asmith

Ok here's a summary of what I have achieved so far:

Linked the SPA3102 to Axon

Configured the FXS port to a handset registered with axon

configured the FXO port with axon

Can dial out fine - no problems

Can dial in - problems occuring

Caller dials in and asks to dial extension number or switchboard e.g 100

call is then blind transfered to that extension which rings for 2 seconds and then silence appears until the person is cut off

In IVM this is the log of what is happening:

10:49:28 Answering call...

10:49:28 Answered line [1 "VoIP Call Attendant"] call number [324] cid [FXO1] did[198] drn[0 (0ms)]

10:49:28 Play file: C:\Documents and Settings\All Users\Application Data\NCH Swift Sound\IVM\OGMs\VoIP Call Attendant Start.wav

10:49:37 Caller pressed key [1]

10:49:38 Caller pressed key [1]

10:49:38 Caller pressed key [1]

10:49:38 Variable number = 111

10:49:38 Command - Transfer

10:49:38 Voip Blind Transfer To: 111

10:49:38 Play system prompt: Ringing

10:49:38 Transferring sip call to: 111

10:49:38 Call has disconnected

10:49:38 Call disconnected

10:49:52 Incoming SIP call

10:49:54 VoIP Voice Mail FXO1 [Answered]

I'm guessing the problem is to do with this FXO1 answering, ive tried to find some options on the SPA3102 with no luck

Can anyone help please!!!

Regards

Adam

Thanks Anuj

 

I'll have a play around with it this evening! I did have another problem but can't remember it anymore, so it couldnt have been that big an issue

 

Adam

Sorry to be a pain again but could you provide me with some instructions on how to do this?

Thankyou

Adam

Thanks Anuj

I'll have a play around with it this evening! I did have another problem but can't remember it anymore, so it couldnt have been that big an issue

Adam

Anybody? Please

Hi guy's sort of another problem,

Call comes into gateway and is redirected to Axon, rings destination but then gets cut off after an x amount of rings, how can you get the call to goto the voicemail

on external calls coming into axon as internal voicemail works fine.

Adam

Excellent!

Thanks Pythonpoole

Adam

FINALLY, I have the Linksys 3102 working properly. It allows calls from the VoIP extensions within Axon by dialing out on the PSTN line as well as answering on the PSTN line. This provides a backup in case the VoIP external network is down.

 

I posted an updated configuration of my 3102 setup at:

spam://www.allnetbiz.com/sipura3102.htm

You can also see screenshots of the Axon external line settings from links on the above webpage.

 

Here is what I discovered:

• Yes, the SIP ports must have different numbers. Make certain you specify ports, which no other device is using.
  In the Axon setup use the same port number assigned to the FXO settings in the Sipura 3102. For example, the IP address in the Axon External Line "Server" name should be 192.168.1.124:5072 if the FXO (PSTN LINE) SIP port is set to 5072.
  

These few notes and additions to the setup shown in the original posting above should be the complete picture for allowing the Sipura 3102 to perform well.

 

Thanks,

Henry

Thanks Henry!

I have got my 3102 working like this with axon through your instructions and alot of pulled hair!

I only have a slight niggle though, the FXS1 line rings once when a call comes in on my phone line then its diverted to my pbx like it should,

does any one have any ideas on how you can stop the FXS1 handset ringing initially when the call comes through?

Adam

Hi Alab

I'm also having the same problem!!

Have you worked out a solution for this yet? Or has anyone else had this problem and found a work-around?