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Hello Technical Support Team, please help me, the ACD does not work, I have Voieguide v7.6.30 VGagent v7.6 HMP DialogicHMP_for_VoiceGuide_393. The call in a normal transfer does arrive. The problem is if the VG transfer to an ACD queue is not routed to the agent's phone. Pop-up only. the PBX is: Fortinet model Fortivoice 200F IP voiceguide 10.10.91.201 IP Pbx 10.10.91.200 Ext Agent / AgentTelephone=4510 ext register VG /hmp/pbx 4500-4507 acd queue redque I attach the logs. and other captures. BR Diego MIno logs_para_caso.zip

hi do you have any experience with Genesys Sip Server? Configuring a sip trunk in Genesys I send the call to voiceguide without problem, but to return or transfer I can only by the conference option. But there I occupy two ports. Is there a way to transfer it to a sip trunk? Or can you create a sip trunk on the voiceguide side? Br Diego

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

Hi could you please check the log let me know y i am not able to call mobile numbers. Also for calling to sip number its connecting sometimes but mostly its getting error please check. WireSharkLog.pcapng 1231_CallEvents.txt 1231_vgService.txt ClusterPkg.log

i would like to know is there a possibility to configure the script based on SIP number using SIP Trunk example: If I receive a call on SIP Number 201 its should played script 1 If I receive a call on SIP Number 202 its should played script 2 if this can be done then please let me know the process.

Hi i am using Transfer to call please check the below image and logs.When press 1 call is hanging up not able to transfer please let me know how could i use transfer to call as i am using SIP Trunk. WireSharkLog.pcapng 1226_CallEvents.txt

Hi, I have downloaded VoiceGuide_7.6.4 and DialogicHMP30_Drivers_for_VoiceGuide_7.5_gen2_375 successfully install on my laptop I am using CiscoCallManager to configure Voice Guide i followed below link https://www.voiceguide.com/vghelp/source/html/config_ciscocallmanager.htm. And also added VoIP Registration and VoIP Authentication details in VoiceGuide config file. Please let me know what is the issue. Attached files are ConfigFile,Logfile,WireShark capture file WireSharklog.pcapng Config.xml 1219_ktTel.txt 1219_0916_vgEngine.txt

SIP Configured but not able to originate calls and receive calls. kindly check the log and revert back log.zip

My system is having trouble connecting to the SIP server. The solution I found was to stop VG and restart it. It is really inconvenient because it implies that I am always ready to do this action. When the system does not receive the correct answer from the SIP, it gets bogged down and dials the numbers one at the back of the other without a pause between each attempt. Is it there any way to ask that during a unsuccessful attempt that the system waits some time before another try? I include the files with this post and before publishing it please remove the IP addresses and city name. Trace_04-23-2018_(2).zip