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OP stated in previous post that he has the dsm16 in his system

What are you wanting to accomplish? Have 1 extension answer most of the calls and allow other extensions to pickup ringing calls Could you diagram your desired call flow Inbound call hits a ring group with 4 extensions 257 251 244 & 229 Ring the first non busy extension and have another...

What happens on Blind Transfer?

OK you are using only an IVR with three paths (does path equal the ext in the IVR?) and no ring group "3 paths 257 (operator) Path 1 251 path 2 244 path 2 229 path 3" How can ext 251 and 244 both be on path 2 without a ring group? I can not find this - "Each is assigned in one hunt group 257 to...

Can you provide a more detailed call flow? Are you trying to pickup a call that is ringing in an ACD queue? FWIK only agents can pickup the call. If you want anyone to pick up an ACD call, then you will have to set an overflow destination, adjust the queue wait times etc for the queue. Create a...

Try this for the trunk settings Name your trunk Enter your outgoing CID Under Outgoing settings- Trunk name - whatever you choose to name it PEER Details- host=IP address of SIP gateway type=friend context=from-trunk disallow=all allow=ulaw dtmfmode=auto insecure=port,invite relaxdtmf=yes...

You can use cascading Call Flow Control Add a new CFC Lunch set the destination for normal(off green) = to the D/N CFC set the destination for override (on red) = to the lunch destination On the inbound route point it to the new Lunch CFC You can program another BLF for the Lunch Break You...

Have you tried 1.Press Feature 812. The display will change to show your call log records. · Use the * key to select which call log records you are viewing. The options are All, Mis (missed calls), Ans (answered incoming calls) and Out (outgoing calls). · Use the speaker up and speaker down...

Should work, since they are sip phones, but will take some effort and research. I've got a CP-8821 working on PBX in a Flash system. Took awhile to find and make a working SEPmac.xml file

I've never used Megapath so I would leave it like they suggest. The SRV record contains additional info like port,priority,and target IP

I always use qualify-yes you can try changing PBX configuration -> Settings -> SIP Settings Registrations (registertimeout) from 20 to 55 from the cli you can try running traceroute to that host name or nslookup, host, or dig

in trunk peer details Do you have qualify=yes host= FQDN or ip address? Instead of a hard reboot, you could have tried sip reload from the cli or bring up the trunk configuration then save and apply without making changes

Most routers allow outgoing connections In your trunk configuration is the address to your SIP provider. The trunk will contact the provider and if successful will open the ports for a return connection. The registration sequence will keep the ports open. I have many sites that use dynamic...

Port forwarding is not needed unless you have remote extensions, or an remote office with only a digital gateway connecting back to the main office. None of my sites have port forwarding, and I use a vpn for remote phones.

Maybe something wrong on their end or with your register string What is your router type? SIP ALG disabled?

Never worked with MegaPath. I prefer providers with unlimited channels. In the trunk settings, the incoming section is usually left blank. Create an inbound route for each DID. Shared Call Group: follow the Emetrotel Documentation in this Link I had a group setup on 1 DID with ten appearances...

Who is the provider? It depends if they will setup two trunks without you incurring additional cost. One provider I know, has 1 channel trunks and if you need more than 1 concurrent call you must purchase additional trunks @ $45 each. Providers with multi channel trunks is the norm. Many DID's...

Hi, I can't find a way to send a PM. One of my local telco's provides a line just for the use of elevators, pools, alarms, etc (can only dial 911 and toll free numbers) for about a third the cost of a PSTN. Depending on regulations in your area, you can use sip, can use pbx extensions or...

You can't go wrong with SIP, a reliable ISP provider and good bandwidth is a necessity. All my client are on SIP except for two

Glad its working for you