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Carl Johnson

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About Carl Johnson

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  1. All patton side .. When you setup the patton make sure you do not have those in the hunt group for SIP -> PSTN destinations other than nothing else.
  2. I agree with the forum admin .. SOAP is the native answer .. all major databases have a native ability now to take input via SOAP without any 3rd party, here are the MSSQL docs .. I think your issue is you may not want to pay for it .. http://msdn.microsoft.com/en-us/library/ms...23(SQL.90).aspx
  3. Could be mis-configured disconnect settings, most US telco use a loop break to provide the disconnect signal (busy signal as well). See below for known working settings for DTMF and disconnects in about 10 cities around the US (all different telco's). profile call-progress-tone US_Dialtone play 1 1000 350 -13 440 -13 profile call-progress-tone US_Alertingtone play 1 1000 440 -19 480 -19 pause 2 3000 profile call-progress-tone US_Busytone play 1 500 480 -24 620 -24 pause 2 500 profile call-progress-tone US_Releasetone play 1 250 480 -24 620 -24 pause 2 250
  4. Does this release resovle issues with VM hanging in mailbox and not sending email? IE. We use 10 exts as some 100+ VM per day exts and about 2-10% of the items in 3.4.3201 hang in the mailbox and are not (we can tell easily as the VM is set to delete .. howerver email failures are not logged?)
  5. Sure .. here you are (this works with minor mod on any patton SN 411X or SN 452X) This config takes calls from the SIP interface (IF_SIP bound to GW_SIP .. set the IP of your PBXnSIP box here .. currently and routes to FXO ports (FXO_HUNT) and takes calls from the FXO ports (IF_FX0-3 using US type tones) and sends to the SIP interface (IF_SIP/GW_SIP) cli version 3.20 administrator administrator password gTwkMZxcw6rVnPpTNxYkuA== encrypted clock local offset -07:00 dns-client server webserver port 80 language en sntp-client sntp-client server primary 192.
  6. After a recent update to the CS4XX we are now having issues passing through G729 .. what changed and why? The proxy now gives a 488 message to the receiving or sending party .. but the phone and gateway advertise 18 being a codec on the list? Of course this works perfect on my Pro versions of the pbx?
  7. Consider a Adtran 900 series, they include a PRI<->SIP, SIP<->FXS, PRI<->FXS in one box and work pretty well and very reasonable. This is the IAD many telco's use to provide SIP to PRI service.
  8. For analog, I am without a doubt 100% sold on Patton as the others just cannot be tweaked for all situations (various tone sets) .. but the grandstream gateways do work and work fairly well (especially for the money) .. I have used the others vegastream, mediatrix, and mulitech and they are more hassle than they are worth .. IMO. For digitial, we have used Adtran, Mediatrix, Patton, and Audiocodes. Mediatrix have been the most reliable and make the most sense in the GUI. ($2500 2 port PRI) Audiocodes is a complete bearcat and whoever wrote the GUI should be shot and it has no rhyme or re
  9. TFTP and HTTP are two different animals and my preference is not to allow TFTP from the real world .. you know?
  10. We have an office pro edition and I know a version of so ago they enabled ad-hoc recording .. so how does it work? IE enter the star code then ? or dial, get connected, enter the star code?
  11. By default, with using just an IP/host when provisioning the Polycom will look for http://IP/mac.cfg .. we had to type in via the keypad the URL http://IP/provisioning and then it at least worked on the applicaiton side .. but not the bootloader as it appears the boot loader will not authenticate when asked.
  12. Pbxnsip .. I really need resolve on this .. please help!!
  13. Carl Johnson


    We have At&t MPLS (via ACC, much better pricing and exact same service) in 4 sites with QOS, we had Qwest MPLS as well with no issues. The service is very good, we route about 160 concurrent calls over our MPLS during anytime of the day so voice quality is key and we have ZERO issues.
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