Do I Need To Install Dialogic Hmp Interface Boards For Voip

[ID2016]

Hi VG Support,

 

I installed VG V7 for VOIP on my local laptop. I have tested HMP 3.0 v 367 yesterday and saw expected result: CIPDevice::processEvent -> receive GCEV_UNBLOCKED on :N_iptB1T1:P_IP:M_ipmB1C1. But today I am getting this:

************************************************************

* *

* IP Media Server - Media services over IP Demo Program. *

* Copyright ⌐ 2007 Dialogic Corporation. *

* *

************************************************************

DTMFMode inband

TxCoder[0]

Capability: g711mulaw

Type: 2

Direction: 1

Payload_type: 255

FramesPerPacket: 20

VAD: 0.

RxCoder[0]

Capability: g711mulaw

Type: 2

Direction: 2

Payload_type: 255

FramesPerPacket: 20

VAD: 0.

[info] CEventRouter::Init: Initializing channels...may take a few seconds!

<<Number of Fax (& Voice) boards found: 1 >>

<<Number of Voice (& Fax boards) found: 1 >>

<<Number of IPT boards found: 1>>

Error: CIPTBoard::Init -> gc_OpenEx failed to get IPT board

handle for :N_iptB1:P_IP

Error: CIPModule::Init -> failed to initialize IPT board :N_iptB1:P_IP

<<Number of IPM boards found: 1>>

Waiting for key:

'Q' - to quit

What could possibly went wrong?

Thank you

 

[SupportTeam]

You do not need to install any interface boards to use VoIP.

 

VoIP comms is done over the PC/servers own network interface.

[ID2016]

1214_1334_vgEngine.txt1214_1419_vgEngine_extract.txt1214_vgService.txt

 

Thank you. I was able to resolve this issue.

However I have problems configuring and placing calls via VOIP.

 

I would appreciate your help as I never worked with VOIP/SIP, etc but need to proceed with evaluation and write scripts.

 

So far I installed HMP 3.0, VG 7.5.7; connected to existing MS SQL (test environment) in Config.xml. Installed SJPhone 1.65 on another PC.

 

Do I need to select free SIP provider for validation? We do have Cisco installed internally but SIP trunk needs to be built by network admins.

Once I selecte a SIP provider, what values should be populated into VOIP regserver and regclient.

 

Also when I run netstat -abno, I see that port 5060 is running PID 10320 without specifics what it is; so I am hoping this related to HMP installation.

Not sure if need to attache any logs as SIP is not configured yet.

 

 

I appreciate your direction in building evaluation of VG. I have few days to complete and come up with suggestions.

 

Thank you !

[ID2016]

I wanted to add that when enteing my IP address on SJPhone (different PC) , I see that in the DOS window where I run HMP test utility (where VG is installed), the output is generated.Please see the attached file.

 

Basically, when I type my IP address in JSPhone and click Dialed Number options, I hear a recording : Main Menu. For VoiceMail press1, For Fax press 2, etc. This all is recorded in this window.

 

Thank you for your help.

Output.txt

[SupportTeam]

Pleae now exit IPMedia server demo and start VoiceGuide service afterwards.

 

Then place call into the system.

 

Please post the vgEngine and ktTel trace fiels that capture the incoming call.

 

We can then see what is going on with the incoming calls into VoiceGuide.

[ID2016]

Done. I am attaching the log files that created now - without config.xml file being populated with SIP information.

 

Today I was provided information for Cisco SIP trunk based on this info http://www.voiceguide.com/vghelp/source/html/config_ciscocallmanager.htm

• IP address of Communications Manager for the VoiceGuide SIP trunk —> 99.999.999.99 (if I can insert two IP address then also use 99.999.999.00)
• Telephone number that routes to this trunk —>345-345-3456

 

Question : how config.xml file should look like for this connection and what needs to be in vg.ini?

How do I place inbound and outbound calls. The purpose of this development is placing outbound calls outside the network on the random phone numbers.

For test purposes I have SJphone; can I test with a mobile phone?

 

<VoIP_Registrations>

<VoIP_Registration>

<Display></Display>

<Protocol></Protocol>

<RegServer></RegServer>

<RegClient></RegClient>

<LocalAlias></LocalAlias>

<Expires></Expires>

</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>

<Display></Display>

<Realm></Realm>

<Identity></Identity>

<AuthUsername></AuthUsername>

<AuthPassword></AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

ktTel file is empty.

 

Thank you.

1215_vgService.txt1215_1058_vgEngine.txt1215_0915_vgEngine.txt1214_vgService.txt

[SupportTeam]

Could you please post the entire vgEngine trace file(s) - as saved by VoiceGuide.

 

Only excerpts were posted above.

[ID2016]

This is all what I have in the files. These are not excerpts.

[SupportTeam]

Has the logging been turned off in VG.INI file?

 

Could you please post the VG.INI file from your system.

 

VG.INI is in VoiceGuide's main directory.

[ID2016]

Thank you for your response! I modified the vg.ini and can see more information now in the logs. I was able to make outgoing calls using sip ip provided to me.

 

So far i have few issues:

1. The error during testing with utility IP Media Server is coming and going without apparent reason and I cannot fix it. One day it is there , and then it works. Is it due to the eval license?

2. VGStatusMonitor generates an error when I try to open it but then opens fine.

3.I do see some errors in logs files and i do not know the explanation.

4. I have few working scripts on V6 for Dialogic card. Do I need to do anything so will work on V7 over VOIP?

5. Should I be also able to make incoming calls with the same setting?

 

Thank you. I am attaching log files and vg.ini

 

vg_settings.txt1216_0937_vgEngine.txt 1216_1057_vgEngine.txt1216_1234_vgEngine.txt1216_1243_vgEngine.txt1216_CallEvents.txt1216_ktTel.txt1216_ktTts.txt1216_vgService.txt

 

Thank you!

[SupportTeam]

1. The error during testing with utility IP Media Server is coming and going without apparent reason and I cannot fix it. One day it is there , and then it works. Is it due to the eval license?

 

It would not be effaced by license status.Do you leave it running for a long time?

If you got VoiceGuide to answer/make calls then there is not need to test with IPMediaServer anymore.

Just start VoiceGuide service (make sure no other VoIP software is running at same time) and place calls direct into VoiceGuide.

(VoiceGuide in evaluation mode does need restarting every hour).

 

2. VGStatusMonitor generates an error when I try to open it but then opens fine.

 

Is VoiceGuide service running when you start the VoiceGuide Status Monitor ?

What error do you see? Can you post a screenshot?

 

4. I have few working scripts on V6 for Dialogic card. Do I need to do anything so will work on V7 over VOIP?

 

v6 scripts should work on v7

 

5. Should I be also able to make incoming calls with the same setting?

 

Yes.

 

Traces posted show outgoing call attempts but no incoming calls.

Can you please first confirm that incoming calls are answered correctly.

Please post traces capturing a call coming into the system being answered by VoiceGuide.

 

Once you have incoming calls working correctly and being answered by VoiceGuide then you can move onto placing outgoing calls.

[ID2016]

Last week I was able to make a call on eval version VOIP V7 (using DB from V6), today I cannot make any calls.

Please check my config.xml file.

There is VOIP config there and our test DB connection string. I see records in the callque table when I load the phone into Loader, but no calls.

Also I have noticed that status monitor never changes the mapping to the script no matter what I enter into the loader.

 

Can you please confirm if my VOIP setting is correct. Is it OK to user DB from V6 if VG is shutdown there?

 

 

<?xml version="1.0"?>

<VoiceGuideConfig>

<Devices_Dialogic>

<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\Credit Card Payment\Credit Card Payment.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></Display>

<Protocol>SIP</Protocol>

<RegServer>10.248.237.12</RegServer>

<RegClient>6146635430@10.248.237.12</RegClient>

<LocalAlias></LocalAlias>

<Expires></Expires>

</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>

<Display></Display>

<Realm></Realm>

<Identity></Identity>

<AuthUsername></AuthUsername>

<AuthPassword></AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

<Example>

<VoIP_Registrations>

<VoIP_Registration>

<Display>CallCentric (www.callcentric.com) </Display>

<Protocol>SIP</Protocol>

<RegServer>callcentric.com</RegServer>

<RegClient>177711111111@callcentric.com</RegClient>

<LocalAlias>177711111111@10.1.1.9</LocalAlias>

</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>

<Display>CallCentric</Display>

<Realm></Realm>

<Identity></Identity>

<AuthUsername>177711111111</AuthUsername>

<AuthPassword>abcd1234</AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

</Example>

</VoIP_Lines>

<Dialer>

<Notes>

By default VoiceGuide will use SQLite as its backend database.

Other database engines can be used instead if preferred.

Below are some examples showing how to configure VoiceGuide to use other database engines for its backend database.

</Notes>

<OutDialQue_ADODB_Provider>System.Data.SQLClient</OutDialQue_ADODB_Provider>

<OutDialQue_Database>OutDialQue</OutDialQue_Database>

<OutDialQue_ConnectString>Data Source=cocbaptdead1.co.trinity-health.org;Database=OutDialQue;User ID=voiceguide;Password=voiceguide;</OutDialQue_ConnectString>

<OutDialQue_PortToUse_LinkField>Disable</OutDialQue_PortToUse_LinkField>

<OutDialQue_SqlPrefix></OutDialQue_SqlPrefix>

<OutDialQue_SqlSuffix></OutDialQue_SqlSuffix>

</Dialer>

<acd>

<Example>

<queues>

<queue>

<name>redque</name>

<paths>

on {timeout 60} goto [Voicemail box 0001]

on {1} goto [redque 1]

</paths>

</queue>

<queue>

<name>bluque</name>

<paths>

on {timeout 60} goto [Voicemail box 0002]

on {1} goto [bluque 1]

</paths>

</queue>

</queues>

</Example>

</acd>

</VoiceGuideConfig>1219_1732_vgEngine.txt1219_1732_vgEngine.txt1219_CallEvents.txt1219_vgService.txt

Thank you very much !

[ID2016]

I have not received a response from you about VOIP configuration.

I am attaching more log files. 1220_1335_vgEngine.txt1220_ktTel.txt1220_CallEvents.txt

 

Thank you.

[ID2016]

Adding Dialogic logs from Dialogic/logs ( complete directory) . I am using VOIP.

Apology for attaching seprarte files instaed of zip version.... I was trying to rename/change zip files but the system doe snot allow me to send zip files in type of presentation....

 

I'd need help to figure out VOIP issue....cluster.txtClusterPkg.txtctbbapi.logdm3enumerate.logrtflog-LOCAL-20161219-19h55m51.347s.txtrtflog-LOCAL-20161220-01h56m49.479s.txtrtflog-LOCAL-20161220-08h31m25.485s.txtrtflog-LOCAL-20161220-11h47m22.511s.txtrtflog-LOCAL-20161220-10h27m00.056s.txtrtflog-LOCAL-20161220-01h56m49.479s.txt

rtflog-LOCAL-20161220-09h04m30.415s.txt

[SupportTeam]

Traces show outgoing call attempts to 92489789145

 

As this is a VoIP system, outgoing calls would usually be specified in this format:

 

number@address

 

Could you please describe in more detail your VoIP setup. Through what VoIP switch/device do you intend to send the outgoing VoIP calls? What is the IP address of that device?

 

As advised previously, we recommend testing that incoming calls work first. Do you have incoming calls working reliably now? How are these calls sent into your system? It looks like you have not set up any authentication for any VoIP service provider, so can you please advise in more detail how the VoIP calls would be routed in/out of this system?

[SupportTeam]

Response from user:

 

Yes of course.
The system needs to make only outgoing calls.
The IP address that included in earlier email is an IP for SIP trunk to Cisco. Also there is an internal phone number associated with this IP.
The phone 92489789145 is an example of external outgoing call with prefix 9.

I have sent you config file with settings for server. But i wanted you to look at the config and also at log files and determine whats wrong.

[SupportTeam]

If you have a SIP trunk and an IP address for this SIP trunk then have you tried placing the calls by dialing:

 

92489789145@sip-trunk-ip-address

 

or maybe:

 

2489789145@sip-trunk-ip-address

 

(Do you know whether the dialed numbers need a "9" in front of them when made over that SIP trunk?)

 

Will calls placed over this SIP trunk need to be authorized/authenticated? Or will any call sent over this SIP trunk willjuet be made, without any need for per-call authentication?

 

It may be a good idea to use WireShark (www.wireshark.org) to capture the SIP level communications if you encounter any issues with sending the calls over the SIP trunk.

(Specify sip in the WireShark's Filter text box to only view SIP messages)

[SupportTeam]

Response from user:

 

Yes 9 needs to be infront of the number.

How should i place the calls using mynumber@sip-ip?

Should it be inserted in config.xml file or directly placed in the loader?

I did see SIP calls in wireshark and it all was with status ok but no call was made this week.

Have u seen my dialogic log?

Thanks.

[SupportTeam]

How should i place the calls using mynumber@sip-ip?

 

in place where you before specified 92489789145, now specify:

 

92489789145@sip-trunk-ip-address

 

(replace sip-trunk-ip-address with actual IP address)

 

If you still encounter issued please post ktTel and WireShark logs

[ID2016]

Calls are not getting through....Please see logs from VG and Dialogicrtflog-LOCAL-20161221-17h36m11.552s.txt1221_1756_vgEngine.txt1221_CallEvents.txt1221_ktTel.txtrtflog-LOCAL-20161221-17h32m20.446s.txt

 

Thank you.

[ID2016]

Please read previous email with log files from Dialogic and VoiceGuide.

Here - Wireshark fileSIP_Whireshark_pcapng.txt1221_1756_vgEngine.txt1221_CallEvents.txt1221_ktTel.txtrtflog-LOCAL-20161221-17h32m20.446s.txt

 

 

Please look into log files. The calls are not going through.

rtflog-LOCAL-20161221-17h36m11.552s.txt

[SupportTeam]

Traces show that calls are being placed to:

 

92489789145@101.102.103.104

 

Is the IP address of the SIP trunk 101.102.103.104?

 

WireShark shows no response when SIP INVITE message are sent to IP 101.102.103.104

 

 

Please re-confirm the SIP trunk IP address and post WireShark and ktTel traces if you still encounter issues.

(Please .ZIP up traces before posting them

[ID2016]

No avail - no calls wen through.

Attaching todays' logs - VG, Dailogic and whireshark.

I have two possible IP for Cisco SIP trunk. Traces for each of the IP I have tried via phoneloader.

Please review 4 zip files (i renamed them to txt due to security)VG_logs.txtdialogic_log_12222016.txtwireshark_IP_10_231_234_19.txtwireshark_IP_10.248.237.12.txt

 

Thank you!

[SupportTeam]

WireShark traces show that devices at 10.24.237.12 and 10.231.234.19 both reply with "404 - Not Found" to INVITE request sent to them.

 

Please see screenshot attached of the SIP comms captured in wireshark_IP_10.248.237.12 and wireshark_IP_10_231_234_19 traces.

 

You would need to contact administrators of the Ciscso CUCM s and ask them why the CUCM answered "404 - Not Found" instead of forwarding the call.