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Messages - mjkadel

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Hi all,

We're still trying to roll out our new Mitel phone system with the help of a contractor. We have a MiCC server which acts as the Salesforce connector. The ACD users log into the CTI Softphone on their Salesforce homepage. Theoretically when somebody calls our ACD queue it should open a Salesforce case and fill in the account and contact fields automatically. When calls come in they first land in Nu-Point so the caller can navigate a call flow before ending up in the ACD queue.

There are always multiple call events in the .js logs in Salesforce. When the first call event contains the caller's phone number the integration works great. When the first call event contains the Nu-Point extension the integration does not work. If we aim the call directly at the queue, bypassing Nu-Point, the integration always works.

Does anyone know of a way to filter out the Nu-Point call events from being passed on to Salesforce through MiCC?

Thanks,

Michael

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FWIW, the issue turned out to be the SIP profile. I think this is what we used, but I'm somewhat unfamiliar with the system still. This is a CSV export of the SIP Device Capabilities for the uWarp.

SIP Device Capabilities,vid1.81,20.0.3.21,
0,1,19,8,24,22,17,13,25,30,35,4,57,36,23,12,5,10,11,56,37,6,26,16,2,55,27,34,18,60,52,53,38,41,29,58,21,54,28,31,32,33,14,9,59,48,49,50,15,39,40,63,64,65,66,67,68,
SIP Device Capabilities Number,Comment,Outbound Proxy Server,Replace System based with Device based In-Call Features,Allow MWI Notifications without Subscription,Enable Digit Collection In Busy Or Alerting State,TLS Only,Allow Device To Use Multiple Active M-Lines,Allow Using UPDATE For Early Media Renegotiation,AVP Only Device,Enable Mitel Proprietary SDP,Force sending SDP in initial Invite message,Ignore SDP Answers in Provisional Responses,IP Media Default,Limit to one Offer/Answer per INVITE,Prevent SDP Renegotiation If Peer Initiated Hold,Prevent the Use of IP Address 0.0.0.0 in SDP Messages,Renegotiate SDP To Enforce Symmetric Codec,Repeat SDP Answer If Duplicate Offer Is Received,Send Answer only after renegotiation is complete,Support CTI Hold/Retrieve,Suppress Use of SDP Inactive Media Streams,Allow Display Update,Allow FQDN for Resiliency,Disable Reliable Provisional Responses,Disable Use of User-Agent and Server Headers,Fail REFER To Keep Call Active On Mid-Call Feature,If TLS use 'sips:' Scheme,Mode for Out-of-Band DTMF,Multilingual Name Display,Override Auto-Answer Headers,Override Auto-Answer Headers With,Q.850 Reason Headers,Remove Anonymous User,Require Reliable Provisional Responses on Outgoing Calls,Suppress Redirection Headers,Use P-Asserted Identity Header,Use user=phone,Enable Distinctive Ringing,Internal Ring,External Ring,Callback Ring,Registration Period Minimum,Session Timer,Session Timer: Local as Refresher,Subscription Period,Subscription Period Minimum,Subscription Period Refresh (%),Invite Ringing Response Timer,Allow Out Subscriptions for Remote Digit Monitoring,Force Out Subscriptions for Remote Digit Monitoring,Creator,Date Created,Created with Version,SIP Device,Vendor Notes,Dial Plan,
1,Uwarp,,No,No,No,No,No,No,Yes,No,No,No,ipv4,No,No,No,No,No,No,No,Yes,No,No,No,No,No,No,RFC 4733 DTMF,No,No,,No,No,No,No,Yes,No,No,<http://www.notused.com>;info=alert-internal,<http://www.notused.com>;info=alert-external,<http://www.notused.com>;info=alert-community1,300,0,No,3600,300,80,0,No,No,,,,,,,

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Thanks, Ralph!

4
Mitel MiVoice Business/MCD/3300 / External/Streaming Music on Hold
« on: June 15, 2020, 12:09:39 PM »
I'm in the process of having a new MiVoice Business system installed and are looking for how to get external live "music" on hold. I work for a TV and Radio station, and we're trying to get our broadcast into the phone system live.

The sales engineer for the contractor who is setting it up suggested we get a Pika uWarp Plus
https://www.pikatech.com/music-on-hold-solutions/#1526310543202-effabdba-e1dd

This allows us to configure it with SIP credentials and put in our internet streaming source. It works great for approximately 2 hours and 20 minutes, then disconnects and reconnects. At that point the sound is choppy. It will continue to reconnect every 2:20 with the same choppy sound until it finally disconnects for the final time.

Does anyone have any experience with a device like this, or with a similar device? If there were other ways of delivering the audio to the system we could do that too. We have an MXe III 3300, but it does not have the analog MOH port on the back.

We're moving from an Asterisk based system which allowed our live stream so easily. Before that we had an ancient Millennium which also allowed for this, but all via analog wiring. So 13+ years of this functionality is difficult to give up.

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