Transfer Problem

[Guest zhangmin]

I do a 3Way Conference Call-Blind in the Transfer call module,but it not works.

my sip server is Elastix.what I should do to config my Elastix server or pbx?

thanks

transfer.zip

[SupportTeam]

As you are using SIP lines you must use the "Dialout and Conference" option.

[Guest zhangmin]

"Dialout and Conference" option checked

it returns fail path.I don't konw why?

when I chose PBX Hookflash Transfer ,it works well.the same phonenumber.

could you give me some advices?

0911_1605_vgEngine.txt

0911_ktTel.txt

[SupportTeam]

The destination transfer number was not specified as an IP address.

 

On VoIP systems the destination numbers need to be specified as IP addresses.

 

The number specified was:

 

6000

 

It probably should be:

 

6000@192.168.1.24

 

 

Please .ZIP up any traces before posting them.

[Guest zhangmin]

even though 6000@192.168.1.24

it still return fail path

0911_ktTts.zip

0911_CallEvents.zip

0911_1605_vgEngine.zip

[SupportTeam]

Also can you please add the following line to the VG.INI, in [Log] section:

 

Encoding=Unicode

 

This will save the vgEngine log files in Unicode, instead of ASCII.

[SupportTeam]

Please include the ktTel and WireShark traces as well that capture the "Dial and Conference" attempt.

 

Please .ZIP up any traces before posting them.

 

Also can you please add the following line to the VG.INI, in [Log] section:

 

Encoding=Unicode

 

This will save the vgEngine log files in Unicode, instead of ASCII.

[SupportTeam]

Have you set the CallerID on the outgoing call? Many VoIP switches will require the CallerID on the outbound call to let it know what account the outgoing call should be billed to - and then authenticate using that CallerID before allowing call to proceed.

 

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

[Guest zhangmin]

which file I need to set the CallerID on the outgoing call? config.xml is right?

by the way ,can you read chinese?

[Guest zhangmin]

I try again, for detail please see the attachment.thanks

attachment.zip

[SupportTeam]

Traces show that CallerID on outgoing call was set to: 2000@192.168.1.24

 

The switch is rejecting the outbound call with a "401-Unauthorised", and HMP/VoiceGuide cannot authenticate 2000@192.168.1.24 as it does not have authentication information for it.

 

The only extensions that you have provided the authentication information for in Config.xml are the 5000@192.168.1.24 and 7000@192.168.1.24 extensions.

 

Suggest CallerId is set to one of the registered extensions (5000@192.168.1.24 or 7000@192.168.1.24), or add authentication information for 2000@192.168.1.24 , or set the VoIP switch to not require authentication on outgoing calls.

 

From traces:

 

174215.625  6               voip authentication data idx=0: realm:asterisk, identity:sip:7000@192.168.1.24, username:7000, ******
174215.625  6               voip authentication data idx=1: realm:asterisk, identity:sip:5000@192.168.1.24, username:5000, ******

174458.171  6   3   1       sRvOutboundLeg=, strDialoutOptions=<CallerId>2000@192.168.1.24</CallerId>

 

Screenshot of WireShark attached:

[Guest zhangmin]

thanks I solve this problem by add authentication information for 2000@192.168.1.24