Cant Connect To Sip

[Guest jfsauriol]

I cant connect to my telco. I have tried to connect without any registrations and authentifations, as asked by my provider. He can accept my lines to its system from my IP address. If I do it whitout regis. and auth., but my config did not work. When I do it without regis and auth, I dont see the lines in my Line Status.

We have tried to put regis and auth, but still dont work. What can you suggest to me?

[SupportTeam]

He can accept my lines to its system from my IP address.

So sounds like you do not need to perform any SIP Registration in this case. Great. This makes system setup very straightforward.

 

When I do it without regis and auth, I dont see the lines in my Line Status

Please post the vgEngine and ktTel traces which show system startup when no SIP Registration information is specified in Config.xml.

[Guest jfsauriol]

vgEngine and ktTel log attached.

log1.zip

[SupportTeam]

When I do it without regis and auth, I dont see the lines in my Line Status

The traces show that 6 ports have been opened. The Line Status application should show that there are 6 lines "Waiting for a call...".

 

Can you post a screenshot of what you see when you start the Line Status application ?

 

Trace also shows a number of calls being attempted. To see what happened on those outgoing calls we would need to see the WireShark trace.

[Guest jfsauriol]

The traces show that 6 ports have been opened. The Line Status application should show that there are 6 lines "Waiting for a call...".

 

Can you post a screenshot of what you see when you start the Line Status application ?

 

Trace also shows a number of calls being attempted. To see what happened on those outgoing calls we would need to see the WireShark trace.

[SupportTeam]

No files were attached to the post. Can you try attaching them again?

[SupportTeam]

files supplied by user

trace1.zip

trace2.zip

[SupportTeam]

Trace 1 shows that system is making outbound calls (sends SIP INVITE messages)to IP 66.199.153.77, but SIP server at IP 66.199.153.77 is responding with a "401 Unauthorized" message.

You will need to ask the Telco that is running the SIP server at 66.199.153.77 why they are responding with a "401 Unauthorized" message instead of proceeding with the call.

 

 

Trace 1 shows that system is trying to register itself with SIP server at IP address 70.38.64.8, but the SIP server at IP address 70.38.64.8 is responding with a "401 Unauthorized" message to every REGISTER request.

 

The WWW-Authenticate field in the "401 Unauthorized" response is:

 

WWW-Authenticate: Digest algorithm=MD5, realm="pbx4", nonce="2b2f9282"

 

so you should change the 'realm' field in the Config.xml's VoIP_Authentication entry to be: pbx4

[Guest jfsauriol]

Hi,

 

I've asked my telco to make changes, and he is what they did:

 

"We've given you a standard account configured the same way all of our softphones and IP phones are configured. You have a username, xxx, and a password, yyy, used to register to us. You have G729 enabled. You just need voiceguide to register using this username and password pair".

 

My setup is as follows:

 

VoiceGuide PC ---- router ----- Internet ----- telco serve

 

My voiceguide PC is behind a NAT, so it`s IP is different than my public IP. My telco says the server I am using is on Asterisk 1.8. They claim they have hundred of phones and softphones, or many different models, all working well, but they don`t have a voiceguide to test with. According to them though, this configuration is absolutely typical.

 

the outbound proxy and SIP register server are the same.

 

Is there anything I should know about this sort of setup? Will Voiceguide work behind a NAT?

[Guest jfsauriol]

I've put the following config in, after my provider switched me over to a sip USERNAME / PASSWORD instead of a fixed IP. I have to specify my VG system is behind a NAT.

 

We`ve tried the same authentication with a softphone (Xlite, if it matters, also behind the NAT) and it registers fine. So the provider settings are correct. Why would the credentials work on a simple softphone but not on VG. Can VG work from behind a NAT router?

 

Here is my config, password can be given if needed for testing

66.199.153.77 is the server, jfsauriol2 the username. It seems like a very straightforward setup, but it is always rejected and the provider says if it works with IP phones, a softphone, etc it should work with any SIP-enabled system.

 

 

 

<VoIP_Lines>

<VoIP_Registrations>

 

<VoIP_Registration>

<Protocol>SIP</Protocol>

<RegServer>sip:66.199.153.77</RegServer>

<RegClient>jfsauriol2@66.199.153.77</RegClient>

<LocalAlias>jfsauriol2</LocalAlias>

<Expires>60</Expires>

</VoIP_Registration>

 

</VoIP_Registrations>

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display>Virtutel</Display>

<Realm>66.199.153.77</Realm>

<Identity></Identity>

<AuthUsername>jfsauriol2</AuthUsername>

<AuthPassword>***</AuthPassword>

</VoIP_Authentication>

 

</VoIP_Authentications>

 

</VoIP_Lines>

 

 

 

 

[SupportTeam]

Can you please try what was recommend in the previous answer posted:

 

you should change the 'realm' field in the Config.xml's VoIP_Authentication entry to be: pbx4

 

If you continue having problems after making that change then please post the WireShark trace and the VoiceGuide ktTel trace that capture system startup and reregistration attempts.