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  1. R

    Bandwidth.com: The Buck Stops Here

    David Morken is Bandwidth.com CEO. According to info at this site http://whois.pho.to/bandwidth.com his email address is: Registrant: Bandwidth.com, Inc. David Morken 4001 Weston Parkway Cary, NC 27513 US Phone: 1-919-2971010 Email: [email protected] Registrar Name....: Register.com...
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    Time limit - dropped calls

    After making the suggested changes, I now have random disconnects. The disconnects are no longer after exactly 30 minutes. They vary from 25 to 45 minutes. This is only happening on calls placed thru Flowroute. I have moved them to the back of the list and removed the temporary changes.
  3. R

    Time limit - dropped calls

    Many thanks gregc for this post. I've had calls disconnect after exactly 30 minutes with Flowroute as the provider. The default value for session-expires turns out to be 1800 seconds (30 min). I have increased the value per Flowroute's temporary work around. I am not sure why this is not an...
  4. R

    Problem Freepbx 2.8 and VOIPTalk SIP

    Glad you got it working. Sometimes you just need another set of eyes.
  5. R

    Problem Freepbx 2.8 and VOIPTalk SIP

    When you use "context=default", calls are limited to extensions. The default context code is located in extensions.conf and is defined as: [default] include => ext-local exten => s,1,Playback(vm-goodbye) exten => s,2,Macro(hangupcall) Change the trunk context to "context=from-trunk" and...
  6. R

    POTS lines-Outbound caller ID-workaround idea

    Tariff OneCommunications will have a tariff filed in all states that they operate. I just happen to pull this one for Michigan, but each state will most likely be the same.
  7. R

    2.9 Music on Hold Collapse

    I experienced the same problem after upgrading to FPBX 2.9. The permissions were not set correctly on moh directory. chmod 775 /var/lib/asterisk/moh then "moh reload" from CLI
  8. R

    mute and disconnect

    If rtpholdtimeout is set to 300, change it to value 360 and see if the disconnect occurs after 6 mins.
  9. R

    mute and disconnect

    Does this behavior happen on both incoming and outgoing calls?
  10. R

    I HAVE A DREAM NEW CallerID Superfecta 2.2.4: THE MODULE

    Thanks, that works.
  11. R

    mute and disconnect

    Set rtpkeepalive to value 30, then "amportal restart" from CLI.
  12. R

    I HAVE A DREAM NEW CallerID Superfecta 2.2.4: THE MODULE

    Confirmed. Caller ID Superfecta does NOT NOT NOT work with FreePBX 2.9.
  13. R

    mute and disconnect

    What value do you have set in the phone for Silence Suppression (Yes or No)? It should be set to Yes. If already set to Yes, check the RTP timer values for rtptimeout, rtpholdtimeout and rtpkeepalive.
  14. R

    "all circuits are busy now"

    You could attach the debug file to your ticket and see if they will correct the problem. I was not successful with Flowroute because they could not get their provider to solve it. In Outbound Routes, add additional trunks. If there is a failure on the first trunk, an attempt will be made on...
  15. R

    "all circuits are busy now"

    No, it does not. In the debug that you posted above, voip.ms sent an ACK reply to your INVITE. It appears they do not have a path available to the number you are calling. I experienced a similar problem with Flowroute. I would occassionally get congestion calling Time Warner numbers in the 254...
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    Dial Timing

    You did not mention which ATA that you are using, but most configurable devices have two timers that control the dial time. An interdigit short timer specifies the amount of time available to dial the next digit. An interdigit long timer specifies the amount of time available to dial the...
  17. R

    "all circuits are busy now"

    Did you change the "Device type:" in Account Settings on the voip.ms portal? There is a setting for ATA and one for IP PBX server.
  18. R

    Incoming with DIDFORSALE

    I have a DID from didforsale working on RentPBX. This needs to be added to extensions_custom.conf. [from-did4sale] exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=${CALLERID(num):2}) exten => _1NXXNXXXXXX,n,Goto(from-trunk,${DID},1) You also need to create a second trunk using...
  19. R

    Cordless recommendations???

    Fact: Typical PSTN calls are not yet made over a SIP trunk. While SIP is growing in popularity, a "typical PSTN" call is made using a combination of digital and analog devices. SIP just happens to be a digital protocal that at some point interfaces with the PSTN. The majority of residenial...
  20. R

    Cordless recommendations???

    DECT 6.0 cordless phones work great when connected to an analoge interface. This is afterall, what it is designed to do and where it is used most. In a typical PSTN call, there can and will be multiple A/D & D/A conversions.
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