Vg And Voip Using Hmp

[kumar343]

Hello Support,

 

We have installed VG 7.0.9, HMP 3.0 SU255, .Net3 on WinXP (service pack -2). PC Details: Intel Celeron® CPU 2.80GHz, IGB RAM. But not able to register VG with VoIP. Please find attached config.xml and other relevant file for your reference and suggest the solution. Also while we run IP Media Server Demo using HMP, we get some error which is attached in file.

 

Thanks,

Kamlesh

 

config-files.zip

[SupportTeam]

When installing VoiceGuide the target platform needs to be correctly selected. ie. you need to select whether you want to install for HMP or for Dialogic cards.

Looks like during this installation the "Dialogic cards" options was chosen (which is an incorrect choice).

 

To fix this please open the VG.INI file and in section [Assembly.Load] change field:

 

ktTel=ktTelDialogicSR60.dll

 

to be:

 

ktTel=ktTelDialogicHMP30.dll

 

and restart the VoiceGuide service.

 

Please post traces as before if you still encounter any problems.

 

If problems relate to system not registering with the SIP server then we will need to see the WireShark trace capturing the IP packets sent during the time the registration was attempted. (WireShark can be downloaded from www.wireshark.org)

[SupportTeam]

We'd also recommend updating your VoiceGuide installation to the latest version (v7.1.0 currently) - available from our WWW Downloads page.

[kumar343]

We'd also recommend updating your VoiceGuide installation to the latest version (v7.1.0 currently) - available from our WWW Downloads page.

 

 

We just changed the setting as per your suggestion and that triggers the SIP registration but it doesn't complete. All relevant log files are attached along with tcpdump.

config-files.zip

[SupportTeam]

ktTel trace shows:

 

170400.078 1948 fn VoIPProvider_AuthenticationAdd(auth_realm=202.89.78.23, auth_identity=, auth_username=16463679782, auth_password=******)

170400.093 1948 fn VoIPProvider_AuthenticationSet

170400.093 1948 TelDriver_VoIPProvider_AuthenticationSet start. adding 1 auth entries.

170400.093 1948 adding to auth parmblk: 202.89.78.23 16463679782 ******

170400.093 1948 gc_SetAuthenticationInfo call. iCountofAuthInfoAdded=1

170400.093 1948 gc_SetAuthenticationInfo success. iCountofAuthInfoAdded=1, iHmp_iptB1=1

170400.093 1948 TelDriver_VoIPProvider_AuthenticationSet end

170400.109 1948 fn VoIPProvider_Register(protocol=SIP, reg_server=202.89.78.23, reg_client=16463679782@202.89.78.23:5060, local_alias=16463679782@202.89.78.23, sH323SupportedPrefixes=)

170400.109 1948 TelDriver_VoIPProvider_Register IP_PROTOCOL_SIP reg=[server:202.89.78.23, client:16463679782@202.89.78.23:5060, alias:16463679782@202.89.78.23, ttl=0, ] iptB1=1

...

170400.265 2612 ev GCEV_SERVICERESP (board device)

170400.296 2612 GCEV_SERVICERESP ResultInfo: gcValue=1286(0x506|GCRV_CCLIBSPECIFIC|event caused by cclib specific failure) gcMsg=[Event caused by call control library specific failure] ccLibId=8 ccLibName=[GC_H3R_LIB] ccValue=[0x151f||] ccMsg=[iPEC_SIPReasonStatus407ProxyAuthenticationRequired] additionalinfo=[]

 

and WireShark show this "407 Proxy Authentication Required" message received from 202.89.78.23:

 

Proxy-Authenticate: Digest realm="SysMaster", nonce="1de17b89987fef13b04cb49c9b8add94", opaque="192e682b85cd687d6fa6b20ddb7cf28b", uri="sip:202.89.78.23"

 

The 'Domain' setting in the Config.xml needs to match the 'realm' asked for by the SIP Server. Right now in your Config.xml no entry for SysMaster is specified, so VoiceGuide cannot find the corresponding authentication entry.

 

In the Config.xml file please change:

 

<VoIP_Registration>

<Display>Test Line</Display>

<Protocol>SIP</Protocol>

<RegServer>202.89.78.23</RegServer>

<RegClient>16463679782@202.89.78.23:5060</RegClient>

<LocalAlias>16463679782@202.89.78.23</LocalAlias>

</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>

<Display>Test Line</Display>

<Domain>202.89.78.23</Domain>

<AuthUsername>16463679782</AuthUsername>

<AuthPassword>deleted</AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

 

to:

 

<VoIP_Registration>

<Display>Test Line</Display>

<Protocol>SIP</Protocol>

<RegServer>202.89.78.23</RegServer>

<RegClient>16463679782@202.89.78.23:5060</RegClient>

<LocalAlias>16463679782@202.89.78.23</LocalAlias>

</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>

<Display>Test Line</Display>

<Domain>SysMaster</Domain>

<AuthUsername>16463679782</AuthUsername>

<AuthPassword>deleted</AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

 

and restart the VoiceGuide service. (change shown in red).

[SupportTeam]

From your traces we can see that you want to place outbound calls.

When placing outbound calls over VoIP you need to specify the IP address to/through which to send the call.

ie. dialed number would look something like this: 9312089343@202.89.78.23

[kumar343]

Thanks for your reply.

 

After making the changes, we are able to register with SIP server. But when we place outgoing call, getting 400 Bbad Request, probably VG is not sending the USER ID in Invite request. Tcpdump is attached for your reference.

 

Thanks.

 

config-files.zip

[SupportTeam]

What make/model is the SIP server used here?

 

Do you have a sample WireShark capture of a successful registration+call sent to this switch?

[kumar343]

This is third party(Sysmaster) soft switch. We are running more than 10k customer making VoIP calls through this soft switch.

 

Successful SIP registration and SIP call log is attached for your reference.

 

Thanks.

successful-sip-logs.zip

[SupportTeam]

Example of successful outbound call had these entries in INVITE message:

 

From: "Rawat, Chetan" <sip:16463678864@202.89.78.23:5060>;tag=00AAB578

Contact: "Rawat, Chetan" <sip:16463678864@202.89.65.39:5060>

 

whereas traces show that HMP/VoiceGuide is sending right now:

 

From: <sip:202.89.65.39:5060>;tag=c042cc0-0-13c4-50022-34b8-21a2abf7-34b8

Contact: <sip:202.89.65.39:5060>

 

to set the 'CallerID' on the outbound calls please add this to the "Options" field when loading new outbound calls into the system:

 

<CallerID>16463678864@202.89.65.39</CallerID>

 

or you may need to try:

 

<CallerID>16463678864@202.89.65.39:5060</CallerID>

 

or

 

<CallerID>sip:16463678864@202.89.65.39:5060</CallerID>

 

check WireShark logs of the outbound calls to confirm whether the From: and Contact: fields are being set correctly.

 

Do you know whether the Sysmaster SIP Switch looks at From: or the Contact: fields to extract the User ID ?

[kumar343]

to set the 'CallerID' on the outbound calls please add this to the "Options" field when loading new outbound calls into the system:

 

<CallerID>16463678864@202.89.65.39</CallerID>

 

we could not locate the 'Option' in attached call loader. Please let us know the exact step to enable this tag.

 

Appreciate your help.

 

Thanks,

Kamlesh

call-loader.zip

[SupportTeam]

Click on the 'Variables' Tab.

The second text box lets you specify the 'Call Options'.

(the first one is for script variables).

 

See screenshots here: http://www.voiceguide.com/vgDialer.htm

 

The Call Options field can also be set when loading calls from XML file or WCF/COM or direct into database etc.

[kumar343]

We have tried following three tags in Call Options but VG is not initiating the call.

 

<CallerID>sip:16463678864@202.89.65.39:5060</CallerID>

<CallerID>16463678864@202.89.65.39:5060</CallerID>

<CallerID>16463678864@202.89.65.39</CallerID>

 

Wireshark log and Call Loader screenshot is attached.

 

Secondly, which tag we can use to change the SIP Session Expiration Value which is 3600 by default.

 

 

 

files.zip

[SupportTeam]

WireShark trace only shows that a (successful) registration was made, but no outgoing calls were sent.

 

The telephone number loader trace (0911_1201_vgDialListLoad.txt) shows that it was only opened (at 12:01:52) but no calls were loaded using the telephone number loader app.

 

The vgEngine trace shows that when VoiceGuide was started (at 11:58:26) it tried to immediately dial some entries still in the database, but the entries in the database had malformed expressions in the telephone number fields.

 

Looks like somehow these expressions were loaded as the telephone numbers to be dialed:

 

<,honeNumber>17322260011@202.89.78.23</,honeNumber>

<CallOptions><CallerID>16463678864@202.89.78.23</CallerID></CallOptions>

aaI16463678864@202.89.78.23/aI/a...

 

Traces do not show when or how were these expressions loaded into the database.

 

Are you trying to load calls direct into the database yourself?

 

The best thing to do now is to ensure that the garbage entries are no longer in the database. Do this:

 

1. Stop VG service

2. Delete file vgDb.vdb3 in VoiceGuide's \data\ subdirectory.

3. Start VG service and start WireShark.

4. Open the telephone number loader app

5. Load one call (setting the call options to <CallerID>16463678864@202.89.65.39</CallerID>

6. Load one call (setting the call options to <CallerID>16463678864@202.89.65.39:5060</CallerID>

7. Load one call (setting the call options to <CallerID>sip:16463678864@202.89.65.39:5060</CallerID>

 

We should then see the outbound calls being made. Please .ZIP up the traces and post them here.

 

 

 

115838.640 6 3 1 state Dialing (auto) <,honeNumber>17322260011@202.89.78.23</,honeNumber>

 

115838.656 6 7 2 state Dialing (auto) <CallOptions><CallerID>16463678864@202.89.78.23</CallerID></CallOptions>

 

115838.812 6 7 2 state Dialing aaI16463678864@202.89.78.23/aI/a...

[kumar343]

Thanks....Calls are working now if we add number manually in Outbound Call Loader. But once we upload the OutDial*.xml file, again we get the bad request message i.e VG is not sending the Caller ID in Invite request. Attached is the OutDial file and wireshark log. Please check if the outdial file format is correct?

 

Also let us know how can we set SIP Session Expiration value to 60 in SIP registration.

 

Kamlesh

log-files.zip

[SupportTeam]

The outdial XML file you uploaded contained this:

 

<?xml version="1.0"?>

<OutDialEntry>

<PhoneNumber>9312089343@202.89.78.23</PhoneNumber>

<PhoneNumber>0017322260011@202.89.78.23</PhoneNumber>

<PhoneNumber>0012012550429@202.89.78.23</PhoneNumber>

<PhoneNumber>0012146149000@202.89.78.23</PhoneNumber>

</OutDialEntry>

 

You did not indicate which one of the CallerID options works on your system, but if it was <CallerID>16463678864@202.89.65.39</CallerID> then the outdial XML file should look more like this:

 

<OutDialEntry>

<PhoneNumber>9312089343@202.89.78.23</PhoneNumber>

<OnAnswerLive>c:\MyScripts\myscriptgoeshere.vgs</OnAnswerLive>

<CallOptions><CallerID>16463678864@202.89.65.39</CallerID></CallOptions>

</OutDialEntry>

<OutDialEntry>

<PhoneNumber>0017322260011@202.89.78.23</PhoneNumber>

<OnAnswerLive>c:\MyScripts\myscriptgoeshere.vgs</OnAnswerLive>

<CallOptions><CallerID>16463678864@202.89.65.39</CallerID></CallOptions>

</OutDialEntry>

<OutDialEntry>

<PhoneNumber>0012012550429@202.89.78.23</PhoneNumber>

<OnAnswerLive>c:\MyScripts\myscriptgoeshere.vgs</OnAnswerLive>

<CallOptions><CallerID>16463678864@202.89.65.39</CallerID></CallOptions>

</OutDialEntry>

<OutDialEntry>

<PhoneNumber>0012146149000@202.89.78.23</PhoneNumber>

<OnAnswerLive>c:\MyScripts\myscriptgoeshere.vgs</OnAnswerLive>

<CallOptions><CallerID>16463678864@202.89.65.39</CallerID></CallOptions>

</OutDialEntry>

 

 

Please see: http://www.voiceguide.com/vghelp/source/html/diallistinto.htm for outdial XML file structure definition.

[kumar343]

Below caller ID option works perfectly fine.

<CallerID>16463678864@202.89.65.39</CallerID>

If we create OutDial with format suggested by you in last reply, we get following error in browser.

<P style="FONT: 13pt/15pt verdana">Only one top level element is allowed in an XML document. Error processing resource 'file:///C:/Documents and Settings/Admi...

<OutDialEntry>-^







If we create OutDial file VG dials only first number in list. Attached is the sample OutDial file.

OutDial_1.xml

[SupportTeam]

The attached file is not in the right format to load multiple outbound call entries.

 

Please follow example supplied. That example shows how multiple calls are loaded from one outdial file.

 

If you wish to be able to view the file in the browser then a top level element can be added for the whole file, like this:

 

<OutDialEntries>

<OutDialEntry>

<PhoneNumber>9312089343@202.89.78.23</PhoneNumber>

<OnAnswerLive>c:\MyScripts\myscriptgoeshere.vgs</OnAnswerLive>

<CallOptions><CallerID>16463678864@202.89.65.39</CallerID></CallOptions>

</OutDialEntry>

<OutDialEntry>

<PhoneNumber>0017322260011@202.89.78.23</PhoneNumber>

<OnAnswerLive>c:\MyScripts\myscriptgoeshere.vgs</OnAnswerLive>

<CallOptions><CallerID>16463678864@202.89.65.39</CallerID></CallOptions>

</OutDialEntry>

<OutDialEntry>

<PhoneNumber>0012012550429@202.89.78.23</PhoneNumber>

<OnAnswerLive>c:\MyScripts\myscriptgoeshere.vgs</OnAnswerLive>

<CallOptions><CallerID>16463678864@202.89.65.39</CallerID></CallOptions>

</OutDialEntry>

<OutDialEntry>

<PhoneNumber>0012146149000@202.89.78.23</PhoneNumber>

<OnAnswerLive>c:\MyScripts\myscriptgoeshere.vgs</OnAnswerLive>

<CallOptions><CallerID>16463678864@202.89.65.39</CallerID></CallOptions>

</OutDialEntry>

</OutDialEntries>

[kumar343]

Thanks...OutDial is successfully loaded and dialled all the numbers after making changes suggested by you.

 

We have created one simple script which will play sound file as soon as call is answered.

 

The problem is sound file is not being played immediately after the call is answered. There is delay of 5-10 seconds. Please suggest the changes to be made for delay removal. Secondly, sound file is being played again and again, we want to play it once only after the call is answered. VGS file is attached for your reference.

 

How many lines are provided in evaluation version? As we can see only 2 lines.

 

 

 

pay-rem.vgs

[SupportTeam]

Thanks for letting us know the outbound calls are now loaded and being made.

 

Regarding the running of outbound scripts and time it takes to do Live Person vs Answering Machine detection please read: http://www.voiceguide.com/vghelp/source/html/detectcallanswer.htm

 

If after reading the above you still have questions then please start a new thread for this question as this is a new topic, and in the new post include the VoiceGuide trace files capturing the outgoing call. Trace files are in VoiceGuide's /log/ subdirectory. Please .ZIP up trace files before posting them.

 

The Evaluation HMP license from Dialogic has 2 lines. That is why on HMP systems you are limited to 2 lines until a HMP license is purchased (contact sales@voiceguide.com for purchasing HMP licenses as well).

VoiceGuide itself supports 30 lines in evaluation mode - allowing you to control up to a full T1 or E1 trunk while still evaluating the software.