White Paper Or Instructions For Sip Integration

[iTime]

Hello VoiceGuide,

 

We suggested one of our customers an IVR Solution using VoiceGuide 7 + HMP Driver + VoIP, which will be running on SIP trunks supported by Avaya CS1000.

I have the following two questions from them:

 

1. Now the customer is asking if they could obtain an implementation white paper or written instructions on SIP integration between the VoiceGuide System and an Avaya CS1000 on software release 7.5 and Session Manager.

 

2. And Do you know if our customer needs a 3rd party Avaya SIP license along with DSP resources (apart from HMP driver licenses)?

 

Thanks a lot for your help as always.

[SupportTeam]

SIP connectivity is pretty generic.

 

Instructions on configuring VoiceGuide to do SIP registration is here: http://www.voiceguide.com/vghelp/source/html/config_voip_register.htm

 

A simpler configuration would involve just setting up a direct IP trunk on the PBX to point direct to IP address of the machine on which VoiceGuide is installed.

Then VoiceGuide will just answer any calls that arrive at it's IP address.

 

We do not know whether any SIP licenses and/or DSP resources would be required on the Avaya. Those questions should be directed to the Avaya supplier.

[iTime]

Hello VoiceGuide,

 

The customer tried to configure their Avaya SIP trunk and asked me if this works with Voiceguide + HMP on my end.

It is not working and I got the following logs with lots of errors. I could see the Wireshark shows "No service available" error, and HMP log shows "Compile error"

 

Would you be able to see and let me know any findings? I am attaching Config file (configured based on their explanations) and log files.

Any help is appreciated.

 

Thanks.

LogsAndConfig.zip

[SupportTeam]

Can you please post the WireShark trace that captures the VoiceGuide service startup and any attempted calls.

 

Please .ZIP up the WireShark trace before posting.

[iTime]

I am posting the wireshark log file again because my first attempt was not successful. (If you have received my previous post with wireshark log file, please ignore this one.)

 

Thanks again.SIP_Trace.zip

[SupportTeam]

It looks like the HMP/VoiceGuide system is on machine with IP 10.132.182.98, and the Avaya CS1000 is at IP 10.15.5.25. correct?

 

 

Your current VoIP Registration/Authentication is set to:

 

<VoIP_Registrations>

<VoIP_Registration>

<Display>TopekaIVR</Display>

<Protocol>SIP</Protocol>

<RegServer>10.15.5.25</RegServer>

<RegClient>node1@10.15.5.25</RegClient>

<LocalAlias>node1@10.15.5.25</LocalAlias>

<Expires>3600</Expires>

</VoIP_Registration>

</VoIP_Registrations>

 

<VoIP_Authentications>

<VoIP_Authentication>

<Display>TopekaIVR</Display>

<Realm>Topeka.org</Realm>

<Identity/>

<AuthUsername>node1</AuthUsername>

<AuthPassword>node1</AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

 

We would recommend changing the above to:

 

<LocalAlias>node1@10.132.182.98</LocalAlias>

 

instead of:

 

<LocalAlias>node1@10.15.5.25</LocalAlias>

 

and also changing

 

<Realm>Topeka.org</Realm>

 

to:

 

<Realm></Realm>

 

if after these changes you still see that Avaya CS1000 is responding to a REGISTER request with a 503 Service Unavailable response, then you should speak with the Avaya CS1000 administrator as to why the Avaya is advising that service is unavailable.

 

 

 

149 172843.302 5592 fn VoIPProvider_AuthenticationAdd(auth_realm=Topeka.org, auth_identity=, auth_username=node1, auth_password=******)

 

156 172843.313 5592 fn VoIPProvider_Register(protocol=SIP, reg_server=10.15.5.25, reg_client=node1@10.15.5.25, local_alias=node1@10.15.5.25, sH323SupportedPrefixes=)

 

 

REGISTER sip:10.15.5.25;ttl=10 SIP/2.0

From: <sip:node1@10.15.5.25>;tag=e55ed78-0-13c4-50022-1b3529-1eaa95f9-1b3529

To: <sip:node1@10.15.5.25>

Call-ID: e436b18-0-13c4-50022-1b3529-208f68ab-1b3529

CSeq: 1 REGISTER

Via: SIP/2.0/UDP 10.132.182.98:5060;branch=z9hG4bK-1b3529-6a47a83a-60ccf857

Max-Forwards: 70

Supported: replaces

User-Agent: VoiceGuideHMP

Expires: 3600

Contact: <sip:node1@10.15.5.25>

Content-Length: 0

 

 

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 192.168.100.131:5060;branch=z9hG4bK-1b3529-6a47a83a-60ccf857

To: <sip:node1@10.15.5.25>;tag=AvayaDefaultTag

From: <sip:node1@10.15.5.25>;tag=e55ed78-0-13c4-50022-1b3529-1eaa95f9-1b3529

Call-ID: e436b18-0-13c4-50022-1b3529-208f68ab-1b3529

CSeq: 1 REGISTER

Retry-After: 30

Content-Length: 0

[iTime]

(I have just lost my post so I am composing it again. If you have received this message previously then please ignore...)

 

Hi again,

 

Our customer managed to configure their Avaya server to accept registrtion from our VG server. I see "200 OK" message for SIP registration on Wireshark log upon start of VoiceGuide.

However, when I try to call to the Avaya server using the given number, I get busy signals and VoiceGuide does not respond.

I see that Wireshark shows "SIP Invite" signals (when a call arrives) so those two server are definetely talking to each other, but I also found out that the Wireshark displays a message, "Unrecognised SIP header (P-location)" at this moment.

I am not really sure how serious this message means.

 

Would you look into this issue?

I am posting all possible logs including VG logs, VG config file, Wireshark log, and HMP RTF log.

 

FYI, the Avaya SIP server/Session manager is on 10.15.5.25 and talks to our VG server via 10.132.182.98 (Private IP), then the signals are sent to 192.168.100.131, which is displayed on Wireshark log.

 

Thanks a lot for your help!

HMP_VG_ConnectedButBusySignal.zip

[SupportTeam]

Have you tried setting a direct SIP trunk between the Avaya CS1000 (originally Nortel CS1000) and the HMP/VoiceGuide system?

 

This is usually the simplest way of connecting the systems. Avaya would just send calls down the SIP trunk, and most likely it will use messages adhering closer to standard SIP protocol when sending calls over the SIP trunk.

 

With SIP trunks there is usually no need to perform any registration - calls will just be sent over the trunk as they come.

 

Right now the HMP traces show that it has problems parsing the messages from the Avaya.

 

Please set up a SIP trunk connection that will just have the Avaya CS1000 send calls to HMP/VoiceGuide IP address, and post the traces that capture the incoming calls as before.

[iTime]

I will ask our customer if they can set up as requested.

 

In the mean time, would you be able to see this RTF log again? This is re-collection of RTF Trace log for this case, enabling more options by modifying the RTF Config file.

If might explain which part of the message have caused the failure. (I tried to read but could not find any better clues)

 

Thanks again.

 

I will get back to you once I gather the information.

RTFLog_DebugEnabled.zip

[SupportTeam]

Those RTF logs would really need to be referred to Dialogic.

 

Best next step is to set up a basic SIP trunk from the Avaya/Nortel CS1000 and see how the CS1000 presents call INVITES when it is sending them over a generic SIP trunk.

[iTime]

I have spoken to Dialogic support and showed the rftLog + Wireshark log and received the following comment:

 

Wireshark trace shows SIP message size portion as ~2614k.

 

There was a change made at the stack level reducing the size of the buffer to be allocated for incoming sip messages discovered recently. From 4096 to 2048 as part of changes on XMS, and thus that change came in hmp 3.0 as well.

 

The way to address this is the change the following parm in IP_VIRTBOARD structure:

virtBoards[0].sip_mime_mem.size = 4096;

 

It is an internal HMP data structure available via the Dialogic API http://www.dialogic.com/~/media/manuals/docs/globalcall_for_ip_hmp_v11.pdf

If the toolkit you use does not allow you access to this, you'll need to speak with them about how to change it.

 

It sounds like VoiceGuide I am using (I guess it is the most recent one) does not work with the change for HMP 3.0.

Would you be able to verify this?

Thanks.

[SupportTeam]

This version of VoiceGuide sets .sip_mime_mem.size to 4096 as was suggested:

[old link removed]

To update just stop VG service and install over the to of existing installation.

Please post traces as before if still encountering issues.

Have you tried setting up a basic SIP trunk from the Avaya to HMP/VoiceGuide ?

[iTime]

The link does not work. I get "The file does not exist..." error.

Would you give me another link for the file?

 

Have you tried setting up a basic SIP trunk from the Avaya to HMP/VoiceGuide ?

I have asked my customer another test session with their Avaya Technician. I will let you know when It is done.

 

Thank you.

[iTime]

I tried to download ".....3.3_130405.exe" (not "0404.exe") instead and it worked.

So, if you meant the file name above, I have got it and tested it.

 

It seemed to have resolved that problem but encountered another problem. (Thanks for this fix first of all!)

When I make a call now, on VG Status Monitor I see it is showing the name of my script modules and I also see "Invite" was successful on Wireshark as it did not try 4 times or hangs up (so I see the call is made).

However, I do not hear any sound from the phone, or VG does not recognize any button touches: I tried to enter User# and PWD during [PlayWelcomeGetEmpNumber], VG does not seem to recognize any thouch.

 

Maybe RTP is not transferring to the SIP trunk correctly?

I am attaching the trace just in case you could provide some more help for it.

 

thanks again for the fix for the original issue.

VG_7.3.3_InviteOK_NoSound.zip

[SupportTeam]

WireShark trace shows that HMP/VoiceGuide registers with SIP server at 10.15.5.25, and when call arrives the INVITE packet arrives from 10.15.5.25.

 

During the call HMP/VoiceGuide sends RTP packets to 10.15.5.14 (which looks like is the IP address of the phone handset from where the call is made?) but no RTP packets are sent to HMP/VoiceGuide at all from anywhere.

 

So you may want to ask the Avaya admin why is Avaya not sending any RTP packets to HMP/VoiceGuide after HMP/VoiceGuide answered the call...

 

Use this filter to see in Wireshark just the SIP and RTP packets:

 

sip || rtp

 

 

As previously mentioned, we would recommend setting up a basic SIP trunk from the Avaya/Nortel CS1000 and see how the CS1000 presents call INVITES when it is sending them over a generic SIP trunk.

Also; when a SIP trunk is in place there should be no need to perform any registrations. PBX just sends calls over the trunk and HMP/VoiceGuide just answers the calls.

[iTime]

Thanks for your explanation.

I have sent these questions to the Avaya technician and am waiting for his response.

I will keep you posted on this.

 

In the mean time I got the following comment from Dialogic support team.

It may sound interesting to you as well. (Not sure if the errors are caused by VG or the SIP from the Avaya)

 

Also, we see a couple application level failures in RTF which may be unrelated in this case.

First was unsupported target type passed to this call just before sending registration request:

04/05/2013 12:44:09.377 4452 3460 gc APPL gclib <:::: gc_SetUserInfo(target_type:4, target_id:1, duration:0)

04/05/2013 12:44:09.377 4452 3460 gc ERR1 gclib ::::> gc_SetUserInfo(target_type:4, target_id:1, duration:0) - unsupported target type

 

Target_type:4 is GCGT_GCLIB_BOARD. They are not sure what you are trying to set here above. But you should not be using this type.

04/05/2013 12:44:09.377 4452 3460 gc APPL gclib <:::: gc_ReqService(target_type:9, target_id:1, mode:async)

Also they see some unsupported or bad parm passed here:

04/05/2013 12:44:24.459 4452 3164 gc APPL gclib <:::: gc_GetCallInfo(crn:0x8000001h, info_id:16)

04/05/2013 12:44:24.459 4452 3164 gc_h3r SH_MGR DEBG sm_data.cpp:179 ! 1 ! >> GetCallInfo...with info_id: 16

04/05/2013 12:44:24.459 4452 3164 gc_h3r ERR1 sm_data.cpp:307 ! 1 ! << GetCallInfo: bad_param / un_supported info_id = 0x10

04/05/2013 12:44:24.459 4452 3164 gc ERR1 gclib ::::> gc_GetCallInfo(crn:0x8000001h, info_id:16) - returns;-1

This would be expected to be either origination / destination addresses or IP call id which is supported for SIP, but nothing maps to 16. So, not sure what you are expecting to pass here.

We suggest you also correct this issue within your application.

[SupportTeam]

The SetUserInfo and GetCallInfo calls mentioned would not be affecting why Avaya is not sending any RTP packets to HMP/VoiceGuide after HMP/VoiceGuide answered the call.

The version below does not make the SetUserInfo and GetCallInfo calls, but do not believe that you will find Avaya behaving differently when those Dialogic HMP driver level internal calls are removed.

[old link removed]

As mentioned before, trying to register HMP/VoiceGudie as an extension is probably not the right approach.
The intention is for this system to handle 24 simultaneous calls.
This is more then what the PBX would be usually simultaneously connecting to just an extension - so this is probably not the most suitable approach.
The most suitable/expected approach to interconnect for 24 simultaneous calls would be by using a SIP trunk.

[iTime]

Thanks for another update.

 

do not believe that you will find Avaya behaving differently when those Dialogic HMP driver level internal calls are removed

 

You are right. I did not think the reason RTP was not coming back is caused by VG. (I just forwarded the message as suggested by Dialogic, hoping that it could be helpful for you.)

I strongly believe the cause is a network config issue and our customer is still working on this on their end.

 

I will keep you posted how it goes.

 

Again thank you.

[iTime]

Hello again,

 

Since our customers are not coming back to us with this blocked IP for RTP, we are trying to test our VoiceGuide script with another SIP trunk provider, Broadvox.

 

I would like to ask you if my Registration information is correct for this service provider.

They said:

BTN/DID Username: xxxx2053

Password: not used

SIP Server Hostname DNS A/SRV: ld01-06.fs.broadvox.net

Static Receive-From, Send-To IP 208.88.116.228

Open the following firewall Ports 5050, 5060, 5061, 1024-65535

Allow these IPs 208.93.224.224/28 208.93.226.208/28 208.93.227.208/28

 

While we are configuring the firewall for this, would you be able to verify that I added <Registration> correctly?

Thank you very much for your help again.

 

<VoIP_Registrations>

 

<VoIP_Registration>

<Display>TopekaIVR</Display>

<Protocol>SIP</Protocol>

<RegServer>208.88.116.228</RegServer>

<RegClient>xxxx2053@208.88.116.228</RegClient>

<LocalAlias>xxxx2053@208.88.116.228</LocalAlias>

<Expires>3600</Expires>

</VoIP_Registration>

 

</VoIP_Registrations>

 

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display>Broadvox</Display>

<Realm>ld01-06.fs.broadvox.net</Realm>

<Identity></Identity>

<AuthUsername>xxxx2053</AuthUsername>

<AuthPassword></AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

[SupportTeam]

Please do not use the Realm entry. ie. change:

 

<Realm>ld01-06.fs.broadvox.net</Realm>

 

to:

 

 

<Realm></Realm>

 

 

After you have set up the firewall the please post the WireShark and ktTel traces that capture system startup and registration, and an incoming call.

 

Please .ZIP up traces before posting.

[iTime]

Thanks for the instrunction.

 

I am having the "Registration" problem with this SIP provider too.

Our hosting company has verified the firewall has been configured as requested and I changed the VG config file not to use <realm> as instructed.

However, Wiershark shows VG keeps trying to "REGISTER" but does not even get any response. (Not even an error message like "authentication failure.")

 

What I noticied from Wireshark information is that it shows our VPN IP address (10.132...) as "VIA" in SIP packet. We provided our public IP address (74.205....) to this provider for this testing and I believe what they see should be this public IP unless I try to contact an IP address in VPN tunnel...

 

I am not sure if what I see in "Via" section is what they also see or there is some other reason why I do not get any response from the other end.

If you could give me an advice, that will be great.

 

Thanks again as always.

SIP_Broadvox.zip

[SupportTeam]

WireShark shows the IP UDP packets are being sent to IP address: 208.88.116.228

 

Can you ask the operator of SIP server that sits at IP address 208.88.116.228 whether they are seeing those packets?

 

If they are seeing them then are they responding to them?

 

You can also ask the ask the Firewall administrator to check whether they are forwarding the IP UDP packets to 208.88.116.228 and if they are seeing any responses from208.88.116.228.

 

(and if they are seeing responses from 208.88.116.228 then are they forwarding them to your server at IP 192.168.100.131)

[iTime]

Please ignore my previous post.

The reason it was not doing anything was because I used incorrect account information.

I was able to "Register" after I had provided the correct authentication information.

Please see the attached information.

 

Now the problem is that I do not hear TTS when it is expected. I see from VG Status monitor describing, "tts generate start..." then it does not proceed and I hear no sound.

I also realized that there is no TTS related log generated (even if I set "ktTts=10" in vg.ini file)

I am using ATT Audrey voice and OS is Windows 2008 64 bit.

 

Thanks for helping me make the progress to this far!

SIP_Broadvox_working.zip

[SupportTeam]

Thanks for letting us know the registration is working now.

 

Regarding TTS:

 

ktTel trace shows:

 

020 182804.442 4740 WARN CreateTTSPort returned NULL. SAPI TTS not enabled.

 

You may want to re-install AT&T SAPI tools.

 

If you continue to have issues with TTS please start a new topic on TTS issue.