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    FOOD FOR THOUGHT Wi-fi candybar IP phone?

    Does anyone know of a wi-fi candybar style IP phone with physical keypad and wideband audio? Something like the Belkin wi-fi phone but for SIP. Bonus points if it has any kind of built-in speech functionality as I'm totally blind and can't see an LCD. Thanks!
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    TIPS Cheapest/best provider for international calls to Australia/Europe from the US?

    Which provider are people using for international calls to Australian/European fixed line and mobile numbers? Preferably with a backup PSTN access number in case data isn't available. Are the betamax sites (Freevoipdeal, Powervoip, etc) still active? Thanks.
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    TIPS FCC RoboCall BlackList

    I've adapted the script for FreeSWITCH systems. More info is here.
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    TIPS Remote Users and dynamic IPs

    Disables Ward's default iptables config. If you ever want to bring it back, just do rm /etc/sysconfig/iptables mv /etc/sysconfig/iptables.ward /etc/sysconfig/iptables iptables-restore /etc/sysconfig/iptables
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    TIPS Remote Users and dynamic IPs

    Try this (I've used this config on several systems, your mileage may vary) - log into your system as root, and run: cp /etc/sysconfig/iptables /etc/sysconfig/iptables.ward iptables -F iptables-save > /etc/sysconfig/iptables service iptables restart
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    TIPS Outbound routes: allowing calls through a route to most users

    I don't want to "allow" certain extensions and block all others. I want to allow all extensions, but only block some.
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    TIPS Outbound routes: allowing calls through a route to most users

    Hello, Asterisk 13.6.0, Free PBX 12 on latest Incredible ISO. How can I configure the PBX to usually allow a user to make calls through a specified route, but block a select group of extensions from doing so, forcing them to fall back to other (secondary) routes? I don't want to set up a...
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    FOOD FOR THOUGHT Help setting up a specific phone

    Asterisk's chan_sip (older) driver has native support for BLF (busy lamp field) keys. Do your phones support that? If so, and your extensions are configured as chan_sip devices, BLF should work out of the box.
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    I HAVE A DREAM broadcast emergency calls

    Hello, Asterisk 13.6.0, Free PBX 12 on the new Incredible PBX ISO. Is there a script, module, etc I can install or config change I can make that broadcasts all calls to 911 to the default page group? Ideally it would announce the extension making the call, the number it was calling, then...
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    TRY THIS internal calling broken chan_sip

    Fixed with new install.
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    TRY THIS internal calling broken chan_sip

    Attempted to use the script at ~/upgrade-asterisk-to-current and a manual recompile (make clean/make/make install) but the same error still occurs. According to the wikipage referenced in the console output, this appears to be a NAT issue (but if that were the case why does pjsip work)?
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    TRY THIS internal calling broken chan_sip

    Hello, Running Asterisk 13.6.0, Freepbx 12 on Incredible PBX ISO. I have a few phones registered to the system, which is running on a VPS. Using Pjsip, the phones can register, and internal calls pass successfully without issue. Using chan_sip, phones appear to register, and outgoing calls work...
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    TUTORIAL Pat Fleet Voice Prompts for Asterisk

    Received the following email from Flowroute: "Hello Bill, We actually don’t offer the voice service at all at this time. Pat is available and may offer different formats as well. http://www.patfleet.com/ [-] Sincerely Christine M. Flow route Support" Sent an email to the address on Pat's...
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    TUTORIAL Pat Fleet Voice Prompts for Asterisk

    Once again, unless the files are of sufficient quality, you'll only make bigger files with no quality gain. Does anyone have the original, full quality versions of the prompts? We could run those through this script to generate g.722 versions.
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    TUTORIAL Pat Fleet Voice Prompts for Asterisk

    Yes, but if you convert from a low-quality format to a high-quality format, no quality is actually gained. The objective of converting the prompts to g.722 is to improve the voice quality on phones which support it. If you have prompts in their original (before conversion to 8k) format, that...
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    TUTORIAL Pat Fleet Voice Prompts for Asterisk

    I'm curious about this as well.
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    TIPS What Happened to Lenny?

    Setup: Incredible PBX 13 ISO on a VPS. Asterisk 13.6.0, Free PBX 12. Using default dialplan, Lenny worked for a while, but now no longer does. It seems to ring, then it goes busy. Dial string in Freepbx for the 53669 extension is SIP/[email protected] Any suggestions?
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