We are using commercial Paging Pro to do emergency alerts. However, I have the announce count set to 3 with a 60 second recording to play. It only plays one time and then there is dead air till I hang up. Is there a setting somewhere I am missing or any idea what could be happening? I can talk on the page and that still works, it just doesn’t play the recording the second or third time.
Thank you
1 post - 1 participant
I have moved property and my Freepbx was not updated. I installed a fresh copy which is 16.0.40.7
Recreated my previous setup. I can dial between extensions and receive calls from my SIP provider.
I am unable to make outgoing calls. I get the response “I am sorry that is not a valid extension” when dialling out. Seems odd to me that the system thinks I am dialling an extension.
My dial plan is et to X. ie the period after X
I thought it might be Firewall, it is behind a Unifi Dream Router. I turned off the rules to Drop but no difference.
Any suggestions please?
16 posts - 2 participants
I’ve got the VMWare virtual machine which I thought was PBX In A Flash, but the name displays as “Incredible PBX 2027-U”. I used to run it about a decade ago. But the company I had a Vitelity account for no longer has that account and they are not issuing new ones. Where today can one get a DID and service affordably for businesses that host their own? I appreciate your recommendations.
3 posts - 3 participants
Hi All
I attempted to migrate my vm running freepbx from vmware esxi to xcp-ng cluster and this didnt work as there where booting issues after so i have gone down the route of building a new server on xcp-ng
so what i have done is i have performed a backup of the old host. Built a new host along side, updated the modules on the new server so they are up to date. unregistered the old server from the gui then registered the new one. performed the restore from the backup. shut down the old server. Changed the static dhcp lease so the IP address will be recieved by the new host.
on the new host i can connect no issues internally on the network however externally that has worked on the old server is not working.
I can confirm the connection attempts are being recieved because i run tail -f /var/log/fail2ban and i can see the successful connection attempts associated with the external device.
under fail2ban i see
[2024-02-29 05:13:05] SECURITY[2477] res_security_log.c: SecurityEvent=“SuccessfulAuth”,EventTV=“2024-02-29T05:13:05.633+0000”,Severity=“Informational”,Service=“PJSIP”,EventVersion=“1”,AccountID=“101”,SessionID="2817593271@192.168.1.37",LocalAddress=“IPV4/UDP/192.168.0.234/5060”,RemoteAddress=“IPV4/UDP/58.87.6.70/1030”,UsingPassword=“redacted”
under full logs i see it doing this
[2024-02-29 05:11:11] VERBOSE[9285] res_pjsip_registrar.c: Added contact ‘sip:101@58.87.6.70:1030;user=phone’ to AOR ‘101’ with expiration of 60 seconds
[2024-02-29 05:11:14] VERBOSE[9285] res_pjsip/pjsip_options.c: Contact 101/sip:101@58.87.6.70:1030;user=phone is now Unreachable. RTT: 0.000 msec
[2024-02-29 05:14:05] VERBOSE[2415] res_pjsip_registrar.c: Removed contact ‘sip:101@58.87.6.70:1030;user=phone’ from AOR ‘101’ due to expiration
[2024-02-29 05:14:05] VERBOSE[13543] res_pjsip/pjsip_options.c: Contact 101/sip:101@58.87.6.70:1030;user=phone has been deleted
[2024-02-29 05:14:22] VERBOSE[9285] res_pjsip_registrar.c: Added contact ‘sip:101@58.87.6.70:1030;user=phone’ to AOR ‘101’ with expiration of 60 seconds
[2024-02-29 05:14:25] VERBOSE[9285] res_pjsip/pjsip_options.c: Contact 101/sip:101@58.87.6.70:1030;user=phone is now Unreachable. RTT: 0.000 msec
What could be going wrong here?
1 post - 1 participant
I am trying to set up an OBI device as a PJSIP trunk. Any good guide? I found multiple ones for SIP and tried but they are not working. Thanks.
2 posts - 2 participants
Hello everyone,
I want to set up recording in my custom context in extensions_custom.conf
I have registered an extension (e.g. 6001) and checked Force to all recording options in the advanced setting of that extension.
however when I call from this extension to my custom context, everything seems to execute within it, but there is no call record for my 6001 extension.
So I started experimenting with sub-record-check, but I’m not entirely sure if I understand all the arguments, even after checking extension_additional.conf.
The third argument is clear - It declares the priority of recording, so values can be : yes | no | never | force | dontcare
But what about the first and second arguments? What do they represent? I assume they should be a CID and DID but theese argument are changing depending on internal, external or from-trunk calls.
Could someone please explain this to me?
I’ve tried many combinations and for me seems to work this one:
same => n, Gosub(sub-record-check,s,1(exten,${CALLERID(num)},dontcare))
After applying this, my custom extension is recorded. However, is it correct? Could it cause inappropriate behavior elsewhere in FreePBX?
Thank you for any insights.
1 post - 1 participant
I added streaming moh. When I call and put the call on hold, there is silence
1 post - 1 participant
Hi,
I have a server that is running FreePBX. This server also has a 4 port FXODahdi card installed.
FreePBX has detected the analogue port under DADHI config → Analog ports
There is one SIP client registered to the Freepbx.
I created a Trunk that is linked to the FXO port/DAHDI channel.
I have set up an outbound route on the FreePBX that allows me to make a call from the SIP client and route the call out via the Dahdi trunk. When I make a test outbound call it works successfully.
The next test I wanted to do was to send a flash hook from the FXO port to the FXS port after a call is established. When I do the test I make an outbound call from the SIP client to an external number, the call connects via the DAHDI trunk and there is two-way audio. When I press the hold button on the SIP client, on the other end I hear the MOH from Freepbx.
What I would like to see first is that when I press the hold button on the sip client (after the call has been established). The SIP client then sends an INVITE with the SDP attribute a=sendonly and I want the Freepbx to interpret that to send a flash hook signal on the FXO port instead of sending the music on hold generated by FreePBX.
Is this possible with my current setup?
current asterisk and FreePBX version
FreePBX 16.0.33
freepbx*CLI> core show version
Asterisk 18.20.2 built by mockbuild @ jenkins7 on a x86_64 running Linux on 2024-01-02 13:09:10 UTC
I look forward to hearing from you.
Regards,
Shanx499
1 post - 1 participant
Hi there!
Today I tested a FreePBX 17 installation on openSUSE Leap 15.5 on base of actual git repository of FreePBX. I surprisingly managed it to get it working after a few tweaks. Following a rough description:
Adding the php devel repository for php 8.2. The following packages are needed:
php8-phar
php8-curl
php8-zlib
libzip5
php8-zip
apache2-mod_php8
php8-mysql
php8-mbstring
php8-gettext
php8-gd
php8-posix
php8-sysvsem
php8-xmlwriter
php8-xmlreader
php8-tokenizer
php8-sqlite
php8-pdo
php8-openssl
php8-iconv
php8-ctype
php8-dom
php8-cli
php8
Adding the telephony repository for Asterisk.
Just install asterisk and asterisk-odbc, asterisk-spandsp and libspandsp3
Additionally you need apache2, mariadb and nodejs18.
Apache must be run as asterisk:asterisk
A few symlinks have to be set.
I cloned the FreePBX framework git repository and installed it with the provided install script. All other modules have been installed using fwconsole moduleadmin downloadinstall.
Further I used the existing http apache config files from FreePBX 16 to integrate it into apache and you have to enable php8 and rewrite module in /etc/sysconfig/apache2.
The result looks pretty promising. I think, this should work and could be used for further more tests.
Thanks
Dirk
1 post - 1 participant
Hi,
I am using FreePBX for door entry phones, and recently I’ve been getting random calls on a single softphone from an unknown random extension like 254703313643. When I answer, it’s silent and cuts off.
The door phones are connected directly via wired LAN. The soft phones connect into FreePBX via port forwarding. The soft phone app is Groundwire by Acrobits.
External SIP port = 32256
RTSP ports = 5466-5470
In the FreePBX logs I see nothing whatsoever to suggest I am being hacked. Firewall is active and shows no errors.
Any ideas?
Thanks.
4 posts - 2 participants
Hi all, I’ve got a curious problem here - our phone system is refusing to connect calls to certain numbers with seemingly no rhyme or reason to which numbers it likes, and which numbers it dislikes… but always the same numbers. I can have two numbers on the same dialing code and it will consistently allow calls to one, but not the other. Any help would be appreciated. This was happening on 14.x but I’ve just upgraded the server to 15.0.37.4 today (Asterisk 13.38.3). We use Orbtalk for our trunk and have done for many years without too many problems. The curious thing is that when this happens, the phones do ring at the other end, as clients have called us back asking why we’ve given them a one-ring call.
From what I can see in the trace, all is the same until the successful call goes through a couple of rounds of “making progress” before ringing, whereas the failed call jumps straight to “ringing” and then throwing the following:
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@macro-dialout-trunk:37] Dial("PJSIP/extn-00000xxx", "PJSIP/02920123456@Orbtalk_PJ,300,Tb(func-apply-sipheaders^s^1,(1))U(sub-send-obroute-email^02920123456^02920123456^1^1709284901^^01633xxxxxx)") in new stack
[time] VERBOSE[x][C-00000xx] app_stack.c: PJSIP/Orbtalk_PJ-00000xxx Internal Gosub(func-apply-sipheaders,s,1(1)) start
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/Orbtalk_PJ-00000xxx", "Applying SIP Headers to channel PJSIP/Orbtalk_PJ-00000xxx") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:3] Set("PJSIP/Orbtalk_PJ-00000xxx", "TECH=PJSIP") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:4] Set("PJSIP/Orbtalk_PJ-00000xxx", "SIPHEADERKEYS=Alert-Info") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:5] While("PJSIP/Orbtalk_PJ-00000xxx", "1") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:6] Set("PJSIP/Orbtalk_PJ-00000xxx", "sipheader=unset") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:7] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?SIPRemoveHeader(Alert-Info:)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:8] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "1?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
[time] ERROR[1485] res_pjsip_header_funcs.c: No headers had been previously added to this session.
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:9] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?Set(sipheader=<http://127.0.0.1>;info=unset)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:10] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?Set(sipheader=<http://127.0.0.1>unset)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:11] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?SIPAddHeader(Alert-Info:unset)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:12] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?Set(PJSIP_HEADER(add,Alert-Info)=unset)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:13] EndWhile("PJSIP/Orbtalk_PJ-00000xxx", "") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:5] While("PJSIP/Orbtalk_PJ-00000xxx", "0") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:14] Return("PJSIP/Orbtalk_PJ-00000xxx", "") in new stack
[time] VERBOSE[x][C-00000xx] app_stack.c: Spawn extension (from-trunk, 02920123456, 1) exited non-zero on 'PJSIP/Orbtalk_PJ-00000xxx'
[time] VERBOSE[x][C-00000xx] app_stack.c: PJSIP/Orbtalk_PJ-00000xxx Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[time] VERBOSE[x][C-00000xx] app_dial.c: Called PJSIP/02920123456@Orbtalk_PJ
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx is ringing
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx is ringing
[time] ERROR[13940] pjproject: sip_inv.c ....Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx answered PJSIP/extn-00000xxx
[time] VERBOSE[x][C-00000xx] app_stack.c: PJSIP/Orbtalk_PJ-00000xxx Internal Gosub(sub-send-obroute-email,s,1(02920123456,02920123456,1,1709284901,,01633xxxxxx)) start
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@sub-send-obroute-email:1] GotoIf("PJSIP/Orbtalk_PJ-00000xxx", "0?sendEmail") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@sub-send-obroute-email:2] NoOp("PJSIP/Orbtalk_PJ-00000xxx", "email notifications disabled..exiting.") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@sub-send-obroute-email:3] Return("PJSIP/Orbtalk_PJ-00000xxx", "") in new stack
[time] VERBOSE[x][C-00000xx] app_stack.c: Spawn extension (from-trunk, , 1) exited non-zero on 'PJSIP/Orbtalk_PJ-00000xxx'
[time] VERBOSE[x][C-00000xx] app_stack.c: PJSIP/Orbtalk_PJ-00000xxx Internal Gosub(sub-send-obroute-email,s,1(02920123456,02920123456,1,1709284901,,01633xxxxxx)) complete GOSUB_RETVAL=
[time] VERBOSE[31093][C-00000xx] bridge_channel.c: Channel PJSIP/Orbtalk_PJ-00000xxx joined 'simple_bridge' basic-bridge <58195129-f15d-4d45-8abc-89805e98b603>
[time] VERBOSE[x][C-00000xx] bridge_channel.c: Channel PJSIP/extn-00000xxx joined 'simple_bridge' basic-bridge <58195129-f15d-4d45-8abc-89805e98b603>
[time] VERBOSE[31093][C-00000xx] bridge_channel.c: Channel PJSIP/Orbtalk_PJ-00000xxx left 'simple_bridge' basic-bridge <58195129-f15d-4d45-8abc-89805e98b603>
[time] VERBOSE[x][C-00000xx] bridge_channel.c: Channel PJSIP/extn-00000xxx left 'simple_bridge' basic-bridge <58195129-f15d-4d45-8abc-89805e98b603>
On a working call, the same section of the log is:
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@macro-dialout-trunk:37] Dial("PJSIP/extn-00000xxx", "PJSIP/02920123456@Orbtalk_PJ,300,Tb(func-apply-sipheaders^s^1,(1))U(sub-send-obroute-email^02920123456^02920123456^1^1709286729^^01633xxxxxx)") in new stack
[time] VERBOSE[x][C-00000xx] app_stack.c: PJSIP/Orbtalk_PJ-00000xxx Internal Gosub(func-apply-sipheaders,s,1(1)) start
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp("PJSIP/Orbtalk_PJ-00000xxx", "Applying SIP Headers to channel PJSIP/Orbtalk_PJ-00000xxx") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:3] Set("PJSIP/Orbtalk_PJ-00000xxx", "TECH=PJSIP") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:4] Set("PJSIP/Orbtalk_PJ-00000xxx", "SIPHEADERKEYS=Alert-Info") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:5] While("PJSIP/Orbtalk_PJ-00000xxx", "1") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:6] Set("PJSIP/Orbtalk_PJ-00000xxx", "sipheader=unset") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:7] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?SIPRemoveHeader(Alert-Info:)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:8] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "1?Set(PJSIP_HEADER(remove,Alert-Info)=)") in new stack
[time] ERROR[7998] res_pjsip_header_funcs.c: No headers had been previously added to this session.
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:9] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?Set(sipheader=<http://127.0.0.1>;info=unset)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:10] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?Set(sipheader=<http://127.0.0.1>unset)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:11] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?SIPAddHeader(Alert-Info:unset)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:12] ExecIf("PJSIP/Orbtalk_PJ-00000xxx", "0?Set(PJSIP_HEADER(add,Alert-Info)=unset)") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:13] EndWhile("PJSIP/Orbtalk_PJ-00000xxx", "") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:5] While("PJSIP/Orbtalk_PJ-00000xxx", "0") in new stack
[time] VERBOSE[x][C-00000xx] pbx.c: Executing [s@func-apply-sipheaders:14] Return("PJSIP/Orbtalk_PJ-00000xxx", "") in new stack
[time] VERBOSE[x][C-00000xx] app_stack.c: Spawn extension (from-trunk, 02920123456, 1) exited non-zero on 'PJSIP/Orbtalk_PJ-00000xxx'
[time] VERBOSE[x][C-00000xx] app_stack.c: PJSIP/Orbtalk_PJ-00000xxx Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[time] VERBOSE[x][C-00000xx] app_dial.c: Called PJSIP/02920123456@Orbtalk_PJ
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx is making progress passing it to PJSIP/extn-00000xxx
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx is making progress passing it to PJSIP/extn-00000xxx
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx is making progress passing it to PJSIP/extn-00000xxx
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx is making progress passing it to PJSIP/extn-00000xxx
[time] SECURITY[2624] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2024-03-01T09:52:11.593+0000",Severity="Informational",Service="AMI",EventVersion="1",AccountID="myadminuser",SessionID="0x7f13e00d2498",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/41726",UsingPassword="0",SessionTV="2024-03-01T09:52:11.593+0000"
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx is ringing
[time] VERBOSE[x][C-00000xx] app_dial.c: PJSIP/Orbtalk_PJ-00000xxx is ringing
2 posts - 2 participants
Hi,
my target is take a decision before route the calls to queue:
Currently I are using the Queue’s feature Join Announcement and Join Empty (Queue Plus Option) but the join announcement is always played (expected behaviour).
how can I get the number the agents logged in a queue before reach the queue?
My FreePBX version is 15.0.37
thanks by advance.
regards,
3 posts - 2 participants
Latest FreePBX 16 with Sangoma P330 phones with latest 4_18_1 firmware.
FreePBX issue we are seeing recently seems to be related to “Internal Auto Answer” being enabled for an extension in Application/Extensions/Advanced. If you intercom-call an extension that has that enabled and the callee ends the call the your extension stays off-hook with a dead call.
For example, let’s say extension 39 is setup with “Internal Auto Answer=Intercom”.
Extension 31 should automatically end the call if the other party leaves (shouldn’t it?), fairly sure it worked this way in the past. Not sure when the issue began, my phones were using firmware 4_15_4 but upgrading them did not fix it. Not sure if this is a FreePBX or phone issue though.
I had a look at the asterisk logs and they show 39 leaving:
“bridge_channel.c: Channel Local/PAGE39@app-paging-00000464;2 left ‘simple_bridge’ basic-bridge”
I don’t see anything for extension 31 leaving until the user manually ends the call at their end. Note if “Internal Auto Answer=Disabled” for the called phone the issue does not occur.
Not sure if this is a phone firmware issue or not so I thought I’d post it here for comments.
Thanks.
1 post - 1 participant
With the advent of Dynamic Routes, I have been trying to do more in the GUI, without touching custom dial plan. I have a call file, and I am wondering what you guys think the best way to hook into the GUI is to kick off a call flow so that everyone can see what’s happening, regardless of knowing dialplan or not.
Any ideas? A misc. application that goes to the next step? Or something else?
1 post - 1 participant
Hello everyone,
My current local network consists of an on-premise FreePBX server, two Cisco IP phones and a Linphone client (softphone) on my laptop.
Everything in my local network is connected via switch and the phones are able to call each other.
For my next step, I want to bind my Lithuanian mobile phone number to my on-premise FreePBX server, so that when someone calls my mobile phone number - the Cisco IP phones which are sitting in my room and have 201 and 202 extensions start to ring. Also, it’d be cool if I could call someone using my existing mobile phone number but physically make the call using my Cisco IP phone.
I assume the first step is to find & purchase a SIP trunk provider in my country and configure my local network, so that the FreePBX server is able to reach the internet.
Please let me know if I am wrong.
Best regards,
TMR
5 posts - 3 participants
Hello all!
This is probably something simple, but I can’t get it to work. FreePBX v16 with SangomaPhone.
There are 2 queues with 3 extensions in each. The idea is to have the call come into the first queue, and after 1 minute (for example), if no one is available, go to the second queue. This works fine when all 3 extensions in the first queue are on a call or are marked as DnD. However, it doesn’t work if an extension is available, but simply doesn’t pick up for a while.
I’ve set extensions “initial ring time” to 20 and “ring time” to 0; “no answer” is set to “terminate call” with “busy”. From this, I would assume that if it’s not answered in 20 seconds, it would consider the extension as busy and move onto the next one in the queue. But in practice the call keeps ringing this extension past the 20 seconds.
I haven’t explored auto-pause, but seeing there’s no apparent auto-unpause, it won’t work, since people will forget to unpause.
Is the above the correct approach for this scenario?
Thanks!
2 posts - 1 participant
Good part of the day!
I need to get calls from freepbx, and I do this:
$query = “{
fetchAllCdrs(first: 10000, after: 0, orderby: date, startDate: "$date", endDate: "$date") {
cdrs {
id
cnum
did
disposition
duration
calldate
outbound_cnum
}
}
}”;
$response = Http::withToken($this->getAccessToken())
->post(config('free_pbx.url') . '/gql', [
'query' => $query
])
->json();
The problem is that I can’t pass startDate and endDate in format (‘Y-m-d H:i:s’), and also i can’t filter by ‘outbound_cnum’ field in the query.
Maybe there is other way to get calls filtered by date (‘Y-m-d H:i:s’) and outbound_cnum?
Is there is a way to extend this api? in /admin/modules/cdr/Api/Gql/Cdr.php there is a function:
ublic function queryCallback() {
if($this->checkAllReadScope()) {
return function() {
return [
‘fetchAllCdrs’ => […],
‘fetchCdr’ => […],
//can i add here smth like ‘fetchCustomFilterCdr’ => [‘my logic here’]
]
}
}
3 posts - 2 participants
We are struggling to get incoming fax to email working on FreePBX 16 the latest version. I have followed the instructions from other users, we have even bought the Fax Pro module, and we were told by some else that this would make it work.
Does anyone have this working on the latest version of FreePBX 16.0.40.7 please?
4 posts - 2 participants
I have a fully running freepbx and 5 extensions
Unable to register zoiper softphone on my iphone. Registration faulure no transports left to try (503)
The system is on my unifi network and I thought it might be a firewall issue but I have run out of ideas.
Any suggestions please.
1 post - 1 participant
I have an issue with voicemail greetings which are uploaded in the UCP. Even high quality wideband announcements are just squeezed down to an ugly calling .wav file finally - where can I set the formats to which uploaded greetings are converted?
For “normal” announcements uploaded to FreePBX in the admin panel I can choose all target formats to which they should be converted, is there some kind of similar setting for the uploaded greetings in UCP?
1 post - 1 participant