Ward’s issues with all of this were things implemented by the Schmooze team that seems to have mostly went on to form Clearly IP.
Prior to this, Sangoma has been pretty stand off about everything FreePBX related. Too much stnad off IMO.
Of course behind the scenes issues that made those people want to leave is also possible. None of us can know that unless it is made public.
Did you configure the Zulu server with a valid SSL certificate, with a STUN server and port forwarding as described in the wiki?
https://wiki.freepbx.org/plugins/servlet/mobile?contentId=98699558#content/view/98699558
https://wiki.freepbx.org/plugins/servlet/mobile?contentId=110003971#content/view/110003971
Hello
I have a small-sized installation with only 2 extensions so use the free distro of Freepbx running on Debian.
Today i bought x3 Sangoma wired phones ( S705 - S305 ) and x1 Sangoma DECT (DC10M + DC201)
I tryed to install Endpoint manager to set up these new phones in free version as i read Endpoint manager might be free when used with Sangoma phones but process is unable to proceed as “PHP Component Zend Guard Loader” is required.
System Admin is also mandatory to get Endpoint managed installed, but when trying to install System Admin, i get the following message :
Errors with selection:
- System Admin cannot be installed:
* PHP Component Zend Guard Loader is required but missing from you PHP installation.Please try again after the dependencies have been installed.
- The File “/usr/sbin/incrond” must exist.
No actions to perform
So i’m stuck when installing System Admin and PHP Component Zend Guard Loader component is also not listed into available downloads.
Many thanks for your help and advices,
FreePBX: 14.1-1.sng7.x86_64
Asterisk: 13.29.2-1.sng7.x86_64
I have an Analog card (AEX800) where inbound and outbound calls work. Inbound routes to IVR just fine.
I have a Digium SIP trunk and inbound calls reach the PBX but get an error greeting “The number you have dialed is not in service”
I ran asterisk -rvvvvv and watched the trace of inbound calls from the analog pstn trunk and the digium sip trunk. Everything appears the same until the AGI Script sangomacrm.agi line.
Analog (good/working)
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <DAHDI/25-1>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@from-pstn:25] ExecIf(“DAHDI/25-1”, “1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
– Executing [s@from-pstn:26] Goto(“DAHDI/25-1”, “ivr-1,s,1”) in new stack
– Goto (ivr-1,s,1)
SIP (bad/fails to route to IVR)
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
== Begin MixMonitor Recording PJSIP/digium-siptrunk-0000000d
– <PJSIP/digium-siptrunk-0000000d>AGI Script sangomacrm.agi completed, returning 4
== Spawn extension (from-pstn, XXXXXXXXXX, 24) exited non-zero on ‘PJSIP/digium-siptrunk-0000000d’
– Executing [h@from-pstn:1] Macro(“PJSIP/digium-siptrunk-0000000d”, “hangupcall,”) in new stack
Being a Linux person I know return 0 is good and return 4 (or anything non-zero) is bad…but it doesn’t say why.
The inbound analog truck calls to go IVR, the SIP trunk inbound fails. The Digium trunk delivers a CID that is +1XXXXXXXXXX whereas the analog has a CID of XXXXXXXXXX. That doesn’t seem to be it.
Why wont the inbound SIP calls go to IVR? I have the route defined to go to IVR. But the sangomacrm.agi appears to be the breakpoint where the inbound SIP call goes off the rails.
What am I missing?
On Ext 1031 while connected to my cell phone, *2 and ## and will not get you a dial tone. . Also tried if on the Cell phone and same result.
I also don’t see incoming on the CDR also but I’m no expert on analyzing CDR’s
It might be a coincident that it stop when I disable direct dial
Ext 1031 is a Sangoma s500 Zero Touch configured with endpoint manager
Assuming the hacker got the password for ext 1031
And register another ext 1031, I should see his IP address right?
The IP address I’m seeing for 1031 is correct.
How about if I change the the password and then use endpoint manager to reboot the phone
I don’t have physical access to the phone. WIll it reboot and get the new config?
Every time I to pastebin and click on capTCHA, it removes what I paste
Hope this link works
https://drive.google.com/open?id=18FCEIcFpeMIAxf9Htea-mwbUYvDUDWnG
This file is complere but I change the IP address just incase someone else is reading it.
Thanks
I tried deleting the PJSIP trunk and re-adding it as SIP, per the instructions on Digium’s site:
https://support.digium.com/community/s/article/How-to-configure-a-Digium-SIP-Trunking-account-with-AsteriskNOW-and-FreePBX
Same failure. The call is send down my SIP trunk from Digium. FreePBX gets it, I see it in asterisk -rvvvv. It processes the inbound call and again it hits the " sangomacrm.agi completed, returning 4"
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
== Begin MixMonitor Recording SIP/digium-siptrunk-00000000
– <SIP/digium-siptrunk-00000000>AGI Script sangomacrm.agi completed, returning 4
== Spawn extension (from-trunk, XXXXXXXXXX, 24) exited non-zero on ‘SIP/digium-siptrunk-00000000’
– Executing [h@from-trunk:1] Macro(“SIP/digium-siptrunk-00000000”, “hangupcall,”) in new stack
Why in the bleeping does sangomacrm.agi return a 4 on a SIP or PJSIP inbound but returns a 0 and successfully routes the call to IVR for an inbound analog call?!?!?!?!?!
Yes Endpoint Manager is free for use with Sangoma Phones but the module is a Commercial Module which is only supported on the FreePBX Distro not Debian. If you install the official Distro then Zend Guard Loader will be installed and the System Admin module will be there.
Link to Distro: https://www.freepbx.org/downloads/freepbx-distro/
There is a bug in FreePBX (recently fixed for I think 14 & 15 only) which allowed callers to dial any extension in an IVR even it was tied to a directory.
Okay…I disabled the CRM module and that wasn’t it. It still fails.
Inbound from analog-pstn routes to IVR 1.
Inbound from digium sip trunk fails to route to IVR 1. Call gets “The number you have diailed…” error recording.
Working:
– Executing [recordcheck@sub-record-check:3] Return(“DAHDI/25-1”, “”) in new stack
– Executing [in@sub-record-check:5] Return(“DAHDI/25-1”, “”) in new stack
– Executing [s@from-pstn:4] Set(“DAHDI/25-1”, “CHANNEL(tonezone)=us”) in new stack
– Executing [s@from-pstn:5] ExecIf(“DAHDI/25-1”, “1?Set(__FROM_DID=s)”) in new stack
– Executing [s@from-pstn:6] Set(“DAHDI/25-1”, “returnhere=1”) in new stack
– Executing [s@from-pstn:7] Gosub(“DAHDI/25-1”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“DAHDI/25-1”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“DAHDI/25-1”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“DAHDI/25-1”, “”) in new stack
– Executing [s@from-pstn:8] Set(“DAHDI/25-1”, “CDR(did)=s”) in new stack
– Executing [s@from-pstn:9] GotoIf(“DAHDI/25-1”, “0?”) in new stack
– Executing [s@from-pstn:10] ExecIf(“DAHDI/25-1”, “0 ?Set(CALLERID(name)=0987654321)”) in new stack
– Executing [s@from-pstn:11] Set(“DAHDI/25-1”, “__MOHCLASS=”) in new stack
– Executing [s@from-pstn:12] Set(“DAHDI/25-1”, “__REVERSAL_REJECT=FALSE”) in new stack
– Executing [s@from-pstn:13] GotoIf(“DAHDI/25-1”, “1?post-reverse-charge”) in new stack
– Goto (from-pstn,s,15)
– Executing [s@from-pstn:15] NoOp(“DAHDI/25-1”, “”) in new stack
– Executing [s@from-pstn:16] Set(“DAHDI/25-1”, “__CALLINGNAMEPRES_SV=allowed_not_screened”) in new stack
– Executing [s@from-pstn:17] Set(“DAHDI/25-1”, “__CALLINGNUMPRES_SV=allowed_not_screened”) in new stack
– Executing [s@from-pstn:18] Set(“DAHDI/25-1”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
– Executing [s@from-pstn:19] Set(“DAHDI/25-1”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
– Executing [s@from-pstn:20] NoOp(“DAHDI/25-1”, “CallerID Entry Point”) in new stack
– Executing [s@from-pstn:21] Goto(“DAHDI/25-1”, “ivr-1,s,1”) in new stack
– Goto (ivr-1,s,1)
– Executing [s@ivr-1:1] Set(“DAHDI/25-1”, “TIMEOUT_LOOPCOUNT=0”) in new stack
– Executing [s@ivr-1:2] Set(“DAHDI/25-1”, “INVALID_LOOPCOUNT=0”) in new stack
– Executing [s@ivr-1:3] Set(“DAHDI/25-1”, “_IVR_CONTEXT_ivr-1=”) in new stack
– Executing [s@ivr-1:4] Set(“DAHDI/25-1”, “_IVR_CONTEXT=ivr-1”) in new stack
– Executing [s@ivr-1:5] Set(“DAHDI/25-1”, “__IVR_RETVM=”) in new stack
– Executing [s@ivr-1:6] GotoIf(“DAHDI/25-1”, “0?skip”) in new stack
– Executing [s@ivr-1:7] Answer(“DAHDI/25-1”, “”) in new stack
– Executing [s@ivr-1:8] Set(“DAHDI/25-1”, “IVR_MSG=custom/ivmintrojj”) in new stack
– Executing [s@ivr-1:9] Set(“DAHDI/25-1”, “TIMEOUT(digit)=3”) in new stack
– Digit timeout set to 3.000
– Executing [s@ivr-1:10] ExecIf(“DAHDI/25-1”, “1?Background(custom/ivmintrojj)”) in new stack
– <DAHDI/25-1> Playing ‘custom/ivmintrojj.slin’ (language ‘en’)
== Spawn extension (ivr-1, s, 10) exited non-zero on ‘DAHDI/25-1’
– Executing [h@ivr-1:1] Hangup(“DAHDI/25-1”, “”) in new stack
== Spawn extension (ivr-1, h, 1) exited non-zero on ‘DAHDI/25-1’
– Hanging up on ‘DAHDI/25-1’
– Hungup ‘DAHDI/25-1’
Failing:
– Executing [recordcheck@sub-record-check:3] Return(“PJSIP/digium-siptrunk-0000001f”, “”) in new stack
– Executing [in@sub-record-check:5] Return(“PJSIP/digium-siptrunk-0000001f”, “”) in new stack
– Executing [1234567890@from-pstn:4] Set(“PJSIP/digium-siptrunk-0000001f”, “CHANNEL(tonezone)=us”) in new stack
– Executing [1234567890@from-pstn:5] Set(“PJSIP/digium-siptrunk-0000001f”, “__FROM_DID=1234567890”) in new stack
– Executing [1234567890@from-pstn:6] Set(“PJSIP/digium-siptrunk-0000001f”, “returnhere=1”) in new stack
– Executing [1234567890@from-pstn:7] Gosub(“PJSIP/digium-siptrunk-0000001f”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“PJSIP/digium-siptrunk-0000001f”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“PJSIP/digium-siptrunk-0000001f”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“PJSIP/digium-siptrunk-0000001f”, “”) in new stack
– Executing [1234567890@from-pstn:8] Set(“PJSIP/digium-siptrunk-0000001f”, “CDR(did)=8588002052”) in new stack
– Executing [1234567890@from-pstn:9] GotoIf(“PJSIP/digium-siptrunk-0000001f”, “0?”) in new stack
– Executing [1234567890@from-pstn:10] ExecIf(“PJSIP/digium-siptrunk-0000001f”, “0 ?Set(CALLERID(name)=0987654321)”) in new stack
– Executing [1234567890@from-pstn:11] Set(“PJSIP/digium-siptrunk-0000001f”, “__MOHCLASS=”) in new stack
– Executing [1234567890@from-pstn:12] Set(“PJSIP/digium-siptrunk-0000001f”, “__REVERSAL_REJECT=FALSE”) in new stack
– Executing [1234567890@from-pstn:13] GotoIf(“PJSIP/digium-siptrunk-0000001f”, “1?post-reverse-charge”) in new stack
– Goto (from-pstn,1234567890,15)
– Executing [1234567890@from-pstn:15] NoOp(“PJSIP/digium-siptrunk-0000001f”, “”) in new stack
– Executing [1234567890@from-pstn:16] Set(“PJSIP/digium-siptrunk-0000001f”, “__CALLINGNAMEPRES_SV=allowed_not_screened”) in new stack
– Executing [1234567890@from-pstn:17] Set(“PJSIP/digium-siptrunk-0000001f”, “__CALLINGNUMPRES_SV=allowed_not_screened”) in new stack
– Executing [1234567890@from-pstn:18] Set(“PJSIP/digium-siptrunk-0000001f”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
– Executing [1234567890@from-pstn:19] Set(“PJSIP/digium-siptrunk-0000001f”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
– Executing [1234567890@from-pstn:20] NoOp(“PJSIP/digium-siptrunk-0000001f”, “CallerID Entry Point”) in new stack
– Executing [1234567890@from-pstn:21] Goto(“PJSIP/digium-siptrunk-0000001f”, “ivr-1,s,1”) in new stack
– Goto (ivr-1,s,1)
– Executing [s@ivr-1:1] Set(“PJSIP/digium-siptrunk-0000001f”, “TIMEOUT_LOOPCOUNT=0”) in new stack
– Executing [s@ivr-1:2] Set(“PJSIP/digium-siptrunk-0000001f”, “INVALID_LOOPCOUNT=0”) in new stack
– Executing [s@ivr-1:3] Set(“PJSIP/digium-siptrunk-0000001f”, “_IVR_CONTEXT_ivr-1=”) in new stack
– Executing [s@ivr-1:4] Set(“PJSIP/digium-siptrunk-0000001f”, “_IVR_CONTEXT=ivr-1”) in new stack
– Executing [s@ivr-1:5] Set(“PJSIP/digium-siptrunk-0000001f”, “__IVR_RETVM=”) in new stack
– Executing [s@ivr-1:6] GotoIf(“PJSIP/digium-siptrunk-0000001f”, “0?skip”) in new stack
– Executing [s@ivr-1:7] Answer(“PJSIP/digium-siptrunk-0000001f”, “”) in new stack
== Spawn extension (ivr-1, s, 7) exited non-zero on ‘PJSIP/digium-siptrunk-0000001f’
– Executing [h@ivr-1:1] Hangup(“PJSIP/digium-siptrunk-0000001f”, “”) in new stack
== Spawn extension (ivr-1, h, 1) exited non-zero on ‘PJSIP/digium-siptrunk-0000001f’
I don’t get why the pjsip trunk inboound route won’t work.
Try disabling the module?
Latest update…
Still having issues with the registration becoming unreachable from our school sites. But i have setup a softphone that works fine so i’m guessing it’s something on the schools firewall which i’ll have to investigate Monday.
My next issue is i can’t seem to get the PJSIP trunks registered, we use Gamma who are IP auth based so in my mind it should be as simple as;
Trunk Name: School Name
Outbound Caller ID: The DDI we use for that trunk
Max Channels: 3
SIP Server: 109...31 - double checked that this matches with what was provided by Gamma
SIP Port: 5060 - confirmed with Gamma this is correct.
Looking at the logs when making a call i get:
[2020-02-09 11:44:55] VERBOSE[6469][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:34] NoOp("SIP/1056-00000009", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3") in new stack
[2020-02-09 11:44:55] VERBOSE[6469][C-00000009] pbx.c: Executing [s@macro-dialout-trunk:35] GotoIf("SIP/1056-00000009", "0?continue,1:s-CHANUNAVAIL,1") in new stackI tried, but the asterisk did not start.
How did you disable it?
I have put the cell module on the excluded list in the asterisk modules page.
Try disabling the FreePBX module.
fwconsole ma disable cel
fwconsole reload
But I’m honestly not sure if this will do what you want.
This disables the cel module in the freepbx web environment, but does not disable the cel writing service. I need to disable it, so that it does not load the system (I do not use it).
I have solved the problem. I have had to override the general section in the cel_custom.conf file
[general]
enable=no
Earlier it was [general] with the plus sign, as if something was added.
I’m using 15.0.16.42 with Asterisk 16.6.2
I get the above error when trying to “Apply Config”
When I run that I get the following error: [Exception (404)]
Unable to locate the FreePBX BMO Class 'Pjsip’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install pjsip 2) fwconsole ma enable pjsip
When I run the first one I get the following error: Unable to install module pjsip:
When trying to run the second I get: The following error(s) occured:
Does anyone know what could be the problem?
It started after I updated the Modules.
I just have to say that this is very upsetting how Sangoma/Schmooze is handling this matter. I don’t exactly consider it advance notice when you tell someone on a Friday that when you come in on Monday your system may not work. That right there shows that they have no regard for their customers or installers.
Additionally, FreePBX was and is meant to be open source, to allow people to make customizations at their own will, to use third-party modules if they so choose. The fact the Sangoma/Schmooze is now disallowing the use of such modules is terrible and shows that it’s only about the money to them and that they do not care about the community of users.