I'm working on analog integration and its giving me some issues. DTMF tones seem to be cut off and I dont think the Seiemns is sending all the DTMF signaling that it should. This is a dual integration and the call manager part is working fine. Unity is answering calls from the Siemens side and voice quailty seems good and we can light the MWIs.
We have a inline monitor between the dialogic/siemens connection so that we can monitor the connection.
When a call is forwarded from a siemens phone (RNA) to the Unity hunt group ( on the Siemens), unity picks up. The seimens sends the 4 digit extension of the phone that was busy (extension 1935) the inline monitor detects all 4 digits (and we can hear them as well) but unity only catches the last 2 digits . (35) I'm using the Integration Monitor on the unity server to determine what Unity sees.
So 2 questions:
Why is the DTMF getting "cut off" and how do I troubleshoot it.
#2 Shouldnt the siemens send the calling party number with the call as well ? If it doesnt send that when a subscriber calls from their own phone they wont get the "enter you password" prompt and instead will need to hit "*" to start the login process.
Unity 4.0.3, Call manager 3.3.3 SR4a, Siemens 9751 running 9006.5 software. Dialogics D120JCT
We're using the dialogic boards in the unity server to provide the analog integration beteeen Unity and the Siemens. I used this docuemnt as a guide for the integration
I've seen Siemens installs with similar symptoms that were resolved using this...
For your second question, I believe the Siemens only sends the Called number for forwarded calls. On direct calls to voicemail, it does not send any digits at all. Thus, the Easy Message Access feature will not be available. It may be possible to spoof this feature by creating speed dials for your users. Users would need a speed dial key that would dial Unity, pause, dial *, then dial their extension followed by #. This should get them right to the "enter your password" prompt.
Interesting. I seemed to come to the same conclusion while working the issue. Only the called number gets sent across on forwarded calls.
Why does cisco think the calling gparty number will be sent in the integration doc? Its really frustrating to work on and the customer expectation has been set from the cisco sales team it would work that way. Arahhhghghh
I had the PBX tech change the off-hook delay on his end from 100ms to 1000ms. I'll do the change on the unity end as well just be safe.
The Seimens tech didnt seem to think his PBX would send calling party information to a "external voicemai" via DTMF either.
Yep, looks like there's some inconstancy in the integration guides. You'll note though that in the Integration Features section only Forward to Personal Greeting and Message Waiting Indicator are listed. There is no Easy Message Access or Identified Subscriber Messaging, both of which would require Calling party number.
I pointed out the inconstancy to our pubs guys and they say it'll be revised and updated on Cisco.com sometime tomorrow if not by the end of day today.
Of course, if you or your PBX tech do find a way to get both Called and Calling parties out of the Siemens, let me know. I'm sure we'll gladly update the docs again. ;)
For the Siemens Hicom 300E and 300H series it is definately possible to get it to send both the caller and calling ID's. This is programable in the PBX - look up the ZAND-VMI command for more details.
Thanks for the tip. I'll mention it to our Siemens PBX support person and see what they think.
Is the Hicom 300E and 300H the same as the 9751 (9006.5 software) PBX that we are using? I know somewhere along the line Siemens bought Rolm and I dont know when the products started to become different.
Thanks again !!
We still are not able to make the siemens send calling number information. So every VM appears to come from "unknown caller" I've verified with an inline device that the siemens simply isnt sending calling party information.
How is ur MWI setup to work?
How do the users access voicemail from their phone?
Do they press a button which asks their password like in a IP Phone?
Is the MWI button and the VM access button same?
Appreciate your feedback.
I am having the same problem..
I have the Siemens 300E with Unity 4.03, the problem is exactly the same.It doesn't look like the Siemens is sending the DTMF's digits to unity.
Whenever I dial the unity pilot number, it takes me to generic greeting...
Any thoughts and solution to this issue is greatly appreciated.
Did you get this working? If yes, Please let me know the solution.
I greatly appreciate your help..
This is the workaround we have tested.
This install was dual integration install. We had Unity integrated with Call Manager via IP and with Siemens via analog.
Integration with Siemens worked fine with the Analog except for the caller-id.
Siemens PBX was integrated with the call manager via PRI. 8 was configured as access-code to dial between the pbx and ip phones.
Assume the pbx extension is 4315.
Siemens pbx phone (4315) was configured to forward to 84135 for Ring no Answer and Forward busy conditions, since 8 was the access code to send calls to the call manager via the router, calls coming to 4315 and not being answered were forwarded to 84135, which was sent to call manager via the gateway.
We had dummy extension 4315 configured in call manager which for fwd all to voicemail and incoming calls for 4315 was forwarded to the correct voicemail. We were able to get the caller-id working and the voicemails showing up in the email was with the caller-id. The MWI was still over the analog lines.
Though this worked I believe there were lot of configuration involved on pbx as per the pbx engr and also this workaround is not approved by TAC. We are planning to try QSIG between the siemens and call manager to reduce the amount of configuration involved in pbx. Even if we get it working via QSIG, this workaround is also not supported by TAC.
If you don't have dual integration, I don't know if there are any workaround.
Hope this helps.
Thank you, I have Dual integration and I was also thinking of sending the call across the PRI tie between Siemens and call manager.
Did you ever get the Siemens > QSIG > CM working? I have a Siemens/Unity integration coming up and don't want to go the PIMG route. QSIG integration would be great but I believe MWI would go over Analog still. Not sure though.