@nickb wrote:
Can the Pinset Pro module bring pinset functionality as a gateway device for other functions?
For example, being able to have a set of pins function as a gateway for the caller to join a specific queue e.g. as a way of identifying VIP callers.
Posts: 1
Participants: 1
@jhayes wrote:
We are using a remote CDR database and the UCP module controls are no longer available for downloading recorded calls. We are running asterisk 13.17 and FreePBX 14.0.1.4. The only error on the page is a 403 for https://DOMAIN//ucp/assets/js/compiled/main/bootstrap-toggle.min.js.map
Posts: 1
Participants: 1
@ccampbell wrote:
I found this page that describes all the variables that are used in Endpoint Manager.
https://wiki.freepbx.org/display/FPG/Variables-Version+12.X+or+greater
But how do I edit the dialplan variable to make changes to the dialplan?
Thank you,
Court
Posts: 4
Participants: 2
@cfapress wrote:
I've been wracking my brain as to why multicast paging is not functioning for me. We need to page 60+ phones. The default paging was overloading the CPU on my machine so we opted for the commercial module for multicast paging. It installed properly and I've configured a test paging group.
The idea is when calling x1275 it goes to a paging group with multicast added. I have no phones selected at all. I figured there's no need to define which phones to apply this to as the phones themselves are configured to listen for multicast broadcasts on the specified IP and Port.
I've performed some multicast testing on my network and switches to assure nothing was getting blocked by a switch. The software I used is by Valcom. It's free and their testing instructions are easy to follow. I've successfully tested multicast between two computers using the same address and port I defined in my FreePBX paging group.
Below you'll see a dump from Asterisk that shows the Multicast page appears to be functioning. However, nothing comes across on our GrandStream phones. The page comes from extension 1210.
There are several error lines, which I've highlighted in the dump below. Though, I'm unsure if they're significant to my issue.
Also, using a PC that's listening for the same multicast as the phones - no packets arrive there either.
I'm wondering if something about the network card driver on my FreePBX box is preventing a multicast from going out. Is that possible? I figured multicast was a universal thing.
Thanks in advance for any tips to get this working.
=======================
login as: root
Access denied
root@10.2.1.227's password:
Last login: Fri Oct 13 13:55:51 2017
| ___| __ ___ ___| _ \| __ ) \/ / | |_ | '__/ _ \/ _ \ |_) | _ \\ / | _|| | | __/ __/ __/| |_) / \ |_| |_| \___|\___|_| |____/_/\_\ NOTICE! You have 7 notifications! Please log into the UI to see them! Current Network Configuration +-----------+-------------------+--------------------------+ | Interface | MAC Address | IP Addresses | +-----------+-------------------+--------------------------+ | eth1 | 00:19:DB:AF:4B:2A | 10.2.1.227 | | | | fe80::219:dbff:feaf:4b2a | | eth1.20 | 00:19:DB:AF:4B:2A | 10.2.2.227 | | | | fe80::219:dbff:feaf:4b2a | | eth2 | 00:60:08:C3:85:4A | | | eth2.20 | 00:60:08:C3:85:4A | | +-----------+-------------------+--------------------------+ Please note most tasks should be handled through the GUI. You can access the GUI by typing one of the above IPs in to your web browser. For support please visit: http://www.freepbx.org/support-and-professional-services [root@cfafreepbx-sbcc ~]# asterisk -rvvvv Asterisk 13.16.0, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 13.16.0 currently running on cfafreepbx-sbcc (pid = 26263) cfafreepbx-sbcc*CLI> cfafreepbx-sbcc*CLI> == Setting global variable 'SIPDOMAIN' to '10.2.2.227' -- Executing [1275@from-internal:1] Goto("PJSIP/1210-0000190d", "app-pagegroups,1275,1") in new stack -- Goto (app-pagegroups,1275,1) -- Executing [1275@app-pagegroups:1] Set("PJSIP/1210-0000190d", "MCAST=224.0.0.1:7001") in new stack -- Executing [1275@app-pagegroups:2] Macro("PJSIP/1210-0000190d", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("PJSIP/1210-0000190d", "TOUCH_MONITOR=1507918462.11835") in new stack -- Executing [s@macro-user-callerid:2] Set("PJSIP/1210-0000190d", "AMPUSER=1210") in new stack -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/1210-0000190d", "0?report") in new stack -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/1210-0000190d", "1?Set(REALCALLERIDNUM=1210)") in new stack -- Executing [s@macro-user-callerid:5] Set("PJSIP/1210-0000190d", "AMPUSER=1210") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/1210-0000190d", "0?limit") in new stack -- Executing [s@macro-user-callerid:7] Set("PJSIP/1210-0000190d", "AMPUSERCIDNAME=Basement") in new stack -- Executing [s@macro-user-callerid:8] GotoIf("PJSIP/1210-0000190d", "0?report") in new stack -- Executing [s@macro-user-callerid:9] Set("PJSIP/1210-0000190d", "AMPUSERCID=1210") in new stack -- Executing [s@macro-user-callerid:10] Set("PJSIP/1210-0000190d", "__DIAL_OPTIONS=Ttr") in new stack -- Executing [s@macro-user-callerid:11] Set("PJSIP/1210-0000190d", "CALLERID(all)="Basement" <1210>") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("PJSIP/1210-0000190d", "0?limit") in new stack -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/1210-0000190d", "0?Set(GROUP(concurrency_limit)=1210)") in new stack -- Executing [s@macro-user-callerid:14] ExecIf("PJSIP/1210-0000190d", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:15] GotoIf("PJSIP/1210-0000190d", "0?continue") in new stack -- Executing [s@macro-user-callerid:16] ExecIf("PJSIP/1210-0000190d", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack -- Executing [s@macro-user-callerid:17] Set("PJSIP/1210-0000190d", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:18] GotoIf("PJSIP/1210-0000190d", "1?continue") in new stack -- Goto (macro-user-callerid,s,29) -- Executing [s@macro-user-callerid:29] Set("PJSIP/1210-0000190d", "CALLERID(number)=1210") in new stack -- Executing [s@macro-user-callerid:30] Set("PJSIP/1210-0000190d", "CALLERID(name)=Basement") in new stack -- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/1210-0000190d", "0?cnum") in new stack -- Executing [s@macro-user-callerid:32] Set("PJSIP/1210-0000190d", "CDR(cnam)=Basement") in new stack -- Executing [s@macro-user-callerid:33] Set("PJSIP/1210-0000190d", "CDR(cnum)=1210") in new stack -- Executing [s@macro-user-callerid:34] Set("PJSIP/1210-0000190d", "CHANNEL(language)=en") in new stack -- Executing [1275@app-pagegroups:3] Set("PJSIP/1210-0000190d", "_PAGEGROUP=1275") in new stack -- Executing [1275@app-pagegroups:4] GotoIf("PJSIP/1210-0000190d", "1?:busy") in new stack -- Executing [1275@app-pagegroups:5] Set("PJSIP/1210-0000190d", "DEVICE_STATE(Custom:PAGE1275)=INUSE") in new stack -- Executing [1275@app-pagegroups:6] Gosub("PJSIP/1210-0000190d", "app-paging,ssetup,1()") in new stack -- Executing [ssetup@app-paging:1] Set("PJSIP/1210-0000190d", "_SIPURI=") in new stack -- Executing [ssetup@app-paging:2] Set("PJSIP/1210-0000190d", "_ALERTINFO=Ring Answer") in new stack -- Executing [ssetup@app-paging:3] Set("PJSIP/1210-0000190d", "_CALLINFO=<uri>;answer-after=0") in new stack -- Executing [ssetup@app-paging:4] Set("PJSIP/1210-0000190d", "_SIPURI=intercom=true") in new stack -- Executing [ssetup@app-paging:5] Set("PJSIP/1210-0000190d", "_DTIME=5") in new stack -- Executing [ssetup@app-paging:6] Set("PJSIP/1210-0000190d", "_ANSWERMACRO=") in new stack -- Executing [ssetup@app-paging:7] Set("PJSIP/1210-0000190d", "PAGE_CONF=1507918462302") in new stack -- Executing [ssetup@app-paging:8] Return("PJSIP/1210-0000190d", "") in new stack -- Executing [1275@app-pagegroups:7] Set("PJSIP/1210-0000190d", "PAGEMODE=PAGE") in new stack -- Executing [1275@app-pagegroups:8] Set("PJSIP/1210-0000190d", "PAGE_MEMBERS=") in new stack -- Executing [1275@app-pagegroups:9] Set("PJSIP/1210-0000190d", "PAGE_CONF_OPTS=") in new stack -- Executing [1275@app-pagegroups:10] Set("PJSIP/1210-0000190d", "ANNOUNCEMENT=beep") in new stack -- Executing [1275@app-pagegroups:11] Set("PJSIP/1210-0000190d", "STREAM=NONE") in new stack -- Executing [1275@app-pagegroups:12] AGI("PJSIP/1210-0000190d", "page.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/page.agi -- Executing [s@app-page-stream:1] Wait("Local/s@app-page-stream-00000595;2", "1") in new stack -- Called s@app-page-stream -- Called PAGErtp@app-paging/n -- Executing [PAGErtp@app-paging:1] Macro("Local/PAGErtp@app-paging-00000596;2", "autoanswer,rtp") in new stack -- Executing [s@macro-autoanswer:1] GotoIf("Local/PAGErtp@app-paging-00000596;2", "1?knowndial") in new stack -- Goto (macro-autoanswer,s,19) -- Executing [s@macro-autoanswer:19] Set("Local/PAGErtp@app-paging-00000596;2", "DIAL=MulticastRTP/basic/224.0.0.1:7001") in new stack -- Executing [s@macro-autoanswer:20] ExecIf("Local/PAGErtp@app-paging-00000596;2", "0?Set(DIAL=DAHDIticastRTP/basic/224.0.0.1:7001)") in new stack -- Executing [s@macro-autoanswer:21] GotoIf("Local/PAGErtp@app-paging-00000596;2", "0?macro") in new stack -- Executing [s@macro-autoanswer:22] GotoIf("Local/PAGErtp@app-paging-00000596;2", "0?pjsipua") in new stack -- Executing [s@macro-autoanswer:23] Set("Local/PAGErtp@app-paging-00000596;2", "USERAGENT=") in new stack -- Executing [s@macro-autoanswer:24] Goto("Local/PAGErtp@app-paging-00000596;2", "uafin") in new stack -- Goto (macro-autoanswer,s,28) -- Executing [s@macro-autoanswer:28] ExecIf("Local/PAGErtp@app-paging-00000596;2", "1?Set(USERAGENT=rtp)") in new stack -- <PJSIP/1210-0000190d>AGI Script page.agi completed, returning 0 -- Executing [s@macro-autoanswer:29] ExecIf("Local/PAGErtp@app-paging-00000596;2", "0?Set(PAGE_VOL=;volume=)") in new stack -- Executing [s@macro-autoanswer:30] ExecIf("Local/PAGErtp@app-paging-00000596;2", "0?Set(CALLINFO=<sip:broadworks.net>;answer-after=0)") in new stack -- Executing [s@macro-autoanswer:31] ExecIf("Local/PAGErtp@app-paging-00000596;2", "0?Set(ALERTINFO=Intercom)") in new stack -- Executing [s@macro-autoanswer:32] ExecIf("Local/PAGErtp@app-paging-00000596;2", "0?Set(ALERTINFO=info=Auto Answer)") in new stack -- Executing [s@macro-autoanswer:33] ExecIf("Local/PAGErtp@app-paging-00000596;2", "0?Set(ALERTINFO=ring-answer)") in new stack -- Executing [s@macro-autoanswer:34] ExecIf("Local/PAGErtp@app-paging-00000596;2", "0?Set(ALERTINFO=<http://www.sangoma.com>;info=external)") in new stack -- Executing [s@macro-autoanswer:35] ExecIf("Local/PAGErtp@app-paging-00000596;2", "0?Set(ALERTINFO=<http://example.com>;info=alert-autoanswer)") in new stack -- Executing [s@macro-autoanswer:36] ExecIf("Local/PAGErtp@app-paging-00000596;2", "1?Set(__SIP_URI_OPTIONS=intercom=true)") in new stack -- Executing [PAGErtp@app-paging:2] NoOp("Local/PAGErtp@app-paging-00000596;2", "") in new stack -- Executing [PAGErtp@app-paging:3] GotoIf("Local/PAGErtp@app-paging-00000596;2", "1?doptions") in new stack -- Goto (app-paging,PAGErtp,6) -- Executing [PAGErtp@app-paging:6] ExecIf("Local/PAGErtp@app-paging-00000596;2", "1?Set(_DOPTIONS=b(autoanswer^s^1(Ring Answer,<uri>;answer-after=0)))") in new stack -- Executing [PAGErtp@app-paging:7] Dial("Local/PAGErtp@app-paging-00000596;2", "MulticastRTP/basic/224.0.0.1:7001,5,A(beep)b(autoanswer^s^1(Ring Answer,<uri>;answer-after=0))") in new stack -- Executing [1275@app-pagegroups:13] Set("PJSIP/1210-0000190d", "CONFBRIDGE(user,template)=page_user_duplex") in new stack -- Executing [1275@app-pagegroups:14] Set("PJSIP/1210-0000190d", "CONFBRIDGE(user,admin)=yes") in new stack -- Executing [1275@app-pagegroups:15] Set("PJSIP/1210-0000190d", "CONFBRIDGE(user,marked)=yes") in new stack -- Executing [1275@app-pagegroups:16] Answer("PJSIP/1210-0000190d", "") in new stack -- MulticastRTP/0x7f44b90f60f0 Internal Gosub(autoanswer,s,1(Ring Answer,<uri>;answer-after=0)) start -- Executing [s@autoanswer:1] GosubIf("MulticastRTP/0x7f44b90f60f0", "1?func-set-sipheader,s,1(Alert-Info,Ring Answer)") in new stack -- Executing [s@func-set-sipheader:1] NoOp("MulticastRTP/0x7f44b90f60f0", "Sip Add Header function called. Adding Alert-Info = Ring Answer") in new stack -- Executing [s@func-set-sipheader:2] Set("MulticastRTP/0x7f44b90f60f0", "HASH(_SIPHEADERS,Alert-Info)=Ring Answer") in new stack -- Executing [s@func-set-sipheader:3] Return("MulticastRTP/0x7f44b90f60f0", "") in new stack -- Executing [s@autoanswer:2] GosubIf("MulticastRTP/0x7f44b90f60f0", "1?func-set-sipheader,s,1(Call-Info,<uri>;answer-after=0)") in new stack -- Executing [s@func-set-sipheader:1] NoOp("MulticastRTP/0x7f44b90f60f0", "Sip Add Header function called. Adding Call-Info = <uri>;answer-after=0") in new stack -- Executing [s@func-set-sipheader:2] Set("MulticastRTP/0x7f44b90f60f0", "HASH(_SIPHEADERS,Call-Info)=<uri>;answer-after=0") in new stack -- Executing [s@func-set-sipheader:3] Return("MulticastRTP/0x7f44b90f60f0", "") in new stack -- Executing [s@autoanswer:3] Gosub("MulticastRTP/0x7f44b90f60f0", "func-apply-sipheaders,s,1()") in new stack -- Executing [s@func-apply-sipheaders:1] ExecIf("MulticastRTP/0x7f44b90f60f0", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack -- Executing [s@func-apply-sipheaders:2] NoOp("MulticastRTP/0x7f44b90f60f0", "Applying SIP Headers to channel") in new stack -- Executing [s@func-apply-sipheaders:3] Set("MulticastRTP/0x7f44b90f60f0", "SIPHEADERKEYS=Call-Info,Alert-Info") in new stack -- Executing [s@func-apply-sipheaders:4] While("MulticastRTP/0x7f44b90f60f0", "1") in new stack -- Executing [s@func-apply-sipheaders:5] Set("MulticastRTP/0x7f44b90f60f0", "sipheader=<uri>;answer-after=0") in new stack -- Executing [s@func-apply-sipheaders:6] SIPAddHeader("MulticastRTP/0x7f44b90f60f0", "Call-Info: <uri>;answer-after=0") in new stack -- Executing [s@func-apply-sipheaders:7] Set("MulticastRTP/0x7f44b90f60f0", "PJSIP_HEADER(add,Call-Info)=<uri>;answer-after=0") in new stack [2017-10-13 14:14:22] ERROR[26026][C-000018b0]: res_pjsip_header_funcs.c:520 func_write_header: This function requires a PJSIP channel. -- Executing [s@func-apply-sipheaders:8] EndWhile("MulticastRTP/0x7f44b90f60f0", "") in new stack -- Executing [s@func-apply-sipheaders:4] While("MulticastRTP/0x7f44b90f60f0", "1") in new stack -- Executing [s@func-apply-sipheaders:5] Set("MulticastRTP/0x7f44b90f60f0", "sipheader=Ring Answer") in new stack -- Executing [s@func-apply-sipheaders:6] SIPAddHeader("MulticastRTP/0x7f44b90f60f0", "Alert-Info: Ring Answer") in new stack -- Executing [s@func-apply-sipheaders:7] Set("MulticastRTP/0x7f44b90f60f0", "PJSIP_HEADER(add,Alert-Info)=Ring Answer") in new stack [2017-10-13 14:14:22] ERROR[26026][C-000018b0]: res_pjsip_header_funcs.c:520 func_write_header: This function requires a PJSIP channel. -- Executing [s@func-apply-sipheaders:8] EndWhile("MulticastRTP/0x7f44b90f60f0", "") in new stack -- Executing [s@func-apply-sipheaders:4] While("MulticastRTP/0x7f44b90f60f0", "0") in new stack -- Executing [s@func-apply-sipheaders:9] Return("MulticastRTP/0x7f44b90f60f0", "") in new stack -- Executing [s@autoanswer:4] Return("MulticastRTP/0x7f44b90f60f0", "") in new stack == Spawn extension (default, PAGErtp, 1) exited non-zero on 'MulticastRTP/0x7f44b90f60f0' -- MulticastRTP/0x7f44b90f60f0 Internal Gosub(autoanswer,s,1(Ring Answer,<uri>;answer-after=0)) complete GOSUB_RETVAL= -- Called MulticastRTP/basic/224.0.0.1:7001 -- MulticastRTP/0x7f44b90f60f0 answered Local/PAGErtp@app-paging-00000596;2 -- <MulticastRTP/0x7f44b90f60f0> Playing 'beep.ulaw' (language 'en') > 0x7f44b91fe220 -- Probation passed - setting RTP source address to 10.2.2.102:2248 -- Executing [1275@app-pagegroups:17] ConfBridge("PJSIP/1210-0000190d", "1507918462302,,,admin_menu") in new stack -- Channel PJSIP/1210-0000190d joined 'softmix' base-bridge <0bdc6d0e-a5ca-41ff-9487-12de006e15af> -- Channel CBAnn/1507918462302-00000597;2 joined 'softmix' base-bridge <0bdc6d0e-a5ca-41ff-9487-12de006e15af> -- Local/PAGErtp@app-paging-00000596;1 answered > Launching ConfBridge(1507918462302,,,user_menu) on Local/PAGErtp@app-paging-00000596;1 -- Channel Local/PAGErtp@app-paging-00000596;1 joined 'softmix' base-bridge <0bdc6d0e-a5ca-41ff-9487-12de006e15af> -- Channel MulticastRTP/0x7f44b90f60f0 joined 'simple_bridge' basic-bridge <e2d4c651-10bc-4157-b209-de812682d6c9> -- Channel Local/PAGErtp@app-paging-00000596;2 joined 'simple_bridge' basic-bridge <e2d4c651-10bc-4157-b209-de812682d6c9> -- Executing [s@app-page-stream:2] Answer("Local/s@app-page-stream-00000595;2", "") in new stack -- Local/s@app-page-stream-00000595;1 answered > Launching Wait(5) on Local/s@app-page-stream-00000595;1 -- Executing [s@app-page-stream:3] Set("Local/s@app-page-stream-00000595;2", "CONFBRIDGE(user,template)=page_user_duplex") in new stack -- Executing [s@app-page-stream:4] Set("Local/s@app-page-stream-00000595;2", "CONFBRIDGE(user,marked)=yes") in new stack -- Executing [s@app-page-stream:5] ConfBridge("Local/s@app-page-stream-00000595;2", "1507918462302,,,") in new stack -- Channel Local/s@app-page-stream-00000595;2 joined 'softmix' base-bridge <0bdc6d0e-a5ca-41ff-9487-12de006e15af> -- Channel PJSIP/1210-0000190d left 'softmix' base-bridge <0bdc6d0e-a5ca-41ff-9487-12de006e15af> -- Executing [h@app-pagegroups:1] ExecIf("PJSIP/1210-0000190d", "1?Set(DEVICE_STATE(Custom:PAGE1275)=NOT_INUSE)") in new stack -- Executing [h@app-pagegroups:2] GosubIf("PJSIP/1210-0000190d", "0?record-page,1()") in new stack -- Executing [h@app-pagegroups:3] ExecIf("PJSIP/1210-0000190d", "0?System(rm .sln)") in new stack -- Executing [h@app-pagegroups:4] ExecIf("PJSIP/1210-0000190d", "0?System(rm -f /var/spool/asterisk/outgoing/)") in new stack -- Channel Local/s@app-page-stream-00000595;2 left 'softmix' base-bridge <0bdc6d0e-a5ca-41ff-9487-12de006e15af> -- Channel Local/PAGErtp@app-paging-00000596;1 left 'softmix' base-bridge <0bdc6d0e-a5ca-41ff-9487-12de006e15af> -- Channel CBAnn/1507918462302-00000597;2 left 'softmix' base-bridge <0bdc6d0e-a5ca-41ff-9487-12de006e15af> -- Channel Local/PAGErtp@app-paging-00000596;2 left 'simple_bridge' basic-bridge <e2d4c651-10bc-4157-b209-de812682d6c9> -- Channel MulticastRTP/0x7f44b90f60f0 left 'simple_bridge' basic-bridge <e2d4c651-10bc-4157-b209-de812682d6c9> == Spawn extension (app-paging, PAGErtp, 7) exited non-zero on 'Local/PAGErtp@app-paging-00000596;2' cfafreepbx-sbcc*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups [root@cfafreepbx-sbcc ~]#
Posts: 2
Participants: 1
@eknudhol wrote:
First some basics:
* I am recovering a VM host server (ESXi 5.5) that would not start after IRMA
* The server, and guest VM's, was working before IRMA and was not damaged
* The issue is due to a failed motherboard
* I have backups that i recovered to a new VM HostI want to recover the the VM since this instance has several IVR's, IAX2 and Extns etc.
But of course the FreePBX guest vm would not start and I couldn't wait so I:
* setup a new instance of FreePBX
* did HW reset to migrate modules to new install, and that worked well
* setup the basic extns and features to get by
* this temporary deployment now has my old deployment ID from the failed VMI have just been able to recover the VM and it started normally, and I activated a new deployment for this server (I did the HW reset earlier to move to my temp solution) and now has new deployment ID and I purchased new modules.
BUT
When I try to map extensions in EPM, they do not show in the mapping list (EPM->Extension Mapping->Add Extension). Since I migrated the deployment to the temp install, I decided to purchase new modules, as several had expired on the old deployment. My thoughts are that since I had PURCHASED EPM on the old deployment and didn't for the recovered VM (only using Sangoma phones) that EPM may be corrupted or "confused".Is there something I can do to "refresh" EPM? (i.e. r&r the EPM module?)
Recreated multiple extensions and they do not show up in EPM.Thoughts?
Thanks in advance for your help!
Posts: 2
Participants: 1
@freak wrote:
I'm using FreePBX 13.0.192.18. I created a pjsip extension and increased max contacts to 2. I connected a yealink t48G and set it up in extensions under other. Set brand model template... Works fine. Next I set an incom 1000G. It picks up it's config file because the phone extension is set on the phone screen. I cannot make a call however i get the following in the logs... any ideas?
[2017-10-18 13:10:10] NOTICE[3222]: chan_sip.c:28600 handle_request_register: Registration from '"hidden-430" <sip:430@10.0.2.3>' failed for '10.0.2.232:5061' - Wrong password [2017-10-18 13:10:10] NOTICE[3222]: chan_sip.c:28600 handle_request_register: Registration from '"hidden-430" <sip:430@10.0.2.3>' failed for '10.0.2.232:5061' - Wrong password
Posts: 17
Participants: 3
@janknut wrote:
Working in a new FreePBX 14 / Asterisk 13 installed from the July distro. I believe I've followed all the steps / instructions properly but I cannot get any of the phones (all Yealink T46G) to provision. I've created extensions, assigned the extensions to the phones (they all appeared in the 'scan network' EPM UI), and they all subsequently appear in the Exention mapping EPM UI. I used a web browser to connect to the Yealink phone(s), entered the IP of the FreePBX as the Auto Provision Server URL and "Confirmed" i.e. saved the setting. When I then click 'Auto provision now' the Yealink Web UI shows the Configuration updating…” message for around 30-45 seconds, then returns to the auto-provision page. The phone itself shows no indication that anything is happening during (or after) this process. The Yealink web UI shows no changes to settings, e.g. Dsskeys etc.
Some troubleshooting I've done. all resulting in the same (non) result.:
Tried both TFTP and HTTP as the provision server protocol (and rebuilt the config files after making these changes)
Tried specifying the port with the IP in the Yealink GUI (e.g. 192.168.x.120:83 for HTTP)
My original extension were configured as PJSIP, so I deleted one mapping and tried mapping to a new Chan_SIP extension.It seems that (likely many)other people have been able to provision T48G's via EPM, so not sure what the problem is or how to further troubleshoot. I've checked and re-checked my settings many times over against the wiki etc.
These phones are running the latest firmware (28.82.0.20) and I noticed that EPMs latest available firmware is version 28.81.0.70 (as such, I've left the template's "Firmware Version" set to the default (i.e. None). Maybe I need to downgrade the firmware?
Very frustrating experience because we puchased the EPM as it was supposed to be a huge time saver. Instead I've spent hours working on this with no results and I could have manually provisioned these 11 phones and been up and running already. But would like to get this working anyway as I anticipate it will make configuration changes and so forth easier / better in the long run.
Any suggestions, ideas appreciated.
Posts: 3
Participants: 2
@MC1 wrote:
Hello
Before upgrading to FreePBX 14, the RMS Management module was running as a trial. After the upgrade, I needed to load in the node key. However, under the RMS Managment > Status tab I find the following:
Agent = running
Connection = Offline
Management Connection = Offline
Does this mean that the trial period is over and the RMS Management tool can no longer run and it has now become a paid service?
Thanks in advance for your help
Posts: 1
Participants: 1
@avayax wrote:
I have extensions on a remote system that do not send CallerID, they are anonymous.
When a call comes in from one of them to one of my ring groups, it gets denied with cos.agi: Starting Class Of Service checks__cos.agi: CallerID not Parseable - received Anonymous - exiting with DENY
Using the COS module.0x7f2af411f760 -- Probation passed - setting RTP source address to 10.1.1.183:10910
[2017-10-21 15:40:48] NOTICE[114485][C-0000d180]: channel.c:4302 _astread: Dropping >incompatible voice frame on SIP/G200AN-00017497 of format g722 since our native format has changed >to (ulaw)
cos.agi: Starting Class Of Service checks
cos.agi: CallerID not Parseable - received Anonymous - exiting with DENY
-- AGI Script Executing Application: (Goto) Options: (macro-trunk-dial,barred,1)
-- Goto (macro-trunk-dial,barred,1)
-- AGI Script cos.agi completed, returning 0
[2017-10-21 15:40:48] WARNING[114485][C-0000d180]: pbx.c:6863 _astpbx_run: Channel >'SIP/G200AN-00017497' sent to invalid extension but no invalid handler: context,exten,priority=macro->trunk-dial,barred,2
Posts: 11
Participants: 4
@Blutechdata wrote:
Hello,
I am interested in the Zulu UC application for FreePBX. I went ahead and activated the 2 free licenses on my FreePBX system to test out the software before purchasing the 20 user license.
Seems to be pretty slick, the only issue I am having with Zulu is the softphone option on the call tab is not showing up. I can only do 'originate call'. I have the 'enable softphone' set to 'yes' in user management. I followed the wiki. What else is necessary to make this work? Let me know if additional information is required please.
FreePBX Firmware: 10.13.66-21
Zulu Version: 13.0.53.4
Zulu Desktop Client Version: 2.1.14
Zulu UC is installed on Windows 10 OS
Posts: 7
Participants: 3
@jasonmel wrote:
Hello,
We have a new PBXact Appliance. When attempting to configure templates for our Sangoma S700 phones I get a Whoops error
PDOException (42S22)
SQLSTATE[42S22]: Column not found: 1054 Unknown column 'phoneLabel' in 'field list'Anyone know a fix?
Thanks,
Posts: 4
Participants: 2
@Bradbpw wrote:
I am having issues with Zulu connecting to my PBX.
I can originate a call from Zulu but only to send the call to my desk phone. The softphone does not work (It says softphone (offline) - RECONNECT). The behavior is the same whether the computer with Zulu is connecting to my PBX over LAN or WAN
Log when Zulu is on same LAN as PBX
October 25th 2017, 9:58:32 am - debug: Softphone - SipJS Internal - sip.transport - - connecting to WebSocket wss://192.168.1.109:8089/ws
October 25th 2017, 9:58:32 am - debug: Softphone - engineEvent - connecting
October 25th 2017, 9:58:32 am - debug: ZuluApp - Softphone - Received connecting message
October 25th 2017, 9:58:32 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket connection error: {"isTrusted":true}
October 25th 2017, 9:58:32 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket disconnected (code: 1006)
October 25th 2017, 9:58:32 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket abrupt disconnection
October 25th 2017, 9:58:32 am - debug: Softphone - SipJS Internal - sip.ua - - transport wss://192.168.1.109:8089/ws failed | connection state set to 2
October 25th 2017, 9:58:32 am - debug: Softphone - engineEvent - disconnected
October 25th 2017, 9:58:32 am - debug: ZuluApp - Softphone disconnected
October 25th 2017, 9:58:32 am - debug: ZuluCall - Updating incomingCall state
October 25th 2017, 9:58:32 am - info: ZuluCall - Number of active calls - 0
October 25th 2017, 9:58:32 am - debug: Softphone - SipJS Internal - sip.ua - - next connection attempt in 16 secondsLogs when connecting over WAN (xxx.xxx.xxx.xxx is the external IP for my PBX)
October 25th 2017, 10:03:58 am - debug: Softphone - SipJS Internal - sip.transport - - connecting to WebSocket wss://xxx.xxx.xxx.xxx:8089/ws
October 25th 2017, 10:03:59 am - debug: PBXCommunicator - increaseReconnectionTime - Increased reconnection time - 8000ms
October 25th 2017, 10:03:59 am - debug: PBXCommunicator - connect - 192.168.1.109 - 8002 - wss://
October 25th 2017, 10:03:59 am - debug: PBXCommunicator - Connecting to pbx server
October 25th 2017, 10:04:17 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket connection error: {"isTrusted":true}
October 25th 2017, 10:04:17 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket disconnected (code: 1006)
October 25th 2017, 10:04:17 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket abrupt disconnection
October 25th 2017, 10:04:17 am - debug: Softphone - SipJS Internal - sip.ua - - transport wss://xxx.xxx.xxx.xxx:8089/ws failed | connection state set to 2
October 25th 2017, 10:04:17 am - debug: Softphone - engineEvent - disconnected
October 25th 2017, 10:04:17 am - debug: ZuluApp - Softphone disconnected
October 25th 2017, 10:04:17 am - debug: ZuluCall - Updating incomingCall state
October 25th 2017, 10:04:17 am - info: ZuluCall - Number of active calls - 0
October 25th 2017, 10:04:17 am - debug: Softphone - SipJS Internal - sip.ua - - next connection attempt in 4 secondsMy PBX is in the DMZ, I also have port 8089 and 8002 forwarded to the PBX (does that even matter if it's in the DMZ?). My PBX firewall is set to allow Zulu service on Internet, Local, and Other.
I attempted to connect with the PBX firewall disabled, that didn't work either. Logs when firewall was disabled:
October 25th 2017, 10:10:31 am - debug: ZuluMenu - _resizeWindow - 207
October 25th 2017, 10:10:31 am - debug: PBXCommunicator - Connection to PBX Server was closed: Invalid Response From Server
October 25th 2017, 10:10:31 am - debug: ZuluApp - Received pbxStatus - offline
October 25th 2017, 10:10:31 am - debug: ZuluUserInfo - Removing listeners
October 25th 2017, 10:10:31 am - debug: ZuluKeypad - Removing listeners
October 25th 2017, 10:10:31 am - debug: ZuluContactSuggestions - Removing listeners
October 25th 2017, 10:10:31 am - debug: ZuluPBXInfo - Removing listeners
October 25th 2017, 10:10:31 am - debug: ZuluConnecting - Initial message - Invalid Response From Server
October 25th 2017, 10:10:31 am - debug: ZuluConnecting - Received pbxErrorMessage - {"sender":{"domain":null,"events":{"pbxConfig":[null],"zuluConfig":[null]},"eventsCount":21}}
October 25th 2017, 10:10:33 am - debug: Softphone - SipJS Internal - sip.transport - - connecting to WebSocket wss://xxx.xxx.xxx.xxx:8089/ws
October 25th 2017, 10:10:33 am - debug: Softphone - engineEvent - connecting
October 25th 2017, 10:10:33 am - debug: ZuluApp - Softphone - Received connecting message
October 25th 2017, 10:10:33 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket connection error: {"isTrusted":true}
October 25th 2017, 10:10:33 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket disconnected (code: 1006)
October 25th 2017, 10:10:33 am - debug: Softphone - SipJS Internal - sip.transport - - WebSocket abrupt disconnection
October 25th 2017, 10:10:33 am - debug: Softphone - SipJS Internal - sip.ua - - transport wss://xxx.xxx.xxx.xxx:8089/ws failed | connection state set to 2
October 25th 2017, 10:10:33 am - debug: Softphone - engineEvent - disconnected
October 25th 2017, 10:10:33 am - debug: ZuluApp - Softphone disconnectedAny ideas?
Posts: 5
Participants: 2
@tlwilliams wrote:
Is it possible to change the caller ID display for attendees in a conference room so that the room in the UCP shows John Smith <1234567890> instead of WIRELESS CALLER <1234567890>?
I've set the CID superfecta to use the contact manager and assigned that on the options on the conference room. When I run the debug on the CID superfecta scheme, it says that the caller id is set to the entry in the contact manager, but when I test this, it does not set it when looking at the UCP
Posts: 1
Participants: 1
@nickb wrote:
I've purchased an HT-814 to check out, and to my surprise it's not listed in the commercial EPM module. Checking the Wiki, I see that the entire HT-8xx series isn't mentioned at all (not even present as tested or not).
Is there any mechanism to know if/when new device support is added?
Posts: 4
Participants: 3
@garias wrote:
Hello guys, i need some help provisioning T46G Yealink phones, i got a lot of them to work with, i setup the template to use http provision, set my options and i see all the files are correctly built from /tftpboot folder.
i do provision from the scanner, phone reboot but nothing get in, i also set provision url form phone gui, phone start process and said configuration updated but nothing happens.
if anyone got some light on this please help.
Posts: 10
Participants: 4
@JayG30 wrote:
I'm using the Conference Room Pro module. I'm setting up a DID into the Conference Room IVR. I want all Conference Rooms to REQUIRE a ADMIN/USER PIN. However I don't see any way to enforce that, meaning the end user could simply remove it leaving the conference room prone to unauthorized use.
I don't really care if the end users change those PIN codes. I would like to enforce a certain length and complexity to it if possible as well, but first just enforcing it would be nice.
Is this possible? Thanks.
Posts: 1
Participants: 1
@gdi2k wrote:
Up until now we've had FreePBX servers (primary and backup) on our LAN, connected to the VOIP gateways of our VOIP providers, and things have been fine.
Now we're adding a second office in a different location, so we need to start thinking about colocating our FreePBX servers with our ISPs, rather than having them in a particular office. We can't have a situation where both offices go down just because one office has a connectivity problem.
As I don't really trust either of the two ISPs to run a very reliable colocation service (it's a duopoly here), I would ideally like to have one server colocated with each ISP, and configure them with High Availability. I believe I can make it work from a networking standpoint: I would have VPN already set up before the FreePBX install such that both servers could see each other on the same network segment (the FreePBX instances would be virtualized).
But I have no idea about latency and bandwidth requirements for HA. I guess connectivity between the servers would be in the 40 - 60ms latency range, with 10-20 MBPS of bandwidth. Is this viable, or am I better off going with a Primary and Backup server (one on each ISP) and backing up two or three times a day?
Thanks!
Posts: 2
Participants: 2
@itcinc wrote:
I am new to FreePBX and have it setup and working. I have purchased the EM module to make programming the Polycom VVX 410 phones easier.
I have setup each extension with the shared VM box (101@device) and the message light works properly.
However, the one piece that I am missing is how to reprogram the Message button on each extension to check the group VM box. I know that I need to assign *98101 and Send to the message button but can't seem to find a place to do that - at least not in EM. Any pointers would be greatly appreciated.!
Thanks.
Posts: 1
Participants: 1
@tigger1197 wrote:
I would like have appointment reminders pull information from the calendar to make calls. Is this possible or not?
Posts: 3
Participants: 2
@ccampbell wrote:
We've found on the last few installs that we've done with FreePBX and Endpoint Manager that with the Cisco SPA phones that we use, the
__provisionAddress__variable is adding [--uid --pwd ] to the beginning of that variable. For instance, when thespa<MAC>.xmlfile is created for each CiscoSPA phone in the /tftpboot directory, we will see an output like this:
<Profile_Rule ua="na">[--uid --pwd ] tftp://10.1.2.3/spa$MA.xml</Profile_Rule>We've worked around the issue by doing this:
The
<Profile__Rule ua="na">__provisionAddress__/spa$MA.xml</Profile_Rule>Entry needs to be modified to this:
<Profile_Rule ua="na">tftp://__destination__/spa$MA.xml</Profile_Rule>But now we're seeing an issue exactly the same as this one:
The original poster in the post above changed the provision address to a FQDN, but we're running these PBXes on internal networks and that's not something that we are excited about doing at all our sites. Why is the [--uid --pwd ] entry showing up?
This doesn't seem to happen on PBXes that we have single network interfaces on, but only on PBXes where we have multiple network interfaces.
Thank you,
Court
Edited to try and improve formatting.
Posts: 1
Participants: 1
