Thanks for confirming Dicko
↧
Is it safe to delete the contents of /var/spool/asterisk/tmp
↧
No audio on some PJSIP endpoints
I agree that something changed in FreePBX but I don’t know what.
↧
↧
No audio on some PJSIP endpoints
I didn’t say something changed in FreePBX. I said those are the two things I can come up with. That doesn’t mean something did change.
↧
BLF Substitution None Not Working
Using Sangoma phones in FreePBX EPM, BLF substitution is set to none which I understand should replace the BLF that matches the extension with a blank, or remove the BLF. Its not working for me. I can substitute the BLF for another line but it will not remove itself.
↧
No audio on some PJSIP endpoints
Why do you care? If your system doesn’t benefit from direct media, I suggest that you turn it off for all extensions and call it a day. If you have some other reason for troubleshooting (to help the community, to use in a future system, whatever), paste a log of a failing call with pjsip logger turned on and explain how that differs from expected behavior.
If your system does benefit, explain whether the failing calls were expected to have direct media but it didn’t work, or whether direct media was inappropriate but Asterisk tried to invoke it anyway. Paste a log showing what happened.
Not true. Except for those on the same LAN as an on-site PBX, nearly all user endpoints (IP phones, softphones, SIP apps) are behind a NAT. For example, internal calls on a cloud PBX have lower latency with direct media, as do calls within a branch office via a PBX at headquarters. When the users are (at least somewhat) also within earshot, the direct sound is a “pre-echo” of what’s heard over the phone and this can be quite annoying when the delay is significant.
So Asterisk has to be smart about when direct media is used. Unfortunately, SIP ALGs, firewall misconfiguration and other problems may make it impossible for Asterisk to reliably determine whether direct media should be used, which is why the administrator can disable it.
↧
↧
Editing extensions into a backup
Of course, it’s just an exercise to know if it’s doable to produce a backup file that can be restored without any further config manipulation.
Thanks.
↧
Change the DID number format in PJSIP Trunk
I assume that the Context for the problematic trunk is from-pstn-toheader, but because the provider is sending in an ususual (or perhaps malformed) format, a > is being improperly included.
Make a copy of from-pstn-toheader and add it to extensions_custom.conf, using a different name such as from-pstn-mytrunk. Modify it to deliver the number without the trailing >, then change the Context for the trunk to from-pstn-mytrunk. Your normal Inbound Routes will now see DID Number as the actual number called.
↧
Change the DID number format in PJSIP Trunk
You need to write a custom context inside the extensions_custom.conf file. Something like this should do the trick (not tested):
[from-pstn-remove-char]
exten => _.,1,Goto(from-pstn,${CUT(${PJSIP_HEADER(read,From)},>,1)},1)
This removes the character > from the number and sends it to the context from-pstn. Also don’t forget to add the newly created context from-pstn-remove-char in the pjsip settings of your trunk.
↧
Sng_freepbx_debian_install failure
Yep – It was out of memory. I couldn’t grep dmesg because I nuked the instance soon after it failed.
But I’ve (through iterations) determined that the smallest AWS Debian instance that will successfully install is a 2GB / 2vCPU instance.
I appreciate the help; thanks!
↧
↧
Official release date of FreePBX 17?
Bump. It’s now 5/28, I am wondering if there is something more concrete about a forthcoming GA for FreePBX 17?
↧
Editing extensions into a backup
One could extract the files from the .gz tar file, import the mysql tables including filestore into a mysql host, run DELETE queries as approproprate, then export the purged databases and repack them with the CEL and CDR tables similarly purged , the voice mail files and the sundry audio and spool files back into a new .gz tar file.
I can’t think of anyONE actually doing that though 
↧
Change the DID number format in PJSIP Trunk
Probably ‘FILTER’ not ‘CUT’
exten => s,1,Set(foo=${FILTER(+0-9,${variable})})
↧
Issue regarding freepbx local host login
Hey guys
I’ve spent two days now trying to get my Ubuntu instance running Freepbx with asterisk up and running so my postgres db can tie together
I am stuck at this step. Ive not created an account and all the username password combos ive attempted have gotten me nowhere.
Ive looked through previous posts on this and every other solution has worked for others except me.
is there any commands i should be running from my CLI to create this login info? I have no separte admin page from what i can tell.
Thanks so much:
↧
↧
Issue regarding freepbx local host login
Most all Freepbx distros use MYSQL/MARIADB as the backend ODBC server,setting up postgres is not trivial but can work, show us your work so-far , but why do you want to swim upstream against the flow?
↧
Multiple destinations for Inbound Route
Hello. Need help setting up the incoming FreeBPX route.
The call diagram is attached.
When calling number 123321, the call goes to my FreePBX and is redirected to a voice-robot. The voice robot either answers the caller’s questions or redirects to the queue of my FreeBPX. This is a standard work situation.
If the voice-robot is disabled, the call should be redirected to a queue.
It is not possible specify multiple destinations for an incoming route. It turns out that if the voice robot trunk is disconnected, the call will not reach the queue.
What I tried to do:
Initially, the operators were located on FreePBX, and the main work took place on another Asterisk server, where it was written in code like:
exten => 123321, GotoIf($[["${DEVICE_STATE(SIP/robot)}" != "UNAVAILABLE"]?robot1,1:queue,1)
exten => robot1,1,Dial(SIP/robot1/${EXTEN},120,mt)
same => n,ExecIf($[${DIALSTATUS}=BUSY]?queue,1)
exten => queue,1,Dial(SIP/queue/1221)
On FreeBPX I tried to do this using Custom Destination, but the call did not go through. I probably did something wrong, because the FreeBPX dialplan is quite tricky.
Also I tried setting a number for the robot in Misc Applications, and adding it and the operator numbers to ring group, but then the robot always becomes unavailable and the call is redirected to the operators in any case.
What other ways are there to organize routes?
↧
No longer possible to search for contacts from Sangoma phone Desktop
Freepbx 15.0.37.4
Sangoma Desktop 4.0.3
Sangoma Realtime API 15.0.55.3
Sangoma Connect 15.0.59.3
Contact Manager 15.0.13
Asterisk REST Interface Users 15.0.3.20
↧
Domain in extension config
I am trying to setup FreeSBC and FreePBX. The domain is sent to FreeSBC to determine which PBX to send traffic too. How can I add the domain part to the extension config so that it will register.
↧
↧
Transfer license Module "Class of Service" from freepbx 13 to 16
I need Advice.
Now I use freepbx13 , and i have account sangoma with paid licence of plugin “Class of Service”.
Can i login with my account with paid license to freepbx16(test server) to activate class of service future for making tests???
Activation on old server freepbx13 not be lost?
↧
407 Proxy Authentication Required when trying to make call to du.ae
Hi all,
I am trying to connect my FreePBX to du.ae sip trunk.
Trunk details:
type=peer
host=5.32.4.225
outboundproxy=10.59.108.25
qualify=yes
fromdomain=5.32.4.225
fromuser=97145627500p
realm=5.32.4.225
authuser=97145627500p
secret=XXXX
context=from-pstn-toheader
insecure=port,invite
dtmfmode=rfc2833
directmedia=no
disallow=all
allow=ulaw
bindaddr=10.15.47.142
Trunk is reachable and well connected, but when I am trying to call via the trunk I am getting this error:
<--- SIP read from UDP:10.59.108.25:5060 --->
SIP/2.0 407 Proxy Authentication Required
Call-ID: 07593afe579b2a24496ab45a0d14b3a6@151.253.188.7:5060
Via: SIP/2.0/UDP 151.253.188.7:5060;received=10.15.47.142;branch=z9hG4bK186f3623;rport=5060
To: <sip:5.32.4.225>;tag=64dbdea1-6656fa1e614d7c
From: "Unknown" <sip:97145627500p@151.253.188.7>;tag=as64de688c
CSeq: 102 OPTIONS
Date: Wed, 29 May 2024 09:49:18 GMT
Warning: 399 sbc.du.com "IP association no match, user not registered"
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '07593afe579b2a24496ab45a0d14b3a6@151.253.188.7:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 10.59.108.25:5060:
OPTIONS sip:10.59.108.25 SIP/2.0
Via: SIP/2.0/UDP 151.253.188.7:5060;branch=z9hG4bK622fd97a;rport
Max-Forwards: 70
From: "Unknown" <sip:97145627500p@151.253.188.7>;tag=as25bec40e
To: <sip:10.59.108.25>
Contact: <sip:97145627500p@151.253.188.7:5060>
Call-ID: 74c50b3039ac184762fae25c1531be4c@151.253.188.7:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-16.0.40.7(18.16.0)
Date: Wed, 29 May 2024 08:14:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
I am going out with the usable IP recieved from the trunk - 10.15.47.142.
Details in the trunk are correct and no issue with trunk connectivity, only when call is made via this one.
Any ideas? thanks in advance.
↧
Im having trouble with audio on my pbx ever since i started sharing the same public ip with pfsense firewall
Im having trouble with audio on my pbx ever since i started sharing the same public ip with pfsense firewall
↧