@mvogel4949 wrote:
I ran a restore using V14 and everything was great. Once I logged off however all visual signs of freepbx went away. All I can see is the UCP button but I’ve lost the 3 figures where you select admin to login. apache is up and running and I can still access the system via ssh. Any ideas what might have happened?
Posts: 2
Participants: 2
@Voicesense wrote:
I am trying to install FreePBX via the Azure Marketplace install on ubuntu. I am running into several challenges.
- After completing the install - where do I find the deployment Id or the serial number. Either is required to open a case with Sangroma?
- How is the product actually activated?
- I have attached a sip trunk but neither of my inbound or outbound calls have audio on them. When using a freepbx locally, I was able to verify on the network settings in sysadmin, which is not available here.
- Both my contact IP and external IP are yellow, and not green, assuming that is contributing to my issue.
Thanks,
Posts: 2
Participants: 2
@VoIPTek wrote:
I wanted to confirm if we agree this is a bug before adding it to bugs.
Upon updating an extension I get the following error, the information does get saved, but of course we have the error.Nothing has changed in some time, with exception of module updates.
Posts: 3
Participants: 2
@gatorHeel wrote:
We have a user reporting that their calls are disconnected after a few minutes whenever the call is transferred or parked. PJSIP trunk. In the example below, it occurred almost exactly at 5 minutes (talk time after answered from is about 4m 30s).
Noticed hangup code 19 in the logs. Any ideas?
Version: 12.7.6-2002-2.sng7
FreePBX 15.0.16.72Link to logs:
Posts: 2
Participants: 2
@SpeakerSeeker wrote:
Hello,
i read all network traffic on my FreePBX using tcpdump and i noticed the following:
After the PBX recieves an “Invite message” (the called sip-phone is online) it immediately sends two messages back to the caller, Trying and Ringing.
So now i ask, why is it sending back a ringing before it got a ringing from the called phone?
Until now i found out that this is maybe due to Asterisk doing inband-progress
i added “progressinband=no” or “progressinband=never” to sip_general_custom.conf but the problem remainsis there any setting in the gui for this?
Posts: 2
Participants: 2
@ddavis66 wrote:
Pretty much a newbie here on FreePBX - setting up a new system (Sangoma PBXact 60 ) - when leaving a voice message on the system and retrying to retrieve this - the system prompt says the voice message is 5 minutes in length even though the actual message is only seconds long. I am able to listen to the voice message - and then there is a long dial tone after the message itself.
Any ideas or direction is appreciated.
Posts: 2
Participants: 2
@Hawkeye wrote:
In CDR reports, see a lot of inbound calls answered by system fax, but see a lot that show up as HANGUP :
ReceiveFAX s [ext-fax] ANSWERED 01:40
The above looks perfectly normal.Two entries below from same callerID displaying HANGUP as the APP with different times.
|Tue, 25 Aug 2020 10:55| |Hangup| s [ext-fax]|ANSWERED|00:56
|Tue, 25 Aug 2020 10:51| |Hangup| s [ext-fax]|ANSWERED|00:23Perhaps someone can tell me why the Hangup is displayed and what might be causing the PBX to not receive the fax.
udptl loaded and working…
Module Description Use Count Status Support Level
udptl UDPTL 3 Running coreSystem is: FreePBX 15.0.16.72
Asterisk: 16.11.1
Thanks,.
Posts: 1
Participants: 1
@lbrenes wrote:
Hello Team,
I am kinda of a newbie with Freepbx and Asterisk, so far everything has been working fine, except from 1 extension that I cannot make outboud calls.
This extension es tied to my Grandstream ATA FXS port, that port is registered correctly, I can receive calls but if I try to make a call Internal/external I get a busy tone.
I used to ssh to my freepbx box and do: asterisk -vvvvv and that helped me to figure out what the problems were but it is not working anymore.
I am using:
FreePBX 15.0.16.72 and Asterisk 16.11.1So I don’t know how to access the data that asterisk -vvvv used to provide so I can share it with you all.
Some help would be great.
Regards,
Luis
Posts: 1
Participants: 1
@kfeen wrote:
So I have a PBX up and running and I’m registering phones remotely over the internet. I have ports 5060, 5160, and 10000-20000 open on the firewall. When I try to call an extension or say *43 I get no audio. From the logs it looks like the call goes through fine. Here is a sample output from the Asterisk log file. “WAN_ADDRESS” is the IP that my phone is located on and “192.168.1.48” is the phones LAN address.
[2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Got RTP packet from WAN_ADDRESS:12110 (type 00, seq 003583, ts 1024938288, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.48:12110 (type 00, seq 031285, ts 039680, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Got RTP packet from WAN_ADDRESS:12110 (type 00, seq 003584, ts 1024938448, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.48:12110 (type 00, seq 031286, ts 039840, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Got RTP packet from WAN_ADDRESS:12110 (type 00, seq 003585, ts 1024938608, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.48:12110 (type 00, seq 031287, ts 040000, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Got RTP packet from WAN_ADDRESS:12110 (type 00, seq 003586, ts 1024938768, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.48:12110 (type 00, seq 031288, ts 040160, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Got RTP packet from WAN_ADDRESS:12110 (type 00, seq 003587, ts 1024938928, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.48:12110 (type 00, seq 031289, ts 040320, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Got RTP packet from WAN_ADDRESS:12110 (type 00, seq 003588, ts 1024939088, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.48:12110 (type 00, seq 031290, ts 040480, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Got RTP packet from WAN_ADDRESS:12110 (type 00, seq 003589, ts 1024939248, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Sent RTP packet to 192.168.1.48:12110 (type 00, seq 031291, ts 040640, len 000160) [2020-08-25 22:00:25] VERBOSE[32003][C-000000f8] res_rtp_asterisk.c: Got RTP packet from WAN_ADDRESS:12110 (type 00, seq 003590, ts 1024939408, len 000160)
Posts: 5
Participants: 3
@tjthorson wrote:
Hello - this is related to a recent post I made here - Asterisk dies after odd error messages related to voice queue length . This issue still happens with this box. I have tried to watch top and check what is happening, but it happens very slowly over several days. I only see mysql, asterisk, and other normal things popping to the top of the list. Looks normal to me. If we restart asterisk (no need to restart the machine, its physical BTW) everything is fine for the day at least. I tried to setup a cron job to just restart asterisk at 3am figuring some future update will fix this problem - but I cant get the cron job to run. I first tried to set it up under the asterisk cron jobs. Then I tried under root… The line Im using in these cron jobs is -
0 3 * * * /var/lib/asterisk/bin/fwconsole restartWhen put into the root - its the only thing in the list. When put into Asterisk - this is the entire file if it helps point to why it isnt working…
I dont think we can resolve the memory issue altogether (Im going to give the client the option to build their backup box and move the backup over to see if its hardware/software) but in the meantime Id like to get this cron job working so we dont have to remote in every morning and do this manually.
Posts: 4
Participants: 3
@threeeye wrote:
Hi guys,
I have a problem with 1 of my extensions.
When calling in to that ext (2067), everything works good.
But, When calling out from that ext, the call disconnects after 31 sec, like when it can’t connect to RTP port - but I have audio both ways…
#asterisk -vvvvvvvvvvvr | grep 2067
– Executing [2033@from-internal:1] GotoIf(“PJSIP/2067-000008c0”, “1?ext-local,2033,1:followme-check,2033,1”) in new stack
– Executing [2033@ext-local:1] Set(“PJSIP/2067-000008c0”, “__RINGTIMER=25”) in new stack
– Executing [2033@ext-local:2] ExecIf(“PJSIP/2067-000008c0”, “0?Set(__CWIGNORE=)”) in new stack
– Executing [2033@ext-local:3] Macro(“PJSIP/2067-000008c0”, “exten-vm,2033,2033,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“PJSIP/2067-000008c0”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/2067-000008c0”, “TOUCH_MONITOR=1598472689.31647”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/2067-000008c0”, “AMPUSER=2067”) in new stack
– Executing [s@macro-user-callerid:3] Set(“PJSIP/2067-000008c0”, “HOTDESCKCHAN=2067-000008c0”) in new stack
– Executing [s@macro-user-callerid:4] Set(“PJSIP/2067-000008c0”, “HOTDESKEXTEN=2067”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/2067-000008c0”, “HOTDESKCALL=0”) in new stack
– Executing [s@macro-user-callerid:6] ExecIf(“PJSIP/2067-000008c0”, “0?Set(HOTDESKCALL=1)”) in new stack
– Executing [s@macro-user-callerid:7] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CALLERID(name)=)”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“PJSIP/2067-000008c0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] ExecIf(“PJSIP/2067-000008c0”, “1?Set(REALCALLERIDNUM=2067)”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/2067-000008c0”, “AMPUSER=2067”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“PJSIP/2067-000008c0”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/2067-000008c0”, “AMPUSERCIDNAME=Rivka Stein”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“PJSIP/2067-000008c0”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“PJSIP/2067-000008c0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:15] Set(“PJSIP/2067-000008c0”, “AMPUSERCID=2067”) in new stack
– Executing [s@macro-user-callerid:16] Set(“PJSIP/2067-000008c0”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:17] Set(“PJSIP/2067-000008c0”, “CALLERID(all)=“Rivka Stein” <2067>”) in new stack
– Executing [s@macro-user-callerid:18] Set(“PJSIP/2067-000008c0”, “HOTDESCKCHAN=2067-000008c0”) in new stack
– Executing [s@macro-user-callerid:19] Set(“PJSIP/2067-000008c0”, “HOTDESKEXTEN=2067”) in new stack
– Executing [s@macro-user-callerid:20] Set(“PJSIP/2067-000008c0”, “HOTDESKCALL=0”) in new stack
– Executing [s@macro-user-callerid:21] ExecIf(“PJSIP/2067-000008c0”, “0?Set(HOTDESKCALL=1)”) in new stack
– Executing [s@macro-user-callerid:22] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CALLERID(name)=)”) in new stack
– Executing [s@macro-user-callerid:23] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CALLERID(all)=”" < >)") in new stack
– Executing [s@macro-user-callerid:24] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CALLERID(all)=2033)”) in new stack
– Executing [s@macro-user-callerid:25] GotoIf(“PJSIP/2067-000008c0”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:26] ExecIf(“PJSIP/2067-000008c0”, “0?Set(GROUP(concurrency_limit)=2067)”) in new stack
– Executing [s@macro-user-callerid:27] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:28] NoOp(“PJSIP/2067-000008c0”, “Macro Depth is 2”) in new stack
– Executing [s@macro-user-callerid:29] GotoIf(“PJSIP/2067-000008c0”, “1?report2:macroerror”) in new stack
– Executing [s@macro-user-callerid:30] GotoIf(“PJSIP/2067-000008c0”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:31] ExecIf(“PJSIP/2067-000008c0”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
– Executing [s@macro-user-callerid:32] Set(“PJSIP/2067-000008c0”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:33] GotoIf(“PJSIP/2067-000008c0”, “1?continue”) in new stack
– Executing [s@macro-user-callerid:49] Set(“PJSIP/2067-000008c0”, “CALLERID(number)=2067”) in new stack
– Executing [s@macro-user-callerid:50] Set(“PJSIP/2067-000008c0”, “CALLERID(name)=Rivka Stein”) in new stack
– Executing [s@macro-user-callerid:51] GotoIf(“PJSIP/2067-000008c0”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:52] Set(“PJSIP/2067-000008c0”, “CDR(cnam)=Rivka Stein”) in new stack
– Executing [s@macro-user-callerid:53] Set(“PJSIP/2067-000008c0”, “CDR(cnum)=2067”) in new stack
– Executing [s@macro-user-callerid:54] Set(“PJSIP/2067-000008c0”, “CHANNEL(language)=en”) in new stack
– Executing [s@macro-user-callerid:55] GosubIf(“PJSIP/2067-000008c0”, “0?app-check-classofservce,s,1()”) in new stack
– Executing [s@macro-exten-vm:2] Set(“PJSIP/2067-000008c0”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“PJSIP/2067-000008c0”, “__EXTTOCALL=2033”) in new stack
– Executing [s@macro-exten-vm:4] Set(“PJSIP/2067-000008c0”, “__PICKUPMARK=2033”) in new stack
– Executing [s@macro-exten-vm:5] Set(“PJSIP/2067-000008c0”, “RT=25”) in new stack
– Executing [s@macro-exten-vm:6] ExecIf(“PJSIP/2067-000008c0”, “0?Macro(vm,2033,DIRECTDIAL,)”) in new stack
– Executing [s@macro-exten-vm:7] ExecIf(“PJSIP/2067-000008c0”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:8] ExecIf(“PJSIP/2067-000008c0”, “0?Gosub(ext-intercom,*802033,1())”) in new stack
– Executing [s@macro-exten-vm:9] ExecIf(“PJSIP/2067-000008c0”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:10] ExecIf(“PJSIP/2067-000008c0”, “0?ChanSpy(SIP/2033,q)”) in new stack
– Executing [s@macro-exten-vm:11] ExecIf(“PJSIP/2067-000008c0”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:12] ExecIf(“PJSIP/2067-000008c0”, “0?Macro(vm,2033,DIRECTDIAL,)”) in new stack
– Executing [s@macro-exten-vm:13] ExecIf(“PJSIP/2067-000008c0”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:14] ExecIf(“PJSIP/2067-000008c0”, “0?Gosub(ext-intercom,*802033,1())”) in new stack
– Executing [s@macro-exten-vm:15] ExecIf(“PJSIP/2067-000008c0”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:16] ExecIf(“PJSIP/2067-000008c0”, “0?ChanSpy(SIP/2033,q)”) in new stack
– Executing [s@macro-exten-vm:17] ExecIf(“PJSIP/2067-000008c0”, “0?MacroExit()”) in new stack
– Executing [s@macro-exten-vm:18] Gosub(“PJSIP/2067-000008c0”, “sub-record-check,s,1(exten,2033,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“PJSIP/2067-000008c0”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“PJSIP/2067-000008c0”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“PJSIP/2067-000008c0”, “NOW=1598472689”) in new stack
– Executing [s@sub-record-check:4] Set(“PJSIP/2067-000008c0”, “__DAY=26”) in new stack
– Executing [s@sub-record-check:5] Set(“PJSIP/2067-000008c0”, “__MONTH=08”) in new stack
– Executing [s@sub-record-check:6] Set(“PJSIP/2067-000008c0”, “__YEAR=2020”) in new stack
– Executing [s@sub-record-check:7] Set(“PJSIP/2067-000008c0”, “__TIMESTR=20200826-161129”) in new stack
– Executing [s@sub-record-check:8] Set(“PJSIP/2067-000008c0”, “__FROMEXTEN=2067”) in new stack
– Executing [s@sub-record-check:9] Set(“PJSIP/2067-000008c0”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“PJSIP/2067-000008c0”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“PJSIP/2067-000008c0”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“PJSIP/2067-000008c0”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“PJSIP/2067-000008c0”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“PJSIP/2067-000008c0”, “5?checkaction”) in new stack
– Executing [s@sub-record-check:17] GotoIf(“PJSIP/2067-000008c0”, “1?sub-record-check,exten,1”) in new stack
– Executing [exten@sub-record-check:1] NoOp(“PJSIP/2067-000008c0”, “Exten Recording Check between 2067 and 2033”) in new stack
– Executing [exten@sub-record-check:2] Set(“PJSIP/2067-000008c0”, “CALLTYPE=internal”) in new stack
– Executing [exten@sub-record-check:3] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CALLTYPE=)”) in new stack
– Executing [exten@sub-record-check:4] Set(“PJSIP/2067-000008c0”, “CALLEE=dontcare”) in new stack
– Executing [exten@sub-record-check:5] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CALLEE=dontcare)”) in new stack
– Executing [exten@sub-record-check:6] GotoIf(“PJSIP/2067-000008c0”, “0?callee”) in new stack
– Executing [exten@sub-record-check:7] GotoIf(“PJSIP/2067-000008c0”, “1?caller”) in new stack
– Executing [exten@sub-record-check:13] Set(“PJSIP/2067-000008c0”, “RECMODE=dontcare”) in new stack
– Executing [exten@sub-record-check:14] ExecIf(“PJSIP/2067-000008c0”, “0?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:15] ExecIf(“PJSIP/2067-000008c0”, “1?Set(RECMODE=dontcare)”) in new stack
– Executing [exten@sub-record-check:16] Gosub(“PJSIP/2067-000008c0”, “recordcheck,1(dontcare,internal,2033)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“PJSIP/2067-000008c0”, “Starting recording check against dontcare”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“PJSIP/2067-000008c0”, “dontcare”) in new stack
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/2067-000008c0”, “”) in new stack
– Executing [exten@sub-record-check:17] Return(“PJSIP/2067-000008c0”, “”) in new stack
– Executing [s@macro-exten-vm:19] GotoIf(“PJSIP/2067-000008c0”, “1?macrodial”) in new stack
– Executing [s@macro-exten-vm:25] GosubIf(“PJSIP/2067-000008c0”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:26] Macro(“PJSIP/2067-000008c0”, “dial-one,25,Ttr,2033”) in new stack
– Executing [s@macro-dial-one:1] Set(“PJSIP/2067-000008c0”, “DEXTEN=2033”) in new stack
– Executing [s@macro-dial-one:2] ExecIf(“PJSIP/2067-000008c0”, “0?Set(__EXTTOCALL=2033)”) in new stack
– Executing [s@macro-dial-one:3] Set(“PJSIP/2067-000008c0”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“PJSIP/2067-000008c0”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:5] GosubIf(“PJSIP/2067-000008c0”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:6] GotoIf(“PJSIP/2067-000008c0”, “1?skip1”) in new stack
– Executing [s@macro-dial-one:9] GotoIf(“PJSIP/2067-000008c0”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:10] GotoIf(“PJSIP/2067-000008c0”, “0?continue”) in new stack
– Executing [s@macro-dial-one:11] Set(“PJSIP/2067-000008c0”, “EXTHASCW=ENABLED”) in new stack
– Executing [s@macro-dial-one:12] GotoIf(“PJSIP/2067-000008c0”, “0?next1:cwinusebusy”) in new stack
– Executing [s@macro-dial-one:24] GotoIf(“PJSIP/2067-000008c0”, “0?next3:continue”) in new stack
– Executing [s@macro-dial-one:26] GotoIf(“PJSIP/2067-000008c0”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:27] GosubIf(“PJSIP/2067-000008c0”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“PJSIP/2067-000008c0”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“PJSIP/2067-000008c0”, “DEVICES=2033”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“PJSIP/2067-000008c0”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“PJSIP/2067-000008c0”, “0?Set(DEVICES=033)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“PJSIP/2067-000008c0”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“PJSIP/2067-000008c0”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“PJSIP/2067-000008c0”, “THISDIAL=SIP/2033”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“PJSIP/2067-000008c0”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“PJSIP/2067-000008c0”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“PJSIP/2067-000008c0”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“PJSIP/2067-000008c0”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“PJSIP/2067-000008c0”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“PJSIP/2067-000008c0”, “THISPART2=SIP/2033”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“PJSIP/2067-000008c0”, “0?Set(THISPART2=DAHDI/2033)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“PJSIP/2067-000008c0”, “NEWDIAL=SIP/2033&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“PJSIP/2067-000008c0”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“PJSIP/2067-000008c0”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“PJSIP/2067-000008c0”, “THISDIAL=SIP/2033”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“PJSIP/2067-000008c0”, “”) in new stack
– Executing [dstring@macro-dial-one:9] GotoIf(“PJSIP/2067-000008c0”, “1?docheck”) in new stack
– Executing [dstring@macro-dial-one:15] GotoIf(“PJSIP/2067-000008c0”, “0?skipset”) in new stack
– Executing [dstring@macro-dial-one:16] Set(“PJSIP/2067-000008c0”, “DSTRING=SIP/2033&”) in new stack
– Executing [dstring@macro-dial-one:17] Set(“PJSIP/2067-000008c0”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:18] GotoIf(“PJSIP/2067-000008c0”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:19] ExecIf(“PJSIP/2067-000008c0”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:20] Set(“PJSIP/2067-000008c0”, “DSTRING=SIP/2033”) in new stack
– Executing [dstring@macro-dial-one:21] Return(“PJSIP/2067-000008c0”, “”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“PJSIP/2067-000008c0”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:29] GotoIf(“PJSIP/2067-000008c0”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:30] GosubIf(“PJSIP/2067-000008c0”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“PJSIP/2067-000008c0”, “DB(CALLTRACE/2033)=2067”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“PJSIP/2067-000008c0”, “”) in new stack
– Executing [s@macro-dial-one:31] Set(“PJSIP/2067-000008c0”, “D_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dial-one:32] GosubIf(“PJSIP/2067-000008c0”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:33] NoOp(“PJSIP/2067-000008c0”, "Blind Transfer: , Attended Transfer: , User: 2067, Alert Info: ") in new stack
– Executing [s@macro-dial-one:34] ExecIf(“PJSIP/2067-000008c0”, “1?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:35] ExecIf(“PJSIP/2067-000008c0”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:36] ExecIf(“PJSIP/2067-000008c0”, “0?Set(ALERT_INFO=)”) in new stack
– Executing [s@macro-dial-one:37] ExecIf(“PJSIP/2067-000008c0”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:38] ExecIf(“PJSIP/2067-000008c0”, “0?Set(ALERT_INFO=Normal;volume=)”) in new stack
– Executing [s@macro-dial-one:39] GosubIf(“PJSIP/2067-000008c0”, “0?func-set-sipheader,s,1(Alert-Info,)”) in new stack
– Executing [s@macro-dial-one:40] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:41] GosubIf(“PJSIP/2067-000008c0”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:42] Set(“PJSIP/2067-000008c0”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:43] Set(“PJSIP/2067-000008c0”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:44] GotoIf(“PJSIP/2067-000008c0”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:45] GotoIf(“PJSIP/2067-000008c0”, “0?godial”) in new stack
– Executing [s@macro-dial-one:46] Gosub(“PJSIP/2067-000008c0”, “sub-presencestate-display,s,1(2033)”) in new stack
– Executing [s@sub-presencestate-display:1] Goto(“PJSIP/2067-000008c0”, “state-not_set,1”) in new stack
– Executing [state-not_set@sub-presencestate-display:1] Set(“PJSIP/2067-000008c0”, “PRESENCESTATE_DISPLAY=”) in new stack
– Executing [state-not_set@sub-presencestate-display:2] Return(“PJSIP/2067-000008c0”, “”) in new stack
– Executing [s@macro-dial-one:47] Set(“PJSIP/2067-000008c0”, “CONNECTEDLINE(name,i)=Sholem Kleinman”) in new stack
– Executing [s@macro-dial-one:48] Set(“PJSIP/2067-000008c0”, “CONNECTEDLINE(num)=2033”) in new stack
– Executing [s@macro-dial-one:49] Set(“PJSIP/2067-000008c0”, “D_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dial-one:50] Macro(“PJSIP/2067-000008c0”, “dialout-one-predial-hook,”) in new stack
– Executing [s@macro-dialout-one-predial-hook:1] MacroExit(“PJSIP/2067-000008c0”, “”) in new stack
– Executing [s@macro-dial-one:51] ExecIf(“PJSIP/2067-000008c0”, “0?Set(D_OPTIONS=trI)”) in new stack
– Executing [s@macro-dial-one:52] NoOp(“PJSIP/2067-000008c0”, “”) in new stack
– Executing [s@macro-dial-one:53] ExecIf(“PJSIP/2067-000008c0”, “0?Set(D_OPTIONS=Ttrg)”) in new stack
– Executing [s@macro-dial-one:54] Dial(“PJSIP/2067-000008c0”, “SIP/2033,25,Ttrb(func-apply-sipheaders^s^1)”) in new stack
– SIP/2033-00004a61 answered PJSIP/2067-000008c0
– Channel PJSIP/2067-000008c0 joined ‘simple_bridge’ basic-bridge <16636199-b783-42c3-8548-150952474544>
– Channel PJSIP/2067-000008c0 left ‘simple_bridge’ basic-bridge <16636199-b783-42c3-8548-150952474544>
== Spawn extension (macro-dial-one, s, 54) exited non-zero on ‘PJSIP/2067-000008c0’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘PJSIP/2067-000008c0’ in macro ‘exten-vm’
== Spawn extension (ext-local, 2033, 3) exited non-zero on ‘PJSIP/2067-000008c0’
– Executing [h@ext-local:1] Macro(“PJSIP/2067-000008c0”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/2067-000008c0”, “1?theend”) in new stack
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/2067-000008c0”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/2067-000008c0”, "SIP/2033-00004a61 montior file= ") in new stack
– Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/2067-000008c0”, “1?skipagi”) in new stack
– Executing [s@macro-hangupcall:7] Hangup(“PJSIP/2067-000008c0”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/2067-000008c0’ in macro ‘hangupcall’
== Spawn extension (ext-local, h, 1) exited non-zero on ‘PJSIP/2067-000008c0’
Posts: 1
Participants: 1
@GSnover wrote:
Here is my Outbound trunk definition:
username=UserName
fromdomain=a1west.sipregistration.com
type=friend
trustrpid=yes
sendrpid=yes
secret=Squirel
qualify=yes
nat=no
host=a1west.sipregistration.com
dtmfmode=rfc2833
canreinvite=noHere is the relevant Invite Header:
Why is that second From: string there, and why is it a variable substitution that hasn’t happened? The folks on the far end think it’s what the Access SBC is not happy with - anybody know why it is sending a second From: line?
Thanks!
Posts: 1
Participants: 1
@AllianceDoug wrote:
We changed our ISP, so we got a new WAN IP address.
The new WAN IP was entered in Settings -> Asterisk SIP Settings -> General SIP Settings tab -> External Address. Actually the "Detect Network Settings button pulled the correct new WAN IP. I hit Submit and Apply Config. No other settings were changed. No issues existed on the old WAN IP.
The new WAN IP was entered in the SIP provider’s website (Twilio). We can make and receive calls just fine, but all incoming calls (Originating) drop right at 32 seconds, every time.
What are all the places I needed to update with a new WAN IP address? Maybe I missed somewhere.
The Twilio call properties for this call says: “Who Hung Up: callee”
Thank you.
Here are the Asterisk Log Files for the beginning of the call:
[2020-08-07 15:26:32] WARNING[45157] iax2/firmware.c: Error opening firmware directory '/var/lib/asterisk/firmware/iax': No such file or directory [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP)) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'chan_dahdi.so' (DAHDI Telephony w/PRI & SS7 & MFC/R2) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'chan_motif.so' (Motif Jingle Channel Driver) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_adsi.so' (ADSI Resource) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_fax.so' (Generic FAX Applications) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_ari.so' (Asterisk RESTful Interface) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_pjsip_outbound_registration.so' (PJSIP Outbound Registration Support) [2020-08-07 15:26:32] ERROR[45157] res_pjsip_config_wizard.c: Unable to load config file 'pjsip_wizard.conf' [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'app_confbridge.so' (Conference Bridge Application) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_parking.so' (Call Parking Resource) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/71/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/72/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/73/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/74/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/75/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/76/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/77/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/78/1, registrar=res_parking/default; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] pbx.c: Remove parkedcalls/70/1, registrar=res_parking; con=<nil>((nil)); con->root=(nil) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'app_meetme.so' (MeetMe conference bridge) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'cel_manager.so' (Asterisk Manager Interface CEL Backend) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'cel_odbc.so' (ODBC CEL backend) [2020-08-07 15:26:32] VERBOSE[2502] chan_sip.c: Reloading SIP [2020-08-07 15:26:32] VERBOSE[2502] netsock2.c: Using SIP TOS bits 96 [2020-08-07 15:26:32] VERBOSE[2502] netsock2.c: Using SIP CoS mark 4 [2020-08-07 15:26:32] VERBOSE[45157] cel_odbc.c: Found CEL table cel@asteriskcdrdb. [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'app_amd.so' (Answering Machine Detection Application) [2020-08-07 15:26:32] VERBOSE[45157] app_amd.c: AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] maximumWordLength [5000] [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'app_playback.so' (Sound File Playback Application) [2020-08-07 15:26:32] VERBOSE[45157] loader.c: Reloading module 'res_digium_phone.so' (Digium Phone Module for Asterisk) [2020-08-07 15:26:33] VERBOSE[45157] loader.c: Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System)) [2020-08-07 15:26:33] WARNING[45157] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages [2020-08-07 15:26:33] VERBOSE[45157] loader.c: Reloading module 'codec_dahdi.so' (Generic DAHDI Transcoder Codec Translator) [2020-08-07 15:26:33] VERBOSE[45157] loader.c: Reloading module 'app_queue.so' (True Call Queueing) [2020-08-07 15:26:33] VERBOSE[45157] asterisk.c: Remote UNIX connection disconnected [2020-08-07 15:26:34] VERBOSE[6241] pbx_variables.c: Setting global variable 'SIPDOMAIN' to '136.228.122.109' [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [+14055737881@from-pstn-e164-us:1] Set("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "CALLERID(number)=4054010812") in new stack [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [+14055737881@from-pstn-e164-us:2] Goto("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "from-pstn,4055737881,1") in new stack [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx_builtins.c: Goto (from-pstn,4055737881,1) [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [4055737881@from-pstn:1] NoOp("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "Catch-All DID Match - Found 4055737881 - You probably want a DID for this.") in new stack [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [4055737881@from-pstn:2] Set("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "__FROM_DID=4055737881") in new stack [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [4055737881@from-pstn:3] Goto("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "ext-did,s,1") in new stack [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx_builtins.c: Goto (ext-did,s,1) [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [s@ext-did:1] Set("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "__DIRECTION=INBOUND") in new stack [2020-08-07 15:26:34] VERBOSE[45274][C-00000010] pbx.c: Executing [s@ext-did:2] Gosub("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "sub-record-check,s,1(in,s,no)") in new stackHere are the Asterisk Log Files for the very end of the call:
[2020-08-07 15:27:15] VERBOSE[45274][C-00000010] bridge_channel.c: Channel PJSIP/Twilio_PJSip_US2_Oregon-00000030 left 'simple_bridge' basic-bridge <bca56c76-ab91-4c39-93f4-0e0cd354c94a> [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] app_macro.c: Spawn extension (macro-dial, s, 53) exited non-zero on 'PJSIP/Twilio_PJSip_US2_Oregon-00000030' in macro 'dial' [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Spawn extension (ext-group, 602, 22) exited non-zero on 'PJSIP/Twilio_PJSip_US2_Oregon-00000030' [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [h@ext-group:1] Macro("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "hangupcall,") in new stack [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "1?theend") in new stack [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx_builtins.c: Goto (macro-hangupcall,s,3) [2020-08-07 15:27:15] VERBOSE[45364][C-00000010] bridge_channel.c: Channel PJSIP/202-00000032 left 'simple_bridge' basic-bridge <bca56c76-ab91-4c39-93f4-0e0cd354c94a> [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "0?Set(CDR(recordingfile)=)") in new stack [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [s@macro-hangupcall:4] NoOp("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "PJSIP/202-00000032 montior file= ") in new stack [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [s@macro-hangupcall:5] GotoIf("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "1?skipagi") in new stack [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx_builtins.c: Goto (macro-hangupcall,s,7) [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Executing [s@macro-hangupcall:7] Hangup("PJSIP/Twilio_PJSip_US2_Oregon-00000030", "") in new stack [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/Twilio_PJSip_US2_Oregon-00000030' in macro 'hangupcall' [2020-08-07 15:27:15] VERBOSE[45274][C-00000010] pbx.c: Spawn extension (ext-group, h, 1) exited non-zero on 'PJSIP/Twilio_PJSip_US2_Oregon-00000030'
Posts: 4
Participants: 2
@Mesh wrote:
How can I check what audio codec was used during a session? Asterisk and a SIP phone have a choice of cedecs to use. But is there a way to find which codec they actually used for a particular connection?
Is there a log file with this info? I ran Asterisk in verbose but still could’t see codecs in any of the in FreePBX’s Asterisk Log Files .
Posts: 1
Participants: 1
@djgaidel wrote:
Hi,
I am having problems removing the " Person at extension *** is unavailable" before going to ext/voicemail.
I have tried disabling this already via dial plan in VM admin but still its present
Posts: 2
Participants: 2
@leo85 wrote:
Hello guys,
Having new VoIP challenge today.
How to make softphone work in background – Zoiper 5 specifically, even when device rebooted, or app never started (like after OS reboot).
Interested in iOS or Android.It seems that an extension I did test today, Asterisk transport was UDP (they recommend it to be TCP). It caused app to work only when app is on the screen, but when minimized no joy, it was iOS.
Is it possible to achieve near 100% app readiness in the WLAN with their Push Proxy paid service? Or, share how you work around this!
Thank you!
Posts: 1
Participants: 1
@davids1 wrote:
Hi,
I am trying to monitor a problematic firewall on a fully up to date FreePBX 15.0.16.72 virtual machine but unsure how to do so.
I am using a CheckMK agent to monitor lots of items including Systemd Service SummaryEvery few days we start getting hundreds of Fail2Ban emails and when we log into the PBX we see the System Firewall is disabled. CheckMK [Systemd Service Summary] still shows the same information as when the firewall was enabled.
OK - 108 services in total, 15 disabled servicesI have read other forum ports about checking the firewall and iptables service so I have started trying to run manual commands to see what results I get.
On a fresh reboot of FreePBX, it shows the System Firewall and Firewall Config both have green ticks in the dashboard, however if I run these commands I get the results below them
service firewalld status
Active: inactive (dead)service iptables status
Active: inactive (dead)service fail2ban status
Active: active (running)service freepbx status
Active: active (exited)Q1 - How can firewalld and iptables be inactive (dead) if the firewall is working correctly?
fwconsole firewall stop - the firewall stops and the pbx is accessable
fwconsole firewall start - the firewall starts and the pbx is secured again
service firewalld status - Active: inactive (dead)
service iptables status - Active: inactive (dead)If I disable and enable the firewall via the GUI then we get -
service firewalld status - Active: inactive (dead)
service iptables status - Active: active (exited)if I run service firewalld stop
and then service firewalld start
Redirecting to /bin/systemctl start firewalld.service
we get
Firewall Rules corrupted! Restarting in 5 seconds
More information available in /tmp/firewall.logBut finally we have
service firewalld status - Active: active (running)When I look at the firewall log I can see
'Firewall Rules corrupted! Restarting in 5 seconds
No fpbx-rtp in ipv6Could this be why the firewall keeps disabling itself every few days? Sorry im a bit stuck.
I have lots of other FreePBXs and PBXacts running with no issues and I may even just rebuild this one, but I would like to learn how to fix it rather than just rebuild.
Any help would be greatly appreciated,
Sorry this is a bit rushed, I can add more detail if needed in the next few days.
Thx
Dave
Posts: 5
Participants: 2
@FOGG wrote:
Hi freePBX community,
We’ve been receiving “Critical Storage Alerts” hourly for several weeks. I have no idea where to find the drive /dev/sda3. Can someone please point me in the right direction?
Posts: 2
Participants: 2
@dan_ce wrote:
DIY/Home project
Is this possible with FreePBX? I have some old rotary phones with unreliable pulse to tone converters fitted and I’d like to get around the problem of them not really working by implementing speak to dial?Is this possible? I think not, but thought it worth an ask.
Thanks
Posts: 3
Participants: 3
@dmaxti wrote:
Hi,
For the first time in any of the hundreds of FreePBX deployments I’ve managed, I am unable to get the CLI to show any of the normal verbose messages for debug purposes.
core set verbose ___ has no impact
Tried so far:
reload asterisk
switch asterisk version (cycled between 13, 16, and 17)
reboot machineThe system is otherwise asymptomatic. log files exist in /var/log/asterisk/ under full.0 and full-date ,but there’s no current full file. The activity ends yesterday afternoon.
Has anyone seen this before and do you have a resolution?
Posts: 2
Participants: 2
