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Could be that your self signed cert expired and you dont have auto renew checked or a third party cert expired and you need to pay money to get the cert renewed. Try to connect to web manager and examine the certs to see which one expired.

The Merlin network reference says: Loudspeaker and voice paging calls cannot be made to non-local dial plan extensions or paging groups. With a wire to the garage you have a couple options but without a wire you would need to work out a wireless solution first.

It is true that you have to default the j series phone to pickup a new cert. For some reason I thought you have done that. The phone remembers the cert it is first given on reset or when first connected. Since you changed domain you would need to reset those phones to get the new cert.

Sounds like Site A is not sending the correct digits to site B. You should be able to see the digits sent from site A and the digits received at site B. Turn on monitor for each site and make a test call. Pause the traces to see what is sent and what is received.

Remove the old revoked cert with the old domain name? Doesn't really make sense since the users would need to enter the new domain name to connect using the new cert.

Recreate the SD card can often fix this issue at least until it happens again.

Chad, you could try O365 direct send relay which has never failed for me. BTW Westi and Sizbut are at the top of the the IPO Tech ladder in IPO land and are well respected here. And you instead have failed to even provide an smtp trace of the failure for review and went ALL CAPS in your...

Since you dont have a semicolon to end the short code and since you have bypassed ARS then I suspect that you are sending one digit at a time directly to the line. Set monitor traces to default and add ISDN layer 2 and 3 traces. Also check Call\ short code messages and Call\MonCM then test...

Hosted puts all of your eggs in one basket provider wise. If they go down then everyone goes down with them. Hosted wont work well for sites with access to only small bandwidth connections.

Wireshark is the answer for SIP or h323 calls. You have to convert h323 packets before they can be translated correctly but SIP needs no translation. You are better off collecting packet traces then replay the actual call. You will be able to see packet loss for the call and two way audio so you...

A member of the group may have forward configured or perhaps a vmpro call flow. If you have zeacom then that app distributes the calls and you can see the interaction when CTI-TAPI traces are on in monitor. Run Monitor with default plus trunking filter and make a call. You should be able to see...

Could be corruption of a file. When looking at the dbgtrail file in verbose do you see standard traces including housekeeping and call flows? There is an entry point for calls and callers that define the session number and other call information. Do you see the sixty second updates that are...

Sounds like grp 300 is not available for whatever reason. What does monitor show as the failure?

Comcast modem replacements have caused loss of ID. Our digit grabber sees the ID from the modem but Partner system no longer detects the ID. We think that is a result of the ID being eleven digits long(+1) instead of ten and below.

Your SIP provider should only send dtmf out of band and not inband. Out of band is preferred since rfc 2833 should be more reliable. The IPO SIP trunk can only be set to one type of dtmf and vmpro will not detect correctly if BOTH inband and out of band dtmf are received. Inband for some calls...

Nah they punted. They say that the problem is with the pbx and they wont make any change.

Lately I have seen two Merlin sites with analog loop start lines where ID stopped after telco replaced equipment. ID stopped being seen on phone displays exactly after Comcast replaced a cable modem which feeds a cisco IAD. A digit grabber sees 1 plus ten digits. The digits grabber did not see a...

Why would you call yourself? Is that a European thing?

Your ISDN L2 traces are not on. I ususally like to see L1 and L2. Your call is here: SYS SC: 813086272671 3 13086272671 sc=type=Dial code=8N;, num=N callinfop->sending_complete=1 secondary_dialtone= Can see here where call is sent to line 5:CD: CALL: 0.93680.0 BState=Idle Cut=1 Music=0.0...

sysmon traces with ISDN L2 and L3 filters and short codes and some others on will tell the tale but then you need someone who can read the traces. BTW IPO does not convert anything to PRI. Instead an IAD(Integrated Access Device) can take a SIP trunk from the provider and convert to mixed media...