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VoIP

VoIP

Voice Over IP

FusionPBX Rejecting Incoming Calls From SIP Trunk

January 17, 2019arstech Leave a Comment
fusionpbx

FusionPBX Rejecting Incoming Calls From SIP Trunk Usually fresh installed FusionPBX Rejecting Incoming Calls From SIP Trunk by default. To fix problem you need add carrier to the domains acl set the node CIDR to allow. Like ProviderIP/32 under CIDR.… Continue Reading →

FreeSWITCH, FusionPBX, VoIP FreeSWITCH, FusionPBX, Rejecting Incoming Calls, SIP, SIP trunk

T38 FAX Cisco SPA122 Configuration

December 22, 2017arstech Leave a Comment

T38 FAX. Cisco SPA122 Configuration How to enable T38 FAX support on Cisco SPA122 VoIP adapters. Proxy and Registration   In  “Proxy and Registration” and “Subscriber Information” fill Proxy: YourSIP.Server User ID: SIP Username Password: SIP Password     NAT… Continue Reading →

VoIP T38 FAX, t38fax, voip t38

How To Configure Distinctive Ring on Grandstream

December 8, 2017arstech Leave a Comment

Configure Distinctive Ring on Grandstream For set up Distinctive Ring  on Grandstream GXP2130, 2140, 2160 phones Goto: Accounts -> Select the account (1,2,3 or 4) -> Call Settings In section Match Incoming Caller ID fill Matching Rule fields For example:… Continue Reading →

VoIP Distinctive Ring, Grandstream, Matching Rule

Cisco SPA504G Time 1 Hour Behind

June 13, 2017arstech Leave a Comment

How to fix Cisco SPA504G Time In “Regional” tab Set: Ignore DHCP Time Offset: yes Daylight Saving Time Enable: yes Daylight Saving Time Rule: start=10/1/7/03:0:0;end=4/1/7/02:0:0;save=1

Cisco, VoIP cisco, Daylight Saving Time, SPA504G, time

How To Enable Web GUI on Adtran Totall Access TA 924 from console

March 29, 2017arstech Leave a Comment
adtran TA 924

How To Enable Web GUI on Adtran Totall Access TA 924 from console, and configure static IP on Ethernet port First you need connect your Adtran TA to your PC via console port. Also make sure your computer and Adtran… Continue Reading →

VoIP 900, 924, adtran, Adtran TA 924, TA, Total Access 924, WEB GUI

Using tcpdump for capture SIP packets

July 25, 2016arstech Leave a Comment

How to capture SIP packets with tcpdump (example) tcpdump -nt -s 0 -A -i eth0 port 5060 -w call.cap Tcpdump will create file call.cap Capture DHCP requests with tcpdump

Linux, VoIP capture, SIP, tcpdump

Install Godaddy SSL certificate on Nginx, Verto, FusionPBX

March 30, 2016arstech Leave a Comment

Verto (VER-to) RTC is a FreeSWITCH endpoint that implements a subset of a JSON-RPC connection designed for use over secure websockets. This allows a web browser or other WebRTC client to originate a call using Verto into a FreeSWITCH installation… Continue Reading →

FreeSWITCH, FusionPBX, Linux, VoIP FreeSWITCH, FusionPBX, Nginx, SSL certificate, Verto, WebRTC

You must install libldns-dev to build mod_enum

March 30, 2015arstech Leave a Comment

You must install libldns-dev to build mod_enum During the install FreeSWITCH on CentOS (Centos/RHEL 6.4 and 6.5, 6.6) you may getting an error: “You must install libldns-dev to build mod_enum.  Stop.” yum install libldns-dev can’t help: No package libldns-dev available… Continue Reading →

FreeSWITCH, Linux, VoIP

Elastix admin password reset

October 29, 2013arstech

Elastix admin password reset For Elastix admin password reset just execute from Linux terminal: # sqlite3 /var/www/db/acl.db “update acl_user set md5_password=’7a5210c173ea40c03205a5de7dcd4cb0′ where id=1” You will be able login to WEB GUI with new password: palosanto

Asterisk, elastix, VoIP admin, elastix, GUI, password, reset, WEB

sipXecs port forwarding on modem

October 29, 2013arstech Leave a Comment

 sipXecs port forwarding Sometimes you may have one way audio (one way RTP) problem on sipXecs phones. Correct functioning sipXecs registered phones behind of NAT requires ports forwarding. Or if network has firewall next ports should be opened: 5060 TCP… Continue Reading →

sipXecs, VoIP one way audio, port forwarding, ports, RTP, sipXecs, sipXecs ports

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