSION_STATE: 8 (RINGING)
[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: Extension 101 has ExtensionState: 8
[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: Checking CW and CFB status for extension 101
[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: Extension 101 is not available to be called
[2018-01-02 16:33:18] VERBOSE[11540][C-000001d1] res_agi.c: dialparties.agi: Extension 101 has call waiting disabled

Regards,

Robbert

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@zigge wrote:

Hi,

Recently upgraded from FreePBX v13 to v14. Experiencing an issue retrieving parked calls. User’s handset is set with two soft keys Park (*270) and Retrieve (*8571). The user is able to park the call, the caller gets parked but upon attempting to retrieve user gets a fast busy . I found a recent support topic with same symptoms and solution was to rid SIP Trunk Name of any spaces. I have confirmed no spaces exist with my trunks.

**Restapps and Parking modules uninstalled and installed fresh
**Current Asterisk Version: 13.18.4

Test Call Output:

    indent preformatted text by 4 spaces

SIP/301-00000043 answered SIP/customer-00000041
– Executing [s@macro-auto-blkvm:1] Set(“SIP/301-00000043”, “__MACRO_RESULT=”) in new stack
– Executing [s@macro-auto-blkvm:2] Set(“SIP/301-00000043”, “CFIGNORE=”) in new stack
– Executing [s@macro-auto-blkvm:3] Set(“SIP/301-00000043”, “MASTER_CHANNEL(CFIGNORE)=”) in new stack
– Executing [s@macro-auto-blkvm:4] Set(“SIP/301-00000043”, “FORWARD_CONTEXT=from-internal”) in new stack
– Executing [s@macro-auto-blkvm:5] Set(“SIP/301-00000043”, “MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal”) in new stack
– Executing [s@macro-auto-blkvm:6] Macro(“SIP/301-00000043”, “blkvm-clr,”) in new stack
– Executing [s@macro-blkvm-clr:1] Set(“SIP/301-00000043”, “SHARED(BLKVM,SIP/customer-00000041)=”) in new stack
– Executing [s@macro-blkvm-clr:2] Set(“SIP/301-00000043”, “GOSUB_RETVAL=”) in new stack
– Executing [s@macro-blkvm-clr:3] MacroExit(“SIP/301-00000043”, “”) in new stack
– Executing [s@macro-auto-blkvm:7] ExecIf(“SIP/301-00000043”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=301)”) in new stack
– Executing [s@macro-auto-blkvm:8] ExecIf(“SIP/301-00000043”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Bev)”) in new stack
– Channel SIP/301-00000043 joined ‘simple_bridge’ basic-bridge <6a943a28-4116-4c28-928d-fdc9ad4fbafb>
– Channel SIP/customer-00000041 joined ‘simple_bridge’ basic-bridge <6a943a28-4116-4c28-928d-fdc9ad4fbafb>
[2018-01-03 11:10:42] NOTICE[14902]: manager.c:3381 authenticate: 127.0.0.1 tried to authenticate with nonexistent user ‘cxpanel’
[2018-01-03 11:10:42] NOTICE[14902]: manager.c:3418 authenticate: 127.0.0.1 failed to authenticate as ‘cxpanel’
– Started music on hold, class ‘default’, on channel ‘SIP/customer-00000041’
– <SIP/301-00000043> Playing ‘pbx-transfer.ulaw’ (language ‘en’)
– Channel Local/70@from-internal-xfer-00000003;1 joined ‘simple_bridge’ basic-bridge
– Executing [70@from-internal-xfer:1] Park(“Local/70@from-internal-xfer-00000003;2”, “”) in new stack
– Channel SIP/301-00000043 left ‘simple_bridge’ basic-bridge <6a943a28-4116-4c28-928d-fdc9ad4fbafb>
– Channel SIP/301-00000043 joined ‘simple_bridge’ basic-bridge
– Parking ‘Local/70@from-internal-xfer-00000003;2’ in ‘default’ at space 71
– Channel Local/70@from-internal-xfer-00000003;2 joined ‘holding_bridge’ parking-bridge
== Extension Changed 71[park-hints] new state InUse for Notify User 307
– <Local/70@from-internal-xfer-00000003;2> Playing ‘digits/7.ulaw’ (language ‘en’)
– <Local/70@from-internal-xfer-00000003;2> Playing ‘digits/1.ulaw’ (language ‘en’)
– Started music on hold, class ‘default’, on channel ‘Local/70@from-internal-xfer-00000003;2’
[2018-01-03 11:10:58] NOTICE[14907]: manager.c:3381 authenticate: 127.0.0.1 tried to authenticate with nonexistent user ‘cxpanel’
[2018-01-03 11:10:58] NOTICE[14907]: manager.c:3418 authenticate: 127.0.0.1 failed to authenticate as ‘cxpanel’
[2018-01-03 11:11:00] NOTICE[9903]: chan_sip.c:28410 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 228
[2018-01-03 11:11:01] NOTICE[9903]: chan_sip.c:28410 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 235
– Channel Local/70@from-internal-xfer-00000003;1 left ‘simple_bridge’ basic-bridge
– Channel Local/70@from-internal-xfer-00000003;1 joined ‘simple_bridge’ basic-bridge <6a943a28-4116-4c28-928d-fdc9ad4fbafb>
– Channel SIP/301-00000043 left ‘simple_bridge’ basic-bridge
– Stopped music on hold on SIP/customer-00000041
– <Local/70@from-internal-xfer-00000003;1> Playing ‘beep.ulaw’ (language ‘en’)
– Channel SIP/customer-00000041 left ‘simple_bridge’ basic-bridge <6a943a28-4116-4c28-928d-fdc9ad4fbafb>
– Channel Local/70@from-internal-xfer-00000003;2 left ‘holding_bridge’ parking-bridge
– Stopped music on hold on Local/70@from-internal-xfer-00000003;2
– Channel SIP/customer-00000041 swapped with Local/70@from-internal-xfer-00000003;2 into ‘holding_bridge’ parking-bridge
– Channel Local/70@from-internal-xfer-00000003;1 left ‘simple_bridge’ basic-bridge <6a943a28-4116-4c28-928d-fdc9ad4fbafb>
== Spawn extension (from-internal-xfer, 70, 1) exited non-zero on ‘Local/70@from-internal-xfer-00000003;2’
– Executing [h@from-internal-xfer:1] Macro(“Local/70@from-internal-xfer-00000003;2”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“Local/70@from-internal-xfer-00000003;2”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“Local/70@from-internal-xfer-00000003;2”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“Local/70@from-internal-xfer-00000003;2”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“Local/70@from-internal-xfer-00000003;2”, “attendedtransfer-rec-restart.php,”) in new stack
– Started music on hold, class ‘default’, on channel ‘SIP/customer-00000041’
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_request: attendedtransfer-rec-restart.php
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_channel: Local/70@from-internal-xfer-00000003;2
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_language: en
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_type: Local
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_uniqueid: 1514995855.543
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_version: 13.18.4
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_callerid: 5195555555
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_calleridname: Zigge
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_callingpres: 0
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_callingani2: 0
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_callington: 0
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_callingtns: 0
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_dnid: unknown
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_rdnis: unknown
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_context: macro-hangupcall
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_extension: s
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_priority: 5
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_enhanced: 0.0
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_accountcode:
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_threadid: 140065625523968
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_arg_1:
<Local/70@from-internal-xfer-00000003;2>AGI Tx >> agi_arg_2:
<Local/70@from-internal-xfer-00000003;2>AGI Tx >>
– <Local/70@from-internal-xfer-00000003;2>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“Local/70@from-internal-xfer-00000003;2”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘Local/70@from-internal-xfer-00000003;2’ in macro ‘hangupcall’
== Spawn extension (from-internal-xfer, h, 1) exited non-zero on ‘Local/70@from-internal-xfer-00000003;2’

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@christianjensen wrote:

No calls allowed, other than between extensions.
Normally this is solved by typing in time groups or time conditions.

This time is different.
Freepbx does not remember the data entered in timegroupe
And connects a empty timegroupe to all outbound routes.
All outbound routes are listed in false links above the timegroupe, showing them to be using the timegroupe.
All outbound are on permanent route
I have done nothing but updating the system before christmas.
Running the latest distro version 13.18.4 on VMware.

Any suggestions?

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@jfall wrote:

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@elarsen wrote:

Hello all,

new freepbx 13 install. Everything else is working as expected.

Having an issue with inbound faxing. Outbound fax works fine from UCP

I have a DID and inbound route assigned to fax recipient 999 (fax enabled user) email address is set to my company email. Email server setup is correct and all other email related items work (voicemail, new user creation etc…)

PBX is behind a nat. All required ports forwarded. 4000-4999. 1:1 nat

Here is the output from the console when I try to send a test fax from faxzero.

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f18dc106520 – Strict RTP learning after remote address set to: 8.38.41.138:37750
– Executing [2677932925@from-trunk-sip-BBD_1:1] Set(“SIP/BBD_1-0000002f”, “GROUP()=OUT_1”) in new stack
– Executing [2677932925@from-trunk-sip-BBD_1:2] Goto(“SIP/BBD_1-0000002f”, “from-trunk,2677932925,1”) in new stack
– Goto (from-trunk,2677932925,1)
– Executing [2677932925@from-trunk:1] Set(“SIP/BBD_1-0000002f”, “__DIRECTION=INBOUND”) in new stack
– Executing [2677932925@from-trunk:2] Gosub(“SIP/BBD_1-0000002f”, “sub-record-check,s,1(in,2677932925,no)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/BBD_1-0000002f”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/BBD_1-0000002f”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/BBD_1-0000002f”, “NOW=1515039207”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/BBD_1-0000002f”, “__DAY=04”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/BBD_1-0000002f”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/BBD_1-0000002f”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/BBD_1-0000002f”, “__TIMESTR=20180104-041327”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/BBD_1-0000002f”, “__FROMEXTEN=unknown”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/BBD_1-0000002f”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/BBD_1-0000002f”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/BBD_1-0000002f”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/BBD_1-0000002f”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/BBD_1-0000002f”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/BBD_1-0000002f”, “2?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/BBD_1-0000002f”, “1?sub-record-check,in,1”) in new stack
– Goto (sub-record-check,in,1)
– Executing [in@sub-record-check:1] NoOp(“SIP/BBD_1-0000002f”, “Inbound Recording Check to 2677932925”) in new stack
– Executing [in@sub-record-check:2] Set(“SIP/BBD_1-0000002f”, “FROMEXTEN=unknown”) in new stack
– Executing [in@sub-record-check:3] ExecIf(“SIP/BBD_1-0000002f”, “10?Set(FROMEXTEN=7708240759)”) in new stack
– Executing [in@sub-record-check:4] Gosub(“SIP/BBD_1-0000002f”, “recordcheck,1(no,in,2677932925)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/BBD_1-0000002f”, “Starting recording check against no”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/BBD_1-0000002f”, “no”) in new stack
– Goto (sub-record-check,recordcheck,12)
– Executing [recordcheck@sub-record-check:12] Set(“SIP/BBD_1-0000002f”, “__REC_POLICY_MODE=NO”) in new stack
– Executing [recordcheck@sub-record-check:13] Return(“SIP/BBD_1-0000002f”, “”) in new stack
– Executing [in@sub-record-check:5] Return(“SIP/BBD_1-0000002f”, “”) in new stack
– Executing [2677932925@from-trunk:3] Gosub(“SIP/BBD_1-0000002f”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/BBD_1-0000002f”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/BBD_1-0000002f”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/BBD_1-0000002f”, “”) in new stack
– Executing [2677932925@from-trunk:4] Set(“SIP/BBD_1-0000002f”, “__FROM_DID=2677932925”) in new stack
– Executing [2677932925@from-trunk:5] Set(“SIP/BBD_1-0000002f”, “CDR(did)=2677932925”) in new stack
– Executing [2677932925@from-trunk:6] ExecIf(“SIP/BBD_1-0000002f”, “1 ?Set(CALLERID(name)=7708240759)”) in new stack
– Executing [2677932925@from-trunk:7] Set(“SIP/BBD_1-0000002f”, “__MOHCLASS=”) in new stack
– Executing [2677932925@from-trunk:8] Set(“SIP/BBD_1-0000002f”, “__REVERSAL_REJECT=FALSE”) in new stack
– Executing [2677932925@from-trunk:9] GotoIf(“SIP/BBD_1-0000002f”, “1?post-reverse-charge”) in new stack
– Goto (from-trunk,2677932925,11)
– Executing [2677932925@from-trunk:11] NoOp(“SIP/BBD_1-0000002f”, “”) in new stack
– Executing [2677932925@from-trunk:12] Set(“SIP/BBD_1-0000002f”, “__CALLINGNAMEPRES_SV=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:13] Set(“SIP/BBD_1-0000002f”, “__CALLINGNUMPRES_SV=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:14] Set(“SIP/BBD_1-0000002f”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:15] Set(“SIP/BBD_1-0000002f”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:16] NoOp(“SIP/BBD_1-0000002f”, “CallerID Entry Point”) in new stack
– Executing [2677932925@from-trunk:17] Set(“SIP/BBD_1-0000002f”, “FAX_DEST=ext-fax^102^1”) in new stack
– Executing [2677932925@from-trunk:18] Set(“SIP/BBD_1-0000002f”, “FAXOPT(faxdetect)=yes”) in new stack
– Executing [2677932925@from-trunk:19] Answer(“SIP/BBD_1-0000002f”, “”) in new stack
> 0x7f18dc106520 – Strict RTP learning after remote address set to: 8.38.41.138:37750
> 0x7f18dc106520 – Strict RTP switching to RTP target address 8.38.41.138:37750 as source
– Executing [2677932925@from-trunk:20] PlayTones(“SIP/BBD_1-0000002f”, “ring”) in new stack
– Executing [2677932925@from-trunk:21] Wait(“SIP/BBD_1-0000002f”, “4”) in new stack
> 0x7f18dc106520 – Strict RTP learning complete - Locking on source address 8.38.41.138:37750
– Executing [2677932925@from-trunk:22] Goto(“SIP/BBD_1-0000002f”, “ext-fax,102,1”) in new stack
– Goto (ext-fax,102,1)
– Executing [102@ext-fax:1] Set(“SIP/BBD_1-0000002f”, “FAX_FOR=Fax (102)”) in new stack
– Executing [102@ext-fax:2] NoOp(“SIP/BBD_1-0000002f”, “Receiving Fax for: Fax (102), From: “7708240759” <7708240759>”) in new stack
– Executing [102@ext-fax:3] Set(“SIP/BBD_1-0000002f”, “FAX_RX_USER=102”) in new stack
– Executing [102@ext-fax:4] Set(“SIP/BBD_1-0000002f”, “FAX_RX_EMAIL_LEN=27”) in new stack
– Executing [102@ext-fax:5] ExecIf(“SIP/BBD_1-0000002f”, “1?Set(ARIUSER=102)”) in new stack
– Executing [102@ext-fax:6] ExecIf(“SIP/BBD_1-0000002f”, “1?AGI(fax.agi)”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fax.agi
– <SIP/BBD_1-0000002f>AGI Script fax.agi completed, returning 0
– Executing [102@ext-fax:7] Goto(“SIP/BBD_1-0000002f”, “s,receivefax”) in new stack
– Goto (ext-fax,s,3)
– Executing [s@ext-fax:3] StopPlayTones(“SIP/BBD_1-0000002f”, “”) in new stack
– Executing [s@ext-fax:4] ReceiveFAX(“SIP/BBD_1-0000002f”, “/var/spool/asterisk/fax/1515039207.89.tif,f”) in new stack
– Channel ‘SIP/BBD_1-0000002f’ receiving FAX ‘/var/spool/asterisk/fax/1515039207.89.tif’
[2018-01-04 04:13:37] WARNING[9847][C-00000044]: chan_sip.c:10649 process_sdp: Failed to initialize UDPTL, declining image stream
[2018-01-04 04:13:37] WARNING[9847][C-00000044]: chan_sip.c:10828 process_sdp: Failing due to no acceptable offer found
== Spawn extension (ext-fax, s, 4) exited non-zero on ‘SIP/BBD_1-0000002f’
– Executing [h@ext-fax:1] GotoIf(“SIP/BBD_1-0000002f”, “1?failed”) in new stack
– Goto (ext-fax,h,104)
– Executing [h@ext-fax:104] NoOp(“SIP/BBD_1-0000002f”, “FAX FAILED for: Fax (102) , From: “7708240759” <7708240759>”) in new stack
– Executing [h@ext-fax:105] Macro(“SIP/BBD_1-0000002f”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/BBD_1-0000002f”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/BBD_1-0000002f”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“SIP/BBD_1-0000002f”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“SIP/BBD_1-0000002f”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <SIP/BBD_1-0000002f>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“SIP/BBD_1-0000002f”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/BBD_1-0000002f’ in macro ‘hangupcall’
== Spawn extension (ext-fax, h, 105) exited non-zero on ‘SIP/BBD_1-0000002f’
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f18dc106520 – Strict RTP learning after remote address set to: 8.38.41.138:41742
– Executing [2677932925@from-trunk-sip-BBD_1:1] Set(“SIP/BBD_1-00000030”, “GROUP()=OUT_1”) in new stack
– Executing [2677932925@from-trunk-sip-BBD_1:2] Goto(“SIP/BBD_1-00000030”, “from-trunk,2677932925,1”) in new stack
– Goto (from-trunk,2677932925,1)
– Executing [2677932925@from-trunk:1] Set(“SIP/BBD_1-00000030”, “__DIRECTION=INBOUND”) in new stack
– Executing [2677932925@from-trunk:2] Gosub(“SIP/BBD_1-00000030”, “sub-record-check,s,1(in,2677932925,no)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/BBD_1-00000030”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/BBD_1-00000030”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/BBD_1-00000030”, “NOW=1515039441”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/BBD_1-00000030”, “__DAY=04”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/BBD_1-00000030”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/BBD_1-00000030”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/BBD_1-00000030”, “__TIMESTR=20180104-041721”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/BBD_1-00000030”, “__FROMEXTEN=unknown”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/BBD_1-00000030”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/BBD_1-00000030”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/BBD_1-00000030”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/BBD_1-00000030”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/BBD_1-00000030”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/BBD_1-00000030”, “2?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/BBD_1-00000030”, “1?sub-record-check,in,1”) in new stack
– Goto (sub-record-check,in,1)
– Executing [in@sub-record-check:1] NoOp(“SIP/BBD_1-00000030”, “Inbound Recording Check to 2677932925”) in new stack
– Executing [in@sub-record-check:2] Set(“SIP/BBD_1-00000030”, “FROMEXTEN=unknown”) in new stack
– Executing [in@sub-record-check:3] ExecIf(“SIP/BBD_1-00000030”, “10?Set(FROMEXTEN=9252484037)”) in new stack
– Executing [in@sub-record-check:4] Gosub(“SIP/BBD_1-00000030”, “recordcheck,1(no,in,2677932925)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/BBD_1-00000030”, “Starting recording check against no”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/BBD_1-00000030”, “no”) in new stack
– Goto (sub-record-check,recordcheck,12)
– Executing [recordcheck@sub-record-check:12] Set(“SIP/BBD_1-00000030”, “__REC_POLICY_MODE=NO”) in new stack
– Executing [recordcheck@sub-record-check:13] Return(“SIP/BBD_1-00000030”, “”) in new stack
– Executing [in@sub-record-check:5] Return(“SIP/BBD_1-00000030”, “”) in new stack
– Executing [2677932925@from-trunk:3] Gosub(“SIP/BBD_1-00000030”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/BBD_1-00000030”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/BBD_1-00000030”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/BBD_1-00000030”, “”) in new stack
– Executing [2677932925@from-trunk:4] Set(“SIP/BBD_1-00000030”, “__FROM_DID=2677932925”) in new stack
– Executing [2677932925@from-trunk:5] Set(“SIP/BBD_1-00000030”, “CDR(did)=2677932925”) in new stack
– Executing [2677932925@from-trunk:6] ExecIf(“SIP/BBD_1-00000030”, “1 ?Set(CALLERID(name)=9252484037)”) in new stack
– Executing [2677932925@from-trunk:7] Set(“SIP/BBD_1-00000030”, “__MOHCLASS=”) in new stack
– Executing [2677932925@from-trunk:8] Set(“SIP/BBD_1-00000030”, “__REVERSAL_REJECT=FALSE”) in new stack
– Executing [2677932925@from-trunk:9] GotoIf(“SIP/BBD_1-00000030”, “1?post-reverse-charge”) in new stack
– Goto (from-trunk,2677932925,11)
– Executing [2677932925@from-trunk:11] NoOp(“SIP/BBD_1-00000030”, “”) in new stack
– Executing [2677932925@from-trunk:12] Set(“SIP/BBD_1-00000030”, “__CALLINGNAMEPRES_SV=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:13] Set(“SIP/BBD_1-00000030”, “__CALLINGNUMPRES_SV=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:14] Set(“SIP/BBD_1-00000030”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:15] Set(“SIP/BBD_1-00000030”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:16] NoOp(“SIP/BBD_1-00000030”, “CallerID Entry Point”) in new stack
– Executing [2677932925@from-trunk:17] Set(“SIP/BBD_1-00000030”, “FAX_DEST=ext-fax^102^1”) in new stack
– Executing [2677932925@from-trunk:18] Set(“SIP/BBD_1-00000030”, “FAXOPT(faxdetect)=yes”) in new stack
– Executing [2677932925@from-trunk:19] Answer(“SIP/BBD_1-00000030”, “”) in new stack
> 0x7f18dc106520 – Strict RTP switching to RTP target address 8.38.41.138:41742 as source
– Executing [2677932925@from-trunk:20] PlayTones(“SIP/BBD_1-00000030”, “ring”) in new stack
– Executing [2677932925@from-trunk:21] Wait(“SIP/BBD_1-00000030”, “4”) in new stack
> 0x7f18dc106520 – Strict RTP learning complete - Locking on source address 8.38.41.138:41742
– Executing [2677932925@from-trunk:22] Goto(“SIP/BBD_1-00000030”, “ext-fax,102,1”) in new stack
– Goto (ext-fax,102,1)
– Executing [102@ext-fax:1] Set(“SIP/BBD_1-00000030”, “FAX_FOR=Fax (102)”) in new stack
– Executing [102@ext-fax:2] NoOp(“SIP/BBD_1-00000030”, “Receiving Fax for: Fax (102), From: “9252484037” <9252484037>”) in new stack
– Executing [102@ext-fax:3] Set(“SIP/BBD_1-00000030”, “FAX_RX_USER=102”) in new stack
– Executing [102@ext-fax:4] Set(“SIP/BBD_1-00000030”, “FAX_RX_EMAIL_LEN=27”) in new stack
– Executing [102@ext-fax:5] ExecIf(“SIP/BBD_1-00000030”, “1?Set(ARIUSER=102)”) in new stack
– Executing [102@ext-fax:6] ExecIf(“SIP/BBD_1-00000030”, “1?AGI(fax.agi)”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fax.agi
– <SIP/BBD_1-00000030>AGI Script fax.agi completed, returning 0
– Executing [102@ext-fax:7] Goto(“SIP/BBD_1-00000030”, “s,receivefax”) in new stack
– Goto (ext-fax,s,3)
– Executing [s@ext-fax:3] StopPlayTones(“SIP/BBD_1-00000030”, “”) in new stack
– Executing [s@ext-fax:4] ReceiveFAX(“SIP/BBD_1-00000030”, “/var/spool/asterisk/fax/1515039441.90.tif,f”) in new stack
– Channel ‘SIP/BBD_1-00000030’ receiving FAX ‘/var/spool/asterisk/fax/1515039441.90.tif’
[2018-01-04 04:17:31] WARNING[9847][C-00000045]: chan_sip.c:10649 process_sdp: Failed to initialize UDPTL, declining image stream
[2018-01-04 04:17:31] WARNING[9847][C-00000045]: chan_sip.c:10828 process_sdp: Failing due to no acceptable offer found
[2018-01-04 04:17:31] WARNING[9847][C-00000045]: chan_sip.c:10649 process_sdp: Failed to initialize UDPTL, declining image stream
[2018-01-04 04:17:31] WARNING[9847][C-00000045]: chan_sip.c:10828 process_sdp: Failing due to no acceptable offer found
[2018-01-04 04:17:31] WARNING[9847][C-00000045]: chan_sip.c:10649 process_sdp: Failed to initialize UDPTL, declining image stream
[2018-01-04 04:17:31] WARNING[9847][C-00000045]: chan_sip.c:10828 process_sdp: Failing due to no acceptable offer found
– Executing [s@ext-fax:5] ExecIf(“SIP/BBD_1-00000030”, “1?Set(FAXSTATUS=“FAILED: error: Unexpected message received statusstr: Unexpected message received”)”) in new stack
– Executing [s@ext-fax:6] Hangup(“SIP/BBD_1-00000030”, “”) in new stack
== Spawn extension (ext-fax, s, 6) exited non-zero on ‘SIP/BBD_1-00000030’
– Executing [h@ext-fax:1] GotoIf(“SIP/BBD_1-00000030”, “1?failed”) in new stack
– Goto (ext-fax,h,104)
– Executing [h@ext-fax:104] NoOp(“SIP/BBD_1-00000030”, “FAX “FAILED: error: Unexpected message received statusstr: Unexpected message received” for: Fax (102) , From: “9252484037” <9252484037>”) in new stack
– Executing [h@ext-fax:105] Macro(“SIP/BBD_1-00000030”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/BBD_1-00000030”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/BBD_1-00000030”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“SIP/BBD_1-00000030”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“SIP/BBD_1-00000030”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <SIP/BBD_1-00000030>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“SIP/BBD_1-00000030”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/BBD_1-00000030’ in macro ‘hangupcall’
== Spawn extension (ext-fax, h, 105) exited non-zero on ‘SIP/BBD_1-00000030’
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f18dc106520 – Strict RTP learning after remote address set to: 8.38.41.138:12638
– Executing [2677932925@from-trunk-sip-BBD_1:1] Set(“SIP/BBD_1-00000031”, “GROUP()=OUT_1”) in new stack
– Executing [2677932925@from-trunk-sip-BBD_1:2] Goto(“SIP/BBD_1-00000031”, “from-trunk,2677932925,1”) in new stack
– Goto (from-trunk,2677932925,1)
– Executing [2677932925@from-trunk:1] Set(“SIP/BBD_1-00000031”, “__DIRECTION=INBOUND”) in new stack
– Executing [2677932925@from-trunk:2] Gosub(“SIP/BBD_1-00000031”, “sub-record-check,s,1(in,2677932925,no)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/BBD_1-00000031”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/BBD_1-00000031”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/BBD_1-00000031”, “NOW=1515039675”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/BBD_1-00000031”, “__DAY=04”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/BBD_1-00000031”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/BBD_1-00000031”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/BBD_1-00000031”, “__TIMESTR=20180104-042115”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/BBD_1-00000031”, “__FROMEXTEN=unknown”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/BBD_1-00000031”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/BBD_1-00000031”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/BBD_1-00000031”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/BBD_1-00000031”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/BBD_1-00000031”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/BBD_1-00000031”, “2?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/BBD_1-00000031”, “1?sub-record-check,in,1”) in new stack
– Goto (sub-record-check,in,1)
– Executing [in@sub-record-check:1] NoOp(“SIP/BBD_1-00000031”, “Inbound Recording Check to 2677932925”) in new stack
– Executing [in@sub-record-check:2] Set(“SIP/BBD_1-00000031”, “FROMEXTEN=unknown”) in new stack
– Executing [in@sub-record-check:3] ExecIf(“SIP/BBD_1-00000031”, “10?Set(FROMEXTEN=7867895398)”) in new stack
– Executing [in@sub-record-check:4] Gosub(“SIP/BBD_1-00000031”, “recordcheck,1(no,in,2677932925)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/BBD_1-00000031”, “Starting recording check against no”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/BBD_1-00000031”, “no”) in new stack
– Goto (sub-record-check,recordcheck,12)
– Executing [recordcheck@sub-record-check:12] Set(“SIP/BBD_1-00000031”, “__REC_POLICY_MODE=NO”) in new stack
– Executing [recordcheck@sub-record-check:13] Return(“SIP/BBD_1-00000031”, “”) in new stack
– Executing [in@sub-record-check:5] Return(“SIP/BBD_1-00000031”, “”) in new stack
– Executing [2677932925@from-trunk:3] Gosub(“SIP/BBD_1-00000031”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/BBD_1-00000031”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/BBD_1-00000031”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/BBD_1-00000031”, “”) in new stack
– Executing [2677932925@from-trunk:4] Set(“SIP/BBD_1-00000031”, “__FROM_DID=2677932925”) in new stack
– Executing [2677932925@from-trunk:5] Set(“SIP/BBD_1-00000031”, “CDR(did)=2677932925”) in new stack
– Executing [2677932925@from-trunk:6] ExecIf(“SIP/BBD_1-00000031”, “1 ?Set(CALLERID(name)=7867895398)”) in new stack
– Executing [2677932925@from-trunk:7] Set(“SIP/BBD_1-00000031”, “__MOHCLASS=”) in new stack
– Executing [2677932925@from-trunk:8] Set(“SIP/BBD_1-00000031”, “__REVERSAL_REJECT=FALSE”) in new stack
– Executing [2677932925@from-trunk:9] GotoIf(“SIP/BBD_1-00000031”, “1?post-reverse-charge”) in new stack
– Goto (from-trunk,2677932925,11)
– Executing [2677932925@from-trunk:11] NoOp(“SIP/BBD_1-00000031”, “”) in new stack
– Executing [2677932925@from-trunk:12] Set(“SIP/BBD_1-00000031”, “__CALLINGNAMEPRES_SV=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:13] Set(“SIP/BBD_1-00000031”, “__CALLINGNUMPRES_SV=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:14] Set(“SIP/BBD_1-00000031”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:15] Set(“SIP/BBD_1-00000031”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
– Executing [2677932925@from-trunk:16] NoOp(“SIP/BBD_1-00000031”, “CallerID Entry Point”) in new stack
– Executing [2677932925@from-trunk:17] Set(“SIP/BBD_1-00000031”, “FAX_DEST=ext-fax^102^1”) in new stack
– Executing [2677932925@from-trunk:18] Set(“SIP/BBD_1-00000031”, “FAXOPT(faxdetect)=yes”) in new stack
– Executing [2677932925@from-trunk:19] Answer(“SIP/BBD_1-00000031”, “”) in new stack
> 0x7f18dc106520 – Strict RTP switching to RTP target address 8.38.41.138:12638 as source
– Executing [2677932925@from-trunk:20] PlayTones(“SIP/BBD_1-00000031”, “ring”) in new stack
– Executing [2677932925@from-trunk:21] Wait(“SIP/BBD_1-00000031”, “4”) in new stack
> 0x7f18dc106520 – Strict RTP learning complete - Locking on source address 8.38.41.138:12638
– Executing [2677932925@from-trunk:22] Goto(“SIP/BBD_1-00000031”, “ext-fax,102,1”) in new stack
– Goto (ext-fax,102,1)
– Executing [102@ext-fax:1] Set(“SIP/BBD_1-00000031”, “FAX_FOR=Fax (102)”) in new stack
– Executing [102@ext-fax:2] NoOp(“SIP/BBD_1-00000031”, “Receiving Fax for: Fax (102), From: “7867895398” <7867895398>”) in new stack
– Executing [102@ext-fax:3] Set(“SIP/BBD_1-00000031”, “FAX_RX_USER=102”) in new stack
– Executing [102@ext-fax:4] Set(“SIP/BBD_1-00000031”, “FAX_RX_EMAIL_LEN=27”) in new stack
– Executing [102@ext-fax:5] ExecIf(“SIP/BBD_1-00000031”, “1?Set(ARIUSER=102)”) in new stack
– Executing [102@ext-fax:6] ExecIf(“SIP/BBD_1-00000031”, “1?AGI(fax.agi)”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fax.agi
– <SIP/BBD_1-00000031>AGI Script fax.agi completed, returning 0
– Executing [102@ext-fax:7] Goto(“SIP/BBD_1-00000031”, “s,receivefax”) in new stack
– Goto (ext-fax,s,3)
– Executing [s@ext-fax:3] StopPlayTones(“SIP/BBD_1-00000031”, “”) in new stack
– Executing [s@ext-fax:4] ReceiveFAX(“SIP/BBD_1-00000031”, “/var/spool/asterisk/fax/1515039675.91.tif,f”) in new stack
– Channel ‘SIP/BBD_1-00000031’ receiving FAX ‘/var/spool/asterisk/fax/1515039675.91.tif’
[2018-01-04 04:21:24] WARNING[9847][C-00000046]: chan_sip.c:10649 process_sdp: Failed to initialize UDPTL, declining image stream
[2018-01-04 04:21:24] WARNING[9847][C-00000046]: chan_sip.c:10828 process_sdp: Failing due to no acceptable offer found
== Spawn extension (ext-fax, s, 4) exited non-zero on ‘SIP/BBD_1-00000031’
– Executing [h@ext-fax:1] GotoIf(“SIP/BBD_1-00000031”, “1?failed”) in new stack
– Goto (ext-fax,h,104)
– Executing [h@ext-fax:104] NoOp(“SIP/BBD_1-00000031”, “FAX FAILED for: Fax (102) , From: “7867895398” <7867895398>”) in new stack
– Executing [h@ext-fax:105] Macro(“SIP/BBD_1-00000031”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/BBD_1-00000031”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/BBD_1-00000031”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“SIP/BBD_1-00000031”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“SIP/BBD_1-00000031”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <SIP/BBD_1-00000031>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“SIP/BBD_1-00000031”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/BBD_1-00000031’ in macro ‘hangupcall’
== Spawn extension (ext-fax, h, 105) exited non-zero on ‘SIP/BBD_1-00000031’

Please advise

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@bajramia wrote:

Hi Team,
I have install FreePBX 14 and Asterisk 13.18.4 im trying to install ooh323 channel
when i run module load chan_ooh323.so i get an error
chan_ooh323.c:2834 reload_config: Unable to load config ooh323.conf, OOH323 disabled
when i create the ooh323.conf
run module load chan_ooh323.so
i get disconnect form astersik i cant reconnect till i delete the ooh323.conf then it connects back

please help
what im doing wrong thank you.

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@comtech wrote:

I saw this post and wondered if the landscape has changed? Hold has been implemented. Is transfer and conference an option now? If yes, is it already on the roadmap? Such a slick product, I can see many uses for it.

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@GameGamer43 wrote:

As we embark on FreePBX 15, we’ve come up with two main themes which we’ve shared with the community at this past years Astricon event. The first is a completely redesigned backup and restore, making it easier to maintain, more flexible, and allow for between version restores. The second things being added to FreePBX for 15 is a GraphQL based API allowing for greater integration into FreePBX.

While we work on both backup and restore and the API, we want to hear from you as to your biggest pain points or just features that would make your life easier. So what would you like to see added to FreePBX 15?

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@Arunbalannv wrote:

I am using Asterisk 13 and Enabled ICE support configured STUN and TURN. That time I was able to make phone calls from all types of Networks including most restrictive NATs. The audio was transferring via Asterisk. Everything was working here.

But Now I need to enable directmedia in order to have direct RTP handover between extensions.
After enabling directmedia, the Asterisk server is not forwarding the ICE candidates from the dialing extension to the destination extension. So the call is only working in local network. And here ICE is not working in this situation.

So how to enable directmedia when using ICE?

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@kd2dwj wrote:

So I just setup a new server, using the new FreePBX 14.0.1.20, setup the responsive firewall for chan_sip (not using pjsip) and everything working wonderfully

Made sure SIP Guests was set to no, and set Allow Anonymous Inbound SIP Calls to no.

Now I’m getting this on my CDR

Call Date Recording System CallerID Outbound CallerID DID App Destination Disposition Duration Userfield Account CDR Table CDR Graph
Fri, Jan 5 2018 4:20 AM 1515161999.4504 “6318489673” <6318489673> 6319570101 Dial 121 ANSWERED 01:04
Fri, Jan 5 2018 4:20 AM 1515161999.4504 “6318489673” <6318489673> 6319570101 Dial 9001 NO ANSWER 00:03
Fri, Jan 5 2018 4:20 AM 1515161999.4504 “6318489673” <6318489673> 6319570101 Dial 9001 NO ANSWER 00:03
Fri, Jan 5 2018 4:19 AM 1515161999.4504 “6318489673” <6318489673> 6319570101 Dial 9001 ANSWERED 00:50
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 ANSWERED 01:28
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:08
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9002 NO ANSWER 00:09
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9001 NO ANSWER 00:15
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9001 NO ANSWER 00:15
Fri, Jan 5 2018 4:01 AM 1515160883.4489 “2102503767” <2102503767> 6319570101 Dial 9001 NO ANSWER 00:20

No idea what this 515161999 number is

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@aeciolemos wrote:

Hi everyone,

I just noticed that under the Reports menu, the “Asterisk Info” page is missing. Anyone know where it has gone?

I am running FreePBX 14.0.1.24
PBX Firmware:12.7.4-1712-2.sng7

Is this by desing? It was very useful to see what peers had registered, their IP addresses, etc.

Thanks

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@haljbr wrote:

Is there a way to do it on GUI?

I need to remove 3 digits when I receive any call on a specific trunk.
I know how to do it when dialing out.
How to do it when there is an inbound call?

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@intrigue wrote:

Hello,

I’m attempting to broadcast an emergency announcement using the Paging and Intercom module through all of the phones. Everything works as expected except I’ve found that the phones do not automatically terminate the call after the announcement is complete. Through further investigation, I’ve found that if the caller initiating the announcement remains on for the duration of the announcement then hangs up, it will disconnect as expected. If the caller hangs up while the announcement is still playing the phones all remain connected indefinitely. Does anyone know of a work around for this?
Any help is greatly appreciated.

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