Logicwrath

I was able to get the call flow for the EXT -> Dynamic CID -> DISA working.  I had to check the box that says "Keep the original Caller-ID".   So far that has been reliable. However, the call logs and history will only show calling this local extension instead of the actual number getting sent into DISA.  While this is better than buying 2 or 3 times the devices it still does not feel like a "permanent fix".  I hope there is a better way to do this soon. Additionally, not being able to provision the DTMF key in the Advanced Provisioner adds complexity.

I assume you are referring to the PREPEND + DISA method.  I wanted to add that my testing with this technique was using a handset registered to the system and not from a cell phone outside the system.

esoare: I have tried using dynamic_cid -> DISA, and it does not have any effect on the caller ID?  I also ensured that the user, device, and the company did not have a default caller ID set to test.  I also tried with and without a "blank" prepend ahead of the dynamic CID object to see if that would essentially "reset" the CID prior to hitting the other call flow modules. http://prntscr.com/clg2ot I did have the PREPEND + DISA mostly working, it was just not 100% reliable.  It seems that 1-3 out of 15 calls will show incorrect caller ID. I was hoping somebody could confirm that creating a single device and registering it on many phones will not cause issues.  This would allow me to register one device per DID if I needed to as a last resort. I am not sure why it is so hard to change outbound caller ID.  This seems like a basic feature.  What is the point of having multiple DID's if you can't easily use them?

https://2600hz.atlassian.net/browse/KAZOO-5088 I created a JIRA ticket if anyone wants to upvote and follow for DTMF DSSKeys for Yealink phones.

In advanced call flows, menu, advanced you can see this: http://prntscr.com/clfct6

Can we create 1 device for each DID and register it on many phones without issue?  The only use would be to change the outgoing caller-id with 1 click.  Would this introduce any kind of problems?

Are we allowed to post lists in hosted? http://api.zswitch.net:8000/v2/accounts/{ACCOUNT_ID}/lists {     "data":{         "name":"Dynamic CID",       "description":"List of new callerid name and number entries ",       "list_type":"dynamic_cid"    } } I get: {  "data": {     "message": "not found"   },   "error": "404",   "message": "not_found",   "status": "error",   "request_id": "sanitized",   "auth_token": "value" }

I am trying to get this to work.  Here is some info: https://github.com/2600hz/kazoo/blob/master/applications/callflow/doc/cf_dynamic_cid.md I have tried posting a call flow like below but I may have mangled it some how. {    "data":{         "name": "dynamic_cid_outbound",         "numbers": [],         "patterns":["^\\*9([0-9]{2,})$"],             "flow":{ "module": "dynamic_cid" ,"data": {     "_id": "cidlist",     "length" : 2,     "entries": {        "00": {            "number": "15173335555",            "name": "COMPANY NAME"        },        "01": {            "number": "15175553333",            "name": "NAME COMPANY"         }     } }             }    } } In this example I dial *900 + the 10 or 11 digit phone number and the dynamic_cid module is still asking me for the caller id to use.  If I enter one it ends up failing. Is anyone using dynamic_cid in a call flow?  Can you sanitize and post a working example I can hack/tweak?

I wanted to use PREPEND + DISA to change the outbound phone number.  It seems like it works 90% of the time.  Then about 10% of the time it shows either restricted or 919-692-6062 or 919-692-6063. Any idea why this happens?  Does it have to do with the route the call takes or the 2600 LCR?

We have this issue too.  We have auto dealerships that fax large packets 20-30 pages.  In our case the finance company does not really complain about it too much so it has not become a a major issue.   However, I consider this a "known issue" where the fax quality is not as good as it could be.  I do not currently think this is related to the page counts.  I also do not think this is an issue with the original PDF resolution or scanner settings.  If I were to guess, this is an issue with the conversion process that PDFs go through during the email to fax and fax to email process.  Perhaps some ghostscript settings could be changed that would have a drastic improvement.

We have been trying to avoid giving end user's access to the Smart PBX and some of the other applications for this specific reason. It would be best case if we could turn off the ability for clients to add Devices, Users, and DIDs to their accounts with out us knowing.  If this is difficult, it would be helpful if we could at least be notified that a new DID, User, or Device has been created on an account so we could update our agreements.  I do not see any kind of branding email template to this affect.  I think all account administrators do get an email about a new user and perhaps new devices as it is now.  However, I do not think any emails get generated for new phone numbers.

When we setup a fax box, if we want to get fax sent receipts sent to the "original sender" we have to specify a recipient in the "Fax sent receipts" box in advanced call flows.  In most cases we do not want to specify any specific email to receive all of the fax sent receipts.  We typically only want the person sending the fax to get a fax sent receipt.  The only way the sender gets the success email is if I enter an email into that box. I am tempted to enter a dummy email to a domain I own, just so it gets bounced, and the original user still gets the sent receipt.  However, this seems like a terrible idea. Can you allow us to leave this global field "Fax sent receipts" blank or null and still send fax sent receipts to the original sender?

I believe Team Mobile let's their customers change their caller ID to whatever they want pretty easily.  I was talking to an auto dealer client and they said they get crazy names all the time from them.

Yea, the more I think about it, I don't see that use case either.  You could easily have separate call flows and if transfer between account extensions.

It would be nice if the emails were stored for 14 days with a web interface for re-sending them.  Email can certainly unreliable.

A use case for us is management companies that own multiple businesses.  So for example, we have one company with 4 separate car lots and a corporate office. We are also in talks with a medical company that has 3-4 business entities and will will be opening up "nerve centers" across the country. It would be nice if we could always create separate accounts for billing, limits, and tracking yet still create and manage call flows with multiple accounts and perform extension to extension dialing when necessary.

In most cases clients are interested in having their voice mails emailed to them instead of having to manage the voice mail box on the phone system.  If there is an issue where a 2600hz SMTP server gets blacklisted or some other kind of delivery issue, is there a way to re-send voice mail messages that did not get delivered?   I do not have a current issue, I am just thinking ahead.  When I was managing my own multi-tenant PBX, I had an issue similar to this at one point.  I was thankfully able to pull the raw emails out of the database and deliver them.  Is there any kind of safeguard like this in place on Kazoo?

Rick, Thank you for your W52P experience.  I feel more comfortable recommending it. My most recent use case will be in a shop floor.  It can get noisy at times.  I wonder if there is a device that does well reducing background noise.

I understand they are limited in features at times.  However, did you like the reliability of a particular handset in terms of call quality, battery life, and range?

Hello, I am trying to decide which wireless DECT handsets to standardize on.  I am attracted to the Yealink W52P as their appears to be provisioning support for it now.  However, the reviews do not seem as positive as the Panasonic TGP500. I have experience with the snom M700 base stations and M65 handsets.  These are a bit too pricey for simple installations where only a couple handsets are needed.  However, I do not have any current complaints with them. I have seen some reviews that suggest the audio is not quite as good on the W52P over the TGP 500 and I have also seen some people suggest that the standby time on the W52P was half that of the TGP500.  I cannot confirm. Is there some other gold standard for SIP based DECT handsets?  I need to quote one and I would prefer to get something that is not going to create a lot of complaints or service requests

"Yealink RDS have sometimes failed me when I am doing a roll out" In my experience, if you are using the RDS service, the phone will get redirected correctly during initial provisioning, however, the 2600hz provisioning server does not configure the "provisioning server" field when it sends the configuration to the phone the first time.  When the phone reboots it will not get a new or updated configuration. Perhaps, we should open a JIRA ticket on this.  I suspect we want this field set during initial provisioning.  I can't think of a reason why leaving this out is intentional.

I am not sure how client is defined.  Is a client a reseller or is a client a sub client of a reseller?  I would prefer to save all the necessary addresses into my configuration file that gets loaded into the firewall and not have to manually check and change configuration items on a per client basis.

I am not currently using the PBX connector, however, I suspect we will at some point for having them all would be best. I will open a support ticket.

I also wanted to add that in cases where end users are on a softphone it becomes harder to do good QoS without the complete list of public IPs as the local IP will be the PC on the local network.  I am also not a big fan of tagging connections based on port or protocol as other services can sometimes get into the wrong priority.

Hello, I was fixing some issues with my golden firewall configuration today and realized that their are quite a few IP addresses getting used in the SIP calls that I did not have flagged.  I am taking other steps to ensure QoS in all situations regardless of 2600hz IP address, however, I would very much like 2600hz to create and maintain a public wiki document of some kind that gives all the address ranges in use. Is this something we can get posted?