Carl Johnson

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.

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

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 profile tone-set default map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Releasetone map call-progress-tone congestion-tone US_Busytone profile tone-set US map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Releasetone map call-progress-tone congestion-tone US_Busytone .. Example of Port 3 on a 4114 we have interface fxo IF_CO3 route call dest-interface IF_PBXNSIP loop-break-duration min 300 max 1500 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id mute-dialing use profile tone-set US

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?)

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 192.168.46.210) 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 192.168.45.203 webserver port 80 language en sntp-client sntp-client server primary 192.168.41.20 port 123 version 4 system hostname rcp-ks-voip-gw2 system ic voice 0 low-bitrate-codec g729 profile ppp default 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 profile tone-set default map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Releasetone map call-progress-tone congestion-tone US_Busytone profile tone-set US map call-progress-tone dial-tone US_Dialtone map call-progress-tone ringback-tone US_Alertingtone map call-progress-tone busy-tone US_Busytone map call-progress-tone release-tone US_Releasetone map call-progress-tone congestion-tone US_Busytone profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 profile pstn default output-gain 2 profile sip default profile aaa default method 1 local method 2 none context ip router interface eth0 ipaddress 192.168.46.212 255.255.255.0 tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 192.168.46.1 1 traffic-class default route 0.0.0.0 0.0.0.0 192.168.46.1 1 context cs switch digit-collection timeout 2 interface sip IF_SIP bind gateway GW_SIP service default route call dest-service FXO_Hunt remote-party-id calling-party interface fxo IF_CO1 route call dest-interface IF_SIP loop-break-duration min 300 max 1500 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id mute-dialing use profile tone-set US interface fxo IF_CO2 route call dest-interface IF_SIP loop-break-duration min 300 max 1500 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id mute-dialing use profile tone-set US interface fxo IF_CO3 route call dest-interface IF_SIP loop-break-duration min 300 max 1500 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id mute-dialing use profile tone-set US interface fxo IF_CO4 route call dest-interface IF_SIP loop-break-duration min 300 max 1500 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id mute-dialing use profile tone-set US service hunt-group FXO_Hunt cyclic drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_CO1 route call 2 dest-interface IF_CO2 route call 2 dest-interface IF_CO3 route call 2 dest-interface IF_CO4 context cs switch no shutdown gateway sip GW_SIP bind interface eth0 router service default domain rcp.local defaultserver manual 192.168.46.210 5060 loose-router session-timer 1600 gateway sip GW_SIP no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface eth0 router vlan 10 shutdown port ethernet 0 0 no shutdown port fxo 0 0 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_CO1 switch no shutdown port fxo 0 1 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_CO2 switch no shutdown port fxo 0 2 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_CO3 switch no shutdown port fxo 0 3 use profile fxo us caller-id format bell encapsulation cc-fxo bind interface IF_CO4 switch no shutdown

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?

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.

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 reason but they work. Patton and Adtran have ok GUI (better than audiocodes) and are cisco like, however the patton is far more configurable than an adtran but the price on an is very right (TA 908e $2000 2 port PRI, Patton $4500 2 PRI)

TFTP and HTTP are two different animals and my preference is not to allow TFTP from the real world .. you know?

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?

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.

What bootloader?

Pbxnsip .. I really need resolve on this .. please help!!

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.

This is v4.1.12.0037 boottom and v3.1.1 SIP