Baze

I believe you can't do this, no. Most of the carriers now are rejecting any non-normal caller ID because of the huge spam calling issues we've been seeing. We recommend clients turn on the "Press 1 to accept the call" feature or use a softphone which you can send callerID info to.

You can definitely use them, just need to contact 2600 to get setup with a contract and then they will get them added to your cluster.

Those apps are closed source iirc.

Ahhhh - yeah, unfortunately not to on the system to my knowledge. Beauty of 2600hz is that basically everything is possible if you have someone that knows how to do it.

Not out of the box, but that would be a really cool thing to build out in a pivot and could likely be done that way I bet. Cool idea!

We just put the PR in for the front end call flow widget here: https://github.com/2600hz/monster-ui-callflows/pull/49

Here is the commit for the backend. Looks like we didn't do a PR on the front end; I've asked our team to get that added, so should be merged shortly after that. https://github.com/2600hz/kazoo/commit/3a75fa9531a721412beb2f78fd75d1f34cc716d8#diff-9dc4e41be30a38fa6bae368a6bc08846

We just put in a commit to allow ring tones to be selected from a callflow, however you might have to remove the internal/external from the device, because that would likely override this. I'll see if I can track down the commit.

Both polycom and yealink have config settings that allow you to set an alert tone for hold which is very handy. It won’t ring back but at least you get some type of notification.

Lol. Well two answers for the price of one I guess!

Call in to their voicemail and hit star during the message and follow prompts. You can also use advanced callflows with the ‘check voicemail’ object if you want to create a dedicated call in number.

Unfortunately we haven't ever been able to get a workaround on those cordless models. We've had to train the clients to just dial *31, etc.

Add another vote for check the box for enable consistent NAT.  From what we noticed, the older (several years that is) versions of the firmware did different things, so that is why sometimes you see conflicting docs.  Other things we do are: -change the UDP timeout to 3600 (well, depending on how your phones are config'd) -turn OFF sip alg -set up bandwidth management and prioritize by IP or another method. We've also noticed many times people will turn on QoS settings thinking that will prioritize, but if I recall all that really does is just select to leave existing tags on the packets or manually add 46 and such, but it does NOT do anything in the way of prioritizing.

Hi guys - I just went in and created a JIRA ticket (hopefully correctly!!!) for this.  You never realize how much work goes into a simple request until you have to type out the specs, and I'm sure there are many steps I missed.  ;)  Would be a cool thing, though, and we have had a request or two over the years for it too.  Looks like there are around 794 other tickets, though.  The guys over there look to be pretty slammed! https://2600hz.atlassian.net/browse/ABP-27

Hey Rick - I spoke to my guys about this, and definitely interested and would love to contribute back as a PR to master if we came up with something good. We are pretty slammed right now, so probably a few months before we could help. My provisioning guys said they would buy me lunch for a month if we did this for them, so not sure what the value would translate to there. ha!

I just went through our tickets and don't see anything - maybe a different client? If you could shoot over to me that would be much appreciated. - brandonb@audian.com

Hi Darren - I think I may have completely missed where you answered those Pivot questions!  Was it a different thread maybe?  I don't think I ever heard anything on this, but we'd love to report back once we get something working.  

Hi Everyone - we've had a few requests from clients on this (mainly from old MSFT responsepoint clients or skype for biz) and always thought it was a great idea.  Anyone here done anything like it?   Idea is: Call comes in and goes direct to message"Thank you for calling, please tell me who you are trying to reach?"(caller) - "Brandon""Transferring to Brandon Last Name now"Behind the scenes: Message is played to callergoes to a "waiting to record" callflow where we are recording that leg of the call. the recording is then converted to WAVA directory has already been selected at setup, and the first name / last names are snagged and saved dynamically (so could also add "Sales" , "Support" , etc.) so that the list of potential matches is up to date with smartPBX directoryAPI Call is made to iSpeech They provide a method to send the API call with a WAV along with a list of names to compare.  Our main issue with iSpeech ASR is it is horrible at 8bit audio, but with a list to compare against and a simple name, we think it should be good.iSpeech returns the namename is matched against directory and a blind transfer is initiated to the user callflow or group.

ha! goooooood point. oops.

May be worthwhile to check on the networking side also. #1, do they have the bandwidth, because that is essentially 15 concurrent calls coming through, and #2, is their firewall set to bypass any type of flood control rules or otherwise that may be causing problems. Our next step would be to pcap their router to figure out whether the invites aren't getting to their location, or they are, but they aren't then being passed to the phones.

Nice - yeah, provided the config for the phones is good, it should just work.  set a callflow, dial that extension for the flow, and phones beep then pickup.  config on the phones is the tricky part, but I'd imagine 2600 already fixed that.

By intercom you mean 2-way, correct?  Polycom and Yealink can use the paging feature (in advanced callflow) which is 1-way and easy.  To get fancier you'd have to use the multicast paging which allows subscribing, but then can allow for Push to Talk for two way paging.  Another benefit there is in VERY large installations you can do internal paging without sending all of those calls over the WAN.  I can send a dummy config if you'd like.

OHHHHH - I did NOT see that in the callflow before.  that little checkbox will be VERY, VERY handy.  Thanks Rick!

right, but don't you have the 20 second timeout issue?  You can't even change it manually in the advanced callflow app because then any change to the smartpbx section will re-write it to 60 seconds. 

Wanted to see if anyone else has a good solution here.  In cases where you want the main callflow to go to a forwarded device (say a call center or cell phone in case of power outage), there isn't an easy way to set.  If you use the option to go to a user, it sets the default ring time to 20 seconds, which isn't enough usually.  We were talking about adding a fourth option which is just "call forward" and sets the callflow to a 60 second timeout to the device instead.  Anyone have a better solution?  Side note - would be lovely to be able to set the ring time from the incoming call handling page too.