@dannyprecise wrote:
I have the queue set to Ring All and set to Yes for Skip busy agent
Agent A is on a call on line 1 and their Line 2 will ring for the next caller in the queue even though they are from the queue set to skip. Will this work with Ring All?
Posts: 5
Participants: 3
@GSnover wrote:
mysqldump: Error 2013: Lost connection to MySQL server during query when dumping table
kvblobstoreat row: 29
Building manifest…
PHP Fatal error: Cannot use object of type DB_Error as array in /var/www/html/admin/libraries/modulefunctions.class.php on line 2394
Whoops\Exception\ErrorException: Cannot use object of type DB_Error as array in file /var/www/html/admin/libraries/modulefunctions.class.php on line 2394
Stack trace:
- Whoops\Exception\ErrorException->() /var/www/html/admin/libraries/modulefunctions.class.php:2394
When I try and run a backup and move to another load - Anyone else seen this error?
Posts: 1
Participants: 1
@twohuck wrote:
Hi
I’m running FreePBX 13.0.195.18 with Asterisk 13.23.1. I’m seeing this error on several machines I run after doing the latest round of module updates. I also ran Yum Update from the CLI to update Asterisk from 13.19.1 to 13.23.1. I’m not seeing any issues on the PBXs or getting any complaints. When I run this command from the CLI “/var/lib/asterisk/bin/module_admin listonline > /dev/null 2>&1” , it runs with no errors. Anybody know if this is something I need to worry about?Thanks
The following commands failed with the listed error
/var/lib/asterisk/bin/module_admin listonline > /dev/null 2>&1 (1)
Posts: 1
Participants: 1
@digitalo wrote:
After updating and upgrading my brand new system I’m presented with following error when opening the webui:
Exception (404)Unable to locate the FreePBX BMO Class 'Oembranding’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install oembranding 2) fwconsole ma enable oembranding
When I try to install the oembranding module I get the following error:
[FATAL] Unable to load oembranding as no license was found for this product.
Posts: 6
Participants: 2
@mficzere wrote:
Hello,
I am hoping that someone can give me a hand here. Here is what we have.
Sangoma Appliance 40, a Sangoma Vega 60G with 8FXO ports and 16 Sangoma s405 phones.
Problem we are having is when people call in through our PSTN lines into the Vega the person receiving the call on the SIP side is very quite in volume to the person calling them. The person on the SIP side can hear the caller with no issues. Also if the user on the SIP side calls out they are very quiet to the person receiving the call yet on the SIP side the volume of the person they called is totally fine. Internal calls within the office from IP phone to IP phone is not a problem. Volumes are just fine.
Any any clue or help would be greatly appreciated.
Thanks.
Posts: 3
Participants: 3
@bajramia wrote:
Hi All,
I have Pbxact UC 2000 fully licensed i want to see if is there anything like CRM which connects with asterisk so my agants can keep notes of the calls so next time when they call i can see all those notes thank you all.Another good module for hospital would be so the patient can call in and be promted to eneter their phone number and check when their appointments is
Almos like and appointments reminder but this they call in.Thank you
Posts: 1
Participants: 1
@andrewaitken wrote:
Hello
I have a user who cannot transfer external calls to other users. Other users in the same group are fine, internal calls are fine, if a external call is transferred to her she can the transfer them on.
I don’t know where to start looking for this one?
Andrew
Posts: 1
Participants: 1
@koreyb wrote:
Hi There,
I was just wondering if anyone has had any luck provisioning AASTRA 9133i, and programming the “Idle Display Name 1” and “Idle Display Name 2” lines?
If so, what PARAMETER = VALUE are you using in the config file to make it populate? I haven’t had much luck funding this online or adding commands to the basefile template.
I’m using freepbx endpoint manager to provision.
Posts: 1
Participants: 1
@electropolis wrote:
When put fwconsole start show this error.
[root@localhost ~]# fwconsole start
Asterisk already running
Running FreePBX startup…
Taking too long? Customize the chown command,
Setting Permissions…
Setting base permissions…Done
Setting specific permissions…
27778 [============================]
Finished setting permissions
Unable to run Pre-Asterisk hooks, because Asterisk is already running on PID 1525 and has been running for 3 minutes, 42 seconds
Running Asterisk post from Dahdiconfig module
Running Asterisk post from Ucpnode module
UCP Node Server has already been running on PID 2712 for 3 minutes, 37 seconds
Running Asterisk post from Xmpp module
Resetting PBX Users Failed: The command “node /var/www/html/admin/modules/xmpp/node/resetpbxusers.js” faile d.Exit Code: 1(General error)
Working directory: /root
Output:
Error Output:
module.js:338
throw err;
^
Error: Cannot find module ‘async’
at Function.Module._resolveFilename (module.js:336:15)
at Function.Module._load (module.js:278:25)
at Module.require (module.js:365:17)
at require (module.js:384:17)
at Object. (/var/www/html/admin/modules/xmpp/node/resetpbxusers.js:3:13)
at Module._compile (module.js:460:26)
at Object.Module._extensions…js (module.js:478:10)
at Module.load (module.js:355:32)
at Function.Module._load (module.js:310:12)
at Function.Module.runMain (module.js:501:10)
Posts: 1
Participants: 1
@stan_dm wrote:
FreePBX 13.0.195.18
trying to login via ssh (not root user) using Putty ssh client
login as: stan
stan @ 192.168.1.82’s password:in server logs I can see what sshd succesfully authenticated user and shell started
but i cant see anything on client endany suggestions?
wbr
stan
Posts: 1
Participants: 1
@sentinelace wrote:
Brand new install, and I get this screen. I’m not sure what is wrong? I am using PJsip on freepbx 14
Posts: 1
Participants: 1
@smbenson wrote:
Since GV removed support for Motif/GV, I am trying to switch to Incredible PBX with GVSIP on vmware. I have a SPA232D and SPA122 for 3 extensions using analog phones. Everything was configured and working on FreePBX calling outbound via pstn line. I have set up Incredible PBX with the same configuration as FreePBX with trunks, incoming / outgoing routes, and extensions. SPA232D PSTN and Line 1, SPA122 Line 1 and LIne 2 all register fine. I can receive incoming PSTN and GV calls through the SPA232D and make GV calls out. I am having problems with calling out through SPA232D to standard POTS line. Here is the end of the log file that shows outgoing sip connecting to trunk and bridging to line 1, but exiting due to dialout-trunk macro and hanging up.
[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-dialout-trunk:21] ExecIf("SIP/100-00000040", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)2188942441)") in new stack [2018-10-30 12:20:19] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-dialout-trunk:22] GotoIf("SIP/100-00000040", "0?customtrunk") in new stack [2018-10-30 12:20:19] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-dialout-trunk:23] Dial("SIP/100-00000040", "SIP/1-pstn/2182960712,300,T") in new stack [2018-10-30 12:20:19] VERBOSE[23081][C-00000021] netsock2.c: Using SIP RTP TOS bits 184 [2018-10-30 12:20:19] VERBOSE[23081][C-00000021] netsock2.c: Using SIP RTP CoS mark 5 [2018-10-30 12:20:19] VERBOSE[23081][C-00000021] app_dial.c: Called SIP/1-pstn/2182960712 [2018-10-30 12:20:19] VERBOSE[23081][C-00000021] app_dial.c: SIP/1-pstn-00000041 answered SIP/100-00000040 [2018-10-30 12:20:19] VERBOSE[23082][C-00000021] bridge_channel.c: Channel SIP/1-pstn-00000041 joined 'simple_bridge' basic-bridge <f8543099-ac21-4989-a665-3a795fd6c9fc> [2018-10-30 12:20:19] VERBOSE[23081][C-00000021] bridge_channel.c: Channel SIP/100-00000040 joined 'simple_bridge' basic-bridge <f8543099-ac21-4989-a665-3a795fd6c9fc> [2018-10-30 12:20:20] VERBOSE[23082][C-00000021] bridge_channel.c: Channel SIP/1-pstn-00000041 left 'simple_bridge' basic-bridge <f8543099-ac21-4989-a665-3a795fd6c9fc> [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] bridge_channel.c: Channel SIP/100-00000040 left 'simple_bridge' basic-bridge <f8543099-ac21-4989-a665-3a795fd6c9fc> [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] app_macro.c: Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'SIP/100-00000040' in macro 'dialout-trunk' [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Spawn extension (from-internal, 2182960712, 6) exited non-zero on 'SIP/100-00000040' [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Executing [h@from-internal:1] Macro("SIP/100-00000040", "hangupcall") in new stack [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000040", "1?theend") in new stack [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx_builtins.c: Goto (macro-hangupcall,s,3) [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/100-00000040", "0?Set(CDR(recordingfile)=)") in new stack [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/100-00000040", "") in new stack [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-00000040' in macro 'hangupcall' [2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000040'Any assistance would be greatly appreciated.
Posts: 1
Participants: 1
@nkramer wrote:
Hello,
We have been using our system for a roughly a year almost and one thing I have been unable to figure out is voice mail time stamp. I checked the server time and it is right, however, when one of us receive a voicemail it is 5 hours ahead. Any assistance Would be greatly appreciated.
Thanks in advance
Posts: 3
Participants: 3
@rnrstar wrote:
FreePBX 13.0.195.17
I’m trying to add a DID for an extension and when I point it at one particular extension I’m getting a 503 response from the server to the SIP provider. If I redirect that DID to any other extension it works no problem. When I use the GUI to search the log file for the DID phone number, nothing shows up.
Any suggestions?
Posts: 4
Participants: 3
@travis_farmer wrote:
bit of background, i have two phone lines connected. one is from the phone company, and serves the household, and the other is via my cellular Home Phone device, and serves my phones. Both go through my phone system for the ability to transfer and direct dial between my phones, and the household phones.
the phone line for the household does not have CallerID service. when the household line rings through to the household phones, that are connected via a FXS ATA (GrandStream GS-HT814), the phone CallerID announcer spells out the IP address of the Asterisk/FreePBX server. internal CallerID works great. but i have been struggling with the inbound CallerID. i have tried a variaty of things, and don’t remember all of what was done. but the result is the same.
i want the internal CallerID to still function, but remain silent for the inbound CallerID (or lack of).
i have tried passing the call through Set CallerID from the inbound route, and then on to the ring group. i blanked out the values in Set CallerID, and it changed nothing.forgive my lack of initial info, as my frustration levels are high from the pressure from the other household members wanting me to fix it.
~Travis
Posts: 1
Participants: 1
@kg4ysy wrote:
Is there a place to see when Asterisk versions will make it into the FreePBX Distro yum repos or a repo that has newer Asterisk versions? I updated to asterisk-13-core.x86_64 13.22.0-1.sng7 today. Unfortunately, that version of Asterisk has a bug in it (https://issues.asterisk.org/jira/browse/ASTERISK-27881) that is causing issues with some of our trunks. I need to get up to Asterisk 13.23.0 or higher to get that bugfix, but I can’t seem to get that version via the yum repos. I enabled a couple other repos I found in /etc/yum.repos.d, but they didn’t have any newer Asterisk 13 versions. 13.23 has been out for a month or two.
Posts: 1
Participants: 1
@nkramer wrote:
Hello,
Ok so I have a very small business with 20 extensions. I am wanting to set two specific extenions (207 and 208) to use one particular Outbound Route and all other use the other route. How can I accomplish this? I don’t see an option to set the route in the extension tab or add extensions to the route tab. Running FreePBX 14.0.4.1. Any help would be awesome.
Thank you in advance.
Posts: 5
Participants: 4
@matt8919 wrote:
I may be reaching out in the wrong place but I figured I’d start here and see if any of the community has any experience with this issue.
I have a client that uses Grandstream GXP 2170 VoIP phones. The phones are currently on firmware version 1.0.9.26. The PBX version we are using is 14.0.3.2. The issue they are having is that randomly throughout the day, their BLF keys show inaccurate behavior. For example, the BLF keys blink red even though the device tied to that BLF is not actually receiving a call at the moment. What also happens is their BLF keys will show solid red even though the device tied to the BLF is not actually on a call. Anyone got any insight on this?
Just so you are all aware, this issue was happening when we first installed them a while back. When they first reported the issue to me I noticed there firmware wasn’t even at 1.0.9.26. I did a factory reset on the devices and then updated them to 1.0.9.26. The issue went away for approx. 1 day and then returned.
Posts: 2
Participants: 2
@mvogel4949 wrote:
I am attempting to look through the asterisk logfiles for some details on a call only to find there is absolutely zero information from calls being written to the logs? I’m seeing phones go unreachable and then reachable but absolutely zero info on calls. Any ideas?
Posts: 8
Participants: 3
@XcamaroX wrote:
Hey guys,
I have an issue that prevents 2 different PBX’s dial internal extensions from trunk to trunk.
I have Trunk A at one location and Trunk B at another location.
Both are connected as I can see the link is healthy but when I dial internal extensions I get a message that the extension is not available.
For example in Trunk A, all extensions are in the 3000 range and in Trunk B are all on 4000.
If from Trunk B I try to call 3001 it says that, that “Your call cannot be completed as dialed, please check…”).
I have setup the Outbound Route but still have this issue.
This is what I find in the terminal, but I cant seem to find the Unkowns.
Reliably Transmitting (NAT) to providerIP:5060: OPTIONS sip:voip.com SIP/2.0 Via: SIP/2.0/UDP publicIP:5160;branch=z9hG4bK283b84f3;rport Max-Forwards: 70 From: "Unknown" <sip:Unknown@publicIP:5160>;tag=as304ae73c To: <sip:voipprovider.com> Contact: <sip:Unknown@publicIP:5160> Call-ID: 7ea75a5c3514465f4522735e3a9e065f@ XXX.XXX.XXX.XXX:5160 CSeq: 102 OPTIONS User-Agent: FPBX-13.0.195.4(13.17.0) Date: Wed, 31 Oct 2018 19:06:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:providerIP:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP publicIP:5160;branch=z9hG4bK283b84f3;received= publicIP;rport=16237 From: "Unknown" <sip:Unknown@ XXX.XXX.XXX.XXX:5160>;tag=as304ae73c To: <sip:voipprovider.com>;tag=as1ecbff6c Call-ID: 7ea75a5c3514465f4522735e3a9e065f@ XXX.XXX.XXX.XXX:5160 CSeq: 102 OPTIONS Server: voip.com Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:providerIP:5060> Accept: application/sdp Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '7ea75a5c3514465f4522735e3a9e065f@publicIP:5160' Method: OPTIONS Reliably Transmitting (NAT) to 192.168.XX4.5:5060: OPTIONS sip:192.168.XX4.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.XX3.5:5160;branch=z9hG4bK35c747b5;rport Max-Forwards: 70 From: "Unknown" <sip:Unknown@192.168.XX3.5:5160>;tag=as6d1317a5 To: <sip:192.168.XX4.5> Contact: <sip:Unknown@192.168.XX3.5:5160> Call-ID: 00138aa3610be7db2cc831d669c3d15f@192.168.XX3.5:5160 CSeq: 102 OPTIONS User-Agent: FPBX-13.0.195.4(13.17.0) Date: Wed, 31 Oct 2018 19:06:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- <--- SIP read from UDP:192.168.XX4.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.XX3.5:5160;branch=z9hG4bK35c747b5;received=192.168.XX3.5;rport=5160 From: "Unknown" <sip:Unknown@192.168.XX3.5:5160>;tag=as6d1317a5 To: <sip:192.168.XX4.5>;tag=as40aed762 Call-ID: 00138aa3610be7db2cc831d669c3d15f@192.168.XX3.5:5160 CSeq: 102 OPTIONS Server: FPBX-13.0.195.13(13.23.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:192.168.XX4.5:5060> Accept: application/sdp Content-Length: 0
Posts: 1
Participants: 1
