VoiceGuide IVR Software Main Page
Jump to content

Vg7 Sip Not Registering Correctly?

Recommended Posts

I'm evaluating VoiceGuide on a new VM, so I need to set up VOIP support.


Currently, I've installed the Dialogic HMP software and the VoiceGuide trial, and believe I've set things up properly, but unfortunately when I dial the number from an outside line, I immediately get a busy signal.

  • When dialing from a softphone (Blink) to the server running VG directly, VoiceGuide appears to pick up correctly
  • When dialing from a softphone to the SIP server (32334@ipaddress), the softphone immediately reports back "Forbidden"
  • When dialing from a phone (outside line), I immediately get a busy signal

Looking at the wireshark traffic, all I'm seeing is:

  • A "REGISTER" command from the VG server to the SIP server
  • A 200 OK response from the SIP server back to the VG server

I'm told that after the 200 OK, we're supposed to send back a "SUBSCRIBE" message, but that does not occur. I'm also told that they combed through there firewall and could not find any records of traffic being blocked to the server.


One other thing I saw is that when I had two VoIP_Registration elements in the config, only the first one would get a REGISTER command. Is that normal or could that be a symptom of the problem as well?


Any thoughts on next steps for diagnosis?





Share this post

Link to post

There should be no need to send any SUBSCRIBE message after the REGISTER is OK'd.


As per SIP standard, no SUBSCRIBE messages of any kind are needed to start receiving calls after the Registration.



What SIP server is at address ?


You should look at that SIP server to determine why that SIP server is not forwarding on the SIP INVITE to the registered HMP/VoiceGuide system when it receives a call that is intended for the HMP/VoiceGuide system


Could you please post the vgEngine and ktTel traces of a system startup when two <VoIP_Registration> entries are in Config.xml ?

Share this post

Link to post

Attached are the VG logs from when there are two registrations in the config file.



Share this post

Link to post

No files seem to have been attached. Could you please try attaching the logs again?


Please .ZIP up log files as before before attaching them.

Share this post

Link to post

Trace shows that only one registration entry was defined in Config.xml


There were two Authentication entries, but only one registration entry.


If you would like to register multiple accounts then each account must have a Registration entry.


Authentication entries are only used if VoIP server requests authentication as part of registration.


Please create a registration entry for the second account that you wish to register.

084633.693   6               VoIP_Authentications_EntriesCount=2
084633.694   6               VoIP_Registrations_EntriesCount=1

Share this post

Link to post



I was able to make more progress with this issue - After updating the LocalAlias setting in the config.xml to point at the correct place, traffic is now directed to the server.


However, the same high-level symptoms still remain:


1. Calls made to the system result in either infinite ringtones (Google Voice) or a fast busy signal (Normal phone line)

2. REGISTER message only seems to be sent for the first VoIP_Registration element in the config file; subsequent ones are ignored


Attached are updated VoiceGuide and Wireshark logs with the trace from a session with a single attempted call via a normal phone line.



Share this post

Link to post

WireShark trace shows incoming call was answered OK by VoiceGuide.


But after the cal is answered (which is acknowledged by SIP server) the SIP server at then sends additional INVITES. Those new INVITES don't even have any SPD information. Purpose of those INVITES is unclear.


And then the SIP server at inexplicably hangs up the answered call (!)


What type of sip server is at


Whatever SIP server/switch is at it is not behaving according to SIP standard. It probably uses some propriety VoIP protocol that is specifically designed to only work with it's own branded handsets - but not with normal standard SIP handsets.




Also, Looks like you are using the default 1 port HMP license. That HMP license could be limiting the number of registered SIP extensions to 1.

Please try using the 2 port Evaluation license that Dialogic makes available online.


Share this post

Link to post



The SIP server is an NEC SV8500 PBX. Is there any additional setup needed on that end, or are there any settings that should be looked at?


Right now it is providing SIP stations, not SIP trunks. Is that supported by VoiceGuide and is any additional setup involved?


I'll take a look at the 2 port license for Dialogic to see if that resolves the registration issue.



Share this post

Link to post

VoiceGuide can connect to PBXs/Switches by both registering itself as SIP extensions and/or handling calls over a SIP trunk.


Setting up a SIP trunk is usually the better approach. PBXs which use propriety way of connecting to their handsets will usually still use standard SIP over a SIP trunk - to allow inter-connectivity with other VoIP systems.


Recommend that you look into setting up a SIP trunk from this PBX.


To use VoiceGuide as an 'extension' you will need determine what configuration changes on the PBX need to be made in order for the PBX to establish calls VoiceGuide's extension using the SIP standard (this may involve determining why right now the PBX disconnects the call after acknowledging that it knows that VoiceGuide has answered it...)

Share this post

Link to post

Create an account or sign in to comment

You need to be a member in order to leave a comment

Create an account

Sign up for a new account in our community. It's easy!

Register a new account

Sign in

Already have an account? Sign in here.

Sign In Now