@cnissarte wrote:

Hello,

i am very new in the freepbx world.

if anyone could kindly advise me.

We have IVR in french and English. works fine, some site are only in English, other only in French. Each “Invalid Retry Recording language” parameter is configured with the “default” value.
When someone enter an digit not corresponding to our ivr option, the “we have not received a valid response please try again” announce is always in French, never in English.
How can we get this announce in English for English IVR please?

thanks a lot

best regards

Posts: 1

Participants: 1

Read full topic

@edricksmith wrote:

Well hello again lovely community;

So this time I have decided to open a support ticket instead of bothering the community with VPN configuration issues with some Sangoma Desk Phones.

However… every time I try to launch the sangoma support desk from my partners portal.

exception.png800×312

Unable to locate file in ./__swift/library/Sangoma/class.SWIFT_SangomaAPI.php OR ./__swift/apps/core/library/Sangoma/class.SWIFT_SangomaAPI.php OR ./__swift/apps/base/library/Sangoma/class.SWIFT_SangomaAPI.php

Posts: 2

Participants: 1

Read full topic

@kb9mfd wrote:

I already have bricked one phone and for the life of me I cannot get these to work. In EPM I did all the normal stuff, created a template etc… and the DHCP server is set properly for provisioning as the Sangoma phones work fine. I factory reset a IP6000 and it did grab its config, but only partly. It grabbed the SIP user information but not the the SIP server or the dialplan. So I turned on firmware update, now its in a reboot loop. I tried following the instructions from Polycom to update the firmware, horribly confusing, and removed the firmware from EPM, That bricked one phone, it now reads “no application” on boot and no matter what I do its stuck on that. The other just boots, gets to loading the config and then reboots. Anyone have any idea how to get these first on the latest firmware without bricking them, then get them on EPM, then actually get them to work?

Posts: 1

Participants: 1

Read full topic

@sentinelace wrote:

I have noticed the last two PBX’s I have deployed that when marked yes, phones are still showing missed calls. Yealink T46S, freepbx 14. All current Modules and updates. Is there additional configuration now?

Posts: 1

Participants: 1

Read full topic

@dsirota wrote:

I just upgraded FreePBX to 15.0.11, and I can’t press “Apply Config” because when I do, I get this message: Unknown Error. Please Run: fwconsole reload --verbose I’ve run that command at least 10 times, nothing changes.

Also, when running the above command, this error message comes back: [Exception] Cron line added didn't remain in crontab on final check. Check /tmp/cron.error for reason.

Posts: 1

Participants: 1

Read full topic

@axl2 wrote:

Hello,
I have a couple of questions but here is the first one. When I go to the Admin-System Admin-Network Settings, we have eth0 set up with ip 10.34.x.x (Cisco Trunk). Our system has been working fine and our guru retired. Now, IT needs us to change the ip address and I’m trying to figure out the system. When I type in the new ip address of 10.43.x.x and hit the Save Setting, the old ip appears again. I’ve rebooted and the Network Settings still status the old 10.34.x.x. When I open the terminal and type ifconfig eth0, its show the new ip of 10.43.x.x. The system works, phone calls getting through. Just don’t know why the WebGUI status the old ip and not the new IP address.
FreePBX 2.11.0.43 and Asterisk 11.9.0

Thanks

Posts: 4

Participants: 2

Read full topic

@pasi wrote:

Every now and then I experience audio choppy issue during my call.
It happens randomly with both outgoing & incoming calls.
Network utilization at the time of the incident is only at 2-3Mbps out of 50Mbps capability.
I observe these Warning and Notice in from the asterisk log.

[2019-05-21 16:20:19] WARNING[28599][C-00000212]: chan_iax2.c:1239 jb_warning_output: Resyncing the jb. last_delay 0, this delay -3474, threshold 1000, new offset 3474
[2019-05-21 16:20:19] WARNING[28618][C-00000212]: chan_iax2.c:1239 jb_warning_output: Resyncing the jb. last_delay 0, this delay -166641430, threshold 1000, new offset 166641430
[2019-05-21 16:25:09] NOTICE[15207]: chan_sip.c:24592 handle_response_peerpoke: Peer ‘VoIP-MS’ is now Lagged. (2017ms / 2000ms)
[2019-05-21 16:25:19] NOTICE[15207]: chan_sip.c:24592 handle_response_peerpoke: Peer ‘VoIP-MS’ is now Reachable. (16ms / 2000ms)

These messages is appearing periodically.
I am running pings to the VoIP trunk’s IP at the same time on the same FreePBX server and there is no abnormal ping delay. They are all within <17ms.

I tried reboot the FreePBX server, disable all other un-use trunks. But the issue is still persisting.

Does anyone have any clue of what is going on?

Thank you.

Posts: 1

Participants: 1

Read full topic

@badams wrote:

I have a Ring Group with several extensions, among them are several Grandstream GXP-2135, 2170, and a DP752 with two DP720 phones paired.
It seems when a call comes into the ring group, all the GXP-2135 and 2170s rings are synced, however the cordless phones (DP720) have rings that are opposite of the “hardwired” phones. So while the 2135s are ringing, the DP720 are silent. When the 2135 stop ringing, the DP720 starts.

I know the community doesnt have anything nice to say about Grandstream, but I’m hoping there is a simple fix somewhere for this that I have been unable to find

Posts: 1

Participants: 1

Read full topic

@comtech wrote:

I have a blended Avaya/Asterisk environment. Asterisk for call treatments and Avaya for agents and queues.

FreePBX/Asterisk 14.

I want to create an app,

• That collected information from the caller
• Stores them as channel variables
• Lets the caller hang up, but keeps the call up and dials the agents queue (in Avaya).
• Once the agent answers they will hear a message to press 1 to begin a callback.
• Once the agent pushes 1, the original caller is called back and connected with the agent.

Question:
I know how to do most of this, what I want to avoid is on the first leg of the call, how do I stop the caller’s hang up from wiping out the entire call?

Another way to ask, is how do I disconnect the caller, and keep the call alive so it can call Avaya?

If you have alternate advice on how to accomplish something similar (other than putting the agents in Asterisk), I am open to that feedback as well. Thanks in advance!

Posts: 2

Participants: 2

Read full topic

@grefranger wrote:

Hello,

Without making any changes recently my system firewall started having a problem today where the dashboard reported “firewall service not running.” The firewall log as this error:
PHP Fatal error: Uncaught exception ‘Exception’ with message ‘Firewall is not running!’ in /var/www/html/admin/modules/firewall/Attacks.class.php:17
Stack trace:
#0 phar:///var/www/html/admin/modules/firewall/hooks/voipfirewalld/firewall.php(549): FreePBX\modules\Firewall\Attacks->__construct(1000)
#1 phar:///var/www/html/admin/modules/firewall/hooks/voipfirewalld/firewall.php(225): updateFirewallRules(true)
#2 /var/www/html/admin/modules/firewall/hooks/voipfirewalld(3): include(‘phar:///var/www…’)
#3 {main}
thrown in /var/www/html/admin/modules/firewall/Attacks.class.php on line 17
1558481127: Monitoring parent (voipfirewalld) died. Shutting down!

I tried various things to fix it including rebooting and restarting the service and the server, upgrading the modules, upgrading Asterisk from 12 to 13, reinstalling the firewall, and running yum update, no luck. Downgrading the system firewall from 13.0.57.1 to 13.0.57 removed the error from the dashboard, however I am still seeing some odd behavior as shown in the picture below:

freepbx system firewall error.JPG1850×504 87.9 KB

My distro is 10.13.66-22

Any help is greatly appreciated!

Posts: 1

Participants: 1

Read full topic

@digiteltlc wrote:

How should sip NAT setting be , when a pbx has more trunks in NAT and no NAT connections ?

i.e. three trunks , one with a same LAN pbx, one with a pbx over vpn, and one with ISP being behind a natted router, and maybe , some ip phones registering from internet to this pbx behind nat (just examples)

Any guideline ?

Posts: 2

Participants: 2

Read full topic

@Jake321 wrote:

Hello,

I have trouble upgrading the freepbx server. It runs into a conflict, you can see it at this paste i made: https://pastebin.com/zrbFQr3i

I also noticed this topic where something like it discussed, but this doesn’t work for me:

I hope that someone can point me in the right direction.

Thanks,
Jake

Posts: 1

Participants: 1

Read full topic

@dan_ce wrote:

I’m looking for either…

a) a script which would check the state of call flow (day/night mode) periodically and report it to an MQTT broker
or
b) a script which would ANSWER a query from another script with the state of call flow with a 0 or a 1…

anyone know if this is doable?

The reason I’m asking is I want to tie it in to my Home Assistant install on another Pi so HA “KNOWS” whether day/night mode is activated or not.

Thank you!!

Posts: 1

Participants: 1

Read full topic

@zosselp wrote:

Hi,

My current FreePBX installation has been working fine for over a year now until today.
I can receive incoming external calls but I cannot make outgoing external calls.

Trunk settings:

username=***
type=peer
secret=***
qualify=no
port=5260
nat=yes
insecure=very
host=***
fromuser=***
fromdomain=***
dtmfmode=rfc2833
allow=all

Failed call log:

> == Setting global variable 'SIPDOMAIN' to '192.168.10.2'
> -- Executing [791279486@from-internal:1] Macro("PJSIP/01-00000043", "user-callerid,LIMIT,EXTERNAL,") in new stack
> -- Executing [s@macro-user-callerid:1] Set("PJSIP/01-00000043", "TOUCH_MONITOR=1558526866.114") in new stack
> -- Executing [s@macro-user-callerid:2] Set("PJSIP/01-00000043", "AMPUSER=01") in new stack
> -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/01-00000043", "0?report") in new stack
> -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/01-00000043", "1?Set(REALCALLERIDNUM=01)") in new stack
> -- Executing [s@macro-user-callerid:5] Set("PJSIP/01-00000043", "AMPUSER=01") in new stack
> -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/01-00000043", "0?limit") in new stack
> -- Executing [s@macro-user-callerid:7] Set("PJSIP/01-00000043", "AMPUSERCIDNAME=Sekretariat") in new stack
> -- Executing [s@macro-user-callerid:8] GotoIf("PJSIP/01-00000043", "0?report") in new stack
> -- Executing [s@macro-user-callerid:9] Set("PJSIP/01-00000043", "AMPUSERCID=01") in new stack
> -- Executing [s@macro-user-callerid:10] Set("PJSIP/01-00000043", "__DIAL_OPTIONS=Ttr") in new stack
> -- Executing [s@macro-user-callerid:11] Set("PJSIP/01-00000043", "CALLERID(all)="Sekretariat" <01>") in new stack
> -- Executing [s@macro-user-callerid:12] GotoIf("PJSIP/01-00000043", "0?limit") in new stack
> -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/01-00000043", "1?Set(GROUP(concurrency_limit)=01)") in new stack
> -- Executing [s@macro-user-callerid:14] ExecIf("PJSIP/01-00000043", "0?Set(CHANNEL(language)=)") in new stack
> -- Executing [s@macro-user-callerid:15] GotoIf("PJSIP/01-00000043", "1?continue") in new stack
> -- Goto (macro-user-callerid,s,29)
> -- Executing [s@macro-user-callerid:29] Set("PJSIP/01-00000043", "CALLERID(number)=01") in new stack
> -- Executing [s@macro-user-callerid:30] Set("PJSIP/01-00000043", "CALLERID(name)=Sekretariat") in new stack
> -- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/01-00000043", "0?cnum") in new stack
> -- Executing [s@macro-user-callerid:32] Set("PJSIP/01-00000043", "CDR(cnam)=Sekretariat") in new stack
> -- Executing [s@macro-user-callerid:33] Set("PJSIP/01-00000043", "CDR(cnum)=01") in new stack
> -- Executing [s@macro-user-callerid:34] Set("PJSIP/01-00000043", "CHANNEL(language)=pl") in new stack
> -- Executing [791279486@from-internal:2] Gosub("PJSIP/01-00000043", "sub-record-check,s,1(out,791279486,dontcare)") in new stack
> -- Executing [s@sub-record-check:1] GotoIf("PJSIP/01-00000043", "0?initialized") in new stack
> -- Executing [s@sub-record-check:2] Set("PJSIP/01-00000043", "__REC_STATUS=INITIALIZED") in new stack
> -- Executing [s@sub-record-check:3] Set("PJSIP/01-00000043", "NOW=1558526866") in new stack
> -- Executing [s@sub-record-check:4] Set("PJSIP/01-00000043", "__DAY=22") in new stack
> -- Executing [s@sub-record-check:5] Set("PJSIP/01-00000043", "__MONTH=05") in new stack
> -- Executing [s@sub-record-check:6] Set("PJSIP/01-00000043", "__YEAR=2019") in new stack
> -- Executing [s@sub-record-check:7] Set("PJSIP/01-00000043", "__TIMESTR=20190522-140746") in new stack
> -- Executing [s@sub-record-check:8] Set("PJSIP/01-00000043", "__FROMEXTEN=01") in new stack
> -- Executing [s@sub-record-check:9] Set("PJSIP/01-00000043", "__MON_FMT=wav") in new stack
> -- Executing [s@sub-record-check:10] NoOp("PJSIP/01-00000043", "Recordings initialized") in new stack
> -- Executing [s@sub-record-check:11] ExecIf("PJSIP/01-00000043", "0?Set(ARG3=dontcare)") in new stack
> -- Executing [s@sub-record-check:12] Set("PJSIP/01-00000043", "REC_POLICY_MODE_SAVE=") in new stack
> -- Executing [s@sub-record-check:13] ExecIf("PJSIP/01-00000043", "0?Set(REC_STATUS=NO)") in new stack
> -- Executing [s@sub-record-check:14] GotoIf("PJSIP/01-00000043", "3?checkaction") in new stack
> -- Goto (sub-record-check,s,17)
> -- Executing [s@sub-record-check:17] GotoIf("PJSIP/01-00000043", "1?sub-record-check,out,1") in new stack
> -- Goto (sub-record-check,out,1)
> -- Executing [out@sub-record-check:1] NoOp("PJSIP/01-00000043", "Outbound Recording Check from 01 to 791279486") in new stack
> -- Executing [out@sub-record-check:2] Set("PJSIP/01-00000043", "RECMODE=dontcare") in new stack
> -- Executing [out@sub-record-check:3] ExecIf("PJSIP/01-00000043", "1?Goto(routewins)") in new stack
> -- Goto (sub-record-check,out,7)
> -- Executing [out@sub-record-check:7] Gosub("PJSIP/01-00000043", "recordcheck,1(dontcare,out,791279486)") in new stack
> -- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/01-00000043", "Starting recording check against dontcare") in new stack
> -- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/01-00000043", "dontcare") in new stack
> -- Goto (sub-record-check,recordcheck,3)
> -- Executing [recordcheck@sub-record-check:3] Return("PJSIP/01-00000043", "") in new stack
> -- Executing [out@sub-record-check:8] Return("PJSIP/01-00000043", "") in new stack
> -- Executing [791279486@from-internal:3] ExecIf("PJSIP/01-00000043", "0 ?Set(CDR(accountcode)=)") in new stack
> -- Executing [791279486@from-internal:4] Set("PJSIP/01-00000043", "MOHCLASS=default") in new stack
> -- Executing [791279486@from-internal:5] Set("PJSIP/01-00000043", "_NODEST=") in new stack
> -- Executing [791279486@from-internal:6] Macro("PJSIP/01-00000043", "dialout-trunk,1,791279486,,off") in new stack
> -- Executing [s@macro-dialout-trunk:1] Set("PJSIP/01-00000043", "DIAL_TRUNK=1") in new stack
> -- Executing [s@macro-dialout-trunk:2] GosubIf("PJSIP/01-00000043", "0?sub-pincheck,s,1()") in new stack
> -- Executing [s@macro-dialout-trunk:3] GotoIf("PJSIP/01-00000043", "0?disabletrunk,1") in new stack
> -- Executing [s@macro-dialout-trunk:4] Set("PJSIP/01-00000043", "DIAL_NUMBER=791279486") in new stack
> -- Executing [s@macro-dialout-trunk:5] Set("PJSIP/01-00000043", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
> -- Executing [s@macro-dialout-trunk:6] Set("PJSIP/01-00000043", "OUTBOUND_GROUP=OUT_1") in new stack
> -- Executing [s@macro-dialout-trunk:7] GotoIf("PJSIP/01-00000043", "0?nomax") in new stack
> -- Executing [s@macro-dialout-trunk:8] GotoIf("PJSIP/01-00000043", "0?chanfull") in new stack
> -- Executing [s@macro-dialout-trunk:9] GotoIf("PJSIP/01-00000043", "0?skipoutcid") in new stack
> -- Executing [s@macro-dialout-trunk:10] Set("PJSIP/01-00000043", "DIAL_TRUNK_OPTIONS=T") in new stack
> -- Executing [s@macro-dialout-trunk:11] Macro("PJSIP/01-00000043", "outbound-callerid,1") in new stack
> -- Executing [s@macro-outbound-callerid:1] ExecIf("PJSIP/01-00000043", "0?Set(CALLERPRES(name-pres)=)") in new stack
> -- Executing [s@macro-outbound-callerid:2] ExecIf("PJSIP/01-00000043", "0?Set(CALLERPRES(num-pres)=)") in new stack
> -- Executing [s@macro-outbound-callerid:3] ExecIf("PJSIP/01-00000043", "0?Set(REALCALLERIDNUM=01)") in new stack
> -- Executing [s@macro-outbound-callerid:4] GotoIf("PJSIP/01-00000043", "1?normcid") in new stack
> -- Goto (macro-outbound-callerid,s,7)
> -- Executing [s@macro-outbound-callerid:7] Set("PJSIP/01-00000043", "USEROUTCID=") in new stack
> -- Executing [s@macro-outbound-callerid:8] Set("PJSIP/01-00000043", "EMERGENCYCID=") in new stack
> -- Executing [s@macro-outbound-callerid:9] Set("PJSIP/01-00000043", "TRUNKOUTCID=146888149") in new stack
> -- Executing [s@macro-outbound-callerid:10] GotoIf("PJSIP/01-00000043", "1?trunkcid") in new stack
> -- Goto (macro-outbound-callerid,s,15)
> -- Executing [s@macro-outbound-callerid:15] ExecIf("PJSIP/01-00000043", "1?Set(CALLERID(all)=146888149)") in new stack
> -- Executing [s@macro-outbound-callerid:16] ExecIf("PJSIP/01-00000043", "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s@macro-outbound-callerid:17] ExecIf("PJSIP/01-00000043", "0?Set(CALLERID(all)=)") in new stack
> -- Executing [s@macro-outbound-callerid:18] ExecIf("PJSIP/01-00000043", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
> -- Executing [s@macro-outbound-callerid:19] ExecIf("PJSIP/01-00000043", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
> -- Executing [s@macro-outbound-callerid:20] Set("PJSIP/01-00000043", "CDR(outbound_cnum)=146888149") in new stack
> -- Executing [s@macro-outbound-callerid:21] Set("PJSIP/01-00000043", "CDR(outbound_cnam)=") in new stack
> [2019-05-22 14:07:46] WARNING[2364]: func_cdr.c:383 cdr_write_callback: CDR requires a value (CDR(variable)=value)
> -- Executing [s@macro-dialout-trunk:12] GosubIf("PJSIP/01-00000043", "1?sub-flp-1,s,1()") in new stack
> -- Executing [s@sub-flp-1:1] ExecIf("PJSIP/01-00000043", "1?Return()") in new stack
> -- Executing [s@macro-dialout-trunk:13] Set("PJSIP/01-00000043", "OUTNUM=791279486") in new stack
> -- Executing [s@macro-dialout-trunk:14] Set("PJSIP/01-00000043", "custom=SIP/TOI") in new stack
> -- Executing [s@macro-dialout-trunk:15] ExecIf("PJSIP/01-00000043", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
> -- Executing [s@macro-dialout-trunk:16] ExecIf("PJSIP/01-00000043", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
> -- Executing [s@macro-dialout-trunk:17] Macro("PJSIP/01-00000043", "dialout-trunk-predial-hook,") in new stack
> -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/01-00000043", "") in new stack
> -- Executing [s@macro-dialout-trunk:18] GotoIf("PJSIP/01-00000043", "0?bypass,1") in new stack
> -- Executing [s@macro-dialout-trunk:19] ExecIf("PJSIP/01-00000043", "1?Set(CONNECTEDLINE(num,i)=791279486)") in new stack
> -- Executing [s@macro-dialout-trunk:20] ExecIf("PJSIP/01-00000043", "1?Set(CONNECTEDLINE(name,i)=CID:146888149)") in new stack
> -- Executing [s@macro-dialout-trunk:21] ExecIf("PJSIP/01-00000043", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)146888149)") in new stack
> -- Executing [s@macro-dialout-trunk:22] GotoIf("PJSIP/01-00000043", "0?customtrunk") in new stack
> -- Executing [s@macro-dialout-trunk:23] Dial("PJSIP/01-00000043", "SIP/TOI/791279486,300,T") in new stack
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
> -- Called SIP/TOI/791279486
> [2019-05-22 14:07:46] WARNING[15408]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:07:52] WARNING[15408]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:07:56] WARNING[12754]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:08:02] WARNING[15408]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:08:06] WARNING[24199]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:08:12] WARNING[15408]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:08:16] WARNING[12754]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:08:17] NOTICE[24199]: res_pjsip_exten_state.c:358 new_subscribe: Extension state subscription failed: Extension 20 does not exist in context 'from-internal' or has no associated hint
> -- SIP/TOI-0000002f is circuit-busy
> [2019-05-22 14:08:18] WARNING[2542]: chan_sip.c:4071 retrans_pkt: Retransmission timeout reached on transmission 4776f8200b94af07153c4e4b6c997577@217.117.142.10 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 32000ms with no response
>   == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing [s@macro-dialout-trunk:24] NoOp("PJSIP/01-00000043", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0") in new stack
> -- Executing [s@macro-dialout-trunk:25] GotoIf("PJSIP/01-00000043", "0?continue,1:s-CONGESTION,1") in new stack
> -- Goto (macro-dialout-trunk,s-CONGESTION,1)
> -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("PJSIP/01-00000043", "RC=0") in new stack
> -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("PJSIP/01-00000043", "0,1") in new stack
> -- Goto (macro-dialout-trunk,0,1)
> -- Executing [0@macro-dialout-trunk:1] Goto("PJSIP/01-00000043", "continue,1") in new stack
> -- Goto (macro-dialout-trunk,continue,1)
> -- Executing [continue@macro-dialout-trunk:1] NoOp("PJSIP/01-00000043", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks") in new stack
> -- Executing [continue@macro-dialout-trunk:2] ExecIf("PJSIP/01-00000043", "1?Set(CALLERID(number)=01)") in new stack
> -- Executing [791279486@from-internal:7] Macro("PJSIP/01-00000043", "outisbusy,") in new stack
> -- Executing [s@macro-outisbusy:1] Progress("PJSIP/01-00000043", "") in new stack
> -- Executing [s@macro-outisbusy:2] GotoIf("PJSIP/01-00000043", "0?emergency,1") in new stack
> -- Executing [s@macro-outisbusy:3] GotoIf("PJSIP/01-00000043", "0?intracompany,1") in new stack
> -- Executing [s@macro-outisbusy:4] Playback("PJSIP/01-00000043", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
> -- <PJSIP/01-00000043> Playing 'all-circuits-busy-now.ulaw' (language 'pl')
>    > 0x7ffb180f9890 -- Probation passed - setting RTP source address to 192.168.10.3:5004
> -- <PJSIP/01-00000043> Playing 'pls-try-call-later.ulaw' (language 'pl')
> -- Executing [h@from-internal:1] Macro("PJSIP/01-00000043", "hangupcall") in new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/01-00000043", "1?theend") in new stack
> -- Goto (macro-hangupcall,s,3)
> -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/01-00000043", "0?Set(CDR(recordingfile)=)") in new stack
> -- Executing [s@macro-hangupcall:4] Hangup("PJSIP/01-00000043", "") in new stack
>   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/01-00000043' in macro 'hangupcall'
>   == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/01-00000043'
> [2019-05-22 14:08:21] NOTICE[12754]: res_pjsip_exten_state.c:358 new_subscribe: Extension state subscription failed: Extension 23 does not exist in context 'from-internal' or has no associated hint
> [2019-05-22 14:08:22] WARNING[24199]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:08:26] WARNING[15408]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info
> [2019-05-22 14:08:32] WARNING[24199]: res_pjsip_pubsub.c:640 subscription_get_handler_from_rdata: No registered subscribe handler for event call-info

Anyone care to help me ?

Posts: 2

Participants: 2

Read full topic

@Hawkeye wrote:

This is the 4th time attempting to run convert.freepbx.org to convert a FreePBX 13 to FreePBX 14.

Start installing a brand new FreePBX 14 server using SNG7-FPBX-64bit-1805-2.iso

step4.jpg1030×776 61.8 KB

step5.jpg1030×776 41.7 KB

Reboot to new FreePBX 14 server, no conversion run as yet.

step6-installed.jpg1144×868 137 KB

No Extensions added as yet

step7-extensions-before-convert_freepbx_org.jpg1144×868 61.7 KB

Start the conversion process as per http://convert.freepbx.org

![step8-start_convert_production_box|643x384]
(upload://1rc8ynMygYOMib6cmNBH18bFeOK.jpeg)

Production (Donor) FreePBX sending…

After sending the conversion_backup.crypt via ssh to the new FreePBX 14 server…

Doesn’t copy over the email addresses assigned to extensions. And you’ll notice later on, all extensions are created as virtual extensions!

Click the Apply Config button, and the error appears. The error in retrieve conf ALWAYS shows its the lowest number extension. (When checked the extension noted in the error, nothing appears wrong. However, note that inbound routes, all extensions that were going to FAX Recipient were all errors. They did not have the fax recipient, just error.

step14-login_after_convert_hit_apply config-retrieve.conf-error.jpg1300×861 156 KB

Note all extensions show as virtual

step15-login_after_convert_extensions_virtual.jpg630×861 86.2 KB

And it show ext 125 is the problem. Like mentioned above, its always the lowest extension number.

Run yum update to see if it fixes the issue (it doesn’t)

step17-run_yum_update_attempt_to_fix.jpg1036×872 173 KB

step18-post_run_yum_update.jpg898×384 70.6 KB

After all this, ran: # fwconsole ma downloadinstall core --force
No repos specified, using: [standard] from last GUI settings

Downloading module ‘core’
Processing core
Downloading…
1111681/1111681 [============================] 100%
Finished downloading
Extracting…Done
Download completed in 5 seconds
Updating tables trunks, pjsip, sip, dahdi, iax, indications_zonelist, devices, users, incoming, outbound_routes, dahdichandids, outbound_route_patterns, outbound_route_sequence, outbound_route_trunks, outbound_routes, trunk_dialpatterns…Done
Migrating pickup groups to named pickup groups
Migrating call groups to named call groups
Checking for possibly invalid emergency caller id fields…none found
Migrating old media encryption values…done
Removing encoding on incoming routes alertinfo values…done
Generating CSS…Done
Module core successfully installed
Updating Hooks…Done

followed by: # fwconsole reload
Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Unable to continue. Invalid argument supplied for foreach() in /var/www/html/admin/modules/core/functions.inc/Driver.class.php on line 113
#0 /var/www/html/admin/modules/core/functions.inc/Driver.class.php(113): Whoops\Run->handleError(2, ‘Invalid argumen…’, ‘/var/www/html/a…’, 113, Array)
#1 /var/www/html/admin/modules/core/functions.inc/Driver.class.php(125): FreePBX\modules\Core\Driver::devicesGetUserMappings()
#2 /var/www/html/admin/modules/core/functions.inc.php(477): FreePBX\modules\Core\Driver::map_dev_user(‘125’, ‘callerid’, ‘device <125>’)
#3 /var/www/html/admin/modules/core/functions.inc.php(73): core_conf->generate_sip_additional(‘13.22.0’)
#4 /var/www/html/admin/libraries/BMO/FileHooks.class.php(65): core_conf->generateConf(‘sip_additional…’)
#5 /var/www/html/admin/libraries/BMO/FileHooks.class.php(24): FreePBX\FileHooks->processOldHooks(Array)
#6 /var/lib/asterisk/bin/retrieve_conf(877): FreePBX\FileHooks->processFileHooks(Array)
#7 {main}

Also removed modules in the new FreePBX 14 to match what was in the production (Donor) FreePBX 13. It made no difference whatsoever. Still get the same retrieve_conf error.

I would love to know what I am doing wrong here. The instructions at http://convert.freepbx.org/ do not mention anything remotely close to these issues.

Thanks for reading this. If you find a fix, please let me know.

Posts: 1

Participants: 1

Read full topic

@SethDM wrote:

I’ve been having an issue with one of my clients using FreePBX. They recently switched and they have a single handset that is having very occasional issues answering a call. Calls are dropped into a queue, generally there’s a single member of the queue and that’s the handset in question. This issue happened when the callers were dropped straight to this handset which resulted in callers hitting voicemail so the queue was my testing/bandaid solution so the handset has more than 1 opportunity to answer. What we’re finding is that when the issue occurs the handset WILL eventually be able to answer but usually after several attempts by the queue.

The latest instances were on 5/8 and 5/9. In both cases the CDR logs on the server are showing the queue rang and there was ‘no answer’ for several 15 second tries but the call was eventually answered. In the call log on the handset it just shows the answer; no missed calls or anything before the answer.

Handset is a Grandstream GXP2170 on the latest firmware. I thought it could be a bug or problem with the handset so I swapped that out. There are about 20 other handsets and none of them have reported similar issues. All the handsets are using OVPN to the off-site FreePBX server so we shouldn’t be looking at some weird firewall or NAT issue.

Anyone have any thoughts? I’ve been googling and poking at this issue for a couple of months and not really gotten anywhere on it. I think my next test is going to be swapping to a different handset model entirely with the thought that maybe I’m dealing with some firmware bug on that model since all the other users with GXP2135s haven’t reported issues.

Posts: 1

Participants: 1

Read full topic

@raymond wrote:

Hi Guys,

I’m trying to set up a MIXMON_POST in the Global conf to run a script but I’m not getting the extension number instead, I’m getting the queue number for INCOMING calls.

This is how the line looks like: MIXMON_POST = /usr/share/asterisk/agi-bin/moverec.sh ^{MIXMONITOR_FILENAME} “^{AMPUSER}” ^{CALLERID(number)} ^{CDR(dst)} ^{CDR(start)} ^{CDR(src)} ^{UNIQUEID} “^{AGENTEXTEN}”

Note: I’m able to get the extension for OUTGOING calls but there are no extensions for INCOMING calls.

Any help will be really appreciated

Thank you

Posts: 1

Participants: 1

Read full topic

@raymond wrote:

Hi Community,

What is the difference between ^{CDR(src)} and “^{CDR(src)}” ???

Thanks

Posts: 1

Participants: 1

Read full topic

@probegtze wrote:

Hello,

I have “time conditions” setup: between 9 and 17H30: calls go to a ring group. From 17H30 until 9AM next morning: the calls go to an announcement (business hours).

I was hoping to use the feature code (*271) to change the conditions when I’m outside of business hours to make the ring group enabled again. That is useful when I’m outside of business hours but I know someone is at the office and I want to join them. Unfortunately: the feature code *271 will only get the announcement to return to the beginning.

What am I doing wrong?

Thanks

Frank

Posts: 1

Participants: 1

Read full topic

@nobby6 wrote:

Hi,
Firstly, is there a way to import CSV, to add-to whats there? Or does the import wipe out everything existing?

Secondly, the export includes column names, must these remain in an import or should that first line be removed, my way of thinking is if this is going into a RDB my mysql/mariadb, then it should be deleted, or does bulk handler take care of this.

Just now getting ready to add over a thousand extensions (thank Larry for perl in making that part easy )

Posts: 1

Participants: 1

Read full topic