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Attended transfer no beep

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@Cyber wrote:

Hi ALL.
I have some issues this attended transfer.

My installation:
FreePBX 15.0.16
Asterisk 13.22.0

We have incoming call. Agent answer it and try to transfer call to another extension using asterisk attended transfer feature code. Extension successfully answered. After talk, extension was hangup the call. Now call return to agent. But no beep after return. This is a problem.

My features_general_custom.conf:
atxfernoanswertimeout = 10
atxferloopdelay = 1
xfersound = beep
xferfailsound = beeperr
pickupsound = beep
pickupfailsound = beeper

What else can i do to resolve this problem?

Best regards,
Sergey.

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SipML5 and FreePBX 13.X

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@NorColorNorName wrote:

Hello,

today i’m trying to configure my freePBX 13 to implement the WebRTC SipML5, but sadly there are few docs about how to configure our FreePBX.

Do someone have a link for me ? like a guide or a tutorial ? I got a fresh install FreePBX 13 and asterisk 13 with a FQDN and a let’sencrypt certificate.

Thank you for any advices.
Regards.
NCNN.

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Cannot configure an IVR

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@AmidouFlorian92 wrote:

Hi
I have a problem when configuring an IVR in Freepbx
This is what I proceeed
1 I created an IVR
2 After that I created an extension for my IVR using the number 2003
3 I created an Inbound route and set the DID number of my inboud route to 2003
4 In the page SET Destination I selected IVR and I associated a recording that I was added previously
5 I login in Xlite with the credentials of my extension 2003
6 When I call my IVR I can respond but I hear a music that is not the recording I have used

Is someone who could help me?
Thanks

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Failed to authenticate device with Server IP Address

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@rsarceno wrote:

FreePBX 13.0.197
Asterisk Version 13.23.1
Freepbx Distro: 10.13.66-22

[2019-08-03 09:37:43] NOTICE[17419][C-00000025] chan_sip.c: Failed to authenticate device sip:30@x.x.x.x;tag=1757554836
[2019-08-01 00:31:20] NOTICE[23370][C-00000000] chan_sip.c: Failed to authenticate device sip:801@x.x.x.x;tag=619170660
[2019-08-02 01:43:54] NOTICE[12702][C-00000006] chan_sip.c: Failed to authenticate device sip:test@x.x.x.x;tag=1200836332

Server crash and have to do a full back and restore.
Since then I’ve been getting a lot of this messages
I have firewall and fail2ban enable
I normally blacklist the IP address but this time the x.x.x.x is the IP address of my server
I’m 100% certain I didn’t set up ext 30, 801 or test

I appreciate any info

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Upgrade 13 to SNG7 - Modules still at 13

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@PitzKey wrote:

Hi guys.

I updated today a machine from FreePBX 13 to SNG7 by following the instructions in the wiki: https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7

After the upgrade was complete, it errored out:

PHP Fatal error: Cannot redeclare class FreePBX\Console\Command\RemoteUnlock in /var/www/html/admin/libraries/Console/RemoteUnlock.class.php on line 23
Whoops\Exception\ErrorException: Cannot redeclare class FreePBX\Console\Command\RemoteUnlock in file /var/www/html/admin/libraries/Console/RemoteUnlock.class.php on line 23
Stack trace:

1. Whoops\Exception\ErrorException->() /var/www/html/admin/libraries/Console/RemoteUnlock.class.php:23
PHP Fatal error: Cannot redeclare class FreePBX\Console\Command\RemoteUnlock in /var/www/html/admin/libraries/Console/RemoteUnlock.class.php on line 23

I fixed it by removing the duplicate file, Thank you, Lorne!

Then it wouldn’t connect to Asterisk, I reminded myself of this post. Thank you again, Lorne!

Finally, it said some modules are broken, I fixed it by re-downloading it.

Now, the issue is, that the OS got updated to SNG, but all modules are still on 13.

[root@localhost ~]# cat /etc/schmooze/pbx-version
12.7.6-1904-1.sng7
[root@localhost schmooze]# rpm -qa | grep sangoma-pbx
sangoma-pbx-1904-1.sng7.noarch

Modules:

[root@localhost ~]# fwconsole ma list
No repos specified, using: [standard,commercial,extended,unsupported] from last GUI settings

+---------------------+------------+-----------------------------------+------------+
| Module              | Version    | Status                            | License    |
+---------------------+------------+-----------------------------------+------------+
| accountcodepreserve | 13.0.2.2   | Enabled                           | GPLv2      |
| amd                 | 13.0.3     | Enabled                           | GPLv3+     |
| announcement        | 13.0.7.7   | Enabled                           | GPLv3+     |
| areminder           | 13.0.10.11 | Enabled                           | Commercial |
| arimanager          | 13.0.5.2   | Enabled                           | GPLv3+     |
| asterisk-cli        | 13.0.4     | Enabled                           | GPLv3+     |
| asteriskinfo        | 13.0.7.1   | Enabled                           | GPLv3+     |
| backup              | 13.0.27.21 | Enabled                           | GPLv3+     |
| blacklist           | 13.0.14.12 | Enabled                           | GPLv3+     |
| broadcast           | 13.0.12.15 | Enabled                           | Commercial |
| builtin             |            | Enabled                           |            |
| bulkdids            | 13.0.2     | Enabled                           | GPLv3+     |
| bulkextensions      | 13.0.3     | Enabled                           | GPLv3+     |
| bulkhandler         | 13.0.14.8  | Enabled                           | GPLv3+     |
| calendar            |            | Not Installed (Locally available) | GPLv3+     |
| callback            | 13.0.5.4   | Enabled                           | GPLv3+     |
| callforward         | 13.0.4.2   | Enabled                           | AGPLv3+    |
| callrecording       | 13.0.11.13 | Enabled                           | AGPLv3+    |
| callwaiting         | 13.0.4.1   | Enabled                           | GPLv3+     |
| campon              | 13.0.4.1   | Enabled                           | GPLv3+     |
| cdr                 | 13.0.33    | Enabled                           | GPLv3+     |
| cel                 | 13.0.26.8  | Enabled                           | GPLv3+     |
| certman             | 13.0.39    | Enabled                           | AGPLv3+    |
| cidlookup           | 13.0.12.3  | Enabled                           | GPLv3+     |
| conferences         | 13.0.23.15 | Enabled                           | GPLv3+     |
| conferencespro      | 13.0.27.11 | Enabled                           | Commercial |
| configedit          | 13.0.7.1   | Enabled                           | AGPLv3+    |
| contactmanager      | 13.0.45.1  | Enabled                           | GPLv3+     |
| core                | 13.0.131   | Enabled                           | GPLv3+     |
| customappsreg       | 13.0.5.7   | Enabled                           | GPLv3+     |
| customcontexts      | 13.0.3.1   | Enabled                           | GPLv2+     |
| cxpanel             | 13.0.5.2   | Enabled                           | GPLv3      |
| dahdiconfig         | 13.0.33.15 | Enabled                           | GPLv3+     |
| dashboard           | 13.0.26.2  | Enabled                           | AGPLv3+    |
| daynight            | 13.0.15.1  | Enabled                           | GPLv3+     |
| dictate             | 13.0.5     | Enabled                           | GPLv3+     |
| digium_phones       | 13.0.7.4   | Enabled                           | GPLv2      |
| directory           | 13.0.19.12 | Enabled                           | GPLv3+     |
| disa                | 13.0.6.12  | Enabled                           | AGPLv3+    |
| donotdisturb        | 13.0.3.1   | Enabled                           | GPLv3+     |
| dundicheck          | 2.11.0.3   | Enabled                           | GPLv3+     |
| endpoint            | 13.0.120   | Enabled                           | Commercial |
| extensionsettings   | 13.0.4     | Enabled                           | GPLv3+     |
| fax                 | 13.0.40.7  | Enabled                           | GPLv3+     |
| faxpro              | 13.0.42    | Enabled                           | Commercial |
| featurecodeadmin    | 13.0.6.4   | Enabled                           | GPLv3+     |
| findmefollow        | 13.0.38.13 | Enabled                           | GPLv3+     |
| firewall            | 13.0.57.1  | Enabled                           | AGPLv3+    |
| framework           | 13.0.197   | Enabled                           | GPLv2+     |
| fw_langpacks        | 12.0.7     | Enabled                           | GPLv3+     |
| hotelwakeup         | 13.0.17.2  | Enabled                           | GPLv2      |
| iaxsettings         | 13.0.6.6   | Enabled                           | AGPLv3     |
| infoservices        | 13.0.1.4   | Enabled                           | GPLv2+     |
| irc                 | 13.0.1     | Enabled                           | GPLv3+     |
| ivr                 | 13.0.27.18 | Enabled                           | GPLv3+     |
| languages           | 13.0.6.4   | Enabled                           | GPLv3+     |
| logfiles            | 13.0.10.5  | Enabled                           | GPLv3+     |
| manager             | 13.0.2.5   | Enabled                           | GPLv2+     |
| miscapps            | 13.0.3.1   | Enabled                           | GPLv3+     |
| miscdests           | 13.0.7     | Enabled                           | GPLv3+     |
| motif               | 13.0.4     | Enabled                           | GPLv3+     |
| music               | 13.0.22.7  | Enabled                           | GPLv3+     |
| outroutemsg         | 13.0.2.2   | Enabled                           | GPLv3+     |
| paging              | 13.0.26.13 | Enabled                           | GPLv3+     |
| pagingpro           | 13.0.19.12 | Enabled                           | Commercial |
| parking             | 13.0.19.11 | Enabled                           | GPLv3+     |
| parkpro             | 13.0.30.21 | Enabled                           | Commercial |
| pbdirectory         | 2.11.0.6   | Enabled                           | GPLv3+     |
| phonebook           | 13.0.6.4   | Enabled                           | GPLv3+     |
| phpagiconf          | 2.11.0.2   | Enabled                           | GPLv3+     |
| phpinfo             | 13.0.2     | Enabled                           | GPLv2+     |
| pinsets             | 13.0.13    | Enabled                           | GPLv3+     |
| pinsetspro          | 13.0.9.14  | Enabled                           | Commercial |
| pm2                 | 13.0.7.1   | Enabled                           | AGPLv3+    |
| presencestate       | 13.0.8.2   | Enabled                           | GPLv3+     |
| printextensions     | 13.0.3.2   | Enabled                           | GPLv3+     |
| queuemetrics        | 2.11.0.3   | Enabled                           | GPLv3+     |
| queueprio           | 13.0.6     | Enabled                           | GPLv3+     |
| queues              | 13.0.34.15 | Enabled                           | GPLv2+     |
| qxact_reports       | 13.0.15.17 | Enabled                           | Commercial |
| recording_report    | 13.0.24.10 | Enabled                           | Commercial |
| recordings          | 13.0.30.13 | Enabled                           | GPLv3+     |
| restapi             | 13.0.21.2  | Enabled                           | AGPLv3     |
| restart             |            | Not Installed (Locally available) | GPLv3+     |
| ringgroups          | 13.0.23.4  | Enabled                           | GPLv3+     |
| setcid              | 13.0.6.3   | Enabled                           | GPLv3+     |
| sipsettings         | 13.0.27.8  | Enabled                           | AGPLv3+    |
| sipstation          | 13.0.16    | Enabled                           | Commercial |
| sms                 | 13.0.12.5  | Enabled                           | Commercial |
| sng_mcu             | 13.0.5     | Enabled                           | Commercial |
| soundlang           | 13.0.26    | Enabled                           | GPLv3+     |
| speeddial           | 2.11.0.4   | Enabled                           | GPLv3+     |
| superfecta          | 13.0.4.7   | Enabled                           | GPLv2+     |
| sysadmin            | 13.0.87    | Enabled                           | Commercial |
| timeconditions      | 13.0.34.11 | Enabled                           | GPLv3+     |
| tts                 | 13.0.13    | Enabled                           | GPLv3+     |
| ttsengines          | 13.0.7.5   | Enabled                           | AGPLv3     |
| ucp                 | 13.0.42.6  | Enabled                           | AGPLv3+    |
| ucpnode             |            | Not Installed (Locally available) | Commercial |
| userman             | 13.0.76.43 | Enabled                           | AGPLv3+    |
| versionupgrade      | 13.0.1.5   | Enabled                           | Commercial |
| vmblast             | 13.0.11    | Enabled                           | GPLv3+     |
| vmnotify            | 13.0.22.4  | Enabled                           | Commercial |
| voicemail           | 13.0.59.3  | Enabled                           | GPLv3+     |
| voicemail_report    | 13.0.13.3  | Enabled                           | Commercial |
| vqplus              | 13.0.42.15 | Enabled                           | Commercial |
| weakpasswords       | 13.0.2     | Enabled                           | GPLv3+     |
| webcallback         | 13.0.11.5  | Enabled                           | Commercial |
| webrtc              | 13.0.32.9  | Enabled                           | GPLv3+     |
| xmpp                | 13.0.19.1  | Enabled                           | AGPLv3     |
+---------------------+------------+-----------------------------------+------------+

I tried running yum update, rebooting, updating modules, no luck.

Any help?

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P-Asserted-Identity with caller withheld "hidden" calls

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@davids1 wrote:

Hi,
I have read and tested quite a few posts about how to insert a PAID Header, P-Asserted-Identity, and although I have it working for normal calls, I can’t get it working for caller withheld “hidden” calls.

I am using a fully up to date New Freepbx installation
PBX Firmware:12.7.6-1904-1.sng7
PBX Service Pack:1.0.0.0
Current Asterisk Version: 13.22.0

My trunks are setup as PJSIP trunks with
Trust RPID/PAI = Yes
Send RPID/PAI = Send P-Asserted-Identity header

Using this configuration with a normal outbound call, it shows this following line in the SIP (INVITE) header (I have replaced the outbound CID number with CID_NUMBER and my server IP address with IPADDRESS for security)

P-Asserted-Identity: sip:CID_NUMBER@IPADDRESS

I do not get any P-Asserted-Identity lines in the Wireshark if I dial out CallerWithheld.
My CallerWithheld outbound route is setup with a prefix of 141 and has “hidden” as the RouteCID which overrides the extension = yes.
Now when I dial out there is no P-Asserted-Identity: in the wireshark dump.

From reading other posts I realise I need to create some custom code in Config Edit - extensions_custom.conf

This one seemed nice and basic but didnt work, it didnt inject any P-Asserted-Identity: in the wireshark dump.
[macro-dialout-trunk-predial-hook]
exten => s,1,NoOp(Adding P-Asserted-Identity)
exten => s,n,SipAddHeader(P-Asserted-Identity: sip:${CALLERID(num)})
exten => s,n(done),MacroExit()

I then moved into more complicated ones like this one - I have commented some lines out as “pai-custom:continue” caused an error.
[macro-dialout-trunk-predial-hook]
;Call With a Hidden Number
exten => s,1,NoOp(The caller id name is: ${CALLERID(name)})
exten => s,n,NoOp(The caller id number is: ${CALLERID(num)})
exten => s,n,NoOp(The ampuser is: ${AMPUSER})
exten => s,n,NoOp(The real caller id is: ${REALCALLERIDNUM})
exten => s,n,NoOp(The outbound cli is: ${DB(AMPUSER/${REALCALLERIDNUM}/outboundcid)})
;exten => s,n,Set(real_cli=${IF($[ “${DB(AMPUSER/${REALCALLERIDNUM}/outboundcid)}” = “” ]?+442xxCID_NUMBER:${DB(AMPUSER/${REALCALLERIDNUM}/outboundcid)})})
;exten => s,n,GotoIf($[ “${CALLERID(name)}” = “anonymous” | “${CALLERID(name)}” = “Anonymous” ]?pai-custom:continue)
;exten => s,n,SipAddHeader(P-Asserted-Identity: “${real_cli}” sip:${real_cli}@IPADDRESS\;user=phone)
;exten => s,n,SipAddHeader(Privacy: id)
exten => s,n,NoOp(Adding P-Asserted-Identity)
exten => s,n,SipAddHeader(P-Asserted-Identity: sip:${DB(AMPUSER/${REALCALLERIDNUM}/outboundcid)})
exten => s,n(done),MacroExit()
exten => s,n,MacroExit()

Using this context I can see the FreePBX logs showing the following lines, is something happening where the PAI info is being injected ok, but then being removed by another piece of code?

pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf("PJSIP/4000-00000070", "0?Set(CALLERID(all)=)") in new stack
pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf("PJSIP/4000-00000070", "1?Set(CALLERID(all)=02xxCID_NUMBER)") in new stack
pbx.c: Executing [s@macro-outbound-callerid:23] ExecIf("PJSIP/4000-00000070", "1?Set(CALLERID(all)=hidden)") in new stack
pbx.c: Executing [s@macro-outbound-callerid:24] ExecIf("PJSIP/4000-00000070", "1?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
pbx.c: Executing [s@macro-outbound-callerid:25] ExecIf("PJSIP/4000-00000070", "1?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
pbx.c: Executing [s@macro-outbound-callerid:26] Set("PJSIP/4000-00000070", "CDR(outbound_cnum)=") in new stack
pbx.c: Executing [s@macro-outbound-callerid:27] Set("PJSIP/4000-00000070", "CDR(outbound_cnam)=hidden") in new stack
pbx.c: Executing [s@macro-dialout-trunk:14] GosubIf("PJSIP/4000-00000070", "0?sub-flp-4,s,1()") in new stack
pbx.c: Executing [s@macro-dialout-trunk:15] Set("PJSIP/4000-00000070", "OUTNUM=07xxMYMOBILE") in new stack
pbx.c: Executing [s@macro-dialout-trunk:16] Set("PJSIP/4000-00000070", "custom=PJSIP") in new stack
pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf("PJSIP/4000-00000070", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
pbx.c: Executing [s@macro-dialout-trunk:18] ExecIf("PJSIP/4000-00000070", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
pbx.c: Executing [s@macro-dialout-trunk:19] Macro("PJSIP/4000-00000070", "dialout-trunk-predial-hook,") in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] NoOp("PJSIP/4000-00000070", "The caller id name is: hidden") in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:2] NoOp("PJSIP/4000-00000070", "The caller id number is: ") in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:3] NoOp("PJSIP/4000-00000070", "The ampuser is: 4000") in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:4] NoOp("PJSIP/4000-00000070", "The real caller id is: 4000") in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:5] NoOp("PJSIP/4000-00000070", "The outbound cli is: 02xxCID_NUMBER") in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:6] NoOp("PJSIP/4000-00000070", "Adding P-Asserted-Identity") in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:7] SIPAddHeader("PJSIP/4000-00000070", "P-Asserted-Identity: sip:02xxCID_NUMBER") in new stack
pbx.c: Executing [s@macro-dialout-trunk-predial-hook:8] MacroExit("PJSIP/4000-00000070", "") in new stack
pbx.c: Executing [s@macro-dialout-trunk:20] GotoIf("PJSIP/4000-00000070", "0?skipcrm") in new stack
pbx.c: Executing [s@macro-dialout-trunk:21] Set("PJSIP/4000-00000070", "__CRM_DIRECTION=OUTBOUND") in new stack
pbx.c: Executing [s@macro-dialout-trunk:22] Set("PJSIP/4000-00000070", "__CRM_DESTINATION=07xxMYMOBILE") in new stack
pbx.c: Executing [s@macro-dialout-trunk:23] Set("PJSIP/4000-00000070", "__CRM_SOURCE=4000") in new stack
pbx.c: Executing [s@macro-dialout-trunk:24] AGI("PJSIP/4000-00000070", "sangomacrm.agi") in new stack
res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
res_agi.c: <PJSIP/4000-00000070>AGI Script sangomacrm.agi completed, returning 0
pbx.c: Executing [s@macro-dialout-trunk:25] Set("PJSIP/4000-00000070", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
pbx.c: Executing [s@macro-dialout-trunk:26] NoOp("PJSIP/4000-00000070", "CRM Finished") in new stack
pbx.c: Executing [s@macro-dialout-trunk:27] GotoIf("PJSIP/4000-00000070", "0?bypass,1") in new stack
pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf("PJSIP/4000-00000070", "1?Set(CONNECTEDLINE(num,i)=07xxMYMOBILE)") in new stack
pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf("PJSIP/4000-00000070", "0?Set(CONNECTEDLINE(name,i)=CID:)") in new stack
pbx.c: Executing [s@macro-dialout-trunk:30] ExecIf("PJSIP/4000-00000070", "1?Set(CONNECTEDLINE(name,i)=CID:(Hidden))") in new stack
pbx.c: Executing [s@macro-dialout-trunk:31] GotoIf("PJSIP/4000-00000070", "0?customtrunk") in new stack

I was hoping to do this myself without having to bother the community but hitting my head against a wall and our main supplier is going to block all incorrect PAI calls soon so I need to get this fixed asap.
Hopefully its something quick you can help with,
Thanks
D

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Unconditional Call Forwarding conflicting with FMFM

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@mulderlr wrote:

Similar to this thread, Call Forward Unconditional (*72) reverts to * VM

I am having an issue with FPBX 13, 10.13.66-21. When users use UCP (13.0.42.6) to change an extensions settings to call forward unconditional enabled to an outside number (eg 8775551234), the forwarding tries for 2 seconds, then tried the FMFM list. According to the documentation, call forward unconditional should be just that - unconditional. It should not be 1. timing out after 2 seconds, and 2, not trying FM/FM after 2 seconds. I am not sure why these two features conflict, but this seems to be a new development for the users of this system. If FMFM is disabled, the unconditional CF seems to work, but it should work correctly regardless IMHO.

Any insight as to whether this is normal behavior, a bug, a setting we are missing or any other help would be appreciated.

Thanks!

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PBX sending RTP to the LAN IP of remote phone

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@mvogel4949 wrote:

I have remote phones registered to a FreePBX 14 system. Calls can be made but there is no audio. I ran a packet capture from the FreePBX and see that it is trying to send audio to the LAN IP of the remote phone. I have NAT set to Yes in the Advanced tab of the extension and NAT set to yes in the Asterisk Chan SIP Settings

66.188.45.2 - WAN IP of Remote phone
172.16.8.100 - LAN IP of FreePBX
192.168.0.160 - LAN IP of Remote Phone

image

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SPA3102 does not detect disconnect tone

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@jamesg224 wrote:

Hi,
This might be in the wrong topic, so feel free to move.

I’m having a bit of trouble trying to get my Linksys SPA3102 to detect the PSTN disconnect tone that is played when the incoming caller hangs up. Everything else on the device is working perfectly. It is connected to a TalkTalk (BT) landline in the UK. I’ve been googling around for the past week or so and tried most of the things that I have found.

I’ve recorded the disconnect tone and measured this at 395Hz at -20, changed the string but no luck. I have been able to get hold of a patton box to also test with and if I set it to detect the busy tone, with the settings below, it works. I have also tried inputting the busy tone string into the linksys but also no luck.
I think I’m missing something obvious here, so any help is appreciated.

Linksys config:

Patton busy tone config:
1: play 375ms 400Hz 0dm
2: pause 375ms

Disconnect tone recording:
https://drive.google.com/open?id=1B3eRsTFmNZAk4jUomQzKPued2QsuT0dW

If anyone can point me in the right direction with this then that would be great. I’m starting to think I have a faulty unit.

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Active Directory and UCP Authentication

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@benjaminrp wrote:

So strange issue I’ve been trying to figure out. We have an active directory setup and have configured FreePBX to use AD for its user database. It pulls all the users and their information, however none of the users can log in to UCP. All of them get authenication error and when I check the freepbx_security log, this is what I see:

Authentication failure for benjaminatdomaincom from IP
Authentication failure for benjaminatrdomaincom from IP
Authentication failure for bjonesatdomaincom from IP
Authentication failure for bjonesatdomaincom from IP
Authentication failure for benjaminatdomaincom from IP
Authentication failure for benjaminatdomaincom from IP

Username and IP address changed, but it lists the username I’m trying along with our IP address.

I have confirmed that the user and group the user is in has enabled UCP.

For the username, I specified to use the ‘mail’ attribute from AD.

Any help would be most appreciated!

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No audio with remote extensions

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@yois wrote:

Fresh install of SNG7-1904 and having the same problem with Asterisk 13.22 and FreePBX 14.0.13.4.

Restarting Asterisk helps for a few minutes, then the problem returns. This was a box that had settings imported from a 10.13.66 box using the conversion tool. I used the Asterisk CLI to verify all settings both on the extension and transport there don’t seem to be any differences.

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Listen to the called party channel before answer

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@dux wrote:

Hello, everyone.
How can I make outbound calls and listen to the called party channel before answer? Now I listened to rings, but, if a call the same number from a regular phone, it, for instance, plays music or says that the subscriber is unavailable.

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Adding a SIP line instead of a SIP Trunk

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@trixie_no5 wrote:

I’ve been provided with SIP account credentials, server, username and secret only. How do I set this up in FreePBX?

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REST API on FreePBX 14

FXO adapter


Maximum simultaneous calls to/from one extension

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@ccts wrote:

I have a Grandstream GXW4216 ata/fxs gateway setup feeding an analog phone system. I presently only have 1 sip account programmed into it & 12 fxs ports using that 1 sip account, that sip account comes off my freepbx as an extension. I can have multiple calls simultaneously thru that extension, as I’ve tested it up to 4 simultaneous calls, but I’m wondering if there’s some limit somewhere either on the ata or in freepbx, as the client reported today receiving a max simultaneous calls reached message when trying to make an outbound call.

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[Solved] Do Not Disturb Confirmation - Change Sound?

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@drummerjoe wrote:

I remember reading somewhere that you can change the Do Not Disturb (*76) confirmation to just a beep (or maybe silent), instead of “Do Not Disturb Activated/Deactivated”. Can you direct me on how to make this change?

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Asterisk Manager Users

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@AnthKay92 wrote:

Is it possible to create an Asterisk Manager User based off of an LDAP user?

I cannot see the option within User Management or Asterisk Management Users.

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Unavailable extension busy tone

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@apel wrote:

A softphone extension works perfectly well but when switched off the caller get a busy tone.
Asterisk status is correct Endpoint: 226/226 Unavailable

If I set a destination unavailable announcement it never plays
if I set a destination busy announcement it plays.

Any ideas how do I fix this please.

FreePBX 13.0.197
Extension 226 This device uses PJSIP technology listening on Port 5060 (UDP)

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Using +E164 numbering format in all SIP headers

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@AnthKay92 wrote:

I have established a new SIP trunk with my SIP provider.

It’s a PJSIP trunk with no out of the ordinary setup. Calls are working both inbound and outbound, however when I dial outbound, I am getting a +1 (US) number show on my mobile phone.

I have set the CallerID on my SIP trunk to the number that should show, however it is still showing as +1

My SIP provider has replied saying:
“Please verify that you are using the +E164 numbering format in all SIP Headers”
“The +E164 is the standard format in Enterprise SIP”

Can anyone point me in the right direction to ensure I am using this format? where would this be set in the PJSIP trunk settings?

Thanks

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