Travis Hawkins

This would be HUGE for us. We try not to roll out faxing too much because we have so many problems with the email-to-fax system. Inbound faxing works flawlessly, but having to email using the specific format is a much bigger problem for our clients than we anticipated. The ability to open the portal, browse for a pdf/tiff/whatever file type, enter the destination number in an input field and hit send would be a godsend.

@fhill Sorry I haven't been on the forums lately but yes our issue was resolved the next day. Apparently there was an issue at a major carrier exchange in Kansas City, MO that was effecting traffic throughout the midwest. It was basically a two-way routing issue where the Internet traffic was taking one route out, but the traffic was trying to take a different route back. It caused all kinds of issues, but voice was obviously the easiest to detect since it's all real-time and browser sessions were able to wait and resolve but it just made the Internet seem a little "slow" to end users. Three of the local IPS all had to work with the regional exchange to resolve the issue, but I never heard the root cause.

@atmosphere617 I haven't performed any captures yet as I've been working directly with 2600hz support thus far, but they still haven't found anything. I'll start investigating more internally now, I was really just curious if anyone else on hosted in the Midwest was seeing issues but thanks for replying and I'll see what other troubleshooting I can do here.

@Brian Dunne Good question on the PSTN side, but yes it's happening on internal ext-to-ext calls in our office as well. From what I can tell we're having no connectivity issues to any of the proxies so I don't *think* it's a Level 3 issue, but I can't confirm that of course.

All- We're located in the midwest and we (along with all of our clients) are having major call quality, mostly one-way audio issues today on the hosted platform. We've opened a ticket and I'm on with support right now but they said they don't see anything wrong so far and they've had no other reports. Is anyone else experiencing issues? We've had reports from about a 300 mile radius thus far and it appears to be effecting every single one of our customers. We're getting no audio on about 9/10 calls and occasionally a call will work, but the audio still sounds horrible when it does come through. Just curious if anyone else is seeing this? Thanks. Travis

Thanks a bunch everybody! Great ideas! We're going to setup a couple of these up and see what we like for ease of use and setup. Again, thanks so much for your help.

We have a couple of clients who have asked if there's an easy way to manually "close" their office for a short time (impromptu meetings, unscheduled lunches, etc) without having to change their office hours schedule or enter a holiday. Lots of traditional systems have a short code for overriding the schedules and that's what they're looking for. Example, there's a company meeting that might last 20 minutes, or maybe 2 hours. How can someone quickly and easily roll the phones to a closed message or menu and then un-roll them once the meeting is done? Thanks.

Is there a procedure for requesting/gaining access to the sandbox system? Thanks in advance. Travis

Interesting. I emailed back on the invite and was told you they were "working out some back end details" and I would receive more info when it was ready for testing, but never heard back after that :-(

Just to add to that Karl... In cases where we need to set the caller id via Monster while waiting on a port, we've been able to open a ticket with 2600hz and have them pre-populate the number for us. It will be nice when that's automated as Darren mentioned, but they can do it for you manually now, just via a ticket. Hope that helps!

We have deployed only one of those, but I remember we had the exact same issue. Unfortunately I don't have access the config remotely, but I'm thinking it was in the auto answer settings where we fixed it. If you don't have any luck with it, I'll see if I can get ahold of the config we have. Good luck!

Darren coming through as usual! That did it, thanks!

Does anyone have a quick guide on setting up a Grandstream HT702 ATA adapter? I've setup several HT704 devices without issue, but I've tried two different HT702 devices and both of them register just fine but can't dial out. Both of them can receive calls without issue, but both receive a fast busy when dialing out and the calls don't show up in the CDRs. I've never had this issue with the HT704s and they're setup identical. Any help would be greatly appreciated! Thanks. Travis

We saw that yesterday too, haven't tried it out yet but very cool!

Looks nice Rick. What are you using?

Is it possible to move an account from one reseller to another or just from a reseller up a level directly under the root (our) account? I know this doesn't effect the functionality, just trying to keep things tidy in our interface. Has anyone ever accomplished this or had support perform an action like this for you?

Good thought Michael. I suppose that might work in a scenario where the main number routes to a single phone, but we have lots of customers whose main number routes to a ring group. Maybe we would just have to be selective about which customers we enabled this feature for.

How will the system determine if the customer's Internet is down?

The user portal would be fantastic.  We've used other sip pbx software before that had the ability to upload and send faxes through a user portal, as well as view and download received faxes.  The entire history of faxes and their associated status were available from within the user portal and that was much less confusing to the end users.

Great explanation Darren, thanks for that.  The customer who called in was actually us as we have the client who works as a call center for companies all over the country and requires the ability to call back their customers using the name and number of the company their representing at the time. We're all for using a new module or a better solution based on customizing outbound caller id.  Unfortunately for us, the only other way to accomplish this currently (as I understand it) is to create a separate line appearance for each of their customers, but doing that requires each to be a new device since we can't provision multiple lines on a single device.  Of course that means paying for each additional device just to be able to add it as a line appearance, and having phones that can handle as many line appearance as we would need. I realize this is all just based on our specific needs, but if there's a way to work on a better solution moving forward that would be fantastic.  Thanks for taking the time to read through our threads and responding! Also, I'm with Karl on not getting the DISA to work by calling in from a cell phone to place a call using the office caller id.  I haven't tried it for quite a while so I'll setup a call flow and try it out again.  Thanks. Travis

Hey Karl, sorry that's not working for you.  Here's a couple of screenshots of a setup that's working for us. Call flow: Here's the DISA config (it's blank but that works for us for this client): Like esoare stated, make sure the 'Use Account Caller ID' is NOT checked.  Also, are you calling from a number that's assigned to the same account you're calling from or are you trying to specify a caller id number from a sub account/client account?  We don't actually use this feature at our office, but we have a few clients who use it without issue. Good luck! Travis

DynamicCLID, I don't think a Prepend on outbound calling will do anything.

We have a client who is a call center for about 40 other companies.  Each company is provided a dedicated DID and we use the prepend functionality to show the name of the company the caller was calling, so whoever answers the phone can respond to the caller appropriately ("Thank you for calling [insert company name here]).  This way the caller doesn't realize they've called an outsourced call center. The other part of that solution is the disa call flow module which we use similarly to what esoare stated.  The internal staff member dials the internal extension of the callflow that directs them to the disa module, they enter the DID they want to dial out with, then dial any number and the callee will see the appropriate caller id information instead of the default caller id which would be a different company.  We've been doing this for over a year and it works great.  The only issue is that the internal staff have to keep a cheat sheet of which DID corresponds which company so they know what number to dial out with, but it's not too big a deal.

We've been doing this for a while with Yealink phones as well, for the same reasons stated by Karl.  The paging in the Kazoo UI is easy to setup and use, but the delay drives people crazy when waiting for the function to answer, and then hearing their delayed voice while they are trying to speak.  We switched to using the multicast features on the Yealink phones about a year ago and it works great, just a little more involved to setup and manage since we can't use the Provisioner to do it.

Perfect, thanks for your help! Do you know if it will eventually be added back to the normal device config page for quicker access?