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I have an issue after an upgrade from R9 to R11 nwith the caller ID names not being controlled by the SIP trunks "Name Priority" from directory? I tried both setting the system> telephony to directory and the SIP Line to directory and both, but it still gives me the trunk info... Anything I'm...

Thank you everyone for your help. I have everything working now and my carrier is happy. Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

I need to disable sip options on an ip500v2 with R9.1. Does anyone know the nouser source? I tried it with SIP_OPTIONS_PERIOD=0, but it’s still sending options to my carrier. Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

I am now changing out a customers IP500v2 for the second time in 2 weeks because WAN port is blown up.... It is plugged directly into a Comcast modem. Has any seen this before? Any possible ways to know why? Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

I'm looking into setup with Skyswitch, so i can control my own SIP trunking, and start to build a base on hosted. Has anyone dealt with them and even know who they are? Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

I found a solution, and the programming is on the Free PBX side not the Avaya. But there was a way to access system/intercom dialtone remotely. Like you have when you create a Breakout on a SCN between to systems. Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

I haven't done this in ages, but how can I set up a short code that give me system dial tone to access outbound dialing from a remote Free PBX system connected thru SIP? Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

LOL... I thought it was. the other site I am having the issue is on 9.1.7 Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

2260812970mS SIP Call Tx: 1 INVITE sip:1704504XXXX@mymtm.us SIP/2.0 Via: SIP/2.0/UDP 172.16.130.120:65476;rport;branch=z9hG4bK3234a7e322e131db7ea043ffcdd5c701 From: "954241XXXX" <sip:954241XXXX@mymtm.us>;tag=79d462cced96d1c5...

This is the trace from system status. 10/10/17 4:10:35 PM-712ms Extension = 314, Digit dialed, Digit = 8 10/10/17 4:10:35 PM-733ms Call Ref = 10181, Originator State = Dialling, Type = User, Destination Type = ARS Form 10/10/17 4:10:35 PM-733ms Call Ref = 10181, Short Code Matched = System...

I have 2 IP500 with R9.1.10 and 9.1.6 with SIP trunks, and sometimes when they call out and get another companies voice mail and dial an extension they get transferred back in to the own system? In the monitor trace after the call is connected and they dial the extension number the to and from...

no sip trunks, PRI. And the company that installed this before us had LAN2 on the internet wide open. I was able to lock manager down before they were hacked, but the hacker used TAPI to dial out even though I had international calling for that extension barred. Now I have the system off the...

how would TAPI overide my ARS restrictions for 011N Barred? Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

Disabled TAPI now. we are changing all our customers over to VPN access for One-X. I have not gotten to them yet. I have not ever seen this type of hack before. Is this common? Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

I think it is somehow. they have one-X so it has to be accessed. Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

Guys I have a weird one here. My customer has an IPO R9.0.7.Someone has hack us 3 times over digital phones. The phone goes off hook and the calls go out over the speaker like a ghost is making the calls. They have digital phones, SCN, and one-X but the phones that are being used are basic user...

I did figure it out. I set the recording on the incoming call route and it records the complete call with transfer. I had to disable the refer on the sip trunk, and it's 100% now! Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

I'm looking? Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com

R9.1 VM Pro call recording after call transfer. I set up a system in VMPro for my customer to call in and with his caller id, verify them and give him dial tone to make calls out. We need this so they can record the calls in and out. I set up 10 SIP trunks with 21 DID's for the stores to call in...

I found it... Its a "Z", you can only send one thing in a field, so in the short code I put NZ"ABC Company" and in the SIP tab under the user I have the number. Jeff "the phone guy" ACSS-SME APSS-SME www.bluestartelephone.com