Get new the pack of firmware for ip phone
I flowed this one. https://wiki.freepbx.org/display/FPG/EPM-Admin+User+Guide#EPM-AdminUserGuide-firmwaremanagementFirmwareManagement
View ArticleYum update still failing
Hi, when trying to “yum update” I still get errors. the collisions I had previously could be removed, now i am astuck at this point: Running transaction check Running transaction test Transaction...
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Digium D70 Phone Apps Issue
Just checked my xml-config file, there‘s a section „alerts“ which says <alerts> <alert alert_info="<http://127.0.0.1>;info=normal" ringtone_id="Digium" ring_type="normal" />...
View ArticleGUI connection timeout,The server at (IP) is taking too long to respond
Thanks so much for your prompt reply. I have tried to disable iptables using the following commands, service iptables stop chkconfig --level 345 iptables off but there are no any results. could there...
View ArticleRemote extensions through VPN
Hi, What must be the settings for remote extensions when there is site-to-site VPN connection between them ? The extensions are considered to be local to the FreePBX system so in Asterisk SIP settings...
View ArticleRemote extensions through VPN
VPN devices are indistinguishable from local network devices in a different broadcast area. nat=yes is deprecated. If Asterisk is on same intranet, the default should be acceptable, as this doesn’t...
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I finally gave up chan_sip for chan_pjsip and it was fine
reraikes: Without manually adding a type=identify section in pjsip_custom_post.conf containing a match=w.x.y. You don’t need to do that. You can set match inbound authentication to Auth Username in...
View ArticleGet new the pack of firmware for ip phone
See installing custom firmware. https://wiki.freepbx.org/display/FPG/EPM-Admin+User+Guide#EPM-AdminUserGuide-InstallingCustomFirmware
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Remote extensions through VPN
david55: deprecated So for remote extensions that are connected through VPN to the main site the only thing that is need to define the network as local in Asterisk SIP settings ? In the following...
View ArticleCall diverts
Good morning all, I have a different configuration that I am trying to make work. I setup call diverts from an extension to a mobile number and this works perfect but I want to be able to configure is...
View ArticleRemote extensions through VPN
nat should be left unset in this case. nat should never be set to yes, but rather to the individual settings actually needed. Setting it to yes may be relatively harmless, as it doesn’t actually...
View ArticleFull HDD FreePBX
Hello, The system reports to me: Storage space is getting high on the following drives of your system: /dev/sda2 is 76% full Unfortunately I don’t have access to SSH consoles. Can I delete old backups...
View ArticleGOOGLE tts
Hi I’m searching a way to download google TTS for freepbx This links seems not be working TTS Engine Custom - Google Cloud Text To Speech - WaveNet and Standard Thanks
View ArticleYum update still failing
yes, i think so: [root@freepbx yum.repos.d]$ pwd /etc/yum.repos.d [root@freepbx yum.repos.d]$ ls -l insgesamt 36 -rw-r--r-- 1 root root 1359 22. Mai 2018 CentOS-Sources.repo -rw-r--r-- 1 root root 106...
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TLS update
hi, i believe this is all you have to do…in settings ->asterisk sip settings->pjsip ->ssl method image.png1138×811 61.3 KB
View ArticleSRV for domain name - cannot connect remote devices
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-ICR-SECT-1.html#components_sipsrv
View ArticleDigium D70 Phone Apps Issue
For my FreePBX setup, I do have the Digium add-ons installed, and I did have the Digium Module. However, I had to disable that module since it conflicted with EPM. In my situation (due to a variety of...
View ArticleConvert System recordings automatically from alaw to G722 and vice versa?
Hi, we use an system recordings as an Endpoint from an IVR (holiday announcement). To change the recording, we use the “Link to Feature Code” feature. Depends of which phone is used, a alaw oder G722...
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