@sentinelace wrote:
We haven’t made any changes to this. When you call in, it should park the call and announce that a call is parked at the lot number. It is just terminates the call. It is supposed to go to the voicemail of an extension if the call is not answered. Here is what the log shows:
I’m not sure what no application means?
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Participants: 1
@sentinelace wrote:
I have lost sysadmin after upgrade to freepbx 14. How do you fix this?
Posts: 1
Participants: 1
@bluesbro1982 wrote:
All:
I used Sangoma’s nifty migration instructions here:
Upgrading+from+FreePBX+10.13.66+to+SNG7 (Sorry, can’t link to post, google it.)
and got everything mostly working on a fresh machine. However, fm/fm would not successfully dial external numbers and was inserting an alert-info: header into any outgoing FM/FM call that originally entered the system though our IVR from the trunk (not internal extensions).
The bug looked pretty close to issue FREEPBX-17927 in the bug tracker.
After beating my head against the wall for a couple hours, I checked my time conditions module. The “alert-info” section was set to “none” HOWEVER…When I used the Sangoma migration script to transfer over from our homebrew distro to SNG7, the script seems to have created on the Inbound routes page an alert info of “none.” This shows up in the dropdown box as a second none after the sangoma ring settings, and should not be confused with the “none” at the top of the dropdown box. Changing the dropdown to the top “none” unsets the alert-info from sip headers and enables fm/fm to call external extensions without AT&T rejecting the malformed SIP header.
Much fun that one. I don’t know if it can be reproduced, but I figured this might help someone.
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Participants: 1
@cloudpbxfuzz wrote:
I can not reply to this thread anymore, but I am having the same issue described in this thread.
I have whitelisted all new POP IP ranges in the Firewall (put them into the trusted zone) as well as whitelisting them in Fail2Ban. IPs from this website (https://support.flowroute.com/SIP_Trunking_and_Voice/Networking_Guides/Set_Firewall_Policies_for_Flowroute’s_SIP_Signaling_and_RTP_Media#RTP_media_(call_audio))
Incoming calls will intermittently fail. I turned on SIP debugging, and every call is coming into the PBX, but the calls that fail, are sending a 401 rejected back to Flowroute. The signaling traffic is coming from one of the IPs in the whitelisted ranges.
Do I need to setup a trunk for each and every Flowroute IP? I would not think (hope) I would need to do that since there are about 64 IPs that the signaling traffic could come from.
Has anyone set this up properly?
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Participants: 4
@Beachtech wrote:
I have a client that used to be with ESI. Can I re-purpose the ESI phones, reprogram them, and put them on the FreePBX system I set up for them? I am not familiar with ESI, but I have done this with other vendors that have offered cloud-based PBX with Yealink phones.
Posts: 1
Participants: 1
@aerodes wrote:
Hi everyone,
I was trying to solve another issue when I went to System Admin-> HTTP Setup->Setting, selected Certificate manager=default and pressed Install.After that I can’t access GUI. How can I get GUI back?
I already tried running systemctl status httpd.service. I tried using Chrome, Explorer and Firefox. Not luck.Error: “Chrome tried to connect to 192.168.1.130 this time, the website sent back unusual and incorrect credentials.”
Any advice is appreciated.
/etc/hosts:
127.0.0.1 uc-50473423 localhost localhost.localdomain localhost4
::1 uc-50473423 localhost localhost6ssl.conf
/etc/httpd/conf.d/ssl.confAutomatically Generated File - 1537389895
{“certconfig”:{“subject”:{“CN”:“localhost.localdomain”,“O”:“localhost.localdomain”},“issuer”:{“CN”:“localhost.localdomain”,“O”:“localhost.localdomain”}},“sslports”:{“sslacp”:{“port”:443,“dir”:"/var/www/html"},“sslucp”:{“port”:“disabled”,“dir”:"/var/www/html/ucp/"},“sslrestapi”:{“port”:“disabled”,“dir”:"/var/www/html/restapi/"},“sslrestapps”:{“port”:“disabled”,“dir”:"/var/www/html/restapps/"},“sslhpro”:{“port”:“disabled”,“dir”:"/tftpboot/"}}}
Do NOT edit this file as it is auto-generated
LoadModule ssl_module modules/mod_ssl.so
SetEnv SSLSETUP true
SSLPassPhraseDialog builtin
SSLSessionCache shmcb:/var/cache/mod_ssl/scache(512000)
SSLSessionCacheTimeout 300Not valid in Apache 2.4
SSLMutex default
SSLRandomSeed startup file:/dev/urandom 256
SSLRandomSeed connect builtin
SSLCryptoDevice builtin
#https://mozilla.github.io/server-side-tls/ssl-config-generator/
SSLProtocol all -SSLv2 -SSLv3
SSLHonorCipherOrder on
SSLCipherSuite ECDHE-ECDSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-RSA-AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA:ECDHE-RSA-AES256-SHA384:ECDHE-RSA-AES128-SHA:ECDHE-ECDSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA:ECDHE-RSA-AES256-SHA:DHE-RSA-AES128-SHA256:DHE-RSA-AES128-SHA:DHE-RSA-AES256-SHA256:DHE-RSA-AES256-SHA:ECDHE-ECDSA-DES-CBC3-SHA:ECDHE-RSA-DES-CBC3-SHA:EDH-RSA-DES-CBC3-SHA:AES128-GCM-SHA256:AES256-GCM-SHA384:AES128-SHA256:AES256-SHA256:AES128-SHA:AES256-SHA:DES-CBC3-SHA:!DSS
Listen 443Skipping sslucp as it is disabled
Skipping sslrestapi as it is disabled
Skipping sslrestapps as it is disabled
Skipping sslhpro as it is disabled
ServerName localhost.localdomain:443 LogLevel warn SSLEngine on SSLCertificateFile /etc/httpd/pki/webserver.crt SSLCertificateKeyFile /etc/httpd/pki/webserver.key # No ca-bundle detected DocumentRoot /var/www/html SetEnvIf User-Agent '.*MSIE.*' nokeepalive ssl-unclean-shutdown downgrade-1.0 force-response-1.0 Alias /.well-known /var/www/html/.well-known Alias /.freepbx-known /var/www/html/.freepbx-known RewriteEngine on RewriteRule ^/\.(well-known|freepbx-known)/ - [H=text/plain,L] RewriteRule (^\.|/\.) - [F]
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Participants: 1
@Elimin wrote:
Hello,
My system:
PBX Firmware:
12.7.5-1807-1.sng7FreePBX 14.0.3.13
Current Asterisk Version: 13.22.0Please help more experienced users how to set codecs.
Incoming calls to the FreePBX are G.711a
Telephones use G.722The “core show channel …” command shows something like this
Incoming call:
– General –
Name: PJSIP/incoming-gw-00000111
Type: PJSIP
UniqueID: 1537443591.557
LinkedID: 1537443591.557
Caller ID: 222222222
Caller ID Name: 222222222
Connected Line ID: 555555555
Connected Line ID Name: 555555555
Eff. Connected Line ID: 555555555
Eff. Connected Line ID Name: 555555555
DNID Digits: 555555555
Language: en
State: Up (6)
NativeFormats: (alaw)
WriteFormat: slin16
ReadFormat: slin16
WriteTranscode: Yes (slin@16000)->(slin@8000)->(alaw@8000)
ReadTranscode: Yes (alaw@8000)->(slin@8000)->(slin@16000)
Time to Hangup: 0
Elapsed Time: 0h0m38sConnection to the phone:
– General –
Name: PJSIP/555555555-00000112
Type: PJSIP
UniqueID: 1537443591.558
LinkedID: 1537443591.557
Caller ID: 555555555
Caller ID Name: 555555555
Connected Line ID: 222222222
Connected Line ID Name: 222222222
Eff. Connected Line ID: 222222222
Eff. Connected Line ID Name: 222222222
DNID Digits: (N/A)
Language: en
State: Up (6)
NativeFormats: (g722)
WriteFormat: slin16
ReadFormat: slin16
WriteTranscode: Yes (slin@16000)->(g722@16000)
ReadTranscode: Yes (g722@16000)->(slin@16000)Is not too much transcoding?
Maybe phones should also use only G.711a
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Participants: 1
@Hawkeye wrote:
Hi, since updating modules with module admin 2 or 3 days ago, have noticed any route which has Route CID set ends up displaying as
"NAME"instead of the name/number.Tried adding the content of the tool tip : “hidden” <#######> and once saved the CID is displayed as
"hidden"This FreePBX is 12.7.5-1807-1.sng7
Will this be fixed?
Thanks
Posts: 1
Participants: 1
@PBXUser1234 wrote:
Hello,
We are having trouble with incoming calls not coming through… We are using Twilio SIP (using pjsip driver) trunks with TLS. Outbound calls work perfectly, but every time I try to make an inbound call (from an external number) all I get is a busy signal.
I setup the NAT rule in our firewall to the FreePBX server and I confirmed that the traffic on port 5061 is being allowed. Also when I run a tcpdump on the FreePBX server it appears that there is traffic coming into the server when I make an inbound call, but I still just get a busy signal. I also tried completely disabling the firewall within FreePBX to see if that would make a difference, but it did not.
I do have the inbound route setup to my extension (softphone)… What am I missing here to get the inbound calling to work?
Thanks in advance!
Posts: 1
Participants: 1
@DougPBX wrote:
Anyone know where I can find firmware file to upload - I have several base stations and they all need to be on the same firmware for roaming to work.
Posts: 1
Participants: 1
@bajramia wrote:
Hi,
Im having some issues with voice on zulu mobil app and when i looked the logs it shows STUN Server unreachable im using google STUN i was wonder if any other recomanditonThank you all.
Posts: 1
Participants: 1
@yelaman wrote:
TO field and the Request-URI are incorrectly
Request-Line: INVITE sip: +77123456789%405.63.114.58@provider_IP SIP / 2.0
To: sip: +77123456789%405.63.114.58@provider_IP
Should be + 77123456789 @provider_IP
it’s not clear where it came from “% 405.63.114.58”trunk connectoin:
Freepbx
trunk
Outgoingtype=peer
qualify=60000
host=195.111.11.119
disallow=all
allow=alaw&g729and i added
useragent=igroup
Posts: 1
Participants: 1
@PitzKey wrote:
Hello,
Wondering if anyone can reproduce this, before submitting a bug report.
FreePBX 14 Asterisk 13.
I updated ‘queues’ from 14.0.2.17 to 14.0.2.22 and calls stopped going to the Periodic Announcement (which in our case is Queue Callback.)
I downgraded ‘queues’ back to 14.0.2.17 and it’s working again.
Anyone else can reproduce this?
Thanks
Posts: 1
Participants: 1
@comtech wrote:
We have Avaya and Asterisk Platform. I can send calls from Avaya to Asterisk, but I would like to share what happened on the Avaya side. It looks like I can pass the UCID from Avaya on the trunk group, but I am not sure how to get it. Once I have the UCID as a variable in Asterisk, I can query Avaya for the rest of the call data.
I assume it is in the SIP header. If it is, is there an easy way to pull and store pieces of SIP header it as a variable for a calls?
I know this crosses a few platforms, but thought I would start here since I usually get good responses here.
Thanks for any guidance.
Posts: 2
Participants: 2
@sylwat wrote:
After the latest module upgrade for Pin Sets from 13.0.8 to 13.0.9 Pin Sets failed to function as intended. Basically when you make a call that has a pin set assigned to the route it prompts for the password followed by the # key however on first try the system reports “password incorrect” then without hanging up dial the same password again this time it accepts it. I have tested this on two systems with exact system, PBX and module versions with the same result. Is there an option to downgrade the module to the previous version? please note i have also uninstalled and re-installed the module with same results.
Posts: 5
Participants: 2
@bksales wrote:
Is this doable? Apparently its too difficult to just dial a feature code, they want to see the status of the flow control or time condition visually. Thanks.
Posts: 2
Participants: 2
@bmartindcs wrote:
We’re running the Freepbx Distro, 13.0.195.12 box with Asterisk 13.17.0. Phones are Sangoma S700. Phones and Sip trunks are using G.729 without issue. If we try to use a feature code to record a recording, it works fine and we can play it back from the featurecode without issue. However, the recording is not updated in the “recordings” module nor on the IVR it is used on.
If I click “play” for the recording in the recordings module, I get the following error:
“Unable to find intermediary converter for /var/lib/asterisk/sounds/en/custom/FILENAME.g729 file:/var/www/html/admin/libraries/media/Media/Media.php:299”
Same happens if I try to use the buttons to convert the file to other formats.
If we record the recording with the desktop or a cell phone as a WAV file then upload that in the recordings module, it will work fine. It’s only when we try to use the feature code for the recording does the problem manifest itself.
I confirmed that the res_convert module is loaded/running.
I show that the format_g729 modules are both showing loaded/running as follows:
codec_g729.so g729 Coder/Decoder, based on Intel IPP 0 Running unknown
format_g729.so Raw G.729 data 0 Running core
res_format_attr_g729.so G.729 Format Attribute Module 1 Running coreMy google-fu shows one bug report that stated after someone upgraded their modules this started and then support bounced saying it’s a support issue not a bug.
Hoping someone can shed some light on this.
Posts: 4
Participants: 3
@earthsubduer wrote:
Hello,
Newer to Freepbx. Is there anyway to autorecord all inbound calls that are then forwarded to another number? My system uses google voice numbers that are then forwarded to other endpoints/numbers. I need to be able to use the auto record feature for all calls that come inbound to the google voice number, forwarded to another number. Need to be able to record entire conversation for all of these calls.
Any help is appreciated!
Regards
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Participants: 1
@bajramia wrote:
I have costumer they want to be promted for voicemail passcode from their extensions is there a way to do this
Thank you
Posts: 5
Participants: 3
@formiche wrote:
Hi,
I’m new in Asterisk. I installed the latest version of FreepbxDistro. I’m trying to implement simple example of AGI. I don’t understand the right sintax of the file test.php. The file that I wrote is executable, it is in /var/lib/asterisk/agi-bin, the owner is asterisk. From the *CLI> console, after the core set verbose 99 and core set debug 99 commands I receive any message from the php server, from the command line of linux the script works.
Someone can help me? The documentation is very poor and not updated
Posts: 1
Participants: 1
