Hello everyone. I am having a issue with fixing the caller ID being sent to my PBXACT server from my Vega 50 GW device. We are currently running 3 FXO pots lines for our service through the GW to the PBX. Incoming and outgoing calls work. Currently we get Anonymous on phones with incoming calls.
Configuration Vega - Currently working but results in anonymous.
I have tried the following on the dialing plan TO_SIP
Source=IF:0201,TEL:<.*> Dest=IF:9901,TEL:<1>
The result is disconnect call cause 2, but more entertaining is when i set this up. If i call in on cell phone (for example) it sounds like the Vega picks up and i get a dial tone and if i dial a number (on cell phone) the vega will starts dialing (resulting in office phones ringing).
I can produce his original issue by setting From Header ‘userinfo’ back to Authentication Username. But if i try his Source - enter “IF:02…,TELC:<.*>” Destination - enter “IF:9901,TELC:<1>”
If i drop the " " from his source and destination to avoid the syntax error. The result when i call in is a constant ringing as if no one is going to pick up the line.
More Detailed configuration information.
Product version Vega 50
Firmware File VEGAEURO_R088S068
Region is set to US
Caller ID Type Information gr30-mdmf
Tone Based Disconnect 1 - checked
On the Vega I have SIP->Sip Profile->From Header ‘userinfo’-> set to Calling Party
On the PBXACT trunk I have Hide CallerID set to NO. SIP trunk is not registered.
Hi,
Your use is far more advanced than I needed to achieve so the following is a poorly educated guess:
To have the Vega talk to my FreePBX I had to have the calls arrive as a ‘trunk’, then tell FreePBX where to send those calls (easy for me as they all went to a single group).
As you describe getting a dial tone and being able to ring office phones when you’ve dialed in using your mobile, I’m wondering if your PBXACT server is waiting for more information before it can handle the call.
So, is there an option for any call arriving from the vega SIP connection to be sent somewhere, regardless of actual caller ID?
I hope that helps,
Steve
I believe your setup is much like mine. I have Vega 50 with three FXO. SIP trunk back to Pbxact server. All incoming calls from that trunk go to a single ring group (Ring All). We are operating without issues send/rec calls all day. All incoming calls show caller ID anonymous. So thats were my problems begin. When trying to adjust my dialing plan “To_SIP” source and destination fields the vega starts acting odd.
It seems that I am doing something wrong with my dialing plan and the Vega then can not make the correct decision on routing. But only when i use a TELC: token.
On your vega is it passing caller ID information to your pbx? And if so what is your dialing plan source and destination? I would like to compare to what i am entering.
I moved from the vega 50 to analog cards to solve some problems I was having (I’m never quite sure if they were caused by my lack of configuration skill or an issue with the Vega, but I’d assume it was the former), however, I do have a backup of the full configuration so I’ll send you a private message with the full file.
sudoroo I want to thank you for your response. I found the file you sent me helpful for comparison to my configs as I worked through the issue. I wish I could say i found a bug or configuration error in this device to explain the caller ID not passing but it was neither. The problem existed with the telephone company and their equipment. It took me over a week of calling and insisting they review the data that I got no caller ID. I even dug up a 10+ year old phone with caller ID on it to prove to the phone company. They are working on it and I hope I will be up and working soon.
