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401 Unauthorized after REGISTER received

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@ls21ann wrote:

Hi

I installed latest version of FreePBX: Stable 10.13.66

If Asterisk receives a REGISTER from Subscriber, Asterisk answers with
401 Unauthorized

In Settings / Advanced Settings under Device Settings
I switched "Require Strong Secrets" to "No"
No change in behavior. Asterisk still send "401 Unauthorized"

In Applications / Extensions / Add Extension / Add New Custom Extension
I generated a new "User Extension" and let the "Password For New User" empty
I klick on Submit and
I get an exception: "Password can not be blank"

Is it not possible to switch password checking off ?
Is there an additional switch somewhere ?

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Upgrade to 10.13.66-13 hangs

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@tebolden wrote:

Attempting to upgrade from 10.13.66-12 to -13 hangs on

Running PHP script "/var/www/html/admin/modules/framework/upgrades//13.0.143/upgrade.php"

This script just keeps spawning "php /usr/sbin/fwconsole ma install framework --force" processes until the system runs out of memory and the script crashes.

freepbx.log show the following entries over and over again:

2016-Jul-06 12:55:07] INFO - installing files to /var/www/html..done
[2016-Jul-06 12:55:07] [INFO] (framework/install.php:102) - installing files to /var/lib/asterisk/bin..done
[2016-Jul-06 12:55:07] [INFO] (framework/install.php:113) - installing files to /var/lib/asterisk/agi-bin..done
[2016-Jul-06 12:55:07] [INFO] (installlib/installer.class.php:17) - Checking for upgrades..
[2016-Jul-06 12:55:07] [INFO] (installlib/installer.class.php:17) - 1 found
[2016-Jul-06 12:55:07] [INFO] (installlib/installer.class.php:17) - Upgrading to 13.0.143..
[2016-Jul-06 12:55:07] [INFO] (installlib/installer.class.php:17) - -> Running PHP script /var/www/html/admin/modules/framework/upgrades//13.0.143/upgrade.php
[2016-Jul-06 12:55:08] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 52
[2016-Jul-06 12:55:08] [WARNING] (libraries/modulefunctions.legacy.php:7) - Depreciated Function module_getinfo detected in /var/www/html/admin/modules/cxpanel/functions.inc.php on line 61
[2016-Jul-06 12:55:10] [INFO] (framework/install.php:90) - installing files to /var/www/html..done
[2016-Jul-06 12:55:10] [INFO] (framework/install.php:102) - installing files to /var/lib/asterisk/bin..done
[2016-Jul-06 12:55:10] [INFO] (framework/install.php:113) - installing files to /var/lib/asterisk/agi-bin..done
[2016-Jul-06 12:55:10] [INFO] (installlib/installer.class.php:17) - Checking for upgrades..
[2016-Jul-06 12:55:10] [INFO] (installlib/installer.class.php:17) - 1 found
[2016-Jul-06 12:55:10] [INFO] (installlib/installer.class.php:17) - Upgrading to 13.0.143..
[2016-Jul-06 12:55:10] [INFO] (installlib/installer.class.php:17) - -> Running PHP script /var/www/html/admin/modules/framework/upgrades//13.0.143/upgrade.php

I assume everytime it spawns a new process it writes the same lines to the log.

I have already tried to manually install the framework module using:

fwconsole ma download framework
fwconsole ma install framework

with the same results. The dashboard shows the framework module as disabled.

How do I get past this issue?

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CentOS 7 Distro

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@xrobau wrote:

For those that have been watching Git (and who hasn't, everyone loves looking at commits, right?) you would have seen some commits hinting at the release of the new FreePBX Distro 7.

This is getting pretty close to being available for beta, and we've been using it internally for a while.

I expect that, barring anything catastrophic, there will be the first public release very soon - early to middle of next week.

There's a couple of changes from our previous Distros that I'd like to mention, that will probably get filled out more in a blog post, but here's a couple of the cool new hotness things that I'm happy about

  1. No more FreePBX Distro Updater scripts. It's just 'yum update'. Always. You can also 'yum downgrade', too. (This doesn't change FreePBX's module versions, as per normal. This is just Distro)
  2. Complete UEFI support
  3. Serial and USB installs just work. (In fact, it's much faster to install from USB than from ISO now -- so much so that installing from USB may be the recommended method of installation, with ISOs as a fallback)
  4. Better development environment - If you want to develop FreePBX, you can just run 'yum install freepbx-devel' and that gets the machine ready for you.
  5. Behind the scenes, all RPM updates are automated, making it a lot easier for us to push out fixes without needing to run 15 different steps to push repos out to all the CDNs.
  6. PHP 5.6 and FreePBX 14

There are a couple of downsides. For one, there will no longer be any i686 support. 64 bit only. Being that it's been impossible to buy 32 bit CPUs for almost 10 years now, we think that's fair. Secondly, as this is a beta release, it will probably be broken in horrible and arcane ways. It may even kick your cat. No promises.

If you have any questions or suggestions, this is the place to post them!

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My calls are not getting recorded in cdr report

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@manjunath wrote:

Hii guys , Im using free pbx distro , I found the my calls are not getting recorded in the cdr reports it look empty , and I can see my previous call reports , Can you guys help me ...............

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Need to call using conference bridge Bridge

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@manjunath wrote:

Hii guys .... is there any way to make a outbound call using conference bridge , ie ., I should be able to make calls to my users, using the conference bridge in order to make a conference call ....

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WebRTC, TLS, SRTP, DTLS and so on

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@madman91 wrote:

Hello everyone. I'm totally confused about configuring WebRTC and TLS/SRTP/DTLS.

I've added certificate on my FreePBX distro. I somehow managed to get WebRTC in UCP working.
After this my softphones (3Cx versioon 6, Eyebeam 1.5.19) lost capability to make calls.

if SRTP in extension settings is YES and eyebeam signalling transport is "prefer to make and accept encrypted calls" = Insufficient information in SDP (c=)...
if SRTP in extension settings is YES, but eyebeam signalling transport is "make normal calls, accept all calls" =
chan_sip.c:10715 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio.

Same when SRTP encription in extension settings is NO

Later i somehow lost working webrtc in UCP, again that red-yellow blinking phone sign...

I enabled TLS in asterisk mini-http server, i enabled tlsv1 in sip settings. = those are for webRTC, right?

but i dont know when to enable/disable these:
AVPF
AVP
Support
SRTP Encryption
DTLS-SRTP

Were you guys able to get UCP/WEBrtc and softphones working?
Question: how to set up WEBrtc without broking anything which is already configured.

P.S. I have a "palmmicro" sip phone with no DTSL or SRTP enabled, it works fine make and accept calls from other sip phones or webrtc. When i answer call from softphone on that phone, the call is established. but when i try to answer call from phone on softphone : Insufficient information in SDP (c=)...
Softphone configured to allow normal and encrypted calls.

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Distro 7 almost out?

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@dbayer wrote:

Hello,

I just saw on the module admin, that the firewall has added support for Distro 7 and FreePBX 14. Does this mean they are about to be released?

I'm in the process of switching to 13, but if 14 is about to come out, I'd be happy to wait a little while.

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What happens if you skip updates?

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@DaileyComputer wrote:

Hey there. So I recently upgraded one of my systems from the 6.12.66 track to the 10.13.66 track. I then went straight from 10.13.66-1 to 10.13.66-13. I wasn't thinking straight that night I guess.

So far the system seems to be working ok. What are the potential problems or downsides? Is there a way to recover from this? Maybe go back and run 10.13.66-2, then each step back up until I get to 13 again? I don't even know if the scripts will let me do that.

Should I back up the config, reinstall, do all the upgrades then restore the config?

Thanks for any advice.

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[SOLVED] Sangoma Card/SIP/DAHDI - unable to dial out on Distro

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@abcym15 wrote:

Hello,
I installed FreePBX Distro fresh this morning on a brand new Dell T20 server via USB install. I downloaded the latest USB image from the FreePBX website, 64 bit 10.13.66, FreePBX 13 and Asterisk 13. I created some SIP extensions (the one I am using to test is 298) and set up a SIP trunk. I have also got a Sangoma Hybrid FlexBRI card with 2 x BRI and 1 x FXO ports.

The Sangoma card was not auto detected and so I have followed online instructions to install Wanpipe and DAHDI again. This is now working fine for incoming calls, the card picks up the call immediately and routes as expected. The incoming calls from the SIP trunk also work as expected.

However, I am unable to dial out on either trunk - I get an 'all circuits are busy' error. I should say that I am a fairly experienced FreePBX user so I know the basics of outbound routes, etc - these have been set up and exactly match those on my other servers.

I am not sure if this could possibly complicate things but this is a second FreePBX server in the same LAN - the other one is my main one and is in a DMZ.

The below is what CLI shows when I try to dial out via SIP trunk. 07XXXXXXXXX is the UK mobile number I am trying to dial, the route is configured so that a prefix of 9 should send calls out via my SIP trunk. 100XXXXXXX is the SIP username for my trunking provider, voipxxxx is their server. 01XXXXXXXXX is the PSTN number linked to the SIP trunk.

= Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [907XXXXXXXXX@from-internal:1] Macro("SIP/298-00000005", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/298-00000005", "TOUCH_MONITOR=1468754447.14") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/298-00000005", "AMPUSER=298") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/298-00000005", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/298-00000005", "1?Set(REALCALLERIDNUM=298)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/298-00000005", "AMPUSER=298") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/298-00000005", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/298-00000005", "AMPUSERCIDNAME=Test") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/298-00000005", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/298-00000005", "AMPUSERCID=298") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/298-00000005", "_DIALOPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/298-00000005", "CALLERID(all)="Test" <298>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/298-00000005", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/298-00000005", "1?Set(GROUP(concurrency_limit)=298)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/298-00000005", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/298-00000005", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("SIP/298-00000005", "CALLERID(number)=298") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/298-00000005", "CALLERID(name)=Test") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/298-00000005", "CDR(cnum)=298") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/298-00000005", "CDR(cnam)=Test") in new stack
-- Executing [s@macro-user-callerid:33] Set("SIP/298-00000005", "CHANNEL(language)=en") in new stack
-- Executing [907XXXXXXXXX@from-internal:2] Gosub("SIP/298-00000005", "sub-record-check,s,1(out,907XXXXXXXXX,force)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/298-00000005", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/298-00000005", "_RECSTATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/298-00000005", "NOW=1468754447") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/298-00000005", "__DAY=17") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/298-00000005", "__MONTH=07") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/298-00000005", "__YEAR=2016") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/298-00000005", "__TIMESTR=20160717-122047") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/298-00000005", "__FROMEXTEN=298") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/298-00000005", "_MONFMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/298-00000005", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/298-00000005", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/298-00000005", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/298-00000005", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/298-00000005", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/298-00000005", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/298-00000005", "Outbound Recording Check from 298 to 907XXXXXXXXX") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/298-00000005", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/298-00000005", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("SIP/298-00000005", "recordcheck,1(force,out,907XXXXXXXXX)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/298-00000005", "Starting recording check against force") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/298-00000005", "force") in new stack
-- Goto (sub-record-check,recordcheck,5)
-- Executing [recordcheck@sub-record-check:5] Set("SIP/298-00000005", "_RECPOLICY_MODE=FORCE") in new stack
-- Executing [recordcheck@sub-record-check:6] GotoIf("SIP/298-00000005", "1?startrec") in new stack
-- Goto (sub-record-check,recordcheck,16)
-- Executing [recordcheck@sub-record-check:16] NoOp("SIP/298-00000005", "Starting recording: out, 907XXXXXXXXX") in new stack
-- Executing [recordcheck@sub-record-check:17] Set("SIP/298-00000005", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [recordcheck@sub-record-check:18] Set("SIP/298-00000005", "__CALLFILENAME=out-907XXXXXXXXX-298-20160717-122047-1468754447.14") in new stack
-- Executing [recordcheck@sub-record-check:19] MixMonitor("SIP/298-00000005", "2016/07/17/out-907XXXXXXXXX-298-20160717-122047-1468754447.14.wav,ai(LOCAL_MIXMON_ID),") in new stack
-- Executing [recordcheck@sub-record-check:20] Set("SIP/298-00000005", "_MIXMONID=0x7f6830b3ef70") in new stack
-- Executing [recordcheck@sub-record-check:21] Set("SIP/298-00000005", "_RECORDID=SIP/298-00000005") in new stack
-- Executing [recordcheck@sub-record-check:22] Set("SIP/298-00000005", "_RECSTATUS=RECORDING") in new stack
-- Executing [recordcheck@sub-record-check:23] Set("SIP/298-00000005", "CDR(recordingfile)=out-907XXXXXXXXX-298-20160717-122047-1468754447.14.wav") in new stack
-- Executing [recordcheck@sub-record-check:24] Return("SIP/298-00000005", "") in new stack
-- Executing [out@sub-record-check:8] Return("SIP/298-00000005", "") in new stack
-- Executing [907XXXXXXXXX@from-internal:3] ExecIf("SIP/298-00000005", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [907XXXXXXXXX@from-internal:4] GosubIf("SIP/298-00000005", "0?sub-diversion-header,s,1()") in new stack
-- Executing [907XXXXXXXXX@from-internal:5] Set("SIP/298-00000005", "MOHCLASS=default") in new stack
-- Executing [907XXXXXXXXX@from-internal:6] Set("SIP/298-00000005", "_NODEST=") in new stack
-- Executing [907XXXXXXXXX@from-internal:7] Macro("SIP/298-00000005", "dialout-trunk,2,07XXXXXXXXX,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/298-00000005", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/298-00000005", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/298-00000005", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/298-00000005", "DIAL_NUMBER=07XXXXXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/298-00000005", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/298-00000005", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/298-00000005", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/298-00000005", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/298-00000005", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/298-00000005", "DIAL_TRUNK_OPTIONS=T") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/298-00000005", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/298-00000005", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/298-00000005", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/298-00000005", "0?Set(REALCALLERIDNUM=298)") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/298-00000005", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,7)
-- Executing [s@macro-outbound-callerid:7] Set("SIP/298-00000005", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/298-00000005", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] Set("SIP/298-00000005", "TRUNKOUTCID=100XXXXXXX") in new stack
-- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/298-00000005", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,15)
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/298-00000005", "1?Set(CALLERID(all)=100XXXXXXX)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/298-00000005", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/298-00000005", "1?Set(CALLERID(all)=100XXXXXXX)") in new stack
-- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/298-00000005", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/298-00000005", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:20] Set("SIP/298-00000005", "CDR(outbound_cnum)=100XXXXXXX") in new stack
-- Executing [s@macro-outbound-callerid:21] Set("SIP/298-00000005", "CDR(outbound_cnam)=") in new stack
[2016-07-17 12:20:47] WARNING[2963]: func_cdr.c:377 cdr_write_callback: CDR requires a value (CDR(variable)=value)
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/298-00000005", "0?sub-flp-2,s,1()") in new stack
== Begin MixMonitor Recording SIP/298-00000005
-- Executing [s@macro-dialout-trunk:13] Set("SIP/298-00000005", "OUTNUM=07XXXXXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/298-00000005", "custom=SIP/01XXXXXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/298-00000005", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/298-00000005", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/298-00000005", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/298-00000005", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/298-00000005", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/298-00000005", "1?Set(CONNECTEDLINE(num,i)=07XXXXXXXXX)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/298-00000005", "1?Set(CONNECTEDLINE(name,i)=CID:100XXXXXXX)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/298-00000005", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)100XXXXXXX)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/298-00000005", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("SIP/298-00000005", "SIP/01XXXXXXXXX/07XXXXXXXXX,300,T") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/01XXXXXXXXX/07XXXXXXXXX
[2016-07-17 12:20:47] NOTICE[4084][C-00000009]: chan_sip.c:23808 handle_response_invite: Failed to authenticate on INVITE to 'sip:100XXXXXXX@sip.voipxxxx.co.uk;tag=as0be1f7b0'
-- SIP/01XXXXXXXXX-00000006 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:24] NoOp("SIP/298-00000005", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/298-00000005", "0?continue,1:s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/298-00000005", "RC=21") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/298-00000005", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/298-00000005", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/298-00000005", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/298-00000005", "1?Set(CALLERID(number)=298)") in new stack
-- Executing [907XXXXXXXXX@from-internal:8] Macro("SIP/298-00000005", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/298-00000005", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/298-00000005", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/298-00000005", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/298-00000005", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')
-- Playing 'pls-try-call-later.ulaw' (language 'en')
-- Executing [h@from-internal:1] Macro("SIP/298-00000005", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/298-00000005", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/298-00000005", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/298-00000005", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/298-00000005' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/298-00000005'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/298-00000005

Furthermore, when I try to dial out via the DAHDI trunk with an 8 prefix, I get this:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [807XXXXXXXXX@from-internal:1] Macro("SIP/298-00000008", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/298-00000008", "TOUCH_MONITOR=1468754855.18") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/298-00000008", "AMPUSER=298") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/298-00000008", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/298-00000008", "1?Set(REALCALLERIDNUM=298)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/298-00000008", "AMPUSER=298") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/298-00000008", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/298-00000008", "AMPUSERCIDNAME=Test") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/298-00000008", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/298-00000008", "AMPUSERCID=298") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/298-00000008", "_DIALOPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/298-00000008", "CALLERID(all)="Test" <298>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/298-00000008", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/298-00000008", "1?Set(GROUP(concurrency_limit)=298)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/298-00000008", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/298-00000008", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("SIP/298-00000008", "CALLERID(number)=298") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/298-00000008", "CALLERID(name)=Test") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/298-00000008", "CDR(cnum)=298") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/298-00000008", "CDR(cnam)=Test") in new stack
-- Executing [s@macro-user-callerid:33] Set("SIP/298-00000008", "CHANNEL(language)=en") in new stack
-- Executing [807XXXXXXXXX@from-internal:2] Gosub("SIP/298-00000008", "sub-record-check,s,1(out,807XXXXXXXXX,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/298-00000008", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/298-00000008", "_RECSTATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/298-00000008", "NOW=1468754855") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/298-00000008", "__DAY=17") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/298-00000008", "__MONTH=07") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/298-00000008", "__YEAR=2016") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/298-00000008", "__TIMESTR=20160717-122735") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/298-00000008", "__FROMEXTEN=298") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/298-00000008", "_MONFMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/298-00000008", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/298-00000008", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/298-00000008", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/298-00000008", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/298-00000008", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/298-00000008", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/298-00000008", "Outbound Recording Check from 298 to 807XXXXXXXXX") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/298-00000008", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/298-00000008", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("SIP/298-00000008", "recordcheck,1(dontcare,out,807XXXXXXXXX)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/298-00000008", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/298-00000008", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/298-00000008", "") in new stack
-- Executing [out@sub-record-check:8] Return("SIP/298-00000008", "") in new stack
-- Executing [807XXXXXXXXX@from-internal:3] ExecIf("SIP/298-00000008", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [807XXXXXXXXX@from-internal:4] GosubIf("SIP/298-00000008", "0?sub-diversion-header,s,1()") in new stack
-- Executing [807XXXXXXXXX@from-internal:5] Set("SIP/298-00000008", "MOHCLASS=default") in new stack
-- Executing [807XXXXXXXXX@from-internal:6] Set("SIP/298-00000008", "_NODEST=") in new stack
-- Executing [807XXXXXXXXX@from-internal:7] Macro("SIP/298-00000008", "dialout-trunk,1,07XXXXXXXXX,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/298-00000008", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/298-00000008", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/298-00000008", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/298-00000008", "DIAL_NUMBER=07XXXXXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/298-00000008", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/298-00000008", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/298-00000008", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/298-00000008", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/298-00000008", "DIAL_TRUNK_OPTIONS=T") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/298-00000008", "outbound-callerid,1") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/298-00000008", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/298-00000008", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/298-00000008", "0?Set(REALCALLERIDNUM=298)") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/298-00000008", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,7)
-- Executing [s@macro-outbound-callerid:7] Set("SIP/298-00000008", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/298-00000008", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] Set("SIP/298-00000008", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/298-00000008", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,15)
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/298-00000008", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/298-00000008", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/298-00000008", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/298-00000008", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/298-00000008", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:20] Set("SIP/298-00000008", "CDR(outbound_cnum)=298") in new stack
-- Executing [s@macro-outbound-callerid:21] Set("SIP/298-00000008", "CDR(outbound_cnam)=Test") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/298-00000008", "0?sub-flp-1,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/298-00000008", "OUTNUM=07XXXXXXXXX") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/298-00000008", "custom=DAHDI/r1") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/298-00000008", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/298-00000008", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/298-00000008", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/298-00000008", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/298-00000008", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/298-00000008", "1?Set(CONNECTEDLINE(num,i)=07XXXXXXXXX)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/298-00000008", "1?Set(CONNECTEDLINE(name,i)=CID:298)") in new stack
-- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/298-00000008", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)298)") in new stack
-- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/298-00000008", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:23] Dial("SIP/298-00000008", "DAHDI/r1/07XXXXXXXXX,300,T") in new stack
-- Called DAHDI/r1/07XXXXXXXXX
-- Hanging up on 'DAHDI/8-1'
-- Hungup 'DAHDI/8-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:24] NoOp("SIP/298-00000008", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0") in new stack
-- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/298-00000008", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/298-00000008", "RC=0") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/298-00000008", "0,1") in new stack
-- Goto (macro-dialout-trunk,0,1)
-- Executing [0@macro-dialout-trunk:1] Goto("SIP/298-00000008", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/298-00000008", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 0 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/298-00000008", "1?Set(CALLERID(number)=298)") in new stack
-- Executing [807XXXXXXXXX@from-internal:8] Macro("SIP/298-00000008", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/298-00000008", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/298-00000008", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/298-00000008", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/298-00000008", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')
-- Playing 'pls-try-call-later.ulaw' (language 'en')
-- Executing [h@from-internal:1] Macro("SIP/298-00000008", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/298-00000008", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/298-00000008", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/298-00000008", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/298-00000008' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/298-00000008'

Except when doing it via DAHDI, between these two lines:
-- Called DAHDI/r1/07XXXXXXXXX
-- Hanging up on 'DAHDI/8-1'
My mobile phone actually starts ringing and I can answer it. DAHDI does not seem to detect that the phone has been answered, and only when I hang up my mobile does the 'Hanging up' message appear in CLI.

My DAHDI config is as follows:

/etc/dahdi/system.conf:

span=1,0,0,CCS,AMI
span=2,0,0,CCS,AMI
bchan=1-2,4-5
hardhdlc=3,6
fxsks=7,8
echocanceller=oslec,7,8
loadzone=uk
defaultzone=uk

etc/asterisk/chan_dahdi.conf

[general]

; generated by module

include chan_dahdi_general.conf

; for user additions not provided by module

include chan_dahdi_general_custom.conf

[channels]
language=en
busydetect=yes
busycount=10
usecallerid=yes
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=no
immediate=yes
faxdetect=no
rxgain=8
txgain=2
progzone=uk
callprogress=yes
ukcallerid=yes
cidsignalling=v23
cidstart=polarity
loadzone=uk
defaultzone=uk

Lastly, out of frustration with this, I tried creating an IAX2 trunk between this server and my other server on the same network. This IAX trunk worked fine and allowed me to dial out using the SIP trunks connected to the other server.

I have exhausted my knowledge of these issues and would really appreciate some advice. I hope I've included everything you need, please let me know if not.

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Saving recording to a remote folder (NFS Share)

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@esarant wrote:

Hello,

I was wondering if anyone has setup a FreePBX distro installation with the recordings being on a remote NFS Share. Are there any implications? Is it a good idea to do that?

Our system has around 60-70 concurent calls so there will be alot of NFS reading / writting.

Regads,
esarant

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More info about security issue SEC-2016-002~004

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@4allbusiness wrote:

Hello FreePBX,

Does anyone know where I can find more detailed information about the last security issues.
SEC-2016-002 SEC-2016-003 SEC-2016-004
We updated all modules offcourse, but we want to know what this issue was.
Many thanks for any onformation about this.

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Asking for deployment id_number... Error! name lookup timed out

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@jayclark wrote:

Using "fsconsole sysadmin activate deploymentid" shows:

Asking for deployment deploymentid...Error!
name lookup timed out

I've verified the following information is correct and valid:

-DNS configuration.
-IP configuration.
-Internet Access
-Deployment Configuration and Deployment ID.
-Verified NAT firewall and nothing is being blocked from PBX.

I currently have FreePBX version 10.13.66-13, FWConsole 13.0.163, SysAdmin 13.0.63.1. Using the SysAdmin module in the GUI presents Activate button and upon pressing basically cycles and then refreshes the page never making it to the deployment section.

What am I missing?

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FreePBX rpm and installation script

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@Stell0 wrote:

Hi everyone,
I'm a NethServer (NS) developer (CentOS 7 based and 100% compatible distribution) and I'm trying to have FreePBX
14 packaged for NS. I took a look to freepbx 14 rpm and I noticed that it has some strange behaviours:
· some directories and files are created during %post
· there are a lot of chmod/chown in %post
· sed of php.ini
Then the installation script is launched and it copies some files, creates symlinks, and do a few others chmod.

Consequences are that:
· rpm -ql freepbx doesn't list real freepbx files but those under /usr/src/...
· rpm -V doesn't show changed files. I could mess with /var/www/html/admin/ files and rpm -V keeps giving me freepbx package as intact
· rpm -V of asterisk* packages says that some files and directories has changed ownership and permission (and updating or reinstalling those packages change back those permissions)

My questions are: why rpm install freepbx like this? Wouldn't be better to install freepbx during rpm %build instead of %post? How do you handle Asterisk upgrades if upgrading Asterisk packages change back file permissions?

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Update to 10.13.66-13 and now cannot log into the gui

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@digitalb wrote:

I have updated from 6.12.65-30 manually running each script (all 15 with reboots) and now I am no longer able to log into the PBX system via the gui. I have try the amportal a u xxxxxxxxxxxx trick but it does not work.

I checked what versions each module is on and found a few of them were on 12.xxx still, manually updated them via the cli.

Any suggestions on where to start troubleshoot this issue?

update: used the new way to unlock and received the following:
fwconsole unlock cvukpqfgmkr7vv1t9lom3sk761
Unlocking: cvukpqfgmkr7vv1t9lom3sk761

It never actually sai "session Should be unlocked now"

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Ansible with FreePBX

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@dbayer wrote:

Has anyone used Ansible to install and maintain the Freepbx Distro? Or even maintain Freepbx?

If so do you have any Ansible roles or playbooks you can share?

Thanks,
Daniel

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Attempting to Activate Distro Kills web gui

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@gene778 wrote:

After attempting to activate my distro with Freepbx 12 and Asterisk 13, the webgui will no longer come up on the screen. All that appears is "Welcome to Freepbx" and the 6 tab buttons at the upper left of the screen. Looking at the log show the following:

[2016-Jul-27 02:34:15] [FATAL] (libraries/sql.functions.php:25) - die_freepbx(): SELECT name, prefix FROM callerid_entries [nativecode=1017 ** Can't find file: 'callerid_entries' (errno: 2)]SQL - <br /> SELECT name, prefix FROM callerid_entries

Please advise. Will provide more of the log if you need it.
Thank you,
Gene in Virginia

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Is it really End of Life date for 6.12.65 was on 12-31-15? Or is that a typo and suppose to be 12-31-16?

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@richelle_vpg wrote:

I remember just upgrading to 6.12.65 few months ago. My teammate and I both think the date for its end of life could be a typo but I wanted to confirm.

Thanks in advance who can clarify this.

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Im getting new error message in asterisk log which

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@manjunath wrote:

Hii guys i've noticed a error message today which is as follows ....
"Jul 28 19:02:33 localhost asterisk[11096]: rc_avpair_new: unknown attribute 1490026597"
I also checked the Mysql password which was correct so please guide me if you ever come accross this issue/error ......

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Upgrade distro

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@andersonhaulage wrote:

I have 2 production systems. Both running 6.12.65-32, FreePBX 13.0.163. One is running Asterix 11.19.0, the other 11.21.2. With PhoneApps and Endpoint MAnager

64Bit distros.

Endpoints are almost exclusively Mitel 6867i, using Chan SIP on port 5060. I want to upgrade to 10.13.66 with Asterisk 13. Is there a non-destructive way to that?

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