VoiceGuide IVR Software Main Page
Jump to content

Hmp Config File

Recommended Posts

Where can I to chceck that VG registered to SIP provider?

That is visible in ktTel trace. Best check is to just try placing a call to the SIP provider number.

 

Would recommend leaving Identity field blank/empty. No need to use this field unless you are registering multiple numbers with same provider.

 

In your config.xml woudl change:

 

<Realm></Realm>

<Identity>48222441593@sip.itc.gtsenergis.pl</Identity>

 

to:

<Realm>sip.itc.gtsenergis.pl</Realm>

<Identity></Identity>

 

 

If you have any problems with SIP registration please post WireShark trace capturing SIP registration.

Share this post


Link to post

in WIRESHARK no logs with sip protocol.

I started wireshark and then run vg and no one line with sip protocol.

Share this post


Link to post

Would have expected to see SIP REGISTER requests at least.

Can you see any SIP activity when trying to dial this machine directly by dialing its IP directly from another SIP phone/softphone?

Share this post


Link to post

You can use a softphone on another computer (on same network) and just dial the IP address of the VoiceGuide machine.

Share this post


Link to post

No success with x-lite calling ip. no trace on wireshark.

 

maybe i made something wrong?

 

I installed HMP Software, and DCM (no extra configuration).

Then I configured config file and start script.

 

Should I configure something else?

post-4030-129837685853_thumb.jpg

Share this post


Link to post
harNo success with x-lite calling ip. no trace on wiresk.

 

Mybe you have a firewall installed on this system that is blocking the SIP packets? Or maybe check if WireShark is monitoring the right network interface?

 

If you have external softphone calling in the you should see incoming SIP INVITE messages messages in the WireShark's network interface trace.

Share this post


Link to post

If VoiceGuide/HMP is running and a SIP call is sent to that system then VoiceGuide/HMP will automatically answer that call.

Share this post


Link to post

I can't configure an sip client (x-lite) to call to my computer with VOICEGUIDE/HMP.

test I made with x-lite registered by my voip provider.

 

Can You help me to test call to my computer IP address?

Witch soft-phone to use and how to configure?

Share this post


Link to post

Now next step:

 

I attach log and config files.

 

Can You to check where is problem because VG doesn't answer when i call to 48222441539 but answered when i call direct ip address.

SISTI.zip

Share this post


Link to post

Can you please post the WireShark trace which captures the VoiceGuide startup and the incoming direct ip call.

Share this post


Link to post

Can you try using IP address of sip.itc.gtsenergis.pl instead, and see if that results in the SIP REGISTER packet being sent out?

Share this post


Link to post

Can you send me an example of working config file?

 

Is my config file properly wrote?

Now main problem is logging by the voip provider...

Share this post


Link to post

No SIP REGISTER been send after changing address to ip.

 

Please send me an example working config file.

 

I see no trace line with vg logging to VoIP server.

Share this post


Link to post

Can we have a quick look at this machine by remotely logging into it? (using logmein.com or similar)

 

Please email login details to support@voiceguide.com and include a link to this support forum thread in your email.

Share this post


Link to post

Trace shows that the REGISTER request sent out by VoiceGuide/HMP is being replied to by the SIP server with a "401 Unauthorized".

 

As there is no followup REGISTER message sent out then this means that VG's Config.xml does not have the matching Realm entry.

 

In 401 response we see:

 

realm="sip.itc.gtsenergis.pl"

 

So there should be an Authentication entry in Config.xml for realm sip.itc.gtsenergis.pl

Please ensure that you have this Authentication entry in your Config.xml

 

If you still have problems with this then you can post your Config.xml here (remove the password) and we can modify it for you.

Share this post


Link to post

WireShark trace shows that the SIP registration is now OK.

 

The SIP server with which you registered should now be sending calls to the VoceGuide/HMP for the account/number that was registered.

 

What happens when you make the call into the system through the SIP provider's number?

Share this post


Link to post

SIP doesn't working.

 

When i try to call i hear that connection can't be established. In my Account by Provider i don't see that VG logged into SIP server.

 

until yesterday no logs to my voip account.

Share this post


Link to post

Can You to send an example of config file with registration to other VoIP provider? I created callcentric account and with your suggestion in example config file I can't registering.

 

can You to write config file with free account in callcentric witch working properly?

Share this post


Link to post
In my Account by Provider i don't see that VG logged into SIP server.

You should contact the SIP service provider and ask them why this is the case, given that WireShark's network trace is showing you that the registration was successful.

Share this post


Link to post
I created callcentric account and with your suggestion in example config file I can't registering

Can you please post VG's Config.xml file and the WireShark trace capturing the SIP registration messages sent out at VoiceGuide service startup.

(change the authentication password in the posted config.xml file).

Share this post


Link to post

I've got suggestion from my VoIP provider: Use in request-uri and Authorization domain (no ip address) and don't use TTL.

 

If You can help me I can to send password per e-mail to test difference configuration.

 

Success my login into VoIP service (wireshark logs) not mean that I can receive calls.

Prompt from Provider : target number is unavailable. Provider see me but can't recognize connected station.

Share this post


Link to post

Can you please post the WireShark trace that captures registration when "sip.itc.gtsenergis.pl" is specified as the registration server's address, instead of the IP address.

 

The previous WireShark trace shows SIP server responding with OK to registration request, which is usually an indication that SIP provider is now ready to send calls to the party that registered the number.

Share this post


Link to post

When config include:

 

<RegServer>sip.itc.gtsenergis.pl</RegServer>

<RegClient>48222441593@sip.itc.gtsenergis.pl</RegClient>

 

or

 

<RegServer>sip.itc.gtsenergis.pl</RegServer>

<RegClient>48222441593@217.153.192.36</RegClient>

 

no trace in wireshark with SIP protocol.

 

I think, you can test it.

 

I send trace with IP addres (to trace number 46990)

Then I started X-LITE with domain address and he login without problems as You can see in traces from 62845.

 

Problem is I don't see any traces in WireShark when I put domain name in config file.

sip20110304.zip

Share this post


Link to post

The posted trace was a .txt file.

 

WireShark trace files are .pcap files. They contain more information then the exported .txt file that was posted.

Share this post


Link to post

Registration works fine from one of our test servers (this one was Win2008 x32 running HMP 3.0 SU296).

 

Three variants below were used, and all resulted in REGISTER messages being sent out and resent with digest authentication upon receipts of 401 response. (a dummy password was used to registration was not successful, but the messages were being sent out).

 

(see attached file for pcap trace)

 

When we had a quick look on your machine it looked like you had many other programs installed on it.

Sounds like you have other VoIP apps (X-lite?) installed on this system as well.

Would recommend that you deploy HMP on its own separate machine to test HMP in isolation.

 

 

 

 

<VoIP_Registrations>

 

<VoIP_Registration>

<Display></Display>

<Protocol>SIP</Protocol>

<RegServer>sip.itc.gtsenergis.pl</RegServer>

<RegClient>48222441593</RegClient>

<LocalAlias>48222441593</LocalAlias>

</VoIP_Registration>

 

</VoIP_Registrations>

 

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display></Display>

<Realm>sip.itc.gtsenergis.pl</Realm>

<Identity></Identity>

<AuthUsername>48222441593</AuthUsername>

<AuthPassword>asd123</AuthPassword>

</VoIP_Authentication>

 

</VoIP_Authentications>

 

 

-------------------------------------------------------------

 

<VoIP_Registrations>

 

<VoIP_Registration>

<Display></Display>

<Protocol>SIP</Protocol>

<RegServer>sip.itc.gtsenergis.pl</RegServer>

<RegClient>48222441593@sip.itc.gtsenergis.pl</RegClient>

<LocalAlias>48222441593</LocalAlias>

</VoIP_Registration>

 

</VoIP_Registrations>

 

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display></Display>

<Realm>sip.itc.gtsenergis.pl</Realm>

<Identity></Identity>

<AuthUsername>48222441593</AuthUsername>

<AuthPassword>asd123</AuthPassword>

</VoIP_Authentication>

 

</VoIP_Authentications>

 

-------------------------------------------------------------

 

 

<VoIP_Registrations>

 

<VoIP_Registration>

<Display></Display>

<Protocol>SIP</Protocol>

<RegServer>sip.itc.gtsenergis.pl</RegServer>

<RegClient>48222441593@sip.itc.gtsenergis.pl</RegClient>

<LocalAlias>sip:48222441593@217.153.192.36:5060</LocalAlias>

</VoIP_Registration>

 

</VoIP_Registrations>

 

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display></Display>

<Realm>sip.itc.gtsenergis.pl</Realm>

<Identity></Identity>

<AuthUsername>48222441593</AuthUsername>

<AuthPassword>asd123</AuthPassword>

</VoIP_Authentication>

 

</VoIP_Authentications>

reg.zip

Share this post


Link to post

I try again with HMP but again without success.

 

trace looks as all working good but SIP showing that after inviting i receive BYE...

 

Where is the problem?

sip.zip

Share this post


Link to post

please post the WireShark format .pcap file.

 

Attached text file does not have enough details about what was sent/received so we are unable to comment on it.

Share this post


Link to post

Trace shows the VoiceGuide system registered itself with the SIP server (sip.itc.gsenergis.pl) and that a call arrived at the VoiceGuide system, which VoiceGuide answered, but soon after answering the call VoiceGuide hung up.

 

To see why VoiceGuide ended the answered call we would need to see the VocieGuide trace file capturing the incoming call. (ktTel and vgEngine traces)

 

Please start a new thread about this as this would be unrelated to the HMP config file.

Share this post


Link to post
This topic is now closed to further replies.
×