Tuly
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Posts
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Resource Library: Monster UI Apps for KAZOO
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Posts posted by Tuly
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The reason why im excited about it, when I have to do a quick API, lets say change The park ring back or change the outbound caller ID, in order to get the token I have to put in the username and password of an extension, hash it then copy paste a few times to get the token, (Maybe there is an easier way, I don't know )
I do it just on a Linux server, -
getting the auth-token takes me longer then doing the API i want (and most of the time i need to change the users password cuz i don't know the password)
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have i just discovered america? is this an official way to get the Auth Token of an account? when you are on any page in kazoo press "cd" it will bring up a page with the account name, account ID, and the Auth Token, makes my life much easier now....
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@esoare Can you take a good long ping in the middle of the day to us-west.p.zswitch.net and to the 173.1 IP?
Also a trace route to both,
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It's very interesting I've already been logged into about 20 Clients and all of them have the same results high pings and packet loss , And it's only in the middle of the day......
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Hi,
Is anyone who has Verizon FIOS having issues lately with the US-east? we’re having for the last I would say 2 weeks or maybe even more, high pings and a lot of packet loos to us-east, and this is only happening on FIOS customers
Interesting part is that pinging .70.1 or 70.10 is good ( I guess it’s in the same datacenter) only the 2600HZ IPs are bad, and its happening only during working hours, about 10-4, we are not getting anywhere with Verizon, they keep saying that up till the IP 70.3 everything looks good, (and it does, the last hop to that IP by level 3 is also very good)
How could we start to troubleshoot to see where the issue is?
Thanks,
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when you have "Call Forwarding Failover" on, the option Require Key Press - Leave voicemail on forwarded numbers is not working, the FW number voicemail will always pickup,
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sorry for bumping this,
i need to turn OFF sip ALG, from the info i got here is that clicking the check enable consistent NAT Turns off ALG, but other websites say that to turn off ALG it should not be checked,
https://support.siteserver.com/kb/a109/sonicwall-disabling-sip-alg.aspx
http://www.surevoip.co.uk/support/wiki/troubleshooting:sip_alg:sonicwall
so what is it?
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On 9/8/2017 at 11:48 AM, Logicwrath said:
Have you tried or seen this yet?
Thanks that worked,
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We have a client constantly complaining that people can't hear the voice because it's very quiet, is it possible to change the voice and make it higher so that people outside the office should hear louder? Or is it possible on a yealink phone to change the outgoing voice to make it higher?
Thank you.
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when creating a Directory from a callflow, it would be easier for users to have the new Directory in the list not needing to closed and reopen the directory box or order to save it, as the video below,
(the same is with adding a new media file in a voicemail)
Thank you!
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@JR^ ummm Very interesting, the only way I can re-created now is if I change the "timeout" to less than the current time, for example the current timeout is 40 seconds, change it to anything less (20 or 10) or change it to anything more but less than 10 seconds (45 or 49) but more then 10 seconds is good,
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Good to know all the educational stuff, but as the subject says "on the hosted system"
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I do not believe the DND changes anything on the server, when you set a phone to DND and call the extension, the phone is getting the call but it only gets rejected,
I like to completely disable the feature of DND on phones, a lot of headaches and frustrated users when they have the phone on DND and they don't even know what it means.
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To make it clear smart PBX should have the same description as in Callflows > users > call forwarding. "Require Key Press"
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Not sure this is the place to discuss, but I got a quite a few complains that voicemails have been deleted, and I also see on our account all voicemails are missing past 30 days,
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whenever you make a change in a callflow you get a reminder to save the changes with a warning " You have modified this Callflow, don't forget to save it!
but when you change anything inside a ring group, it will not say this warning, and the ring group will not be saved only if you click the green button save changes, would be good to have this warning also when you edit a ring group,
Thank you!
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Hi all,Are voicemails being systematically deleted from the hosted platform and if so what are the time limits?
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JR While your there maybe it's also worth changing Smart PBX has "Leave voicemails on forwarded numbers" and advanced callflows has "Require Key Press" (i believe it makes more sense Require Key Press)
Thank you!!
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Something confuses us every time when we need to forward to a cell phone, and decide if we should keep The Originators caller ID or to see the office caller ID,in Smart PBX it says "Keep your Caller-ID" users never know what that means, in advanced callflows it says "Keep Caller ID" (Also a bit confusing)
Can we change the description to something like "show Originators caller ID" or "Keep original caller ID" ?
Thanks!
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Thank you understood, Let me go a step further,
is it possible to transfer back a phone call ( was transferred to a cell) from cell phone to the office phone?
Let's say you get a forwarded call on your cell phone, and you would like to transfer it back to one secretary is possible to press something like *2101?
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I can imagine there's no easy way of doing it but I might as well ask,
a client of 50+ phones is asking "please send me in Excel a list of all names - extension number - VM to email address - and direct numbers to each EXT - is there an easy way to do it?
2600hz Hosted RTP Media Servers
in PSTN, Software, and Services
Posted
I do not believe it has anything to do with SRV records, or with model of phone you are using, when you register your SIP signaling to us-central, and you make a call the server (2600HZ) decides what the RTP media server should be, the phone or the SRV record do not and cannot decide,