Voiceguide Integration With Nortel Cs1000M V6 Using Hmp

[aodecarli]

I'm trying to integrate VoiceGuide running on Dialogic HMP 3.0 with a Nortel CS1000M PBX running V6 using SIP. My questions are:

1) What is the best/easiest option to use in the Nortel CS1000M PBX to have this working?:

a) Static End Point

SIP Third Party

c) SIP Line

d) SIP Trunk

e) SIP Nortel

Which of the methods above require VoiceGuide to register itself with the PBX? Is Authentication needed at all?

2) How do I set Inband Signalling with a Nortel CS1000M PBX?

Thanks in advance.

[SupportTeam]

What is the best/easiest option to use in the Nortel CS1000M PBX to have this working?:

Questions about Nortel configuration should be directed to Nortel support/supplier.

 

Most likely the "a) Static End Point" or " SIP Third Party" would be the easiest setup - you wold just need to indicate the IP address of the VoiceGuide server and the PBX would send out calls over the network to that IP address. VoiceGuide would then answer the SIP calls which arrive.

 

Most likely the "c) SIP Line" option would require for VoiceGuide to register itself with the PBX. Information on how VoiceGuide can register itself with a SIP Server/PBX can be found here: http://www.voiceguide.com/vghelp/source/html/config_voip_register.htm

 

Is Authentication needed at all?

VoiceGuide does not require Authentication to answer or make calls, but will supply Authentication if it is required by the other SIP Switch/PBX.

 

2) How do I set Inband Signalling with a Nortel CS1000M PBX?

Information on how VoiceGuide can be configured to receive and interpret incoming inband signaling can be found here: http://www.voiceguide.com/vghelp/source/html/inbandsignaling.htm

[aodecarli]

The IVR is running VoiceGuide 7 running SIP on HMP 3.0.

We are trying proposed "a) Static End Point" configuration in the Nortel CS1000M PBX.

We try calling the IVR (Extension 4312) from extension 7632 going through the PBX, we see the calls coming in (See "ITsynch_IVR_ScreenShoots.docx" attached), but the status stays as "Waiting for a call…".

 

If we try to call the IVR from SJphone 1.65 (with its default configuration) then everything works fine and the IVR works as intended.

 

Attached please find have:

 

1) System Description: DxDiag_20100927.txt

 

2) Line monitor screenshot: ITsynch_IVR_ScreenShoots.docx

 

3) sys_infoDNA.exe output: sys_inf.htm.htm

 

4) VG Logs: Copy of 0927_2146_vgEngine.txt, Copy of 0927_ktTel.txt, Copy of 0927_ktTts.txt

 

5) VG config files: Config.xml, ConfigLine.xml

 

6) VG scripts and config: AIMS-IVR_Flow.xml, AIMS-IVR_Flow.vgl, Configurations.vgc, AIMS-IVR_Flow.vgs

 

Please advice what may be failing when we make the call thru the PBX, thank you very much.

ITsynch_HMP_VG_Nortel CS1000_ErrorReportFiles_20100927.zip

[SupportTeam]

If we try to call the IVR from SJphone 1.65 (with its default configuration) then everything works fine and the IVR works as intended.

OK, so looks like you now need to figure out how to have the PBX send the call to the VoiceGuide's IP address.

 

 

Did you assign Extension 4312 to route calls to the IP address of the VoiceGuide server? If yes the you should ask the PBX supplier why the PBX is not functioning as intended.

 

Easiest way to check if SIP packets are arriving on the VoiceGuide system from the PBX is to use WireShark (www.wireshark.org) to capture the IP packets coming in from the network.

[aodecarli]

I think my post was not clear as you did not understand it.

When I call through the IVR the call makes it to the IVR (you can see that by looking at “In/Out” column in VG status monitor in "ITsynch_IVR_ScreenShoots.docx” attached), the problem is that VG does not answer the call (you can see that by looking at “Status” column in VG status monitor in "ITsynch_IVR_ScreenShoots.docx” attached).

 

I took the time to produce all the logs and explain the problem in depth, please take a good look at my post again and advice why VG is not answering the call coming through the IVR.

 

Thank you.

 

 

[SupportTeam]

Trace files show that an old version of VG is installed on this system.

Please update your system to this version: [old link removed] and post traces from this version if you still encounter problems.

[aodecarli]

Updated Voice Guide, same result.

We have now sort of isolated the problem, the problem is that VG stops answering the calls right after more than one concurrent call is done through the IVR. Then, if we reset the PBX server, calls start working again. Then stops again after more than one concurrent call is done. The strange thing is that only stops answering the calls that go through the IVR, if we call using SJphone the call go through just as expected.

Find attached related logs.

Best regards,

20100928_Logs.zip

[aodecarli]

Attached also find the wireshark log of multiple calls, at some point the issue start happening as the Nortel PBX gets:

14073 670.384507 10.84.16.4 10.80.201.11 ICMP Destination unreachable (Port unreachable).

Please advice.

 

 

FirstCallFrom the cabin 7632_2 Calls_Busy tone.zip

[aodecarli]

Hello, attached also see the D Channel at the PBX side.

This is now holding off the implementation of the product, please help.

Thanks.

 

wonder27.zip

[SupportTeam]

VoiceGuide traces show about 19 calls arriving into system, with the first 15 being answered OK and the last 4 do not appear to progress.

 

From VoiceGuide's point of view VoiceGuide is responding with an OK to an incoming INVITE request, but then VoiceGuide never sees an "ACK" which it expects to receive to confirm the connection.

 

not sure why we would see the Destination unreachable ICMP, given that a split second before the SIP INVITE arrived OK.

 

Are there any other SIP applications installed on the VoiceGuide/HMP system?

[aodecarli]

No other SIP applications installed on the VoiceGuide/HMP system.

When closely looking at the log in file “FirstCallFrom the cabin 7632 - Hall-Shoreside Calls_Busy tone.zip” sent today we see the following sequence :

 

1) Successful call made from the cabin through the PBX to the IVR:

 

No Time Source Destination Protocol Info

 

168 8.640048 10.84.16.4 10.80.201.11 SIP/SDP Request: INVITE sip:1010;phone-context=cdp.udp@10.80.201.11;user=phone, with session description (application/x-nt-mcdn-frag-hex) (application/x-nt-epid-frag-hex)

 

169 8.646489 10.80.201.11 10.84.16.4 SIP/SDP Status: 200 OK, with session description

 

170 8.648448 10.84.16.4 10.80.201.11 SIP Request: ACK sip:1010;phone-context=cdp.udp@10.80.201.11;user=phone

 

171 8.649061 10.84.16.4 10.80.201.11 SIP Request: OPTIONS sip:1010;phone-context=cdp.udp@10.80.201.11;user=phone

 

172 8.649627 10.80.201.11 10.84.16.4 SIP/SDP Status: 200 OK, with session description

 

3970 49.671088 10.80.201.11 10.84.16.4 SIP Request: BYE sip:7632;phone-context=cdp.udp@10.80.201.11:5060;maddr=10.84.16.4;transport=udp;user=phone

 

 

2) Then, after some calls, we start continuously seeing:

 

No Time Source Destination Protocol Info

 

89411 831.947517 10.84.16.4 10.80.201.11 ICMP Destination unreachable (Port unreachable)

 

and eventually calls don’t go through.

 

 

3) If then we turn off the IVR we still continuously get:

 

No Time Source Destination Protocol Info

 

534 20.309283 10.84.16.4 10.80.201.11 SIP Request: OPTIONS sip:10.80.201.11:5060

 

which does not make sense as there is no ongoing call.

 

 

As we understand it, point 2) above means that the PBX is informing the IVR that the port the IVR is trying to use for establishing the call is unreachable. Point 3) can only indicate a looping condition as we find no other explanation for it.

 

Resuming, our conclusion is that for some reason, after some time of operation the PBX start failing making SIP port 5060 unreachable, it then enters a looping condition which is only terminated when the PBX is rebooted.

 

Please advice if you aggree with this.

 

Could the problem be Cisco's Hot Standby Router Protocol (HSRP) ?

 

 

[SupportTeam]

A different WireShark trace was attached so we cannot view the messages that you are referring to.

 

If calls from PBX are accepted by the IVR again after the PBX is rebooted then this suggests that PBX vendor/supplier would be the better people to contact to resolve this issue. Especially if calls to IVR form other SIP sources (SJPhone sofphone etc) are always fine.

 

If then we turn off the IVR we still continuously get:

These messages could be sent by the HMP drivers. The HMP service is shut down separately.

 

 

Could the problem be Cisco's Hot Standby Router Protocol (HSRP) ?

Would not think so. I think that only way you could implicate HSRP is if you have the gateways failing over at around the same time that problems arise and you also see the ARP requests all of a sudden returning a different MAC address for the same IP. Don't think that this is what's happening here.

 

The HMP driers on your system are not up to date as well.

We recommend 'Service Update 275' be used with the version that you currently have installed, and newer versions are ok as well. Current latest is 'Service Update 286'.

Don't think that would change anything though...

 

Suggest you contact PBX vendor/supplier to resolve the issue with their PBX.

[aodecarli]

Hello,
Thank you very much for the prompt responses, we really appreciate them as the project is now under a lot of pressure since it is on hold due to this issue.
We have now applied HMP latest release and -as you said- we are still getting the same results.
A question we are now been asked for: "Are you aware of if any VG client has ever successfully integrated Voice Guide 7/HMP 3.0 with a Nortel Cs1000M PBX running V6 sw?"
Also, please find next a link to download the logs for “FirstCallFrom the cabin 7632 - Hall-Shoreside Calls_Busy tone.zip” mentioned in my thread below so you can take a good look at them and help us trouble shoot what is going on:

[old link removed]

We really need a hand here, again we appreciate a prompt response.
Thank you very much.

[aodecarli]

I have an update, PBX guys solved an IP address conflict in the PBX signaling service and everything seems to be working fine now, I'll confirm it once we tested it a little bit longer.

Regards.

[SupportTeam]

Thanks for update. Please post confirmation when you are happy that all has been resolved.

 

In relation to previous questions we do not keep track of what PBXs/Firmware versions our customers use VoiceGuide with, but we have many VoiceGuide installations and HMP is a widely used platform that has been around for a few years now.

 

Another thing that we were going to suggest is to have VoiceGuide register itself with PBX instead of setting up a static route in PBX. On most of installations VG would register itself with PBX, like softphones etc. register themselves. More info on setting up VG to self-register is here: http://www.voiceguide.com/vghelp/source/html/config_voip_register.htm - but if you have gotten the static IP route approach working then you should probably just stick with that.

[aodecarli]

Hello,

After 10 days with no issues we can now say things are working stably.

Thanks again for the prompt responses to our questions.<BR style="mso-special-character: line-break"><BR style="mso-special-character: line-break">

 

On another note, we will soon need to do another installation (licenses still to be purchased) but this time integrating with “Cisco Unified Communications Manager version 7.1(3b)SU2”, any experience with it or recommendation on what is the best way to set it up (E.g. Static End Point, etc.)?

 

 

[SupportTeam]

OK, thanks for the update confirming all is OK now.

 

If you have the option to set up the Cisco UCM with a "Static End Point" (or equivalent) option then that is the easiest option to configure.

 

Most clients using Cisco UCM end up setting up VoiceGuide to register itself with the UCM however.

 

More info on setting up VG to self-register is here: http://www.voiceguide.com/vghelp/source/html/config_voip_register.htm