@anavasis wrote:

Hello,
I am using an OpenVox GSM100 in an ubudo machine having installed two sim cards. The contracts have 3000 sms and i want to use them to send sms. I need also to have the opportunity to receive and read them.
The issue is that i want a module or a software to send. Is there any?

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@anavasis wrote:

i have the latest version of Freepbx and want to import all my contacts to the system. I am also using Jitsi as a client. When someone calls me being in the Phonebook should be visible in Jitsi. How is that possible?
Thanks

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@oscarenzo wrote:

Hello,

Actually I'm running freepbx 12.0.7 and have enabled the fail2ban utility from the control panel, I receive correctly the emails each time that attacker is blocked but the problem is on the email date, it show that come from 1970, I checked in the textplain or code mode of the email and not see errors in the date, show as correct, I test downloading the email, then importing again in the email and still showing from 1970, this is the header.

Return-Path: <noc@empresa.com>
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	by pi.empresaserver.net (Dovecot) with LMTP id pJ33MwkBm1mVJAAAUvdU4g
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Return-path: <noc@empresa.com>
Envelope-to: noc@empresa.com
Delivery-date: Mon, 21 Aug 2017 17:49:29 +0200
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Received: by pbx.empresa.com (Postfix, from userid 0)
	id 70DDE21846; Mon, 21 Aug 2017 17:49:21 +0200 (CEST)
Subject: [Fail2Ban] SIP: banned 84.78.18.90 on pbx.empresa.com
Date: lun, 21 ago 2017 15:49:21 +0200
From: Fail2Ban <noc@empresa.com>
To: noc@empresa.com
Message-Id: <20170821154921.70DDE21846@pbx.empresa.com>
Received-SPF: softfail (mx51.antispamcloud.com: transitioning domain of empresa.com does not designate 192.168.1.108 as permitted sender) client-ip=192.168.1.108; envelope-from=noc@empresa.com; helo=pbx.empresa.com;
X-SPF-Result: mx51.antispamcloud.com: transitioning domain of empresa.com does not designate 192.168.1.108 as permitted sender
Authentication-Results: antispamcloud.com; spf=softfail smtp.mailfrom=noc@empresa.com
X-SpamExperts-Class: whitelisted
X-SpamExperts-Evidence: recipient
X-Recommended-Action: accept
X-Filter-ID: PqwsvolAWURa0gwxuN3S5Wor9+jlOIHEL/GUFVYcxyxEDBffVZVjmVaNbG4ZJG7FZ9vqpOVucmu0
 IHxi3XxG/uwZJJQJmttVN2Ott7dvjkf6F1AQqiIBMBqGsPh5qIzmBlJQaGFGRFOljnZk72M9ATut
 t44JEJY0kC3Q0y7BUmpUB1hEs4DLn1J5+WsqHSqLR4FD3XF/iW8bfMZ25n9jsaH7/dpoQCPbZ2kw
 H1aAsIg7nMBG4+jTFep5h93NA7w/12LvyHECblh5W5zDrIOtObsPqj+vfwEIbG3ra9OVNGS8Ksk+
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X-Report-Abuse-To: spam@quarantine1.antispamcloud.com

Hi,

The IP 84.78.18.90 has just been banned by Fail2Ban after
4 attempts against SIP on pbx.empresa.com.

Regards,

Fail2Ban

For email use thunderbird, thank you advance.

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@Nemetrk wrote:

i have installed FreePBX 13.0.192.16 ( Asterisk 13.12.1 ). Also i registered SSL certififate ( lets encrypt ) to my server for webrtc.

i choose "both" sip drivers at advanced settings.

i'm fighting with this issue like 2 weeks. i read and tried a lot of things and now I'm so confused.

My problems are;

when i add a new pjsip extension with default values everything works fine. i can register to my server with a softphone and no problem with sound but when i try to connect over webrtc ( sipml5 ) i always get this error.

[2017-08-21 20:01:28] NOTICE[2446]: chan_sip.c:28486 handle_request_register: Registration from '"1000" 'sip:1000@xxx.com' failed for 'xxx.xxx.xxx.xxx:19668' - Wrong password

i search google, asterisk , freepbx forums, etc. i tried all solutions which i found nothing change. i always get that error with pjsip.

So after that i tried to add a chansip extension. i can call and speak with softphone also i can register and call over webrtc but problem is i have no sound with webrtc on both sides

i'm using default FreePBX defined ports ( This device uses PJSIP technology listening on Port 5060 (UDP), Port 5060 (TCP), Port 5061 (TLS) )

Why did i always get this error ? How can i register psjip extension with webrtc or how can i fix sound issue over chansip.

Thanks for help.

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@tattosti wrote:

Hi Guys, im working on a new server but i got a problem, i have 3 inbound routes with diferent dadhi channels each one, in 2 of the routes the IVR works fine, but in one of the channels the IVR doesnt take the call, but if you redirect the route to a extension the call is answer,

This is the output of the cli:

== Starting post polarity CID detection on channel 7
-- Starting simple switch on 'DAHDI/7-1'

[INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,extra) VALUES ('CHAN_START',{ts '2017-08-21 16:24:56.281923'},'','','','','','s','from-analog','DAHDI/7-1','','',3,'','1503347096.44','1503347096.44','','','')]
-- Executing [s@from-analog:1] NoOp("DAHDI/7-1", "Entering from-dahdi with DID == ") in new stack
-- Executing [s@from-analog:2] Ringing("DAHDI/7-1", "") in new stack
-- Executing [s@from-analog:3] Set("DAHDI/7-1", "DID=s") in new stack
-- Executing [s@from-analog:4] NoOp("DAHDI/7-1", "DID is now s") in new stack
-- Executing [s@from-analog:5] GotoIf("DAHDI/7-1", "1?dahdiok:checkzap") in new stack
-- Goto (from-analog,s,9)
-- Executing [s@from-analog:9] NoOp("DAHDI/7-1", "Is a DAHDi Channel") in new stack
-- Executing [s@from-analog:10] Set("DAHDI/7-1", "CHAN=7-1") in new stack
-- Executing [s@from-analog:11] Set("DAHDI/7-1", "CHAN=7") in new stack
-- Executing [s@from-analog:12] Macro("DAHDI/7-1", "from-dahdi-7,s,1") in new stack
-- Executing [s@macro-from-dahdi-7:1] NoOp("DAHDI/7-1", "Entering macro-from-dahdi-7 with DID = s and setting to: 71541234") in new stack
-- Executing [s@macro-from-dahdi-7:2] Set("DAHDI/7-1", "_FROMDID=71541234") in new stack
-- Executing [s@macro-from-dahdi-7:3] Goto("DAHDI/7-1", "from-trunk,71541234,1") in new stack
-- Goto (from-trunk,71541234,1)
== Channel 'DAHDI/7-1' jumping out of macro 'from-dahdi-7'
-- Executing [71541234@from-trunk:1] Set("DAHDI/7-1", "__DIRECTION=INBOUND") in new stack
-- Executing [71541234@from-trunk:2] Set("DAHDI/7-1", "CHANNEL(language)=es") in new stack
-- Executing [71541234@from-trunk:3] Gosub("DAHDI/7-1", "sub-record-check,s,1(in,71541234,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("DAHDI/7-1", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("DAHDI/7-1", "_RECSTATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("DAHDI/7-1", "NOW=1503347096") in new stack
-- Executing [s@sub-record-check:4] Set("DAHDI/7-1", "__DAY=21") in new stack
-- Executing [s@sub-record-check:5] Set("DAHDI/7-1", "__MONTH=08") in new stack
-- Executing [s@sub-record-check:6] Set("DAHDI/7-1", "__YEAR=2017") in new stack
-- Executing [s@sub-record-check:7] Set("DAHDI/7-1", "__TIMESTR=20170821-162456") in new stack
-- Executing [s@sub-record-check:8] Set("DAHDI/7-1", "__FROMEXTEN=unknown") in new stack
-- Executing [s@sub-record-check:9] Set("DAHDI/7-1", "_MONFMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("DAHDI/7-1", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("DAHDI/7-1", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("DAHDI/7-1", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("DAHDI/7-1", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("DAHDI/7-1", "2?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("DAHDI/7-1", "1?sub-record-check,in,1") in new stack
-- Goto (sub-record-check,in,1)
-- Executing [in@sub-record-check:1] NoOp("DAHDI/7-1", "Inbound Recording Check to 71541234") in new stack
-- Executing [in@sub-record-check:2] Set("DAHDI/7-1", "FROMEXTEN=unknown") in new stack
-- Executing [in@sub-record-check:3] ExecIf("DAHDI/7-1", "0?Set(FROMEXTEN=)") in new stack
-- Executing [in@sub-record-check:4] Gosub("DAHDI/7-1", "recordcheck,1(dontcare,in,71541234)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("DAHDI/7-1", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("DAHDI/7-1", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("DAHDI/7-1", "") in new stack
-- Executing [in@sub-record-check:5] Return("DAHDI/7-1", "") in new stack
-- Executing [71541234@from-trunk:4] Gosub("DAHDI/7-1", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("DAHDI/7-1", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("DAHDI/7-1", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("DAHDI/7-1", "") in new stack
-- Executing [71541234@from-trunk:5] Set("DAHDI/7-1", "_FROMDID=71541234") in new stack
-- Executing [71541234@from-trunk:6] Set("DAHDI/7-1", "CDR(did)=71541234") in new stack
-- Executing [71541234@from-trunk:7] ExecIf("DAHDI/7-1", "1 ?Set(CALLERID(name)=)") in new stack
-- Executing [71541234@from-trunk:8] Set("DAHDI/7-1", "__MOHCLASS=") in new stack
-- Executing [71541234@from-trunk:9] Set("DAHDI/7-1", "_REVERSALREJECT=FALSE") in new stack
-- Executing [71541234@from-trunk:10] GotoIf("DAHDI/7-1", "1?post-reverse-charge") in new stack
-- Goto (from-trunk,71541234,12)
-- Executing [71541234@from-trunk:12] NoOp("DAHDI/7-1", "") in new stack
-- Executing [71541234@from-trunk:13] Set("DAHDI/7-1", "_CALLINGNAMEPRESSV=allowed_not_screened") in new stack
-- Executing [71541234@from-trunk:14] Set("DAHDI/7-1", "_CALLINGNUMPRESSV=allowed_not_screened") in new stack
-- Executing [71541234@from-trunk:15] Set("DAHDI/7-1", "CALLERID(name-pres)=allowed_not_screened") in new stack
-- Executing [71541234@from-trunk:16] Set("DAHDI/7-1", "CALLERID(num-pres)=allowed_not_screened") in new stack
-- Executing [71541234@from-trunk:17] NoOp("DAHDI/7-1", "CallerID Entry Point") in new stack
-- Executing [71541234@from-trunk:18] Goto("DAHDI/7-1", "ivr-1,s,1") in new stack
-- Goto (ivr-1,s,1)
-- Executing [s@ivr-1:1] Set("DAHDI/7-1", "TIMEOUT_LOOPCOUNT=0") in new stack
-- Executing [s@ivr-1:2] Set("DAHDI/7-1", "INVALID_LOOPCOUNT=0") in new stack
-- Executing [s@ivr-1:3] Set("DAHDI/7-1", "IVRCONTEXT_ivr-1=") in new stack
-- Executing [s@ivr-1:4] Set("DAHDI/7-1", "IVRCONTEXT=ivr-1") in new stack
-- Executing [s@ivr-1:5] Set("DAHDI/7-1", "_IVRRETVM=") in new stack
-- Executing [s@ivr-1:6] GotoIf("DAHDI/7-1", "0?skip") in new stack
-- Executing [s@ivr-1:7] Answer("DAHDI/7-1", "") in new stack
[INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,extra) VALUES ('ANSWER',{ts '2017-08-21 16:24:56.292064'},'','','','','','s','ivr-1','DAHDI/7-1','Answer','',3,'','1503347096.44','1503347096.44','','','')]
-- Executing [s@ivr-1:8] Wait("DAHDI/7-1", "1") in new stack
-- Executing [s@ivr-1:9] Set("DAHDI/7-1", "IVR_MSG=custom/ivr_pfm_2017") in new stack
-- Executing [s@ivr-1:10] Set("DAHDI/7-1", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3.000
-- Executing [s@ivr-1:11] ExecIf("DAHDI/7-1", "1?Background(custom/ivr_pfm_2017)") in new stack
-- Playing 'custom/ivr_pfm_2017.slin' (language 'es')
== CDR updated on DAHDI/7-1
-- Executing [013713#@ivr-1:1] GotoIf("DAHDI/7-1", "1?i,1") in new stack
-- Goto (ivr-1,i,1)
-- Executing [i@ivr-1:1] Set("DAHDI/7-1", "INVALID_LOOPCOUNT=1") in new stack
-- Executing [i@ivr-1:2] GotoIf("DAHDI/7-1", "0?final") in new stack
-- Executing [i@ivr-1:3] Set("DAHDI/7-1", "IVR_MSG=no-valid-responce-pls-try-again") in new stack
-- Executing [i@ivr-1:4] Goto("DAHDI/7-1", "s,start") in new stack
-- Goto (ivr-1,s,10)
-- Executing [s@ivr-1:10] Set("DAHDI/7-1", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3.000
-- Executing [s@ivr-1:11] ExecIf("DAHDI/7-1", "1?Background(no-valid-responce-pls-try-again)") in new stack
-- Playing 'no-valid-responce-pls-try-again.slin' (language 'es')
-- Executing [s@ivr-1:12] WaitExten("DAHDI/7-1", "10,") in new stack
-- Timeout on DAHDI/7-1, going to 't'
-- Executing [t@ivr-1:1] Set("DAHDI/7-1", "TIMEOUT_LOOPCOUNT=1") in new stack
-- Executing [t@ivr-1:2] GotoIf("DAHDI/7-1", "0?final") in new stack
-- Executing [t@ivr-1:3] Set("DAHDI/7-1", "IVR_MSG=no-valid-responce-pls-try-again") in new stack
-- Executing [t@ivr-1:4] Goto("DAHDI/7-1", "s,start") in new stack
-- Goto (ivr-1,s,10)
-- Executing [s@ivr-1:10] Set("DAHDI/7-1", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3.000
-- Executing [s@ivr-1:11] ExecIf("DAHDI/7-1", "1?Background(no-valid-responce-pls-try-again)") in new stack
-- Playing 'no-valid-responce-pls-try-again.slin' (language 'es')
-- Executing [s@ivr-1:12] WaitExten("DAHDI/7-1", "10,") in new stack
-- Timeout on DAHDI/7-1, going to 't'
-- Executing [t@ivr-1:1] Set("DAHDI/7-1", "TIMEOUT_LOOPCOUNT=2") in new stack
-- Executing [t@ivr-1:2] GotoIf("DAHDI/7-1", "0?final") in new stack
-- Executing [t@ivr-1:3] Set("DAHDI/7-1", "IVR_MSG=no-valid-responce-pls-try-again") in new stack
-- Executing [t@ivr-1:4] Goto("DAHDI/7-1", "s,start") in new stack
-- Goto (ivr-1,s,10)
-- Executing [s@ivr-1:10] Set("DAHDI/7-1", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3.000
-- Executing [s@ivr-1:11] ExecIf("DAHDI/7-1", "1?Background(no-valid-responce-pls-try-again)") in new stack
-- Playing 'no-valid-responce-pls-try-again.slin' (language 'es')
-- Executing [s@ivr-1:12] WaitExten("DAHDI/7-1", "10,") in new stack
-- Timeout on DAHDI/7-1, going to 't'
-- Executing [t@ivr-1:1] Set("DAHDI/7-1", "TIMEOUT_LOOPCOUNT=3") in new stack
-- Executing [t@ivr-1:2] GotoIf("DAHDI/7-1", "0?final") in new stack
-- Executing [t@ivr-1:3] Set("DAHDI/7-1", "IVR_MSG=no-valid-responce-pls-try-again") in new stack
-- Executing [t@ivr-1:4] Goto("DAHDI/7-1", "s,start") in new stack
-- Goto (ivr-1,s,10)
-- Executing [s@ivr-1:10] Set("DAHDI/7-1", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3.000
-- Executing [s@ivr-1:11] ExecIf("DAHDI/7-1", "1?Background(no-valid-responce-pls-try-again)") in new stack
-- Playing 'no-valid-responce-pls-try-again.slin' (language 'es')
-- Executing [s@ivr-1:12] WaitExten("DAHDI/7-1", "10,") in new stack
-- Timeout on DAHDI/7-1, going to 't'
-- Executing [t@ivr-1:1] Set("DAHDI/7-1", "TIMEOUT_LOOPCOUNT=4") in new stack
-- Executing [t@ivr-1:2] GotoIf("DAHDI/7-1", "1?final") in new stack
-- Goto (ivr-1,t,5)
-- Executing [t@ivr-1:5] Playback("DAHDI/7-1", "no-valid-responce-transfering") in new stack
-- Playing 'no-valid-responce-transfering.slin' (language 'es')
-- Executing [t@ivr-1:6] Goto("DAHDI/7-1", "app-blackhole,hangup,1") in new stack
-- Goto (app-blackhole,hangup,1)
-- Executing [hangup@app-blackhole:1] NoOp("DAHDI/7-1", "Blackhole Dest: Hangup") in new stack
-- Executing [hangup@app-blackhole:2] Hangup("DAHDI/7-1", "") in new stack
== Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'DAHDI/7-1'
-- Hanging up on 'DAHDI/7-1'
-- Hungup 'DAHDI/7-1'

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@naamanf wrote:

Trying to register a Vega 50 4 port FXO and it won't register. Followed the guide in the wiki and everything looks to be configured correctly. I have registered a trunk with Flowroute without issue. Any suggestions on where I might have something configured wrong to prevent registration?

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@minkhant wrote:

Hi there,
I have a little problem for installing DISA module on Freepbx 13.
Can explain me detail for installing DISA module.

Centos 6.5 server.
Freepbx 13

Kind Regards,

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@hshamallakh wrote:

Hi,

I have 4 fax lines, all SIP T.38 and working fine. The problem is I can't set a certain fax (outbound) to specific group of users. for examble I have fax number xxxxxxxx is supposed to be used by group A, and fax YYYYYYY to be used by group B, and ZZZZZZ to group C, how can i do that without asking each group to add a prefix to set the route, don't want to leave it up to them to decide.

it would be great if there is way to say if the call is fax and from exten A then route thought outbound route 1 ..etc.

Second Problem is that the faxpro coverpage. Can I have multiple settings for it?

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@sentinelace wrote:

I had a heck of a time yesterday on a deployment and noticed everything has changed in the new version of freepbx and now everything is on port 5160. I changed it back because we have not tested pjsip. What do you guys recommend going forward? What is the difference and benefits of pjsip? How will future upgrades affect existing deployments?

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@ccarr33 wrote:

Hello All,

I have an IVR option linked to the Directory. Once you type first 3 letters, it finds what you typed but instead of announcing the voicemail greeting, it does the text to speech. The text to speech butchers a lot of names so I need it to play the voicemail greetings.

In the directory module, I have it set to announce VM greeting and all extensions have a recorded busy, unavailable, and name message. I have looked for another setting in the VM admin, directory, and the extensions themselves.

What am I missing?

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@bwebersa wrote:

I installed a new Freepbx 13 box at a clients and the are noticing a strange issue and it's very intermittent. They will be in the middle of a call and it will just drop. They said they can either be talking to someone on a cell phone or land line when it will happen so it's not like we can easily blame it other side being a cell phone.

Today was the first time I ever experienced it calling them. I called the office and the receptionist picked up I was on the phone with her for about 5 secs and then it went dead. Because of this I was actually able to look at the logs to see what happened and it looks like the phone system is getting a hangup command from some where when it does it. The phone call in question got a hangup cause: 16. Which my understanding is just a standard hangup.

Could the phone system randomly be issuing this command or is it more likely that it's the phones doing it. I grabbed a section of the log files if you would like to see it.

And to give a little back ground they are running Freepbx 13.0.192.8 and Yealink T-46s phones. This issue doesn't seem to be isolated to a single phone either.

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@stevensedory wrote:

Running FreePBX 13.0.192.16 and Asterisk 13.17.0

Recently used the "warm spare" method to move to a new server (new VM on KVM/proxmox)

The server has about 120 remote extension, and had no real problems before.

However, a couple times now, we've had all or most of the extensions go offline. In one case, they all showed as OK, but couldn't be reached (and endpoints showed unregistered from their perspective), and in another case, they went UNREACHABLE.

After some investigation, it looks like this issue happened right around the time this did:

[2017-08-20 10:33:07] WARNING[5413] media_index.c: Failed to stat /var/lib/asterisk/sounds/ru/intercom.sln16: No such file or directory
[2017-08-20 10:33:07] WARNING[5413] media_index.c: Failed to stat /var/lib/asterisk/sounds/ru/vmnotify-instructions2.sln: No such file or directory
[2017-08-20 10:33:07] WARNING[5413] media_index.c: Failed to stat /var/lib/asterisk/sounds/ru/vmnotify-instructions.sln16: No such file or directory
[2017-08-20 10:33:07] WARNING[5413] media_index.c: Failed to stat /var/lib/asterisk/sounds/ru/vmnotify-thankyou.sln16: No such file or directory
[2017-08-20 10:33:07] WARNING[5413] media_index.c: Failed to stat /var/lib/asterisk/sounds/ru/followme.sln: No such file or directory
[2017-08-20 10:33:07] WARNING[5413] media_index.c: Failed to stat /var/lib/asterisk/sounds/ru/vmnotify-instructions2.sln16: No such file or directory
[2017-08-20 10:33:07] WARNING[5413] media_index.c: Failed to stat /var/lib/asterisk/sounds/ru/vmnotify-followed-by-pound.sln16: No such file or directory
[2017-08-20 10:33:07] WARNING[5413] media_index.c: Failed to stat /var/lib/asterisk/sounds/ru/vmnotify-instructions.sln: No such file or directory

After that, about 550 more similar lines. And then 16 seconds later:

[2017-08-20 10:33:24] ERROR[29345] manager.c: Unable to process manager action 'login'. Asterisk is shutting down.
[2017-08-20 10:33:24] ERROR[29345] manager.c: Unable to process manager action 'Logoff'. Asterisk is shutting down.
[2017-08-20 10:33:36] VERBOSE[5370] asterisk.c: Remote UNIX connection
[2017-08-20 10:33:36] VERBOSE[29847] asterisk.c: Ignoring asterisk shutdown request, already in progress.
[2017-08-20 10:33:36] VERBOSE[29847] asterisk.c: Remote UNIX connection disconnected
[2017-08-20 10:33:36] VERBOSE[5337] asterisk.c: Ignoring asterisk shutdown request, already in progress.
[2017-08-20 10:33:36] VERBOSE[5337] asterisk.c: Ignoring asterisk shutdown request, already in progress.

And then this, which I think is Asterisk restarting:

[ 2017-08-20 10:34:06] Asterisk 13.17.0 built by mockbuild @ jenkins2.schmoozecom.net on a x86_64 running Linux on 2017-07-27 17:42:42 UTC

In one instance, I had force stop asterisk, and then fwconsole restart, to get endpoints to come back (fwconsole restart by itself hung, saying it couldn't stop asterisk).

Do I just need to do what is stated in this article, rm the files listed:

?

Thanks in advance people.

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@mlcadmin wrote:

I have used FreePBX for several years but never with an analog card setup. I am a little confused on the proper procedure for the setup of the card. I see some conflicting information about the correct procedure, and I had a problem getting FreePBX installed with the card installed so I thought I would ask here.

What is the correct procedure for installing FreePBX 14 on a system with a Sangoma A200 card? Do I install the OS and do all the updates, then install the card? Or, should I have been able to install the OS with the card installed in the system?

If I installed the OS first and setup the initial system, then installed the Sangoma card, what is the procedure to get it up and running? Do I need to SSH into the system and run "setup-sangoma" from the command line? Or, do I simply need to setup the card using the DAHDI utility in the gui menu?

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@jerryriggin wrote:

Bulk Handler documentation does not say anything about an action column with add,delete or edit. This used to be required in Bulk Extensions. It appears NOT to be the case with Bulk Handler. It ignores "delete" and happily completes but does nothing.

Do I still need to use Bulk Extensions to modify/delete extensions?

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@neoweb wrote:

I am a new user and I was initially confused about usernames vs extensions.

The username is for webservices hosted by FreePBX, etc...and the extension/secret is the actual login/info for the SIP phone.

Correct?

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@KingBeetle wrote:

I just recently set up my FreePBX 13/Asterisk 13 system, and it has been working faultlessly for about a week. I have rebooted several times for various reasons, and my most recent uptime was about 72 hours. Somewhere in that 72 hour time frame, my system stopped receiving inbound calls. Anyone calling in would receive the standard "call cannot be completed as dialed" message.
I could still call extension to extension and even make outbound calls, but any inbound calls would fail. Calling my own DID from an extension resulted in the same message. I checked with my SIP trunk provider and they hadn't had any outages.

I stopped Astrerisk with "fwconsole stop" and rebooted with "shutdown -r now" and when the system came back up, everything was back to normal. That eliminates my SIP trunk provider as the source of the issue, and it has to be the server.

I suppose I could add a cron job to restart the server once every 48 hours or so, but I would really like to know what caused the error. I'm new to Asterisk and a good portion of the log files content is still Greek to me, so I wondered if someone could take a look at the Asterisk log snippets I assembled from my call attempts before reboot and after reboot, and perhaps offer some advice on what caused the issue.

I tried to include the logs in the body of this post in CODE blocks, but they exceeded the max message size. Here's a link to the before and after snippets.

Any assistance or input would be appreciated.

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@CristianDaniel wrote:

Heloo guys, first of all, i`m new on pbx stuf and truly confuzed.

Here is what i whant to know what do i need and the cost of

1. i got 1 VOIP privider 1 Phone No witch supports 20 calls in the same time
2. FreePBX is in one of my dedicated server installed on an VPS witch is on the datacenter
3. We got 4 Phones Cisco ip phone on ower office witch is in another city
4. When someone ring ower phone no all the phones to ring, first who answear take the call
5. On the same time when one of ower phone is busy if someone calls the rest of the phone to ring so to be able to answear in the same time when any of us is busy
6. To be able to transfer calls from one phone to another if it can if no is ok
7. To be able to set opening h and an custom sound (we make the sound recording)
8. if all phones ar busy to put the customer on que when any of the phones is not busy to take the call

This is all what i whant to do nothink alse and i`m verry confuzed on what do i have to buy to make it work all the thinks i need
we ar only 4 ppl, 4 phones.

Thank you guys. If my english is not well i`m sorry

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@CCSNetTech wrote:

Hello Everyone,

I have been tasked to see if the following scenario is possible. The Campus I work on only wants certain phones to dial outbound during a determined time frame by Administration. I have reviewed the Class of Service and how it blocks members from using outbound routes which is exactly what I am needing. But only during the time frame approved, below is a example...

Administration only wants the Curriculum Dept Head outbound call access between 7:00am-4:00pm. The other phones in that Dept can call Emergency numbers and internal extensions only, no outbound routes. After 4:00pm all phone can dial outbound routes again.

Is this scenario possible? If so where do I start?

Like I mentioned above Class of Service is what I need to use but I cannot figure out if the time restraints can be setup.

Thank you in advance for any assistance!!!

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@lesismore wrote:

This is kind of a weird, but for some reason there is no audio when I call a specific number.

When I try to call my voip service provider (fpl), the line shows its been answered but there is no audio.
If I use that line to call anyone else it works perfectly. Only this one number has issues.

So I tried to use my secondary voip service to call my first voip provider it works (but I have to pay for these calls so not preferable.)

So to make sure it wasn't my first provider having issues, using CSip I entered my fpl credentials and called the number having issues, it worked. Great, so I went back to my voip device and called with fpl, and still doesn't work.

So it seems this voip account can receive and make calls to any number, except when through freepbx it has issues with this one number (no audio).

Any suggestions?

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@PitzKey wrote:

Hello Community,

When making a entry it right away connects to the Extension, is there a possibility that it should let callers confirm the entry before connecting?

I see it was asked here & requested here, but no solution so far..

Thanks

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