Extensions Sip Unavailable

[dminof]

Hello, sip extensions get disconnected, but not all, just a few

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075847.870   6                      vgEngine : 7.5.21 - 7.5.6739.39172
075847.870   6                      compiled : 2018-06-14 20:45:44.72
075847.870   6                      location : C:\Program Files\VoiceGuide\vgEngine.dll
075847.874   6                      written  : 2018-06-14 06:39:56

 

-------CONFIG  --------

<Channels>

<Channel>
<Device_Voice>dxxxB1C1</Device_Voice>
<Device_Network>iptB1T1</Device_Network>
<Device_Media>ipmB1C1</Device_Media>
<Protocol>IP</Protocol>
<Script>C:\Program Files\VoiceGuide\Scripts\recargas\rl_recargas.vgs</Script>
<AllowDialOut>1</AllowDialOut>
</Channel>

<Channel>
<Device_Voice>dxxxB1C2</Device_Voice>
<Device_Network>iptB1T2</Device_Network>
<Device_Media>ipmB1C2</Device_Media>
<Protocol>IP</Protocol>
<Script>C:\Program Files\VoiceGuide\Scripts\recargas\rl_recargas.vgs</Script>
<AllowDialOut>1</AllowDialOut>
</Channel>
<Channel>
<Device_Voice>dxxxB1C3</Device_Voice>
<Device_Network>iptB1T3</Device_Network>
<Device_Media>ipmB1C3</Device_Media>
<Protocol>IP</Protocol>
<Script>C:\Program Files\VoiceGuide\Scripts\recargas\rl_recargas.vgs</Script>
<AllowDialOut>1</AllowDialOut>
</Channel>
<Channel>
<Device_Voice>dxxxB1C4</Device_Voice>
<Device_Network>iptB1T4</Device_Network>
<Device_Media>ipmB1C4</Device_Media>
<Protocol>IP</Protocol>
<Script>C:\Program Files\VoiceGuide\Scripts\recargas\rl_recargas.vgs</Script>
<AllowDialOut>1</AllowDialOut>
</Channel>
<Channel>
<Device_Voice>dxxxB2C1</Device_Voice>
<Device_Network>iptB1T5</Device_Network>
<Device_Media>ipmB1C5</Device_Media>
<Protocol>IP</Protocol>
<Script>C:\Program Files\VoiceGuide\Scripts\recargas\rl_recargas.vgs</Script>
<AllowDialOut>1</AllowDialOut>
</Channel>
<Channel>
<Device_Voice>dxxxB2C2</Device_Voice>
<Device_Network>iptB1T6</Device_Network>
<Device_Media>ipmB1C6</Device_Media>
<Protocol>IP</Protocol>
<Script>C:\Program Files\VoiceGuide\Scripts\recargas\rl_recargas.vgs</Script>
<AllowDialOut>1</AllowDialOut>
</Channel>
<Channel>
<Device_Voice>dxxxB2C3</Device_Voice>
<Device_Network>iptB1T7</Device_Network>
<Device_Media>ipmB1C7</Device_Media>
<Protocol>IP</Protocol>
<Script>C:\Program Files\VoiceGuide\Scripts\recargas\rl_recargas.vgs</Script>
<AllowDialOut>1</AllowDialOut>
</Channel>
<Channel>
<Device_Voice>dxxxB2C4</Device_Voice>
<Device_Network>iptB1T8</Device_Network>
<Device_Media>ipmB1C8</Device_Media>
<Protocol>IP</Protocol>
<Script>C:\Program Files\VoiceGuide\Scripts\recargas\rl_recargas.vgs</Script>
<AllowDialOut>1</AllowDialOut>
</Channel>

</Channels>

<Parms>
</Parms>

</Devices_Dialogic>

<VoIP_Lines>

<Notes>
Pretty much any VoIP SIP line can be registered for use with VoiceGuide. 
This includes extensions from internal VoIP PBXs or lines from VoIP providers.

Here is a description of Registration and Authentication fields:

-------------------------------------------------------------------
VoIP_Registration

Protocol   : "SIP"
RegServer  : Name or IP address of the SIP server. eg: "101.102.103.104" or "sip.somevoipservice.com"
RegClient  : VoIP Username. eg: "5551234" or "5551234@sip.examplevoipservice.com" 
LocalAlias : Local Alias for this line/extension. eg: "BobJones"
-------------------------------------------------------------------
VoIP_Authentication

Realm        : Leave this blank, unless you are registering same account with multiple VoIP providers.    examples: "somevoipservice.com", "asterisk"
Identity     : Leave this blank, unless you are registering multiple accounts with same VoIP provider. examples: "sip:1231238@somevoipservice.com"
AuthUsername : Autnetication Username. eg: "bob"
AuthPassword : Autnetication Password. eg: "password1"
-------------------------------------------------------------------
</Notes>

<VoIP_Registrations>

<VoIP_Registration>
<Display>Ivr1</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.10.5</RegServer>
<RegClient>1004@192.168.10.5</RegClient>
<LocalAlias>1004@192.168.10.137</LocalAlias>

</VoIP_Registration>

<VoIP_Registration>
<Display>Ivr1</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.10.5</RegServer>
<RegClient>1005@192.168.10.5</RegClient>
<LocalAlias>1005@192.168.10.137</LocalAlias>

</VoIP_Registration>

<VoIP_Registration>
<Display>Ivr1</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.10.5</RegServer>
<RegClient>1006@192.168.10.5</RegClient>
<LocalAlias>1006@192.168.10.137</LocalAlias>

</VoIP_Registration>

<VoIP_Registration>
<Display>Ivr1</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.10.5</RegServer>
<RegClient>1007@192.168.10.5</RegClient>
<LocalAlias>1007@192.168.10.137</LocalAlias>

</VoIP_Registration>

<VoIP_Registration>
<Display>Ivr1</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.10.5</RegServer>
<RegClient>1008@192.168.10.5</RegClient>
<LocalAlias>1008@192.168.10.137</LocalAlias>

</VoIP_Registration>

<VoIP_Registration>
<Display>Ivr1</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.10.5</RegServer>
<RegClient>1009@192.168.10.5</RegClient>
<LocalAlias>1009@192.168.10.137</LocalAlias>

</VoIP_Registration>

<VoIP_Registration>
<Display>Ivr1</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.10.5</RegServer>
<RegClient>1010@192.168.10.5</RegClient>
<LocalAlias>1010@192.168.10.137</LocalAlias>

</VoIP_Registration>

<VoIP_Registration>
<Display>Ivr1</Display>
<Protocol>SIP</Protocol>
<RegServer>192.168.10.5</RegServer>
<RegClient>1011@192.168.10.5</RegClient>
<LocalAlias>1011@192.168.10.137</LocalAlias>

</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>
<Display>Ivr1</Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1004</AuthUsername>
<AuthPassword>1004</AuthPassword>
<CallerID>1004@192.168.10.5</CallerID>
</VoIP_Authentication>

<VoIP_Authentication>
<Display>Ivr2</Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1005</AuthUsername>
<AuthPassword>1005</AuthPassword>
<CallerID>1005@192.168.10.5</CallerID>
</VoIP_Authentication>

<VoIP_Authentication>
<Display>Ivr3</Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1006</AuthUsername>
<AuthPassword>1006</AuthPassword>
<CallerID>1006@192.168.10.5</CallerID>
</VoIP_Authentication>

<VoIP_Authentication>
<Display>Ivr4</Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1007</AuthUsername>
<AuthPassword>1007</AuthPassword>
<CallerID>1007@192.168.10.5</CallerID>
</VoIP_Authentication>

<VoIP_Authentication>
<Display>Ivr5</Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1008</AuthUsername>
<AuthPassword>1008</AuthPassword>
<CallerID>1008@192.168.10.5</CallerID>
</VoIP_Authentication>

<VoIP_Authentication>
<Display>Ivr6</Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1009</AuthUsername>
<AuthPassword>1009</AuthPassword>
<CallerID>1009@192.168.10.5</CallerID>
</VoIP_Authentication>

<VoIP_Authentication>
<Display>Ivr7</Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1010</AuthUsername>
<AuthPassword>1010</AuthPassword>
<CallerID>1010@192.168.10.5</CallerID>
</VoIP_Authentication>

<VoIP_Authentication>
<Display>Ivr8</Display>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1011</AuthUsername>
<AuthPassword>1011</AuthPassword>
<CallerID>1011@192.168.10.5</CallerID>
</VoIP_Authentication>

 

 

Br Diego

[SupportTeam]

Please post VoiceGuide's vgEngine and ktTel traces from VoiceGuide's \log\ sub-directory which capture system startup and extensions registration process.

A WireShark capture of the SIP communications between VoiceGuide/HMP and the Grandstream UCM6108 would also be good to see.

Configuration quoted shows that this system is opening 8 VoIP ports, so looks like it must be running at least an 8 channel HMP license. Is this correct?

HMP  only supports as many registrations as there are ports licenses on the system.

 

Also:

A better approach is to just have a 'SIP Trunk' between a PBX or SIP Service provider and VoiceGuide/HMP, instead of registering with individual extensions.

Attaching Grandstream's documentation on how to configure a "Peer Trunk" on the Grandstream CM6XXX series PBXs. You just point the trunk to the IP address of the VoiceGuide/HMP system and the set up the Outbound Route to send calls to certain extension/number down that Peer SIP Trunk. 

No need to set up any Regtraion/Authentication on VoiceGuide/HMP side. All calls arriving at VoiceGuide/HMP IP address will be answered automatically.

 

ucm6xxx_sip_trunk_guide.pdf

[dminof]

Hi, tks for your help.

Done. I already managed to configure the incoming call as a trunk and it goes directly to the IP of the voice guide, it works. For the voiceguide  outbound-dialer calls, what would the configuration be using sip trunk or  I have to configure are registered extensions in config.xml?

br

Diego

[SupportTeam]

To make outgoing calls through the PBX it is usually sufficient to make a call to:

destination_number@pbx_ip_address

and the PBX will then route the call out to the telephone number (or extension) specified.

If PBX sees the call coming in from IP address that it has a Peer Trunk with, then the call will usually be allowed through without any authentication required. (ie: PBX will use "Trusted IP Address Authentication" approach).

 

If PBX is set up to require authentication on every outgoing call request, then you can specify the user/extension which is to be used to authenticate the outgoing call by setting the "CallerID" on the outgoing call. The PBX then goes through authentication process before continuing with the call, and VoiceGuide/HMP performs the authentication using the data provided in Config.xml <VoIP_Registration> and <VoIP_Authentication> entries.

Please see: https://www.voiceguide.com/vghelp/source/html/dial_voip.htm

[dminof]

Hello,   

""To make outgoing calls through the PBX it is usually sufficient to make a call to:

destination_number@pbx_ip_address

and the PBX will then route the call out to the telephone number (or extension) specified.""

 

For this you would not also need to create an incoming route? in the pbx, related to the trunk?

Br

Diego

[SupportTeam]

Probably not. But this may depend on PBX and how the trunks from PBX to the Telco are configured.