Voiceguide V7+HMP : VoIP Outbound

[Guest BlackBird]

Hi,

I Installed HMP 3.0 (with the Dialogic 4 port eval license) and VG v7 (v7.0.6 - configured for HMP) on a Windows 2003 SP1 Server for an outbound call system.

 

After configuring the VG config.xml file to use my sip provider, I confirmed that both services (HMP and VG v7) are running without reported problems. Here is what I noticed:

• I created a simple .vgs script, which I reference in the Outbound Call Loader (OCL)
• After loading the phone number what I noticed is that the phone call is not send/completed.
• I noticed that the VG Line Status Monitor does reference that the call was attempted, however, I never received it.
• To confirm that my sip account is working, as a separate exercise, I configured x-lite with my account information and called my pstn line successfully.

I attached the log file (,zip) for your review, could you please advice me on the resolution in order to make outbound calls via VG v7 w/HMP?

Please let me know if you would like me to send additional information..

 

Regards,

Ralph

log.zip

[SupportTeam]

First it needs to be established if the system is answering incoming calls.

 

1. Have you tried placing calls into system by just directly dialing its IP address from softphone or VoIP handset on the same LAN?

 

2. Have you tried placing calls into system by dialing in through the VoIP service provider?

 

 

When placing outbound calls with VoiceGuide you need to specify the IP address of the VoIP destination node or of the VoIP gateway to use.

 

This usually results in the phone number being specified something like this:

 

17877916831@A.B.C.D

 

Where A.B.C.D is the IP address of the VoIP gateway to use.

 

 

Also note that in evaluation mode you are time limited and number of outbound calls is limited. When you see messages like this in log:

 

140819.312 10 3 WARN Evaluation time expired. VoiceGuide needs to be restarted to continue.

 

it's time to restart the system

[Guest BlackBird]

My main goal is to develop a system for outbound call only. I do not plan to have the system answer inbound calls, thus, I have not tested to see if the system will answer calls.

 

I have successfully made outbound calls from the system via my softphone (x-lite, using sipdiscount.com) to my pstn line and to my cellphone, without any problems and without the need to append an IP address ad the end of the numbers I am calling (as suggested on the previous response)

 

With regards to your comment;

"When placing outbound calls with VoiceGuide you need to specify the IP address of the VoIP destination node or of the VoIP gateway to use."

 

I am not clear about your suggestion here, that is, from the info below the SIP proxy represents the resolved name of the SIP server via which outbound calls are to be routed through, the settings for my SIP provider are as follows:

 

SIP/Proxy registrar : sip1.sipdiscount.com

Domain/Realm (optional): sipdiscount.com

STUN server : stun.sipdiscount.com

 

As you can see from the above settings the are no IP addresses provided, the above settings are specified in the VG config.xml, I would suggest (perhaps a design request ? ) that VG should have the logic to place the calls using the loaded phone numbers and using the settings from the config.xml, else, I don't understand the value for config.xml file.

 

I can obtain the IP address by pinging the sip proxy server, if I understand your suggestion, in VG V7 with HMP when uploading the phone number, rather than stating 17877913232 it should be 17877913232@<sip proxy server IP address> , correct?

 

Regards,

Ralph

[SupportTeam]

The Config.xml entries are used to register the system with SIP provides so that they will send calls to the system and are used in authorization of outgoing calls.

 

It allows for multiple registrations with multiple SIP providers, and letting user specify carrier on a per call basis is usually preferred as well, hence automatic selection of which SIP provider to use for outbound calls is not made.

 

The HMP layer requires the IP address. The HMP layer does not resolve a name like sip1.sipdiscount.com to an IP address. You must explicitly specify the IP address.

 

Multi line systems are not usually deployed over Cable/ADSL, because this setup does not guarantee bandwidth and timely packet delivery.

 

Multiline (4 or more lines) VoIP systems are usually ran over a controlled (local/internal enterprise) network, and in such cases the IP addresses are known and fixed, guess that is why there is no real rush from Dialogic to make HMP resolve the domain names like sip1.sipdiscount.com to an IP address before doing the dial. VoiceGuide could do this before placing each call though HMP, but this is not high on requirements because setups which try to run multiple VoIP connections directly with SIP providers over Cable/ADSL don’t really work well in practice anyway, and if you want to try then you can still place call by specifying the IP address explicitly.

 

Recommend you for now you place calls using the IP address of sip1.sipdiscount.com - which as of last ping test is 194.120.0.198

 

So try specifying your outgoing number as 17877913232@194.120.0.198

 

You may want to contact sipdiscount.com to confirm what IP address you should be using.

 

I'd still try placing calls into the system to verify it's correct operation.

[Guest BlackBird]

Thanks for the clarification about the use and importance of the config.xml.

 

I noticed that the sipdiscount server was timing out with to much frequency, so I switch the sip provider to voipbusters. I tested the new account by making outbound sip calls via x-lite successfully.

 

After updating the config.xml with the new sip provider credentials, I attempted your suggestion about appending the sip provider ip address to the telephone number, for example; 17877916831@208.69.32.132

 

Once the phone entry was loaded, via the VG Line Status app I noticed that the system attempted to dial out, but the call got stock on the queue, exceeded the "Call Length" limit and failed to reach its destination (phone did not ring).

 

As an additional test, to by pass the Outbound Call Loader, I attempted to place the call via a vbs script (included in the attachment) the vbs file calls the "vg.Dialer_MakeCall" function. Either attempt (Out bound loader or the vbs script) yielded the same results, the call length was exceeded during the attempts and the call was never made.

 

To expedite the investigation, I attached a fresh batch of logs, including the vbs script mentioned above and a WireShark trace.

 

Regarding your feedback on these point:

 

"Multi line systems are not usually deployed over Cable/ADSL, because this setup does not guarantee bandwidth and timely packet delivery."

While their might be concerns about packet/b issues during the actual communication/message delivery, via the network we are using we have been able to place multiple concurrent calls via a softphone without issues. However, at the moment I will be a happy camper if we are able to dial out using VG v7 and HMP, thus actually ringing/calling the number being dialed.

 

 

"Multiline (4 or more lines) VoIP systems are usually ran over a controlled (local/internal enterprise) network"

The system we are using is in a controlled business network topology, albeit a small business. Could you please define "a controlled (local/internal enterprise) network"? Do you mean that in order to evaluate VG v7 with HMP I will need a PRI or T3 pipe?

 

 

regards,

Ralph

log.zip

[SupportTeam]

The attached WireShark trace does not show any outgoing SIP calls to 17874782634@208.69.32.132

Was the trace running when the outgoing call was attempted?

 

Have you tried making calls into to the system from another computer on same LAN? (or from a VoIP handset)

 

It's best to verify operation first to confirm the basics are working before trying to get the system working with an external SIP provider.

 

Could you please define "a controlled (local/internal enterprise) network"?

A network which you can configure in such a way that you can be guaranteed that no VoIP packets will be dropped or delayed.

 

[Guest BlackBird]

The attached WireShark trace does not show any outgoing SIP calls to 17874782634@208.69.32.132

Was the trace running when the outgoing call was attempted?

 

Have you tried making calls into to the system from another computer on same LAN? (or from a VoIP handset)

 

It's best to verify operation first to confirm the basics are working before trying to get the system working with an external SIP provider.

 

Could you please define "a controlled (local/internal enterprise) network"?

A network which you can configure in such a way that you can be guaranteed that no VoIP packets will be dropped or delayed.

 

"Was the trace running when the outgoing call was attempted?"

Yes, it was. This time I made sure to initiate WireShark before starting VG v7 in order collect more verbose info. The following are the steps I am taking to capture the WS logs (please let me know if I missed anything):

 

1. Start WireShark

2. Go to 'Capture' menu and select the 'Interfaces' menu option.

3. Select your network interface (press the corresponding Start button)

4. In the Filter window type in these 3 letters: sip

5. Start VoiceGuide.

6. You should now see the SIP registration messages appear in the WireShark trace window.

7. You can save the captured trace and post it here or email it to us to assist in debugging.

 

I opened the new (attached) WS log file, and noticed entries with the following "sip.Request-Line == "INVITE sip:17874782634@208.69.32.132 SIP/2.0" " thus confirming it captured a sip transaction.

 

 

"Have you tried making calls into to the system from another computer on same LAN? (or from a VoIP handset)"

Yes I, called the IVR system using the SIP protocol (I used SJ phone to make the IP call). VG answered the call without any problems, I was able to complete the credit card demo successfully.

 

I attached the VG and WS logs with my latest failed attempt to perform an outbound call, please advice.

 

regards,

Raf

 

 

 

 

 

 

 

 

 

 

log_6_30_2008.zip

[SupportTeam]

Yes I, called the IVR system using the SIP protocol (I used SJ phone to make the IP call). VG answered the call without any problems, I was able to complete the credit card demo successfully.

As a next test can you just try placing a call from VoiceGuide to the system running the SJPhone - to confirm that the outgoing calls from the VG/HMP system are working in general.

 

The WireShark trace shows the SIP registration is not completing properly. Please post the <VoIP_Lines> section from your Config.xml file (blank out the password).

 

The SIP registration needs to be resolved first before you will be able to place calls out through the SIP provider.

[Guest BlackBird]

Yes I, called the IVR system using the SIP protocol (I used SJ phone to make the IP call). VG answered the call without any problems, I was able to complete the credit card demo successfully.

As a next test can you just try placing a call from VoiceGuide to the system running the SJPhone - to confirm that the outgoing calls from the VG/HMP system are working in general.

 

The WireShark trace shows the SIP registration is not completing properly. Please post the <VoIP_Lines> section from your Config.xml file (blank out the password).

 

The SIP registration needs to be resolved first before you will be able to place calls out through the SIP provider.

 

 

I will perform the test (placing the call from VG to SJP) as soon as I get access to both the systems. Could you please confirm if the phone number syntax when making the call from VG into SJPhone will be similar to the syntax used when making the call from the system with SJPhone into the system with VG? that is, I will need to load to VG the number in the following syntax, sip:00.00.00.00 (of course with the real ip), correct?

 

Per your request, the following is the SIP registration section currently found in the config.xml file:

 

<VoIP_Registrations>

 

<VoIP_Registration>

<Display>rafter2008</Display>

<Protocol>SIP</Protocol>

<RegServer1>sip.voipbuster.com</RegServer1>

<RegServer>sip.voipbuster.com</RegServer>

<RegClient>rafter2008</RegClient>

<LocalAlias>administrator</LocalAlias>

</VoIP_Registration>

 

</VoIP_Registrations>

 

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display>voipbuster</Display>

<Domain>sip.voipbuster.com</Domain>

<Username>rafter2008</Username>

<AuthUsername>rafter2008</AuthUsername>

<AuthPassword>Removed</AuthPassword>

</VoIP_Authentication>

 

</VoIP_Authentications>

 

[SupportTeam]

The WireShark trace shows that the realm in the Authentication request is rafter2008:

 

WWW-Authenticate: Digest realm="rafter2008",nonce="3326179735",algorithm=MD5

 

So in the VoIP_Authentications section you should change

 

<Domain>sip.voipbuster.com</Domain>

 

to:

 

<Domain>rafter2008</Domain>

 

And restart VG and see if the registration then successful (need to run WireShark again to confirm this)

 

 

You may want to contact voipbuster.com to confirm what IP address you should be using for the outgoing calls after the registration is successful.

[Guest BlackBird]

I changed the domain as suggested, and stopped and restarted VG v7, the outbound calls failed still.

I attached a newer version of the WS log as well as the config.xml I am currently using.

 

I will send email to voipbuster.com with regards to to the IP address for outgoing calls.

log_v2.zip

[SupportTeam]

Had a closer look at your VoIP registrations. Some of the other fields that you hnve there are not as per the recommended template. Try the below:

 

<VoIP_Registrations>

<VoIP_Registration>

<Display>FWD line</Display>

<Protocol>SIP</Protocol>

<RegServer>sip.voipbuster.com</RegServer>

<RegClient>rafter2008@sip.voipbuster.com</RegClient>

<LocalAlias>rafter2008@10.1.1.9</LocalAlias>

</VoIP_Registration>

</VoIP_Registrations>

 

<VoIP_Authentications>

<VoIP_Authentication>

<Display>FWD</Display>

<Domain>sip.voipbuster.com</Domain>

<Username>sip:rafter2008@sip.voipbuster.com</Username>

<AuthUsername>rafter2008</AuthUsername>

<AuthPassword>asd123</AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

 

(replace asd123 with your own password)

 

Run the WireShark trace to capture the registration as before and please post it here.

 

If the above does not register then try changing:

 

<Domain>sip.voipbuster.com</Domain>

 

to:

 

<Domain>rafter2008</Domain>

 

and post WireShark traces of this as well.

[Guest BlackBird]

I performed the changes suggested above and attached the WS logs as requested, unfortunately the outbound calls failed to be executed.

 

After a bit of research, I think I found the root of the cause for the Authentication/Registration error, via x-lite and SJPhone I tried placing a call to the target phone number, and received a verbal error because it seems that I exceeded the amount of calls allowed under the free trial program.

 

I would like to subscribe to a VG tested Voip service provider to make a heavy volume of outbound calls. With that in mind I looked into the following VG help link http://www.voiceguide.com/vghelp/source/ht...ip_register.htm

 

The links mentions the following voip providers:

Free World Dialup (www.freeworlddialup.com)

iptel.org (www.iptel.org)

voxalot (www.voxalot.com)

CallCentric (www.callcentric.com)

BroadSoft BroadWorks platform (www.broadsoft.com) : iinet.net

 

From the above list, is there a voip provider which stands out in terms of ease to configure for VG and quality of voip service?

 

Beyond the Voip providers listed above, Are there any other VOIP service providers which are known to work well and are easy to configure with VG v7?

 

Would Skype work with VG v7?

 

Regards,

Raf

WireShark_Trace_712008_domain_sip.voicebuster.com.zip

WireShark_Trace_712008_domain_rafter2008.zip

[SupportTeam]

The WireShark_Trace_712008_domain_sip.voicebuster.com.zip trace shows that the SIP registration was successful.

 

Maybe the outgoing calls were not forwarded due to the "exceeded the amount of calls allowed under the free trial program" issue or maybe you should be using 194.120.0.198 in your outbound calls, this is what WS shows is used during registration. To hear the "free calls exceeded" message would only be heard by the user of the outgoing call connected to voicebuster.com in the first place.

 

Finding a VoIP provider is really up to you, we do not put forward VoIP provider recommendations. The examples we have are for some of the larger and better known ones. The best VoIP provider for you is the one to which you can get best connectivity. How many simultaneous calls do you want the system to be ultimately making?

 

Would Skype work with VG v7?

No.

[Guest BlackBird]

"The best VoIP provider for you is the one to which you can get best connectivity. How many simultaneous calls do you want the system to be ultimately making?"

 

Please expand - what is your definition of ""get the best connectivity"? I ask because I get good connectivity and voice quality from voicebuster when using a softphone, yet getting voiceuster and VG to work together proves that the selection criteria goes beyond "get the best connectivity"

 

 

Regarding the following "you should be using 194.120.0.198 in your outbound calls"

When using the ip mentioned above, when making an outbound call, the call lenght = 0, there is no duration at all

please see the attached WS trace, while using the above ip.

WS_Trace_1__722008.zip

[SupportTeam]

Please expand - what is your definition of ""get the best connectivity"?

Connection which has least VoIP packets dropped or delayed.

 

When using the ip mentioned above, when making an outbound call, the call lenght = 0, there is no duration at all

But we see the SIP server respond to the INVITE message at least. With a 401-Unauthorised message, but at least you have successfully tried to make the call. If you did not "exceeded the amount of calls allowed under the free trial program" I think the outgoing call would have been succeeded.

[Guest BlackBird]

Sorry for not responding sooner...

 

In an attempt to simplify things, I went ahead an opened an account with CallCentric and moded the config.xml file following the CallCentric params used in the VG Voip registration sample. Per the WS trace (attached) it seems that the binding was successful, VG was not able to complete the outbound call. I am attaching a fresh copy of the WS log and the latest config.xml file. In the config.xml I used the sip login provided by CallCentric instead of the web login.

 

Could you please clarify for me, is the "LocalAlias" param in the config.xml required? I ask because, per the WS trace file it seems that the argument is generating an error, but I am not sure if the error would prevent the outbound call.

 

Please advise.

 

Regards,

Raf

 

wireshark_Trace_1_772008.zip

[SupportTeam]

Trace shows that the SIP registration completed OK, but when you tried to place the outbound call the CallCentric server responded with: "403 User does not exist".

 

Can you attempt an outbound call from your system though CallCentric using a softphone of your choice (eg: SJPhone) an capture the WireShark trace to compare with the WireShark trace from VoiceGuide’s call. (you will of course need to configure SJPhone to register itself with CallCentric).

[Guest BlackBird]

I took you suggestion and configured Xlite to call via CallCentric, it worked (See attached WS trace log)

 

I attempted the following settings for the outbound call "17877916831@callcentric.com" as shown in the xlite WS log. But it did not work in VG, I also tried "17877916831@204.11.192.34" and "17877916831@204.11.192.23" without success.

 

Part of the problem I am seen is that since the following does not work "17877916831@callcentric.com", then which should be the ip appended to the end of the telephone number to be called?

wireshark_Trace_Xlite.zip

[Guest BlackBird]

Hi,

After further investigation about that failure to make the outbound calls I noticed that the error "403 user does not exist" is being generated by the number I am attempting to call, that is, when I try to call either of these 17877916831@callcentric.com, 17877916831@204.11.192.34, or 17877916831@204.11.192.23

the call fails because VG its trying to use the above entries to authenticate my account, rather than placing the call, please let me know if you would like me to expand further in my theory...

[SupportTeam]

I noticed that the error "403 user does not exist" is being generated by the number I am attempting to call,

How did you determine this? Do you have traces taken at the end recipients VoIP point?

 

I’d expect that to be response from the VoIP provider itself.

[Guest BlackBird]

I sure do, the following is the response from the Callcentric support team:

 

Hello,

 

To address your concern, you did not need to upload a copy of your wireshark pcap, as we were able to capture your outbound call attempt from our own SIP Trace, which is as follows:

 

21:30:40 Rx udp:70.45.92.102:5060

INVITE sip:17877010935@callcentric.com SIP/2.0

From: <sip:10.0.0.61:5060>;tag=c59ee70-0-13c4-17ce1-180c649a-17ce1

To: <sip:17877010935@callcentric.com>

Call-ID: c5aa308-0-13c4-17ce1-346b93de-17ce1@10.0.0.61

CSeq: 1 INVITE

Via: SIP/2.0/UDP 10.0.0.61:5060;branch=z9hG4bK-17ce1-5cfcfeb-ded00e4

Max-Forwards: 70

Supported: replaces

Contact: <sip:10.0.0.61:5060>

Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, REFER, NOTIFY

Allow-Events: refer

Content-Type: application/SDP

Content-Length: 212

 

v=0

o=Intel_IPCCLib 207266568 207266569 IN IP4 127.0.0.1

s=Intel_SIP_CCLLIB

i=session information

c=IN IP4 127.0.0.1

t=0 0

m=audio 49152 RTP/AVP 8 0 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

--------------------------------------------------------------------------------

21:30:40 Tx udp:70.45.92.102:5060

SIP/2.0 403 User does not exist

v: SIP/2.0/UDP 10.0.0.61:5060;branch=z9hG4bK-17ce1-5cfcfeb-ded00e4;rport=5060;received=70.45.92.102

 

f: <sip:10.0.0.61:5060>;tag=c59ee70-0-13c4-17ce1-180c649a-17ce1

t: <sip:17877010935@callcentric.com>

i: c5aa308-0-13c4-17ce1-346b93de-17ce1@10.0.0.61

CSeq: 1 INVITE

l: 0

 

From the above mentioned, this is clearly an issue relating to your configurations of your VoiceGuide.

 

Also, during my own test I noticed that the error "403 user does not exist" goes away if I just provide the outbound number by itself, for example: 17877916831 instead of 17877916831@callcentric.com or 17877916831@204.11.192.23 or any other phone@ip combination, the assumption on my part is that the outbound number format is part of the problem.

 

I noticed that Callcentric is listed in the VG VOIP registration samples, I assume that VG V7 VOIP was tested using Callcentric to place outbound calls to pstn cliens. If my assumption is correct, then what was the outbound number format used during the test?

 

Based on WS traces it seems that the VG registration is occurring correctly, for your review/reference below I am providing the registration metadata found in the config.xml please let me know if you see an unusual/incorrect parameter.

 

 

<VoIP_Registrations>

<VoIP_Registration>

<Display>CallCentric</Display>

<Protocol>SIP</Protocol>

<RegServer>callcentric.com</RegServer>

<RegClient>17772306106@callcentric.com</RegClient>

<LocalAlias>17772306106@70.45.92.102</LocalAlias>

</VoIP_Registration>

 

</VoIP_Registrations>

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display>CallCentric</Display>

<Domain>callcentric.com</Domain>

<Username>sip:17772306106@callcentric.com</Username>

<AuthUsername>17772306106</AuthUsername>

<AuthPassword>REMOVED</AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

 

 

 

 

 

 

 

 

 

[Guest BlackBird]

I wanted to clarify the following statement:

 

"Also, during my own test I noticed that the error "403 user does not exist" goes away if I just provide the outbound number by itself, for example: 17877916831 instead of 17877916831@callcentric.com or 17877916831@204.11.192.23 or any other phone@ip combination, the assumption on my part is that the outbound number format is part of the problem."

 

 

Despite either outbound phone number format; 17877916831 or 17877916831@callcentric.com VG is not able to complete the outbound call. The latest WS trace contains the transaction verbiage when attempting the outbound call in either format.

 

Regards,

Raf

wireshark_Trace_1_7102008.zip

[Guest BlackBird]

Hi team,

Based on the recent information I provided on my last two post (post #22 & post #23)

 

Could you please advice me on whether the problem I encountered with VG v7 for VOIP, lies within my configuration? or, is the issue within the VG code?

 

Please let me know if you need additional information.

 

Note post #22 contains the sip trace your requested from the Voip provider.

 

[Guest BlackBird]

Hello Support Team,

The bug (?) raised in this post remains unresolved, could you please provide us with a resolution?

 

Thanks!

[SupportTeam]

The "403 user does not exist" message comes from CallCentric, so you would really need to ask them to explain why are they sending this message back.

 

As mentioned previously in post #4 of this thread the VoIP deployment scenarios for which HMP was designed are ones where the VoIP switch or media gateway that HMP communicates with are on the same LAN.

Having the VoIP switch or media gateway under your control would also allow you to change the setting in them to make them work properly with HMP.

 

Many sites use Asterisk (or similar) box to perform VoIP routing, with HMP connecting to Asterisk and then Asterisk relaying the calls. Maybe this is another setup that you may want to look into.

 

Overall, you need to have good knowledge of IP routing and VoIP protocol to roll out a reliably working multiline VoIP system. If you are unable to debug VoIP protocols then we'd recommend looking at using T1 ISDN or E1 ISDN systems.

[SupportTeam]

This may be relevant here as well. From http://www.voiceguide.com/vghelp/source/html/dial_voip.htm :

 

When making an outgoing call on a VoIP line it is usually necessary to specify the CallerID to be used on the outgoing call. This advises the switch relaying the VoIP call as to which account/subscriber is making the call.

The CallerID on outgoing calls can be specified using the <CallerID> tag on the Options field when loading the outgoing call. For example, to place a call though a FreeSWITCH system which is installed on a server with IP of 10.1.1.11, and with which the user/extension 1010 has been registered by VoiceGuide, the following entry would need to be placed in the Options field:

<CallerID>1010@10.1.1.11</CallerID>

This approach allows the use of multiple accounts when placing outgoing calls, with the account to be used for a particular call specified at call time.