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Play RTSP Streams on IP Phones

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@jcadman wrote:

I have multiple Cisco 8841 IP Phones that are connected to FreePBX 15 and wondering how I would be able to stream my CCTV cameras to my IP phones when i dial a particular number. The NVR box is currently setup so that I can access the cameras through an RTSP stream using VLC. I have heard about app_rtsp but I can not find any information on how to install this on a centos system. Does anyone know how to get this to work?

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Pro Tip - Using Shell & Curl in Dial plans

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@craig24x wrote:

I see a lot of people solving curl requests via additional scripts.

Typical use case goes something like:

“I need a call context to go get some information, set the result as a variable, and use the variable or argument in my next step.” Knee-jerk reaction is “hey, I can make a script for that and I know how to call a script via dial plan, problem solved!”

Yes, problem solved but not efficiently. Multiple steps makes for more points of failure, in my opinion. Especially if you’re not the only “hands in the pot”.

I’ll use an example of something gaining popularity at the moment, Jitsi meet. For those creating a PBX back-end solution, Asterisk needs to CURL an API in order to retrieve a conference room name. Part of the result is then added as a SIP Header.

API URL: https://jitsi-api.jitsi.net/conferenceMapper?id=YOUR-PIN-HERE

Sending a CURL request with a valid pin (1234567890):

curl --silent https://jitsi-api.jitsi.net/conferenceMapper?id=1234567890

Returns:

{"message":"Successfully retrieved conference mapping","id":1234567890,"conference":"teammeeting@conference.meet.mydomain.com"}

It would be nice if we needed the entire string, but the SIP Header should only contain the room name. In this case, “teammeeting”.

This is where folks try to solution with a script to cut down to the desired info, but it can be done easier all in one Dial plan line.

The terminal version of extracting “teammeeting” would be:

curl --silent https://jitsi-api.jitsi.net/conferenceMapper?id=${confid} | sed -e 's/.*"conference":"\(.*\)@.*/\1/'

Key operator being that we added a stream editor command to read the curl result string and trim out what we want.

The SED example below looks for "conference":" , removes it and everything before it. Then, looks for @ removing @ and everything after it

sed -e 's/.*"conference":"\(.*\)@.*/\1/'

Wrapping this up in an Asterisk Dial plan looks something like this:

exten => s,n,Set(CURL_RESULT=${SHELL(curl --silent https://jitsi-api.jitsi.net/conferenceMapper?id=${confid} | sed -e 's/.*"conference":"\(.*\)@.*/\1/')})
exten => s,n,Verbose(0, ${CURL_RESULT});

In the Dial plan line , we’ve obtained the PIN in a previous step, being injected as ${confid}.

Reading an asterisk log would print out the desired ${CURL_RESULT} as “teammeeting”, which is what we needed. No scripts, all one line.

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Problems with caller ID blocking

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@cinergi wrote:

Hello,

I have a PJSIP trunk with voip.ms and I’m trying to set up outbound caller ID blocking. I’ve set up an outbound route matching calls starting with *67, and configured the caller ID on this route as “hidden” using my valid DID number. The problem is that voip.ms blocks these calls because the caller ID number is not sent, which voip.ms requires. I contacted their support, and they indicated that I must pass a valid Caller ID number and set the “privacy=on” flag in the Remote-Party-ID SIP header to instruct voip.ms to hide my number. I’m not sure how to do this. I enabled Send RPID/PAI in the PJSIP trunk options, but this didn’t help.

Using tcpdump, I’ve confirmed that when I set “hidden” in the Caller ID field, FreePBX does not send the Remote-Party-ID field even if it’s enabled in the trunk PJSIP options:

raspbx.home.lan.sip > <voipms-server>.sip: [udp sum ok] SIP, length: 941
    INVITE sip:<number>@<server>:5060 SIP/2.0
    Via: SIP/2.0/UDP <my-IP>:5060;rport;branch=z9hG4bKPj7911f74a-e1db-471f-991f-75261e935ffb
    From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=52ac39b5-4095-4e6e-a7c2-2d8f39859a2e
    To: <sip:<number>@<server>>
    Contact: <sip:asterisk@<my-IP>:5060>
    Call-ID: <call-ID>
    CSeq: 699 INVITE
    Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
    Supported: 100rel, timer, replaces, norefersub
    Session-Expires: 1800
    Min-SE: 90
    Max-Forwards: 70
    User-Agent: FPBX-15.0.16.49(16.9.0)
    Content-Type: application/sdp
    Content-Length:   241

    v=0
    o=- 1897907294 1897907294 IN IP4 <my-IP>
    s=Asterisk
    c=IN IP4 <my-IP>
    t=0 0
    m=audio 19480 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=maxptime:150
    a=sendrecv

However, if I don’t set “hidden” in the Caller ID field, then the Remote-Party-ID is added as expected but Privacy is set to off, not on. voip.ms needs for it to be ON to hide the caller ID, and I don’t know how to do this.

22:14:57.742357 IP (tos 0x60, ttl 64, id 8587, offset 0, flags [DF], proto UDP (17), length 1050)
    raspbx.home.lan.sip > <voipms-server>.sip: [udp sum ok] SIP, length: 1022
        INVITE sip:<number>@<voipms-server>:5060 SIP/2.0
        Via: SIP/2.0/UDP <my-IP>:5060;rport;branch=z9hG4bKPjfef2c145-ace0-44e2-9c7d-4dff06e43841
        From: "test" <sip:<number>@<my-IP>>;tag=8dbc786f-8138-404a-ae0e-5dbb76a9f421
        To: <sip:<number>@<voipms-server>>
        Contact: <sip:asterisk@<my-ip>:5060>
        Call-ID: <caller-ID string>
        CSeq: 27676 INVITE
        Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
        Supported: 100rel, timer, replaces, norefersub
        Session-Expires: 1800
        Min-SE: 90
        Remote-Party-ID: "test" <sip:<number>@<IP>>;party=calling;privacy=off;screen=yes
        Max-Forwards: 70
        User-Agent: FPBX-15.0.16.49(16.9.0)
        Content-Type: application/sdp
        Content-Length:   239

        v=0
        o=- 178299335 178299335 IN IP4 <my-IP>
        s=Asterisk
        c=IN IP4 <my-IP>
        t=0 0
        m=audio 15588 RTP/AVP 0 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=maxptime:150
        a=sendrecv

Thanks
cinergi

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Firewall Zones & Remote Access

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@mikeaph wrote:

I have a PBXact at my office. It is local on the network with everything configured appropriately. I would like to be able to access the GUI from my home.

As I understand it, one option is to change the SSH port to something random on the PBX, then set my router’s firewall at the office to allow only traffic from my home public IP on that port. Then, in the PBX firewall, I would need to allow SSH traffic on the Internet Zone. Typically, that would be unwise because you are exposing yourself to attacks via SSH. However, am I correct in understanding that since I am blocking all traffic at my router, it should be secure since I am only allowing traffic from my home IP? With this setup, I would be able to SSH into my PBX and, at the same time, create a tunnel so that I can access the GUI.

Another option, I believe, is to add my home public IP to the trusted, local or other firewall zone and then allow access to the GUI in whichever of those zones I choose? Then, of course, I would still block all traffic at the firewall level except for my home public IP. Would that still be secure? And if that is a viable option, which zone would be advisable?

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Inbound RTP not recognised by FreePBX 15.0.16.51

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@andrewr123 wrote:

Have installed FreePBX on RPi4 and banging my head trying to get inbound RTP recognised by the PBX.

Config consists of an RPi4 running PBX 15.0.16.51 (Asterisk 16.9.0) with an internal network (no NATs) comprising around 6-10 phones (a mix of Grandstream, SNOM, and Gigaset). All connections using Chan_PJSIP stack. PSTN trunk made using ObiHai FXO gateway. All working satisfactorily on my legacy config running on a Dell tower with FreePBX 2.11 (using Chan_SIP).

Examples:

  1. Try to use internal extension to leave voicemail - voicemail announcements heard, but voicemail not recorded. Wireshark trace confirms RTP is from PBX only - no RTP visible from extension.
  2. Try to pick up voicemail - no response by PBX to DTMF digits
  3. Place external call to reach internal extension (PSTN -> FXO -> PBX -> Extn). Internal extension rings, but no external voice heard.

Have tried converting Chan_PJSIP to Chan_SIP, but then PBX refuses to register the endpoints.

Any thoughts/suggestions most welcome.

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One extension, two phones, only most recently booted rings

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@bryree wrote:

Encountering an interesting issue: I used the gui to place one extension on two different endpoints (both Grandstream 2135s). Only the most recently booted endpoint rings and if that one is unplugged, the other still does not ring until it is rebooted and becomes the ‘most recent.’

I have another extension that is on two endpoints and calls ring both with no problem, but one of them in that case was set up from from the endpoint’s web interface and the other was set up in the gui for FreePBX (for other reason apart from having the same extension on two endpoints). We want to not have to split the setup process like this in cases that require the same extension ringing in multiple offices/buildings and so want to just use the gui.

Is there something particular to setting up one extension on two different endpoints in the FreePBX gui that I’m missing to cause this ‘only most recently booted rings’ thing?

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SIP Call Simulating

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@Chazz wrote:

Hi,
I want to measure performance differences between RTP and SRTP. To do this, I would have to simulate phone calls.

What programs or scripts are there?
As a tip I have already received SIPP, unfortunately I don’t quite understand it.
I use Asterisk 16 with FreePBX 15 on a Raspberry Pi 4.

As an alternative, I thought of opening a softphone several times to simulate conversations. Unfortunately I have not found one that can be opened multiple times and also allows SRTP.

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Remote workers now having issues with phones

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@BigB wrote:

I have a couple issues with some of our remote workers and am trying to figure out the best solution. When all of our employees were working in our offices we have the network setup with the modem in passthru and a security appliance with all the QoS and traffic shaping rules applied. We rarely ever had any phone issues.

Once the stay at home order was put in place all of our employee took their computers home with them and some started having phone issues. Some of the issues include their soft phone not being able to answer a call that is ringing, when they try to answer it appears to create duplicate incoming calls (See screenshot). Another phone issue is that some of our agents are logged into their queues but they show as unavailable even though their soft phone show they are registered. We know this because we have an optional extension destination setup that transfers the caller into an IVR that gives the caller options for how to proceed since the agent isn’t available. Another issue that has been reported is their soft phone might only ring once or twice and then another agent receives the call and answers it so the initial agent gets a RINGNOANSWER which is used in our agent KPI’s.

My initial guess on the cause of the issues is ISP problems or I suspect their Modem/Router combo provided by their ISP is mangling the packets. I have had a couple of our agents log into their router and see if there was a SIP ALG setting they could turn off but most of these consumer routers provided by the ISP don’t have QoS or traffic shaping rules, also turning SIP ALG is not an option in most of these devices. I am curious what other people do for their remote workforce? Do you have them change out the ISP provided router with something else? Would using a VPN tunnel help? What are some of the troubleshooting steps I can do to help track down where the problem is?

image

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Let'sEncrypt certgen issue with firewall enabled

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@FreerPBXer wrote:

I think the last thread on this is now locked, so I’m reposting.

Just went through an initial LetsEncrypt certgen on a fully updated 10.13.66-22 system. It failed certgen until I disabled the firewall on the Connectivity-Firewall page.

This is at least the fourth system in the last six weeks where this has been the case.

Something is decidedly hinky with the firewall and LetsEncrypt. If someone can give me instructions I will capture logs/data the next time this comes up.

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Asterisk crashes every 3-5 days - related to queue application?

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@inc0gnito wrote:

We’re running FreePBX 14.0.13.6, Asterisk 13.29.2 on device & user mode.
Single Xeon 4110 machine w/ 32GB of RAM. Debian 9.9.
Standalone FPBX & Asterisk install from source.

On average, about 60 chan_sip devices online at any one time, 15-40 concurrent channel usage.
One outlier for us is that we have approximately 180 queues setup on this instance.

What happens is that every 3-5 days (usually depending on total call volume or more laterally - time elapsed) Asterisk “freezes” and becomes unresponsive. What we can see very different with all our other installs over the years is that this install exhibits the following issues :

  1. over the course of a day, no matter what - we can see that the manager stasis application continues to hit 3000 scheduled tasks a day. We can see this in intervals starting from 5 minutes to every 5-10 seconds.

WARNING[99873]: taskprocessor.c:1110 taskprocessor_push: The ‘stasis/m:manager:core-00000006’ task processor queue reached 3000 scheduled tasks again.

  1. In the hour or so before Asterisk begins crashing, the log gets filled with the following entries every second -

chan_sip.c: Autodestruct on dialog ‘0c5de18c334e114d5f830e6953e61854@10.10.10.1:5060’ with owner SIP/digium-gw-000030e9 in place (Method: BYE). Rescheduling destruction for 10000 ms

This doesn’t just apply to the Digium G200 PRI trunk. It happens to all SIP trunks.

This happens until the channels start hitting 300-400 channel utilization with the phantom channels and as a result Asterisk starts becoming unresponsive.

If we try to go into the CLI at that time, when the system is close to becoming fully unresponsive - by typing “asterisk -rvvv”, it loads into CLI and then instantly kicks me back to the shell terminal. The last 3 lines would look as follows -

Connected to Asterisk 13.29.2 currently running on CALLCENTRE01 (pid = 99852)
CALLCENTRE01*CLI>
root@CALLCENTRE01:/usr/src#

There are no other logs present to indicate any other error of sorts besides the Autodestruct you see above.

When the above happens, current calls continue to go on and complete, however all new calls, after going through the IVR will simply hang there without any sound, meaning it doesn’t hit the queues.

We usually have to manually kill the asterisk process and do a fwconsole start for things to go back to normal.

Previously we were running Asterisk 16.6.1 with the exact same issues. We downgraded to 13.29.2 because we thought it may have been an issue with 16 (most of our installs are still on 13 without issues) but they continue to persist.

The typical number of calls handled before asterisk crashes is about 25,000 to 50,000 and 3-4 days on average.
I know it’s not an issue of call handling as we have asterisk instances handling 50k calls a day without issue.

I’m starting to think it may be due to the sheer number of queues that we have (180), that’s the only logical explanation I can think of. I’ve read some reports that the queue application may be unstable at scale, but have yet to read anything that’s truly validated.

How can I begin diagnosing this issue?

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Unknown peer - without anonymous on

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@jazzb wrote:

Hi All,

I’m sure this has been asked a lot, but I am trying to get incoming calls working without turning on ‘anonymous on’. I’ve searched around the forums but cant find a definitive answer.

I have an incoming SIP Trunk from my telco, all incoming calls come from the same IP. Works fine if it turn anonymous calling on. But there must me some setting I can use on the Incoming trunk that will allow the calls without turning this on?

Many Thanks.

freepbx*CLI>
  == Setting global variable 'SIPDOMAIN' to '10.50.102.11'
    -- Executing [0203XXX44XX@from-sip-external:1] NoOp("PJSIP/anonymous-00000000", "Received incoming SIP connection from unknown peer to 0203XXX44XX") in new stack
    -- Executing [0203XXX44XX@from-sip-external:2] Set("PJSIP/anonymous-00000000", "DID=0203XXX44XX") in new stack
    -- Executing [0203XXX44XX@from-sip-external:3] Goto("PJSIP/anonymous-00000000", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("PJSIP/anonymous-00000000", "1?setlanguage:checkanon") in new stack
    -- Goto (from-sip-external,s,2)
    -- Executing [s@from-sip-external:2] Set("PJSIP/anonymous-00000000", "CHANNEL(language)=en") in new stack
    -- Executing [s@from-sip-external:3] GotoIf("PJSIP/anonymous-00000000", "1?noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("PJSIP/anonymous-00000000", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2020-05-01 18:46:17.199 UTC.
    -- Executing [s@from-sip-external:6] Set("PJSIP/anonymous-00000000", "receveip=pjsip,remote_addr") in new stack
    -- Executing [s@from-sip-external:7] Log("PJSIP/anonymous-00000000", "WARNING,"Rejecting unknown SIP connection from 164.39.XX.XX:5060"") in new stack
[2020-05-01 18:46:02] WARNING[13310][C-00000001]: Ext. s:7 @ from-sip-external: "Rejecting unknown SIP connection from 164.39.XX.XX:5060"
    -- Executing [s@from-sip-external:8] Answer("PJSIP/anonymous-00000000", "") in new stack
       > 0x7fb414013220 -- Strict RTP learning after remote address set to: 164.39.XX.XX:18288
    -- Executing [s@from-sip-external:9] Wait("PJSIP/anonymous-00000000", "2") in new stack
       > 0x7fb414013220 -- Strict RTP switching to RTP target address 164.39.XX.XX:18288 as source
    -- Executing [s@from-sip-external:10] Playback("PJSIP/anonymous-00000000", "ss-noservice") in new stack
    -- <PJSIP/anonymous-00000000> Playing 'ss-noservice.alaw' (language 'en')
       > 0x7fb414013220 -- Strict RTP learning complete - Locking on source address 164.39.XX.XX:18288
    -- Executing [h@from-sip-external:1] Hangup("PJSIP/anonymous-00000000", "") in new stack
  == Spawn extension (from-sip-external, h, 1) exited non-zero on 'PJSIP/anonymous-00000000'
freepbx*CLI>

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Dial-In Number?

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@jazzb wrote:

Hi All.

Just wondering if its possible to use FreePBX to bridge two external calls. Maybe via the IVR?
For example, can someone use there mobile to make a international call via the PBX so the company picks up the costs rather than the user?

In Summary:
External Caller —> Spare Number —> Auto-Attendant --> Enter PIN --> Dial Tone for outgoing call.

(P.S. I am aware of toll fraud issues, this is just a temp. workaround)

Many Thanks, J

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Caller inputs account number or similar

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@toborgps wrote:

So I only was able to find a post somewhat relevant to this with not much info. I wanted to use an already existing MySQL database through WHMCS to give amount due etc… Does anyone happen to have a guide or some good resources to tackle such a project?

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Fail2Ban -- Something is off

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@vaibhav wrote:

I received the following email from fail2ban

Hi,
The IP 45.143.220.131 has just been banned by Fail2Ban after
46 attempts against SIP on ast.
Regards,
Fail2Ban

My Fail2Ban settings are as follows:

After 2 failed attempts, I want to ban the IP for a month (this logic works for my setup). But this does not match-up with the above email.

Why was the rogue IP not blocked after 2 attempts? How did it get to make 46 attempts??
Also, my " IP’s that are currently banned." list is not growing.

I think I’m missing something here.
Please help.

Thanks

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BARGE, LISTEN and WHISPER

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@ruben23 wrote:

Hi guys i Followed this process for Barging, Listen and Whisper

But when i tried and tested it i cant hear any voice or anything on the conversing two extensions at all and they can hear either, any idea guys.?

.

Add below snippet to extensions_custom.conf . Reload asterisk and you’re good to go. To listen on extension 123, simply dial *222123.

[ext-local-custom]

;listen
exten => _*222x.#,1,Macro(user-callerid,)
exten => _*222x.#,n,Answer
exten => _*222x.#,n,NoCDR
exten => _*222x.#,n,Wait(1)
exten => _*222x.#,n,ChanSpy(sip/${EXTEN:4},q)
exten => _*222x.#,n,Hangup

;whisper
exten => _*223x.#,1,Macro(user-callerid,)
exten => _*223x.#,n,Answer
exten => _*223x.#,n,NoCDR
exten => _*223x.#,n,Wait(1)
exten => _*223x.#,n,ChanSpy(sip/${EXTEN:4},qw)
exten => _*223x.#,n,Hangup

;barge
exten => _*224x.#,1,Macro(user-callerid,)
exten => _*224x.#,n,Answer
exten => _*224x.#,n,NoCDR
exten => _*224x.#,n,Wait(1)
exten => _*224x.#,n,ChanSpy(SIP/${EXTEN:4},qB)
exten => _*224x.#,n,Hangup

here is the logs of the barging process

== Setting global variable ‘SIPDOMAIN’ to ‘10.10.123.250’
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Executing [*2245150@from-internal:1] Macro(“PJSIP/5110-00000019”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“PJSIP/5110-00000019”, “TOUCH_MONITOR=1588464849.25”) in new stack
– Executing [s@macro-user-callerid:2] Set(“PJSIP/5110-00000019”, “AMPUSER=5110”) in new stack
– Executing [s@macro-user-callerid:3] Set(“PJSIP/5110-00000019”, “HOTDESCKCHAN=5110-00000019”) in new stack
– Executing [s@macro-user-callerid:4] Set(“PJSIP/5110-00000019”, “HOTDESKEXTEN=5110”) in new stack
– Executing [s@macro-user-callerid:5] Set(“PJSIP/5110-00000019”, “HOTDESKCALL=0”) in new stack
– Executing [s@macro-user-callerid:6] ExecIf(“PJSIP/5110-00000019”, “0?Set(HOTDESKCALL=1)”) in new stack
– Executing [s@macro-user-callerid:7] ExecIf(“PJSIP/5110-00000019”, “0?Set(CALLERID(name)=)”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“PJSIP/5110-00000019”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] ExecIf(“PJSIP/5110-00000019”, “1?Set(REALCALLERIDNUM=5110)”) in new stack
– Executing [s@macro-user-callerid:10] Set(“PJSIP/5110-00000019”, “AMPUSER=5110”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“PJSIP/5110-00000019”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:12] Set(“PJSIP/5110-00000019”, “AMPUSERCIDNAME=5110”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“PJSIP/5110-00000019”, “0?Set(__CIDMASQUERADING=TRUE)”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“PJSIP/5110-00000019”, “0?report”) in new stack
– Executing [s@macro-user-callerid:15] Set(“PJSIP/5110-00000019”, “AMPUSERCID=5110”) in new stack
– Executing [s@macro-user-callerid:16] Set(“PJSIP/5110-00000019”, “__DIAL_OPTIONS=HhTtr”) in new stack
– Executing [s@macro-user-callerid:17] Set(“PJSIP/5110-00000019”, “CALLERID(all)=“5110” <5110>”) in new stack
– Executing [s@macro-user-callerid:18] ExecIf(“PJSIP/5110-00000019”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-user-callerid:19] GotoIf(“PJSIP/5110-00000019”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:20] ExecIf(“PJSIP/5110-00000019”, “0?Set(GROUP(concurrency_limit)=5110)”) in new stack
– Executing [s@macro-user-callerid:21] ExecIf(“PJSIP/5110-00000019”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:22] NoOp(“PJSIP/5110-00000019”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:23] GotoIf(“PJSIP/5110-00000019”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,24)
– Executing [s@macro-user-callerid:24] GotoIf(“PJSIP/5110-00000019”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:25] ExecIf(“PJSIP/5110-00000019”, “1?Set(__CALLEE_ACCOUNCODE=)”) in new stack
– Executing [s@macro-user-callerid:26] Set(“PJSIP/5110-00000019”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:27] GotoIf(“PJSIP/5110-00000019”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,43)
– Executing [s@macro-user-callerid:43] Set(“PJSIP/5110-00000019”, “CALLERID(number)=5110”) in new stack
– Executing [s@macro-user-callerid:44] Set(“PJSIP/5110-00000019”, “CALLERID(name)=5110”) in new stack
– Executing [s@macro-user-callerid:45] GotoIf(“PJSIP/5110-00000019”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:46] Set(“PJSIP/5110-00000019”, “CDR(cnam)=5110”) in new stack
– Executing [s@macro-user-callerid:47] Set(“PJSIP/5110-00000019”, “CDR(cnum)=5110”) in new stack
– Executing [s@macro-user-callerid:48] Set(“PJSIP/5110-00000019”, “CHANNEL(language)=en”) in new stack
– Executing [*2245150@from-internal:2] Answer(“PJSIP/5110-00000019”, “”) in new stack
> 0x7f54b4037e70 – Strict RTP learning after remote address set to: 10.10.122.43:49168
> 0x7f54b4037e70 – Strict RTP switching to RTP target address 10.10.122.43:49168 as source
– Executing [*2245150@from-internal:3] NoCDR(“PJSIP/5110-00000019”, “”) in new stack
– Executing [*2245150@from-internal:4] Wait(“PJSIP/5110-00000019”, “1”) in new stack
– Executing [*2245150@from-internal:5] ChanSpy(“PJSIP/5110-00000019”, “SIP/5150,qB”) in new stack
> 0x7f54b4037e70 – Strict RTP learning complete - Locking on source address 10.10.122.43:49168

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Reinstall FREEPBX 14

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@jefbuan wrote:

I accidentally delete /etc/freepbx.conf (stupid me :pensive:) , how can I reinstall freepbx without overwriting my exisiting asterisk configurations ??

root@jefVPN:/home/jeff# tail -f /var/log/apache2/error.log
[Sun May 03 09:55:18.501500 2020] [:error] [pid 29780] [client 51.39.2.227:1177] PHP Fatal error: Class ‘FreePBX’ not found in /var/www/html/admin/config.php on line 136
[Sun May 03 09:59:51.709142 2020] [:error] [pid 29785] [client 51.39.2.227:2318] PHP Warning: include_once(/etc/freepbx.conf): failed to open stream: No such file or directory in /var/www/html/admin/config.php on line 126
[Sun May 03 09:59:51.709177 2020] [:error] [pid 29785] [client 51.39.2.227:2318] PHP Warning: include_once(): Failed opening ‘/etc/freepbx.conf’ for inclusion (include_path=’.:/usr/share/php:/usr/share/pear’) in /var/www/html/admin/config.php on line 126

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Let's Encrypt broken on two FreePBX servers

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@volkswagner wrote:

I have two public-facing (no NAT) servers each with seemingly unique problems
renewing or creating new certificates for Let’s Encrypt. Port 80 and 443 are open
to the world and using Responsive Firewall.

I tried deleting the Let’s Encrypt certificate via Certificate Manager, although it
doesn’t show in the list anymore, the FreePBX server is still using the soon to
expire certificate that I “deleted” (even after making the self-signed the default
and being the only cert in the list).

If I try to create a new certificate I get the following Error:
There was an error updating the certificate: 400 { "type": "urn:ietf:params:acme:error:malformed", "detail": "Unable to update challenge :: authorization must be pending", "status": 400 }

This is on FreePBX 14.0.13.28 and Certificate Manager 14.0.8.
I updated all modules, ran yum update and rebooted. Problem persists.

The second server is also Distro FreePBX 14.0.13.26 (not yet updated) running Certificate Manager 14.0.6.

I still have the soon to expire cert and when trying to renew I get the following:
There was an error updating the certificate: Verification timed out

From apache access_log on second server:

    11.22.33.217 - - [03/May/2020:11:41:06 -0400] "GET /admin/ajax.php?module=search&command=global HTTP/1.1" 200 14103 "http://my.pbx.com/admin/config.php?display=certman&action=view&id=1" "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_14_6) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/81.0.4044.113 Safari/537.36"
    ::1 - - [03/May/2020:11:41:14 -0400] "OPTIONS * HTTP/1.0" 200 - "-" "Apache/2.4.6 (Sangoma) OpenSSL/1.0.2k-fips PHP/5.6.40 (internal dummy connection)"
    199.102.239.170 - - [03/May/2020:11:41:25 -0400] "GET /.freepbx-known/b99328a7bb2299536f878611215bd0db HTTP/1.1" 200 32 "-" "-"
    ::1 - - [03/May/2020:11:41:26 -0400] "GET /.well-known/acme-challenge/OLrAjsFN6favT3rcIqo3c3DVNGxJAJ6Rp_Yo65ngeLQ HTTP/1.0" 200 87 "-" "Wget/1.10.2 (Red Hat modified)"
    66.133.109.36 - - [03/May/2020:11:41:28 -0400] "GET /.well-known/acme-challenge/OLrAjsFN6favT3rcIqo3c3DVNGxJAJ6Rp_Yo65ngeLQ HTTP/1.1" 200 87 "-" "Mozilla/5.0 (compatible; Let's Encrypt validation server; +https://www.letsencrypt.org)"
    66.133.109.36 - - [03/May/2020:11:41:29 -0400] "GET /.well-known/acme-challenge/OLrAjsFN6favT3rcIqo3c3DVNGxJAJ6Rp_Yo65ngeLQ HTTP/1.1" 200 87 "-" "Mozilla/5.0 (compatible; Let's Encrypt validation server; +https://www.letsencrypt.org)"
    66.133.109.36 - - [03/May/2020:11:41:30 -0400] "GET /.well-known/acme-challenge/OLrAjsFN6favT3rcIqo3c3DVNGxJAJ6Rp_Yo65ngeLQ HTTP/1.1" 200 87 "-" "Mozilla/5.0 (compatible; Let's Encrypt validation server; +https://www.letsencrypt.org)"
    64.78.149.164 - - [03/May/2020:11:41:31 -0400] "GET /.well-known/acme-challenge/OLrAjsFN6favT3rcIqo3c3DVNGxJAJ6Rp_Yo65ngeLQ HTTP/1.1" 200 87 "-" "Mozilla/5.0 (compatible; Let's Encrypt validation server; +https://www.letsencrypt.org)"
    64.78.149.164 - - [03/May/2020:11:41:32 -0400] "GET /.well-known/acme-challenge/OLrAjsFN6favT3rcIqo3c3DVNGxJAJ6Rp_Yo65ngeLQ HTTP/1.1" 200 87 "-" "Mozilla/5.0 (compatible; Let's Encrypt validation server; +https://www.letsencrypt.org)"
    11.22.33.217 - - [03/May/2020:11:41:25 -0400] "POST /admin/config.php?display=certman HTTP/1.1" 200 41010 "http://my.pbx.com/admin/config.php?display=certman&action=view&id=1" "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_14_6) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/81.0.4044.113 Safari/537.36"
    11.22.33.217 - - [03/May/2020:11:41:33 -0400] "GET /admin/assets/certman/less/cache/lessphp_c2126c9c8880c85219c33383fec832da3dcd1707.css HTTP/1.1" 200 260 "http://my.pbx.com/admin/config.php?display=certman" "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_14_6) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/81.0.4044.113 Safari/537.36"
    11.22.33.217 - - [03/May/2020:11:41:33 -0400] "GET /admin/assets/less/cache/lessphp_24bcf3c8d621d820ac16f650e93b3cb2f1544dcf.css HTTP/1.1" 200 87893 "http://my.pbx.com/admin/config.php?display=certman" "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_14_6) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/81.0.4044.113 Safari/537.36"
    11.22.33.217 - - [03/May/2020:11:41:34 -0400] "GET /admin/ajax.php?module=search&command=global HTTP/1.1" 200 14103 "http://my.pbx.com/admin/config.php?display=certman" "Mozilla/5.0 (Macintosh; Intel Mac OS X 10_14_6) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/81.0.4044.113 Safari/537.36"

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Outbound call rejected by mobile user and continues to call over and over

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@Glennpayne wrote:

Hi All,

Funny issue on my brand new install of freePBX. everything working using IAX2 trunk with provider, inbound routes and outbound routes all working as expected. One little issue

When dialling a mobile number that hits reject call instead of picking up the call and the mobile service has no VM or forwarding, instead of hanging up the call the pbx calls the number again and again.

I would expect that the call once rejected by the mobile user would hang up on the pbx.

Am I missing a setting?

thanks for your help

Glenn

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FreePBX yum update fails with Error: Multilib version problems found

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@sgseidel wrote:

FreePBX 14.0.13.28 / Asterisk 13.32.0

When running yum update, I see the following error:

=============================
$ sudo yum update
Loaded plugins: fastestmirror, langpacks, versionlock
Loading mirror speeds from cached hostfile

  • elrepo: muug.ca

  • nux-dextop: li.nux.ro
    Resolving Dependencies
    –> Running transaction check
    —> Package ghostscript.x86_64 0:9.07-31.el7_6.11 will be updated
    –> Processing Dependency: ghostscript = 9.07-31.el7_6.11 for package: ghostscript-cups-9.07-31.el7_6.11.x86_64
    –> Processing Dependency: libgs.so.9()(64bit) for package: 2:gimp-2.8.22-1.el7.x86_64
    –> Processing Dependency: libgs.so.9()(64bit) for package: ghostscript-9.25-2.el7_7.3.x86_64
    –> Processing Dependency: libgs.so.9()(64bit) for package: ImageMagick-6.7.8.9-16.el7_6.x86_64
    –> Processing Dependency: libgs.so.9()(64bit) for package: libspectre-0.2.8-1.el7.x86_64
    –> Processing Dependency: libijs-0.35.so()(64bit) for package: gutenprint-5.2.9-18.el7.x86_64
    –> Processing Dependency: libijs-0.35.so()(64bit) for package: cups-filters-1.0.35-22.el7.x86_64
    —> Package ghostscript.x86_64 0:9.25-2.el7_7.3 will be an update
    —> Package google-chrome-stable.x86_64 0:81.0.4044.122-1 will be updated
    —> Package google-chrome-stable.x86_64 0:81.0.4044.129-1 will be an update
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    –> Processing Dependency: urw-base35-p052-fonts for package: urw-base35-fonts-20170801-10.el7.noarch
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    –> Processing Dependency: urw-base35-nimbus-mono-ps-fonts for package: urw-base35-fonts-20170801-10.el7.noarch
    –> Processing Dependency: urw-base35-gothic-fonts for package: urw-base35-fonts-20170801-10.el7.noarch
    –> Processing Dependency: urw-base35-d050000l-fonts for package: urw-base35-fonts-20170801-10.el7.noarch
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    –> Processing Dependency: urw-base35-bookman-fonts for package: urw-base35-fonts-20170801-10.el7.noarch
    —> Package urw-fonts.noarch 0:2.4-16.el7 will be obsoleted
    –> Running transaction check
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    –> Processing Dependency: libpthread.so.0 for package: ghostscript-9.07-31.el7_6.11.i686
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    –> Processing Dependency: libgssapi_krb5.so.2 for package: ghostscript-9.07-31.el7_6.11.i686
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    –> Processing Dependency: libdl.so.2 for package: ghostscript-9.07-31.el7_6.11.i686
    –> Processing Dependency: libcupsimage.so.2 for package: ghostscript-9.07-31.el7_6.11.i686
    –> Processing Dependency: libcups.so.2 for package: ghostscript-9.07-31.el7_6.11.i686
    –> Processing Dependency: libcrypt.so.1 for package: ghostscript-9.07-31.el7_6.11.i686
    –> Processing Dependency: libcom_err.so.2 for package: ghostscript-9.07-31.el7_6.11.i686
    –> Processing Dependency: libc.so.6(GLIBC_2.11) for package: ghostscript-9.07-31.el7_6.11.i686
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    –> Processing Dependency: libSM.so.6 for package: ghostscript-9.07-31.el7_6.11.i686
    –> Processing Dependency: libICE.so.6 for package: ghostscript-9.07-31.el7_6.11.i686
    —> Package ghostscript.x86_64 0:9.07-31.el7_6.11 will be updated
    —> Package libgs.x86_64 0:9.25-2.el7_7.3 will be installed
    –> Processing Dependency: adobe-mappings-pdf for package: libgs-9.25-2.el7_7.3.x86_64
    –> Processing Dependency: adobe-mappings-cmap-deprecated for package: libgs-9.25-2.el7_7.3.x86_64
    –> Processing Dependency: adobe-mappings-cmap for package: libgs-9.25-2.el7_7.3.x86_64
    –> Processing Dependency: libpaper.so.1()(64bit) for package: libgs-9.25-2.el7_7.3.x86_64
    –> Processing Dependency: libopenjp2.so.7()(64bit) for package: libgs-9.25-2.el7_7.3.x86_64
    —> Package urw-base35-bookman-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-c059-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-d050000l-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-fonts-common.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-gothic-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-nimbus-mono-ps-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-nimbus-roman-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-nimbus-sans-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-p052-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-standard-symbols-ps-fonts.noarch 0:20170801-10.el7 will be installed
    —> Package urw-base35-z003-fonts.noarch 0:20170801-10.el7 will be installed
    –> Running transaction check
    —> Package adobe-mappings-cmap.noarch 0:20171205-3.el7 will be installed
    —> Package adobe-mappings-cmap-deprecated.noarch 0:20171205-3.el7 will be installed
    —> Package adobe-mappings-pdf.noarch 0:20180407-1.el7 will be installed
    —> Package cups-libs.i686 1:1.6.3-35.el7 will be installed
    –> Processing Dependency: libstdc++.so.6(GLIBCXX_3.4) for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libstdc++.so.6(CXXABI_1.3) for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libstdc++.so.6 for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libssl.so.10(libssl.so.10) for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libssl.so.10 for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libselinux.so.1 for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libgcc_s.so.1(GCC_3.0) for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libgcc_s.so.1 for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libcrypto.so.10(libcrypto.so.10) for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libcrypto.so.10 for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libavahi-common.so.3 for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libavahi-client.so.3 for package: 1:cups-libs-1.6.3-35.el7.i686
    –> Processing Dependency: libaudit.so.1 for package: 1:cups-libs-1.6.3-35.el7.i686
    —> Package fontconfig.i686 0:2.13.0-4.3.el7 will be installed
    –> Processing Dependency: libuuid.so.1(UUID_1.0) for package: fontconfig-2.13.0-4.3.el7.i686
    –> Processing Dependency: libuuid.so.1 for package: fontconfig-2.13.0-4.3.el7.i686
    –> Processing Dependency: libexpat.so.1 for package: fontconfig-2.13.0-4.3.el7.i686
    —> Package freetype.i686 0:2.8-12.el7_6.1 will be installed
    –> Processing Dependency: libbz2.so.1 for package: freetype-2.8-12.el7_6.1.i686
    —> Package glibc.i686 0:2.17-260.el7_6.6 will be installed
    –> Processing Dependency: libfreebl3.so(NSSRAWHASH_3.12.3) for package: glibc-2.17-260.el7_6.6.i686
    –> Processing Dependency: libfreebl3.so for package: glibc-2.17-260.el7_6.6.i686
    —> Package krb5-libs.i686 0:1.15.1-37.el7_6 will be installed
    –> Processing Dependency: libverto.so.1 for package: krb5-libs-1.15.1-37.el7_6.i686
    –> Processing Dependency: libkeyutils.so.1(KEYUTILS_1.5) for package: krb5-libs-1.15.1-37.el7_6.i686
    –> Processing Dependency: libkeyutils.so.1(KEYUTILS_1.0) for package: krb5-libs-1.15.1-37.el7_6.i686
    –> Processing Dependency: libkeyutils.so.1(KEYUTILS_0.3) for package: krb5-libs-1.15.1-37.el7_6.i686
    –> Processing Dependency: libkeyutils.so.1 for package: krb5-libs-1.15.1-37.el7_6.i686
    —> Package lcms2.i686 0:2.6-3.el7 will be installed
    —> Package libICE.i686 0:1.0.9-9.el7 will be installed
    —> Package libSM.i686 0:1.2.2-2.el7 will be installed
    —> Package libX11.i686 0:1.6.5-2.el7 will be installed
    –> Processing Dependency: libxcb.so.1 for package: libX11-1.6.5-2.el7.i686
    —> Package libXext.i686 0:1.3.3-3.el7 will be installed
    —> Package libXt.i686 0:1.1.5-3.el7 will be installed
    —> Package libcom_err.i686 0:1.42.9-13.el7 will be installed
    —> Package libjpeg-turbo.i686 0:1.2.90-6.el7 will be installed
    —> Package libpaper.x86_64 0:1.1.24-8.el7 will be installed
    —> Package libpng.i686 2:1.5.13-7.el7_2 will be installed
    —> Package libtiff.i686 0:4.0.3-27.el7_3 will be installed
    –> Processing Dependency: libjbig.so.2.0 for package: libtiff-4.0.3-27.el7_3.i686
    —> Package openjpeg2.x86_64 0:2.3.1-1.el7 will be installed
    —> Package zlib.i686 0:1.2.7-18.el7 will be installed
    –> Running transaction check
    —> Package audit-libs.i686 0:2.8.4-4.el7 will be installed
    –> Processing Dependency: libcap-ng.so.0 for package: audit-libs-2.8.4-4.el7.i686
    —> Package avahi-libs.i686 0:0.6.31-19.el7 will be installed
    –> Processing Dependency: libdbus-1.so.3 for package: avahi-libs-0.6.31-19.el7.i686
    —> Package bzip2-libs.i686 0:1.0.6-13.el7 will be installed
    —> Package expat.i686 0:2.1.0-10.el7_3 will be installed
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    —> Package keyutils-libs.i686 0:1.5.8-3.el7 will be installed
    —> Package libgcc.i686 0:4.8.5-36.el7_6.2 will be installed
    —> Package libselinux.i686 0:2.5-14.1.el7 will be installed
    –> Processing Dependency: libsepol(x86-32) >= 2.5-10 for package: libselinux-2.5-14.1.el7.i686
    –> Processing Dependency: libsepol.so.1(LIBSEPOL_1.0) for package: libselinux-2.5-14.1.el7.i686
    –> Processing Dependency: libsepol.so.1 for package: libselinux-2.5-14.1.el7.i686
    –> Processing Dependency: libpcre.so.1 for package: libselinux-2.5-14.1.el7.i686
    —> Package libstdc++.i686 0:4.8.5-36.el7_6.2 will be installed
    —> Package libuuid.i686 0:2.23.2-59.el7_6.1 will be installed
    —> Package libverto.i686 0:0.2.5-4.el7 will be installed
    —> Package libxcb.i686 0:1.13-1.el7 will be installed
    –> Processing Dependency: libXau.so.6 for package: libxcb-1.13-1.el7.i686
    —> Package nss-softokn-freebl.i686 0:3.36.0-5.el7_5 will be installed
    —> Package openssl-libs.i686 1:1.0.2k-16.el7_6.1 will be installed
    –> Running transaction check
    —> Package dbus-libs.i686 1:1.10.24-13.el7_6 will be installed
    –> Processing Dependency: libsystemd.so.0(LIBSYSTEMD_209) for package: 1:dbus-libs-1.10.24-13.el7_6.i686
    –> Processing Dependency: libsystemd.so.0 for package: 1:dbus-libs-1.10.24-13.el7_6.i686
    —> Package libXau.i686 0:1.0.8-2.1.el7 will be installed
    —> Package libcap-ng.i686 0:0.7.5-4.el7 will be installed
    —> Package libsepol.i686 0:2.5-10.el7 will be installed
    —> Package pcre.i686 0:8.32-17.el7 will be installed
    –> Running transaction check
    —> Package systemd-libs.i686 0:219-62.el7_6.7 will be installed
    –> Processing Dependency: libpam_misc.so.0(LIBPAM_MISC_1.0) for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libpam_misc.so.0 for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libpam.so.0(LIBPAM_MODUTIL_1.0) for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libpam.so.0(LIBPAM_EXTENSION_1.0) for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libpam.so.0(LIBPAM_1.0) for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libpam.so.0 for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: liblzma.so.5(XZ_5.0) for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: liblzma.so.5 for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: liblz4.so.1 for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libgpg-error.so.0 for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libgcrypt.so.11(GCRYPT_1.2) for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libgcrypt.so.11 for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libdw.so.1 for package: systemd-libs-219-62.el7_6.7.i686
    –> Processing Dependency: libcap.so.2 for package: systemd-libs-219-62.el7_6.7.i686
    –> Running transaction check
    —> Package elfutils-libs.i686 0:0.172-2.el7 will be installed
    –> Processing Dependency: elfutils-libelf(x86-32) = 0.172-2.el7 for package: elfutils-libs-0.172-2.el7.i686
    –> Processing Dependency: libelf.so.1(ELFUTILS_1.7) for package: elfutils-libs-0.172-2.el7.i686
    –> Processing Dependency: libelf.so.1(ELFUTILS_1.6) for package: elfutils-libs-0.172-2.el7.i686
    –> Processing Dependency: libelf.so.1(ELFUTILS_1.5) for package: elfutils-libs-0.172-2.el7.i686
    –> Processing Dependency: libelf.so.1(ELFUTILS_1.4) for package: elfutils-libs-0.172-2.el7.i686
    –> Processing Dependency: libelf.so.1(ELFUTILS_1.3) for package: elfutils-libs-0.172-2.el7.i686
    –> Processing Dependency: libelf.so.1(ELFUTILS_1.1.1) for package: elfutils-libs-0.172-2.el7.i686
    –> Processing Dependency: libelf.so.1(ELFUTILS_1.0) for package: elfutils-libs-0.172-2.el7.i686
    –> Processing Dependency: libelf.so.1 for package: elfutils-libs-0.172-2.el7.i686
    —> Package libcap.i686 0:2.22-9.el7 will be installed
    –> Processing Dependency: libattr.so.1(ATTR_1.0) for package: libcap-2.22-9.el7.i686
    –> Processing Dependency: libattr.so.1 for package: libcap-2.22-9.el7.i686
    —> Package libgcrypt.i686 0:1.5.3-14.el7 will be installed
    —> Package libgpg-error.i686 0:1.12-3.el7 will be installed
    —> Package lz4.i686 0:1.7.5-2.el7 will be installed
    —> Package pam.i686 0:1.1.8-22.el7 will be installed
    –> Processing Dependency: libdb-5.3.so for package: pam-1.1.8-22.el7.i686
    –> Processing Dependency: libcrack.so.2 for package: pam-1.1.8-22.el7.i686
    —> Package xz-libs.i686 0:5.2.2-1.el7 will be installed
    –> Running transaction check
    —> Package cracklib.i686 0:2.9.0-11.el7 will be installed
    —> Package elfutils-libelf.i686 0:0.172-2.el7 will be installed
    —> Package libattr.i686 0:2.4.46-13.el7 will be installed
    —> Package libdb.i686 0:5.3.21-24.el7 will be installed
    –> Finished Dependency Resolution
    Error: Multilib version problems found. This often means that the root
    cause is something else and multilib version checking is just
    pointing out that there is a problem. Eg.:

      1. You have an upgrade for ghostscript which is missing some
         dependency that another package requires. Yum is trying to
         solve this by installing an older version of ghostscript of the
         different architecture. If you exclude the bad architecture
         yum will tell you what the root cause is (which package
         requires what). You can try redoing the upgrade with
         --exclude ghostscript.otherarch ... this should give you an error
         message showing the root cause of the problem.
    
      2. You have multiple architectures of ghostscript installed, but
         yum can only see an upgrade for one of those architectures.
         If you don't want/need both architectures anymore then you
         can remove the one with the missing update and everything
         will work.
    
      3. You have duplicate versions of ghostscript installed already.
         You can use "yum check" to get yum show these errors.
    
    ...you can also use --setopt=protected_multilib=false to remove
    this checking, however this is almost never the correct thing to
    do as something else is very likely to go wrong (often causing
    much more problems).
    
    Protected multilib versions: ghostscript-9.07-31.el7_6.11.i686 != ghostscript-9.25-2.el7_7.3.x86_64
    

$ sudo yum list installed | grep -i ghost
ghostscript.x86_64 9.07-31.el7_6.11 @sng-updates/7.5
ghostscript-cups.x86_64 9.07-31.el7_6.11 @sng-updates
ghostscript-fonts.noarch 5.50-32.el7 @anaconda/1712
$

=========================
I’m not sure how to progress this further. Anyone have suggestions?

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Problem adding custom application for MOH

$
0
0

@idiotzoo wrote:

Doing some experiments with streaming MOH and I’ve run into a show stopper. When attempting to add the custom application for streaming MOH I get this error:

SQLSTATE[22007]: Invalid datetime format: 1366 Incorrect integer value: ‘’ for column asterisk.music.random at row 1
File:/var/www/html/admin/modules/music/Music.class.php:135

It’s a fresh install on top of Debian 10 following the FreePBX wiki guidelines. Everything else seems to be working. Is this a bug or something I’ve done wrong?

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