@sentinelace wrote:
I am trying to set the Blind Transfer Alert Info and Attended Transfer Alert info to a custom ringtone on a yealink phone as follows:
<http://127.0.0.1/Vin/Ring6.wav>;info=TransferThis is setup in the base Edit. When I test a call and do a call, it terminates right away. I have also tried ring3.wav and it just ignores it all together.
How is this supposed to be setup?
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Participants: 1
@jpark1205 wrote:
Hello!
Currently, I am working on a project creating the callcenter program. Everything was setup up and the call worked for both inbound and outbound but, after the certificate update the phone call does not work anymore. It displays the error saying its busy on the inspect console. I have no idea what to do next…
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Participants: 1
@indamax wrote:
Hi, I’ve created a Misc destination to a movile. When the call is transfered, the mobile show the number of FreePBX, not who called.
Example: I’ve the 999999999 movile number, i call to the 888888888 (freepbx) where i’ve the misc destination and it is transfered to 777777777. On 777777777 appears the number 888888888 (freepbx) and I want the 999999999.
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Participants: 1
@Flurd wrote:
The login button on the phone gives me the message “Error” no matter what I enter. Here’s my setup:
- FreePBX 14.0.13.4
- Asterisk 13.22.0
- Sangoma S305 IP Phone
- Config server path is set to t he IP of FreePBX
- PJSIP extension 2204 created under Application->Extensions with secret 2204 and the user password set to 2204
I press the login button and key in 2204 for both the username and password, hit done, and the screen says ‘Error’ but nothing else. What do I need to do to allow this to log in? Where are the error logs? Is there any documentation that explains this?
I’ve been going through the Sangoma PBXact videos but it seems too different to FreePBX and doesn’t explain how to get the login working. There doesn’t seem to be any up to date information anywhere else that I can find either.
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Participants: 1
@comtech wrote:
FreePBX/Asterisk 14
In custom dial plan I am collecting three variables from the caller.
${MONTH}
${DAY}
${YEAR}An example a caller could input is:
${MONTH}=01
${DAY}=05
${YEAR}=2019What command could I use to string the variables together and speak back the date in a format like:
January 5th, 2019Not sure if this is possible, but thought I would check.
Thanks in advance for any help or guidance here.
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Participants: 2
@trixie_no5 wrote:
Do the results of sip set debug on or pjsip set logger on output results to a file? It is difficult to capture the outputs from the command line.
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Participants: 3
@alexj5555 wrote:
Hello
FreePBX 14.0.13.4 is installed.
connected 1 trunk with 5 numbers and 30 channels.
I created 1 extension number (example 1999), set up call forwarding in the incoming rules by DID from the provider to the extension number. but when 2 calls arrive, it’s busy there.
- Does freepbx on internal numbers support multi-channel?
- is it possible to redirect from 1 external number from the provider to 1 internal number with multi-channel support ?
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Participants: 2
@AmidouFlorian92 wrote:
Hi
After installing tts module I got this error and I cannot access to freepbx admin GUI
Please help me
- Whoops\Exception\ErrorException->() /var/www/html/admin/modules/texttospeec h/functions.inc.php:25
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Participants: 2
@AmidouFlorian92 wrote:
When installing TTS module on freepbx I got this error and I when I type a command on CLI I got this error and I cannot acess to freepbx admin GUI
which: no swift in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin)
which: no flite in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin)
which: no text2wave in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin)
which: no espeak in (/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/root/bin)
Whoops\Exception\ErrorException: mkdir(): Aucun fichier ou dossier de ce type in file /var/www/html/admin/modules/texttospeech/functions.inc.php on line 25
Stack trace:
- Whoops\Exception\ErrorException->() /var/www/html/admin/modules/texttospeech/functions.inc.php:25
- mkdir() /var/www/html/admin/modules/texttospeech/functions.inc.php:25
- require_once() /var/www/html/admin/bootstrap.php:364
- require_once() /etc/freepbx.conf:9
- include_once() /var/lib/asterisk/bin/fwconsole:12
Posts: 3
Participants: 1
@dejv wrote:
I am trying to remove the CFU I currently have however even though it says that database entry was removed it still remains there.
Any ideas?
freepbxCLI> database show cfu
/cfu/409 : 3000
/cfu/410 : 3000
/cfu/412 : 3000
/cfu/413 : 3000
/cfu/414 : 3000
/cfu/415 : 3000
/cfu/416 : 3000
/cfu/417 : 3000
/cfu/428 : 3000
9 results found.
freepbxCLI> Database del CFU 428
Database entry removed.
freepbxCLI> database show cfu
/cfu/409 : 3000
/cfu/410 : 3000
/cfu/412 : 3000
/cfu/413 : 3000
/cfu/414 : 3000
/cfu/415 : 3000
/cfu/416 : 3000
/cfu/417 : 3000
/cfu/428 : 3000
9 results found.
freepbxCLI>
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Participants: 2
@Modemdwk wrote:
Hi folks.
I am not sure this forum is the correct place.With Fop2 in our environment is missing the “Calls Waiting list” in the image is an example, take a look the green color, “<601> Recepcion” Before it was working,but honestly, I don’t know why disappear .
I don’t know which plug in of FOP2 is linked with this feature. or how to enable to show the “Calls Waiting List”
This must show in Queues
Thanks for any hint
Daniel W
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Participants: 2
@completehvac wrote:
I wonder if anyone can suggest where i can look, i have setup a new install of freepbx using the latest distro. I have added a trunk with the same provider and connection details as we have working on another system, just a different sip account.
Inbound is working as expected, however when i make an outbound call it is not arriving at the trunk and instead is entered into the queue for inbound! Its driving me crazy, i realise i will need to post the logs, but im on the train currently and just wondered if anyone could suggest anything i can check!
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Participants: 2
@jpark1205 wrote:
Hello! i tried to fix this problem by looking at the past topic that are similar to my problem and tried to fix it base on the answer that was given by other users. But, it seems like i cannot fix this… is there any other suggestion to fixt this problem??
Thank You
[2019-08-21 10:53:44] ERROR[22141] res_pjsip.c: Unable to retrieve PJSIP transport ‘udp,tcp,ws,wss’
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Participants: 1
@foovax wrote:
Inbound routes are set to record NEVER
Extension is set to record FORCEDInbound call arrives and prior to extension answering [sub-record-check.recordcheck] context is executed and creates a MixMonitor on the Inbound call channel.
This works as expected until the extension performs a transfer to an outbound route.
There are two issues:
The recording has the hold music the Inbound caller is hearing and nothing from the Extensions channel. Which is where the extension is talking prior to bringing inbound call onto the line.
When extension hangs up after completing a transfer the recording continues until the inbound caller hangs up.
How can I get 1 recording of only the extension channel from an inbound/outbound call that starts when extension answers and ends when extension hangs up?
Thanks for your time!
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Participants: 1
@jltaft wrote:
I purchased System Admin Pro 2 days ago. When I try to update the activation for my deployment number, the connection times out. The system worked fine last week when I purchased a different module. It behaves as if a server is off-line.
Called Sangoma and was told to submit a bug report, which I did. The email response to the bug report said that was not the correct procedure. Submit a commercial ticket instead. I tried but the web site does not recognize my deployment ID and won’t process the ticket.
I’m stuck. Many thanks in advance for any thoughts on how I should proceed?
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Participants: 1
@Jordack wrote:
I have two separate installs of FreePBX distro, both experience the same issue.
Create a Dashboard
Add the Device Manager Widget For The Extension
Make Changes
Click Save
Nothing HappensCan close out, go back in and nothing is saved.
I see this in the Apache access.log
192.168.1.111 - - [21/Aug/2019:15:37:12 -0400] “POST /ucp/index.php?quietmode=1&module=endpoint&command=savesettings HTTP/1.1” 200 29 “https://eros.mydomain.com/ucp/” “Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:68.0) Gecko/20100101 Firefox/68.0”/var/log/asterisk/freepbx.log contains nothing.
Browser debug console show no errors or anything when I click Save
I’m not sure where else to look for the cause. I can not seem to find any sort of error message.
FreePBX Framework14.0.13.4
User Control Panel14.0.3.3These are for Sangoma S500 phones, but still bought the EMP
Tried updating the UCP Edge, did not resolved so reverted back.
Not sure if this is something I could even call Sangoma for assistance on. (Assuming paid for assistance)
Posts: 4
Participants: 3
@kyiu wrote:
FreePBX 14.0.13.4
Cisco CP-7940 phone unable to register to the server:
Aug 21 21:38:52 freepbx in.tftpd[16853]: Client 192.168.168.65 File not found SEP00070E36404B.cnf.xml
Aug 21 21:38:52 freepbx in.tftpd[16854]: RRQ from 192.168.168.65 filename SIP00070E36404B.cnf
Aug 21 21:38:52 freepbx in.tftpd[16854]: Client 192.168.168.65 finished SIP00070E36404B.cnf
Aug 21 21:38:52 freepbx in.tftpd[16855]: RRQ from 192.168.168.65 filename SIPDefault.cnf
Aug 21 21:38:52 freepbx in.tftpd[16855]: Client 192.168.168.65 finished SIPDefault.cnf
Aug 21 21:39:03 freepbx in.tftpd[16904]: RRQ from 192.168.168.65 filename CTLSEP00070E36404B.tlv
Aug 21 21:39:03 freepbx in.tftpd[16904]: Client 192.168.168.65 File not found CTLSEP00070E36404B.tlv
Aug 21 21:39:04 freepbx in.tftpd[16905]: RRQ from 192.168.168.65 filename SEP00070E36404B.cnf.xml
Aug 21 21:39:04 freepbx in.tftpd[16905]: Client 192.168.168.65 File not found SEP00070E36404B.cnf.xml
Aug 21 21:39:04 freepbx in.tftpd[16906]: RRQ from 192.168.168.65 filename XMLDefault.cnf.xml
Aug 21 21:39:04 freepbx in.tftpd[16906]: Client 192.168.168.65 File not found XMLDefault.cnf.xmlChecked the /tftpboot directory…I can find these files created by the EPM:
dialplan-00070E36404B.xml
SIPDefault.cnf
SIP00070E36404B.cnfIt seem the phone is looking for different files to perform the provisioning.
Please help…
Posts: 1
Participants: 1
@jelani wrote:
Hi I’ve just setup FreePBX. I have a FXO card installed and it’s been detected by dahdi. I’ve setup a ring group to hangup for testing purposes but when i call the POTS line, it just rings out. I have a hutch it’s caused by the tone region setting.
I live in the caribbean (st. lucia) and there is no region setting for the island. I’ve been testing a few tone settings but with no luck… I know the island’s tone settings but i don’t know how to configure Dahdi to use them… Please assist.
Tone settings required
COUNTRY/TONE FREQUENCY in Hz CADENCE in secondsSaint Lucia Busy tone - 425 0.5 on 0.5 off
Congestion tone - 425 0.25 on 0.25 off
Dial tone - 425 continuous
Special dial tone - 425+330 continuous
Pay tone - 622 0.136 on 0.136 off
Ringing tone - 425 0.375 on 0.25 off 0.375 on 2.0 off
Route tone - 425 0.05 on 0.05 off
Warning tone - 425 0.1 on 4.9 off
Call waiting tone - 425 0.2 on 0.2 off 0.2 on 4.0 off
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Participants: 1
@jpark1205 wrote:
Hi!
I know i have been posting too much question… im sorry if this may bother someone. If i try to fix one problem another problem comes up…
so i get this error on my dashboard it says “An unknown port conflict has been detected in PJSIP. Please check and validate your PJSIP Ports to ensure they’re not overlapping”. I think it happend when i change from no to yes on the udp tcp ws ans wss.
Thank You
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Participants: 1
@rpmed wrote:
Hello all, why CDR not show names of internal numbers? on incoming and internal calls
.and in sip header none of user name
INVITE sip:410@172.16.0.10:5160 SIP/2.0
Via: SIP/2.0/UDP 172.16.20.62:5160;branch=z9hG4bK2888115155
From: “412-USER XXX” sip:412@172.16.0.10:5160;tag=2197917500
To: sip:410@172.16.0.10:5160
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Participants: 1
so i get this error on my dashboard it says “An unknown port conflict has been detected in PJSIP. Please check and validate your PJSIP Ports to ensure they’re not overlapping”. I think it happend when i change from no to yes on the udp tcp ws ans wss.