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Setup Vg7 With Sip

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I'm evaluating vg7 using SIP. I downloaded the evaluation HMP license and the evaluation version of vg7 and I entered the sip credentials into the config file, but I can't see it working. When I call the sip phone number, the the voice mail picks up right away, as if there's no sip client connected. I tested the sip line with xlite (a free softphone) on the same computer, and it works fine. I can't figure out what I'm missing.

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Could you please post a copy of VoiceGuide's Trace Logs which captures the problem, this will allow us to see what happened.

 

Trace files are created in VG's \log\ subdirectory.

 

When posting traces/scripts please .ZIP them up and post them as attachments.

 

Please also of possible email to us the WireShark trace which captures the registration process.

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I am having the same problem using vg7 demo I have set up a SIP account with FWD. Installed HMP. I thing I followed your example in config.xml.

 

 

I’m trying to call in from A other computer on the same LAN with a soft-phone but voice guide dose not pickup

 

The line status monitor shows “waiting for call “

 

I included the log file and config.xml

 

TIA_Yona

Config.zip

Log.zip

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VoiceGuide initialization looks fine

 

I’m trying to call in from A other computer on the same LAN with a soft-phone but voice guide dose not pickup

We would need to see a WireShark trace capturing the SIP messages for the incoming call. Please post this trace. It will confirm whether the call is arriving on the system and what response is given by HMP/VoiceGuide.

 

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Unzipped the attached log but was unable to open it in WireShark. It does not look to be a valid .pcap capture file.

 

How did you perform the capture and which version of WireShark did you use?

 

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Joewe:

 

None of the traces show any SIP activity. Have you tried calling into the system using a softphone on the same network, using direct IP dialing?

 

On outbound calls you need to specify the VoIP provider through which the call needs to be made. eg:

 

123456@fwd.pulver.com

 

not just a phone number.

 

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This trace shows a successful registration of the number 877571@fwd.pulver.com from system at IP 192.168.0.106

 

And then a call arriving and being answered. Call (to 877571@fwd.pulver.com) originated from a system with IP 192.168.0.200

 

Call was answered fine and the system answering the call hung up about 3 seconds later.

 

You would now need to look at VoiceGuide traces to see why VoiceGuide hung up the call 3 seconds after answering.

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How do I dial by IP address, do I just enter the IP address instead of the phone number? Do I need to register the softphone with a sip provider in order to make this call?

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How do I dial by IP address, do I just enter the IP address instead of the phone number?

Yes. That's the simplest way.

 

Do I need to register the softphone with a sip provider in order to make this call?

No need to register with anybody if you just want to place a call direct to/from another IP number. This only usually works if you are making a call to/from another IP point on the same local area network, not across routers/gateways. Making a call to another IP point on same LAN is usually the first type of test calls you would do, before trying to place calls to IP outside your LAN.

 

After confirming calls to/from local IP work you can then make calls to external numbers. You may need to register with a VoIP provider at that stage.

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I tested it for incoming calls and it went thru, now I want to test it for outgoing calls, since that's what I actually want to use it for. How do I do that? X-Lite doesn't seem to respond to IP calls.

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Have you tried using SJPhone?

 

You can select the "SIP Direct" mode in SJPhone which will allow you to make direct IP calls to/from the SJPhone.

 

If you are still having problems please provide VoiceGuide and WireShark traces from the system showing the outgoing calls.

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Trace shows that the outbound call is answered successfully but nothing is played as the module specified as the starting module: [sayValueFromOrigScript] (in script MakeCall.vgs) is not told to play anything and after about 10 seconds waiting for some input from the caller VoiceGuide hangs up.

 

If you change [sayValueFromOrigScript] to actually speak a number or play some sound files you should hear that module play the specified number/files after the call is answered.

 

Relevant log lines from the trace:

 

194951.152 11 3 state Dialing 192.168.1.44

195006.402 9 3 state Dialing 192.168.1.44 Ringing...

195009.355 9 3 state Human answer. Start [C:\Program Files\VoiceGuide\Scripts\MakeCallAndPassParams\MakeCall.vgs]

195009.480 9 3 state [sayValueFromOrigScript] Say Numbers: as Digits

195021.762 15 3 state Hanging up call... [LsSayNbrPlay_SayNbrsVBSRunTimeout]

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