Thanks for the feedback guys, it really is appreciated.
We're under a bit of pressure all around at the moment with network expansion and everyone is head down bum up and hard at work.
We're under a bit of pressure all around at the moment with network expansion
Is that why registration is a bit flaky lately?
Is that why registration is a bit flaky lately?
No, services shouldn't be affected. Have you reported issues to support?
No, services shouldn't be affected.
2015-10-12 02:44:44.919152 Failed Registration with status Request Timeout [408]. failure #1
2015-10-12 02:44:45.899154 Failed Registration [408], setting retry to 30 seconds.
2015-10-12 02:44:52.899153 [WARNING] sofia.c:5724 Ping failed with code 408 � count 1/0/1, state DOWN
2015-10-12 02:45:16.899150 [NOTICE] sofia_reg.c:448 Registering 1fae1e8d-90d8-446e-bc50-775f7fd53447
2015-10-12 02:45:48.959148 Failed Registration with status Request Timeout [408]. failure #2
2015-10-12 02:45:49.919152 Failed Registration [408], setting retry to 60 seconds.
2015-10-12 02:46:50.919156 [NOTICE] sofia_reg.c:448 Registering 1fae1e8d-90d8-446e-bc50-775f7fd53447
2015-10-12 02:46:52.059148 [DEBUG] sofia_reg.c:2355 Changing expire time to 240 by request of proxy sip:sip.telecube.net.au
2015-10-12 02:46:53.979155 [WARNING] sofia.c:5699 Ping succeeded with code 404 � count 1/1/1, state UP
Same thing repeats intermittently.......
Have you reported issues to support?
I just did ;)
2015-10-12 02:44:52.899153 [WARNING] sofia.c:5724 Ping failed with code 408 � count 1/0/1, state DOWN
Might be a dns issue, can you change to using the ip and see if your problem persists please?
Might be a dns issue,
Same problem at two different sites using different ISPs.
can you change to using the ip and see if your problem persists please?
I'll change to IP and report back.....
Same thing repeats intermittently.......
No registration problems for about two weeks and it started again Thu Oct 8 03:58:41 2015.
Mon Oct 12 02:45:17 2015 3 SIP registration failure Could not Register user 100nnn1 on server sip.telecube.net.au:5060. Reason: Timeout occurred
Mon Oct 12 02:50:32 2015 1 SIP registration succeeded Successfully registered user 100nnn1 on server sip.telecube.net.au:5060
Mon Oct 12 05:29:30 2015 3 SIP registration failure Could not Register user 100nnn1 on server sip.telecube.net.au:5060. Reason: Timeout occurred
Mon Oct 12 05:37:37 2015 1 SIP registration succeeded Successfully registered user 100nnn1 on server sip.vic.telecube.net.au:5060
Mon Oct 12 06:08:10 2015 3 SIP registration failure Could not Register user 100nnn1 on server sip.telecube.net.au:5060. Reason: Timeout occurred
Mon Oct 12 06:08:26 2015 1 SIP registration succeeded Successfully registered user 100nnn1 on server sip.telecube.net.au:5060
No more since 0608 this morning.
I'll see if I can change the FQDN to an IP address and monitor.
S.
I'll see if I can change the FQDN to an IP address and monitor.
I have changed the registration to
Mon Oct 12 18:13:03 2015 1 SIP registration succeeded Successfully registered user 100nnn1 on server 103.193.167.41:5060
I get an email alert each time it fails so I'll pass on the results.
S.
I get an email alert each time it fails so I'll pass on the results.
My script checks gateway status every 30 min; tries a reset 3 times before giving up. An email alert is issued with the result. I guess without some sort of alert people won't notice these failures since the gateway re-registers on the second attempt.
Happy to share the script for FusionPBX/FreeSWITCH if anyone is interested.
I've changed the proxy address to its IP too.
I need you both to please send details to support so I can look at the ISPs you are connecting from and trace routes back to you.
I just did ;)
I can't find any emails at support about this, can you whim me the ticket number please?
I can't find any emails at support about this, can you whim me the ticket number please?
Oh......I was talking about contacting the head of Telecube support here on WP :)
I can look at the ISPs you are connecting from
What sort of details are you looking for John � just the extension numbers?
What sort of details are you looking for John � just the extension numbers?
Yes please and mention the registration errors too please so I know it's related to this
Oh......I was talking about contacting the head of Telecube support here on WP :)
Ideally email communication is best for support queries please.
Yes please and mention the registration errors too please so I know it's related to this
Hi John, the ticket number is #913-367-147
Might be a dns issue, can you change to using the ip
This seems to be a common point of failure for this (and other) "lookups". What about providing IP addresses by location here, and update as required so those whom are happy to monitor and change as necessary can bypass the DNS?
I realise that this defeats the whole reason for the DNS but it's a persistent problem.
Edit: and I might add, I have always kept a valid host file for just this reason. Brings me undone now and again but at least I know where to lay blame.
Phil
What about providing IP addresses by location here, and update as required so those whom are happy to monitor and change as necessary can bypass the DNS?
Yes, I think this is a good idea and will implement something to publish IP addresses and maintain a mailing list.
I need you both to please send details to support so I can look at the ISPs you are connecting from and trace routes back to you.
Telecube:[#463-927-628]
Yes, I think this is a good idea and will implement something to publish IP addresses and maintain a mailing list
Changing to IP addresses is only a short term solution but at some stage I wont be around reading whirlpool when the IP address changes and then the phones will stop working again.
S.
Changing to IP addresses is
I think you misunderstood. There is no suggestion to use only IP addresses instead of names. It was merely to make the addresses more obvious to anyone happy to use as an alternative.
My hosts file entry: 103.193.167.41 sip.telecube.net.au
@John.M
How do I remove a DID (Delete) ?
Probably something simple I am missing.
How do I remove a DID (Delete) ?
I believe that currently you need to submit a support ticket.
I've also had a few errors logged starting on Oct 9 as shown below from my AVM 7490
This is setup to register to the name sip.telecube.net.au
12.10.15 02:45:56 Registration of Internet telephone number 1xxxxxx failed. Reason for error: Remote site not responding. Timeout.
11.10.15 10:29:15 Registration of Internet telephone number 1xxxxxx failed. Reason for error: Remote site not responding. Timeout.
11.10.15 10:27:51 Registration of Internet telephone number 1xxxxxx failed. Reason for error: Remote site not responding. Timeout.
11.10.15 10:26:37 Registration of Internet telephone number 1xxxxxx failed. Reason for error: Remote site not responding. Timeout.
10.10.15 11:08:59 Registration of Internet telephone number 1xxxxxx failed. Reason for error: Remote site not responding. Timeout.
09.10.15 19:38:17 Registration of Internet telephone number 1xxxxxx failed. Reason for error: Remote site not responding. Timeout.
09.10.15 15:27:41 Registration of Internet telephone number 1xxxxxx failed. Reason for error: Remote site not responding. Timeout.
I'm on Telstra cable and here are some possibly useful stats from my end.
traceroute to sip.telecube.net.au (103.193.167.41), 30 hops max, 60 byte packets
1 fritz.box (192.168.1.150) 0.283 ms 0.362 ms 0.411 ms
2 10.230.188.1 (10.230.188.1) 14.691 ms 14.704 ms 14.690 ms
3 58.160.0.161 (58.160.0.161) 21.492 ms 22.019 ms 21.997 ms
4 58.160.7.242 (58.160.7.242) 21.933 ms 21.991 ms 21.942 ms
5 bundle-ether4.lon-edge901.melbourne.telstra.net (203.50.76.2) 21.264 ms 21.804 ms 21.787 ms
6 bundle-ether2.win-edge901.melbourne.telstra.net (203.50.11.114) 21.809 ms 16.681 ms 16.614 ms
7 equ1651503.lnk.telstra.net (165.228.53.66) 15.836 ms 12.048 ms 12.019 ms
8 183.177.59.26 (183.177.59.26) 11.403 ms 11.313 ms 11.279 ms
9 103.193.167.41 (103.193.167.41) 11.386 ms 15.246 ms 15.253 ms
PING sip.telecube.net.au (103.193.167.41) 56(84) bytes of data.
64 bytes from 103.193.167.41: icmp_req=1 ttl=56 time=9.72 ms
64 bytes from 103.193.167.41: icmp_req=2 ttl=56 time=11.5 ms
64 bytes from 103.193.167.41: icmp_req=3 ttl=56 time=9.64 ms
64 bytes from 103.193.167.41: icmp_req=4 ttl=56 time=6.40 ms
64 bytes from 103.193.167.41: icmp_req=5 ttl=56 time=14.5 ms
64 bytes from 103.193.167.41: icmp_req=6 ttl=56 time=11.2 ms
64 bytes from 103.193.167.41: icmp_req=7 ttl=56 time=6.93 ms
64 bytes from 103.193.167.41: icmp_req=8 ttl=56 time=9.51 ms
64 bytes from 103.193.167.41: icmp_req=9 ttl=56 time=7.09 ms
64 bytes from 103.193.167.41: icmp_req=10 ttl=56 time=9.75 ms
--- sip.telecube.net.au ping statistics ---
10 packets transmitted, 10 received, 0% packet loss, time 45118ms
rtt min/avg/max/mdev = 6.402/9.644/14.579/2.343 ms
At this point we're assuming it's a DNS issue, the TTL for sip.telecube.net.au was set to 60 seconds while we were migrating across to the new network and I hadn't updated it. I have set the TTL now out to 5 days which will greatly reduce the DNS lookups and hopefully stabilise things for those experiencing problems.
I've a plan in place to publish registration server ip addresses on the website as well as alerting changes in advance by email to all voip users.
Early investigations are suggesting that registering direct to ip is resolving the problem but it's a little early to be completely sure.
If you are experiencing these issues please try registering to 103.193.167.41 directly and report back to us whether it resolves the problem or not.
At this point we're assuming it's a DNS issue, the TTL for sip.telecube.net.au was set to 60 seconds while we were migrating across to the new network and I hadn't updated it. I have set the TTL now out to 5 days which will greatly reduce the DNS lookups and hopefully stabilise things for those experiencing problems.
["...
Early investigations are suggesting that registering direct to ip is resolving the problem but it's a little early to be completely sure.
It is now over 24 hours since I changed to the IP address instead of the FQDN for both systems without any registration errors.
I'll change it back to sip.telecube.net.au and see what happens.
S
I'll change it back to sip.telecube.net.au and see what happens.
... our preliminary findings indicate that your onboard niner-triple-zero computer is in error predicting the fault ... ;)
Early investigations are suggesting that registering direct to ip is resolving the problem but it's a little early to be completely sure.
I set one of my two PBXes to IP yesterday. There are no registration failure on either servers since 20151012 2.45 am so thats good.
Thanks for the Whirlpool Offer(s) 2015.
Apart from checking the bill, how can I tell if I am on the Telecube Whirlpool Offer 2015. I believe I am but I can't see anywhere that I can confirm on Web site.
It would be great to have something (excuse me if I have missed it) on some page that indicates which "plan" / "special offer" one is on.
Cheers,
Ashley.
Apart from checking the bill, how can I tell if I am on the Telecube Whirlpool Offer 2015. I believe I am but I can't see anywhere that I can confirm on Web site.
Look on any extension for the rates. Can tell by the rates.
nm
Call Forwarding
Is call forward immediate working?
I have set an extn to call fwd to my mobile using the *21* code.
It confirms the fwd, but when I dial the extn I get a 503 error.
Just wondering if Telecube charge for calls within their own network.
I set my in-laws VoIP router (7800VDPX) to register to Telecube as a secondary extension of my account, and during the testing I called my home landline DID which is ported to Telecube (on the same account) and noticed the call was charged for.
When I call from extension to extension within the same account call is not charged and is classified as 'digital' in the call records.
The call from my in-laws to my home DID was classified as 'National' and charged for, even though the caller and callee are registered to the same Telecube account.
I called my home landline DID which is ported to Telecube (on the same account) and noticed the call was charged for.
If you called via the pstn it will be charged but if the call stays on-net it will be free. I havent tried with TC but usually if you call your DID via another TC extn it will stay on-net. I dont know if ported numbers are the same.
If you called via the pstn it will be charged but if the call stays on-net it will be free. I havent tried with TC but usually if you call your DID via another TC extn it will stay on-net. I dont know if ported numbers are the same.
Yes, and this has been my experience with other providers which is why I asked the question in this instance.
Interestingly, I had a quick look on the TC website and there's no mention of free calls within their network which many other providers advertise.
Perhaps this is an error, or maybe it's intentional but it would be good to get a definitive answer either way.
but it would be good to get a definitive answer either way.
I went back in 'history' and found an old (very old) posting of Johns.
I use the free calls often, from one extension to another extension, not the PSTN/DID number.
VF
Extension to extension (even between different accounts) are free. If you dial a phone number (DID), the call leaves the Telecube network as an external call and is charged.
Note that you can assign a three-digit quick dial number to your extensions. So you could assign your in-laws, say, 333 and that's all you have to dial from your phone.
whrl.pl/RelgHL
you could assign your in-laws, say, 333
Or a slightly different number may be more appropriate. :P
What about 686 (spells MUM)
666 is reserved for the hotline to Hell
666 is reserved for the hotline to Hell
Isn't that what we were talking about? :D
So you could assign your in-laws, say, 333 and that's all you have to dial from your phone.
Or you could adjust their ATA dial plan and direct calls to your DID to an appropriate TC extension number....
H.
Actually it looks like you are still on the standard rates, you should go to the opening post in this thread and click through the rate you want.
yeah � the person who looked into my support email mentioned the same thing. If i click through � I could see the "untimed" code in the box. I remember doing it back on Day 1 but mustn't have clicked apply.
Seems to be in there now. Thanks to you and your staff for sorting it out though!
Is there anyway to see if that code is applied through the 'my account' page?
Is there anyway to see if that code is applied through the 'my account' page?
Services
Extensions
Manage
.... you'll see the rates there.
H.
Not sure if this would fit in with Telecube's business model � just an idea....
Anyway, if you don't ask you won't know hey!
Is there any way to have a Non Geographic number search added to accounts? http://www.e164.org/non-search.php
You dial a 13 or 1300 number and Telecube automatically searches e164 for the geographic number and dials that.
You dial a 13 or 1300 number and Telecube automatically searches e164 for the geographic number and dials that.
The catch is if the user has timed calls to landlines it could be a lot more expensive. I think the user should just dial what they want to call.
Is there any way to have a Non Geographic number search added to accounts?
If they had an API we could get info from it's technically possible .. not sure whether it would be of much benefit though and there's too much going on at the moment to consider it tbh.
I think the user should just dial what they want to call.
Yep, I think if the user wants to do the lookup and dial the alternate number it's probably best left up to the user to do so.
If they had an API we could get info from it's technically possible .. not sure whether it would be of much benefit though and there's too much going on at the moment to consider it tbh.
No probs. Thanks.
I am trying to replicate a config from Faktortel and move to Telecube.
We have a DID that goes straight to an IVR that plays a greeting message, then on timeout or no key-press it routes a hunt list, and then a queue.
So setting up IVR is ok, but there is no timeout or no-keypress option in the IVR. I there some way I can have the IVR route to a huntlist or queue on no keypress?
Hi telecube!
Currently an mvoice hosted Pabx user but with the offer available to whirlpool users might be hard to pass !
Just a few questions -
1. Currently all my handsets are snom 320s. � is there a guide out there to help reconfigure those ?
2. With my current mvoice plan I have 5 handsets registered which allows me to make 5 concurrent calls, will this be the same with telecube?
3. With my internet provider I'm thinking of Telstra or Vodafone mobile internet 4G � any potential issues of SIP blocking of late or is this not applicable to hosted Pabx services ?
Thanks!
Yep, I think if the user wants to do the lookup and dial the alternate number it's probably best left up to the user to do so.
I think so, too, mainly because the e164.org search function is far from perfect and can result in getting connected to an unsuitable number for your location. Someone who regularly calls a certain 13 number can do the lookup themselves. Someone who makes a lot of calls to a variety of 13 numbers can open an account with a VSP that charges 9c or 10c untimed for these calls. This is a Telecube thread so I will refrain from mentioning these providers (I am becoming very diplomatic in my old age, aren't I?).
I think so, too, mainly because the e164.org search function is far from perfect and can result in getting connected to an unsuitable number for your location
Even when the numbers are location appropriate they are not necessarily equivalent. The only role I could see Telecube providing is implementing a function to allow arbitrary mapping of numbers. More generally perhaps, custom dial plans.
BTW John, is there some way to make 1300 numbers work similarly as dialed from the PSTN? For example, so that people ring their local pizza store.
1. Currently all my handsets are snom 320s. � is there a guide out there to help reconfigure those ?
Are they provisioned from the hosted server or set up manually from the web interface?
Under Identity1:
Display name: Telecube extension number or your name
Account: Telecube extension Number
Password: Telecube extension password
Registrar: sip.vic.telecube.net.au
You may want to set up BLF/SLA depending on your set up; use the 'function keys' page for that.
2. With my current mvoice plan I have 5 handsets registered which allows me to make 5 concurrent calls, will this be the same with telecube?
Telecube doesn't restrict the number of channels (lines) per extension (except international). Since a Snom 320 is capable of 3 way conference, your maximum usage would be 5 x 3 = 15 channels (lines) provided your internet connection can support the required bandwidth.
3. With my internet provider I'm thinking of Telstra or Vodafone mobile internet 4G � any potential issues of SIP blocking of late or is this not applicable to hosted Pabx services ?
You may use either Telstra or Voda 4G but keep in mind that Voda 4G has a NATed public IP, so you won't be able to run any servers, not even VPN.
Telecube give us solid pricing, reasonable quality support and reliable up-time with a large amount of customisable options that we can sort out ourselves.
Overall I can't fault the service. John is a man of his word which I also appreciate.
Thanks Telecube.
Is this Whirlpool offer just for home use? I'm looking to port in 10 DIDs and buy 7 handsets. Happy to go regular pricing if that's not the deal, but thought I'd put it out there :)
there some way I can have the IVR route to a huntlist or queue on no keypress?
Currently there's not but having a default route on timeout or no keypress is being added and is considered high in priority.
Currently all my handsets are snom 320s
EasyBB has answered your questions pretty well, I don't think I can offer anything other info.
Finite State Machine writes...
BTW John, is there some way to make 1300 numbers work similarly as dialed from the PSTN? For example, so that people ring their local pizza store.
Yes, we can dummy a DID into the 1300 routing platform.
Is this Whirlpool offer just for home use?
Nope, it's for everyone. Home or business users all have access to the same prices, features and functionality.
Currently there's not but having a default route on timeout or no keypress is being added and is considered high in priority.
Hi John, great thanks for the update.
So far I have had a great experience moving to Telecube
Maybe it was just a slight S[L]IP-up in services :)
:-)
We had a db node fail, the registration servers should have failed over to alternate nodes but they didn't. Still working on finding out why.
I'm digging into the flow of events and will have a statement out by email through the day.
Sincere apologies for the inconvenience.
Make the selection (SIP/IAX) while the extension is deactivated.
Thanks VK2XXY.
Samsung S4 there were no dots but selecting the "button" to the left of the home button bought up a settings menu
Thanks for this. Interestingly, I do not get the settings menu but I do get the option to "Exit" the cloud softphone which is still useful to know.
Has anyone found a way to access the settings in the cloud softphone if the 3 dots cannot be located anywhere on the screen? I'm using the Telstra Buzz phone.
Still working on finding out why.
Gremlins. It's always the gremlins. Sneaky little buggers they are. :)
Sincere apologies for the inconvenience.
Thanks for the great timing! :-)
I had not yet fully commissioned the IVR i have set up on Telecube replacing a dead asterisk box, mainly because i had been on leave and got back yesterday, all our inbound calls were coming in on a couple of PSTN handsets.
Last week one PSTN line died. Yesterday the 2nd Line died. I organised the Telco to divert our incoming calls to my Telecube service and within literally 2 mins of me testing that was working and informing my staff everythig is back to normal this outage occurred.
I was almost certain, the plumbing and then the electricity was going to be the next to go, just before the rubber bus arrived to take me away.
Last week one PSTN line died
Yesterday the 2nd Line died
divert our incoming calls to my Telecube service and within literally 2 mins of me testing that was working and informing my staff everythig is back to normal this outage occurred.
I was wondering who the jinx was .. now I know ;-)
I'm digging into the flow of events and will have a statement out by email through the day.
Hi John,
I have a Telecube account but didn't receive any communication. However, I have never put any credit on my account (yet).
Was it only sent to active customers?
Cheers,
Chris
Was it only sent to active customers?
It was sent to a list of users with active voip extensions. It was a list generated at the end of last month and some newer accounts may have been excluded.
Hi John,
Hope you can help me with this one,
Just set up an account for my mum remotely as she lives far from me.
Topped up with credit, Used whirlpool code, Got a DID number and setup Digital (VoIP) Extensions which are Active and online.
The thing is I tried to ring the DID number provided and it goes to someone else that i do not know and seems like the number is already in use
My mum can ring out, But i can't get the number ID when she rings me as it is set to Private.
So we have no idea what her number is and what we can ring her on.
So we have no idea what her number is and what we can ring her on
What number is set in the portal? Have you configured the ring group correctly?
Number set in the portal is the number i rang and someone picked up (not from my mums redidents) from what i could gather they were in the same region as my mum.
Please explain what the ring group is and how to set up?
Ok Worked out how to set up ring group.
But when i rang this number before shouldn't it of just said to me the number is disconnected and not rang through to a complete stranger? Not sure creating a ring group for this DID is not going to stop it ringing through to that same person.
Will give it a go tommorow.
But when i rang this number before shouldn't it of just said to me the number is disconnected and not rang through to a complete stranger?
Yes it should, please email support with your details asap so we can have a look.
New feature
I've added an address book feature in the Services > Settings section.
At the moment it creates xml for Cisco, Snom and Yealink phones but they are all much the same so it probably works for other makes too. If you need a specific format please email support and ask.
I've added an address book feature in the Services > Settings section.
Will it do a lookup on incoming calls?
Sorry I'm new to Telecube so just figuring things out.
I've set up an account with a number of extensions and have assigned a DiD to my service (thanks TC support for finding me one closer to home than the normal options).
I'd like that number to be presented when any of my extensions calls out. When I go to add a caller-id is there a reason why I'd need to authorise the number when I clearly own it in the system? I'm not home at the moment and so don't want the phone to ring announcing a number to confuse my wife. Is there a better way or do I just have to wait until I'm home and do it then?
Is there a better way
Maybe call forward the DID to a local number(mobile?) then put it back how you want it.
Is there a better way
Please email support and ask to have your DID added as caller id.
Will it do a lookup on incoming calls?
It will be easy enough to add .. just so the name can be added to the caller id when the call hits the handset? Or did you have other uses in mind?
John, I'm currently programming up a new Grandstream HT503 ATA box (we have about a dozen Grandstream HT486 which are excellent voip boxes).
The HT503 supports the (older) version G729-A & G729-B.
It also supports the latest version G729-E which can be provisioned separately.
Do your servers support G729-E ?
If so, I will provision this as the first codec to be selected.
Thanks in advance.
Anyone know the xml syntax required for address books for Gigasets (A510IP) ?
I think you'll find the info in this document. I haven't had a chance to delve into it yet.
Probably been asked before, anyway, trying to set up call forward on busy to an extension on same DID on a Yealink t22p. When it asks for the number, is it the Telecube 7 digit extension number only?
On the Yealink portal it asks for the on (and off) codes, could be *73 & *74 but not sure, anyone know them?
Is there any easy way to configure either the TC admin or Yealink phones to display INCOMING phone numbers with the leading "0" rather than "61"?
Currently showing as:
61733333333
Would like it to show as:
0733333333
Sorry if this has been asked
Is there any easy way to configure either the TC admin or Yealink phones to display INCOMING phone numbers with the leading "0" rather than "61"?
Yes, please email support with your details and I'll fix it
Yes, please email support with your details and I'll fix it
Thanks John � #198-635-832
Hi John,
I've set up a trial account and playing with the extensions before I put credit on.
I've made a call across extensions over different devices and IP's successfully, but I've tried setting up the dial shortcut on device 1 as 200 and device 2 as 201.
Now when I ring, the system says I've got no credit and the call can't be connected. I'm assuming using the dial shortcuts it is still a free call between extensions and this is just a bug in the system?
Next step, dial plans.
Now when I ring, the system says I've got no credit and the call can't be connected. I'm assuming using the dial shortcuts it is still a free call between extensions and this is just a bug in the system?
They are free calls yes, possibly a bug, I'll have a look.
Is there any easy way to configure either the TC admin or Yealink phones to display INCOMING phone numbers with the leading "0" rather than "61"?
Preference for this to be a setting rather than a manual change (same issue on Cisco SPAs). Pretty sure I have an open ticket about it.
I think you'll find the info in this document. I haven't had a chance to delve into it yet.
Thanks.
I had a read, but couldn't work out how to set up an xml file from it.
I tried a to make a couple of files, but they failed to work.
I noticed a new option "live operator answering" however it's greyed out I assume that it's for others to answer calls on your behalf and then forward/etc anyone using this yet? How would I sign up to this?
I noticed a new option "live operator answering" however it's greyed out I assume that it's for others to answer calls on your behalf and then forward/etc anyone using this yet? How would I sign up to this?
It's a new service being added, we'll have operators answering calls and just taking messages to start with but will expand it to full reception services with transferring calls etc over the next few months.
Still doing some testing but hoping to make the service live before the end of the month.
Feature Request:
Please add an option in manage DID to play initial greeting before jumping to a queue or a line hunt
I'd like to put clients onto this, but not everyone necessarily wants a 1300 number for which this feature is available.
Not sure if there is a feature request page on the wiki, or a better place to request features?
Thanks
Please add an option in manage DID to play initial greeting before jumping to a queue or a line hunt
What, an IVR?
Is it just me or the TC site is actually down?
Is it just me or the TC site is actually down?
Phones appear registered.
Portal up.
Main page up.
What, an IVR?
Nope, the IVR can't do what I want...I just want a simple initial greeting to play before the call hits the queue
DID -> Greeting "Hi, thanks for calling my business someone will be with your shortly" -> queue/line hunt
No "please press this or that"
DID -> Greeting "Hi, thanks for calling my business someone will be with your shortly" -> queue/line hunt
I would welcome this feature for After Hours incoming calls instead of going through an IVR, just rings All stations, because there is normally only one or two or none here.
By being able to play a greeting first, the staff who answer the phone after hours can still answer "yeh, what?" instead of "hi this is company xyz customer service, yeh what?"
I have tried to educate the night owls that work after 6pm that if they phone rings they need to introduce our company name first up, but thats kind of impossible.
Please add an option in manage DID to play initial greeting before jumping to a queue or a line hunt
I'd like to put clients onto this, but not everyone necessarily wants a 1300 number for which this feature is available.
I think Telecube can do this with a dummy 1300 number.
I think Telecube can do this with a dummy 1300 number.
How does one go about setting up a dummy 1300 number?
Anyway surely if the option exists for a 1300 number it's a matter of pasting in the same lines of code in the DID section?
I gather his former life did not include Car Salesman, or Politician!
Nope, I come from a fishing background though and have been known to stretch the truth from time to time. ;-)
Yup, was my initial thoughts too. We are on the same network, Telstra.
Are they on the same network (2G, 3G or 4G)?
Hi John, is there any ETA on Google Play purchases, I can't recall it being indicated?
Nothing definite at this stage .. there's work being done on an app which will include other account management functionality.
I'm trying to get a bit more detail on the in app billing and the Telstra credits to make sure there's no "gotchas" I need to look out for.
Are they on the same network (2G, 3G or 4G)?
Telstra 3G to be precise.
I think I have some DNS wierdness happening with my Win2kR2 server, may be thats causing the issue. I've changed to IP addresses for all my trunks and the PDD has improved.
My single ported DID which was from Telstra shows in 61xxx format, as does another Telecube DID
Can you let me know if this is still happening now please?
If so please email support with your details so I can look into it further.
I've made a change that should cover most services but with the current development going on we've got inbound calls routing across different parts of the platform, once everything is consolidated it should all tidy up. Hopefully by mid/end Jan 2016
Can you let me know if this is still happening now please?
Toran had also emailed re my outstanding query.
Confirm that local (02) and mobile (04) are now coming across in 10-digit 0yxxxxxxxx format.
Many thanks.
Can you let me know if this is still happening now please?
All good for me too as mobile and pstn now arrive in local format. A few routing entries to change but no big deal and now consistent with other vsps. The way you have have included DID as a separate SIP_HEADER makes extraction dead easy , thanks.
Paybe this is not the best place for this question but I will try anyway. (maybe a general VOIP question)
We have a 1300 number with Telecube which is routed to an extension as the Primary Answer Point which is then sent out to our Elastix PBX and then sent to an extension of that PBX.
All this was working fine until last week when any call to the 1300 number resulted in an engaged signal...it doesnt work.!
I have changed the PAP to a landline and it works fine. I have changed the PAP to a mobile number and it works fine. My conclusion is that all is well at Telecube. The problem must be with the Elastix PBX. I have checked the settings within Elastix and I have also rebooted the system but a call to the 1300 number always results in an engaged signal.
Any suggestions to located the problem would be appreciated.
Cheers
Any suggestions to located the problem would be appreciated.
If you email support I can take a look
We have a 1300 number with Telecube which is routed to an extension as the Primary Answer Point which is then sent out to our Elastix PBX and then sent to an extension of that PBX.
Inbound Call.
Is the extension registered to TC?
Did you re-generate passwords after after recent security alert?
Is the call arriving at your PBX?
Are you inbound routing on the correct DID... 612/02 or extension number.
Is the call arriving in the 's' context and being lost.?
Lastly... do codecs match at both ends.
As a side note Elastix doesn't appear to get much love on WP. Last time I looked it had lots of extra features like a CRM. If you are not using these Elastix only feature have you thought about a FreePBX system. Uncle Wards recent lean and mean Incredible PBX looks good.
Hey, I just noticed that my account balance is now showing on the Softphone. Was this a recent fix or have I been asleep for the last week?
Was this a recent fix or have I been asleep for the last week?
It's been about a week now yep :-)
Hey guys, all my extensions are down this morning, and calls to support get dropped � is this a general outage?
is this a general outage?
Working fine for me in Brisbane, just called my mobile and it worked.
is this a general outage?
We can make calls, but there are odd things going on in the portal. I can't see any of our extensions and loading anything is slow.
P.S. Brisbane also.
I have a Sydney client down for inbound calls, can dial out ok.
I'm off line too
Nothing registered
cant find any extensions in my portal
Something has gone wrong
I still can't see my extensions in the portal, but it is working more quickly just now.
Later: My extensions are now showing again.
Yes extensions coming back now
I suspect the reason no one is answering support is because they are furiously working on fixing the problem
Hopefully all will be back to normal soon
Has anyone been able to get the "Allowed SSIDs" feature to work in the android softphone?
My work wifi network blocks the softphone operation so I'd like the phone to use 4G when I'm at work (but still keep wifi on so that everything else uses it). I've added my home wifi network SSID to the list in settings however VoIP test calls still fail. I've restarted my device etc but still won't work.
Has anyone gotten this to work successfully?
Is the extension registration status of any value for the mobile app?
If you are using the cloud softphone app then it will be the Acrobits push server maintaining the registration while the app is in the background or closed.
If you are using the cloud softphone app then it will be the Acrobits push server maintaining the registration while the app is in the background or closed.
Yeah that is what I thought. Does the Acrobits API provide any way to see the status of the actual softphone? I'm not 100% familiar with how the push notification system works though so it might not even be possible.
Does the Acrobits API provide any way to see the status of the actual softphone?
It doesn't, it just watches for calls and sends a push notification to the phone to wake up the app when a call comes in.
Has anyone been able to get the "Allowed SSIDs" feature to work in the android softphone?
If you are asking in relation to this question, I don't think you can do what you are trying to. It doesn't seem like it's possible to force 3G/4G in the app on Android. There is an option to "Prefer 3G" in the IOS app.
If you are asking in relation to this question, I don't think you can do what you are trying to. It doesn't seem like it's possible to force 3G/4G in the app on Android. There is an option to "Prefer 3G" in the IOS app.
No my two questions aren't really related.
So what does the Allowed SSIDs option actually do then? Reading the description in the app it seems to only allow operation when connected to one of those SSIDs in the list but doesn't mention anything about cellular data. I've gone searching for a better description of the setting but have come up short. Do you guys have the user guide for the app or access to support on the app?
So what does the Allowed SSIDs option actually do then?
It's not completely clear but it appears to allow you to specify SSIDs that the app will connect through. Although that would imply that your phone would need to be connected to multiple SSIDs for it to be able to work.
The option isn't present in the IOS version.
Do you guys have the user guide for the app or access to support on the app?
Can you send an email to support and ask about the functionality of the setting on Android please and we can fire a question off to Acrobits and ask for clarification.
Can you send an email to support and ask about the functionality of the setting on Android please and we can fire a question off to Acrobits and ask for clarification.
Will do. Just from my quick testing it seems like it basically stops the phone attempting to register when on a wifi network that isn't in the list. If I switch wifi off it registers with 4G immediately. Seems like it is a bit of a blocker to stop it trying to register when on a network that it can't register through. Bit of a strange way to do it by creating a whitelist instead of a blacklist though. It'd be nice to be able to specify some networks to avoid rather than having to specify all of the ones I want it to try on.
This should be working now too, please report bugs to support.
I'm getting all calls going to the AH voicemail regardless of BH time settings.
Are BH still set in EST? (or are they EDT?)
AEDT
Thanks John � That's what I thought.
Mine is set 10:30 to 20:30 in the portal.
But when I make a call to the Queue in right now, it goes directly to VM.
It's currently 10:44 in Perth, so 13:44 AEDT � something is wrong.
But when I make a call to the Queue in right now, it goes directly to VM.
Without specific details I can't investigate .. it would be much more useful if you reported this to support instead.
it would be much more useful if you reported this to support instead.
Yeah I already did that too.
I figure if I post here too then others might check theirs and pipe up if they're having a similar issue � or not.
Without specific details I can't investigate .. it would be much more useful if you reported this to support instead.
Ticket #689-427-530
I built a new queue � just in case the original queue had some sort of defect.
But the new queue does the same � all incoming calls go straight to the AH VM no matter what times I set for BH.
New Feature
You can dial direct to a Queue (Ring Group) if it has been added since 1st Dec 2015 and has an id of 504xxxxx or greater.
Hi John � is this still working?
Extn to queue is currently failing for me.
I have an Avaya IP Office 500. Want to call out on tc. Have purchased an avaya sip license for 5 channels (Existing large installation not cost effective to change systems)
Has anyone configured one of these for tc? Trying to save a lot of reading and the frustration of being locked out by the tc firewall for an hour every time i get the settings wrong.
Trying to save a lot of reading and the frustration of being locked out by the tc firewall for an hour every time i get the settings wrong.
If you have a static IP please email support and we can whitelist it so it doesn't get blocked.
Quick question, set my account up tonite and configured my Fritz Fon, outgoing call on my DID works fine, when I try to call the number from my mobile I get a message that the number has been disconnected!?
EDIT � sorted thanks to the wiki, had to configure the ring group.
when I try to call the number from my mobile I get a message that the number has been disconnected!?
See the "Important Notes" here:
http://wiki.telecube.co
Configure your DID.
A DID is a virtual number so it needs to be connected to extensions or features to receive calls.
To set an answer point for the DID, set its profile type to Call Forward. Then, just input the phone number or digital extension you'd like to route the calls to.
New Feature
You can dial direct to a Queue (Ring Group) if it has been added since 1st Dec 2015 and has an id of 504xxxxx or greater.
Hi John,
Will you be looking to add direct dial to Linehunt also? :)
Thanks John, could you get someone to follow up 526-675-292 then
all i wanted was the call log looked up to find out why it was terminated, and its been nearly a month and still havent got that response.
all i wanted was the call log looked up to find out why it was terminated
That would require sip logging which we don't keep I'm sorry, we just keep the general asterisk log output.
That would require sip logging which we don't keep I'm sorry,
Ok but I am at my wits end over this. My boss is busting my backside because she thinks your service is not reliable (most of it not substantiated and she is paranoid that all voip services are unreliable) and wants me to dump your service. Seemingly half the calls we Park are subsequentially lost, and I have no idea what my colleagues, or the caller is doing.
I have tested the parking procedure and pickup up over and over and it works everytime, including ring back to the original station after 45 mins.
In my view, either the caller has hung up or the colleague has pressed wrong button and hung up on the caller somehow, and I think the only way to tell with any certainty is to inspect the log.
and I think the only way to tell with any certainty is to inspect the log
Yep, unfortunately there isn't a log to inspect here. Apologies.
All of my IAX extensions are failing to register today. Is anyone else having problems?
The app is mostly built but I'm not sure whether or not we will be going ahead with it
I see...
That's unfortunate, would've been handy to have.
John, I have found the fault with call parking.
When the parked call timeouts, it tries to call the original station that parked it. If that station is busy the call terminates.
Also the timeout length of 45 seconds is too short, thus causing the above to happen often. Usually because the 'Parker' is introducing the call and providing some details to the 'parkee'.
Understandable that 45 seconds is a long time to be on hold, but we shouldnt be penalising the caller :)
When the parked call timeouts, it tries to call the original station that parked it. If that station is busy the call terminates.
Hi John, this issue can be easily be fixed by adding something like this to extensions.conf, to send it somewhere after the default fails due to the extension being busy...
[park-dial]
exten => _SIP_.,2,Goto( ...send it somewhere useful... )
Hi John, this issue can be easily be fixed by adding something like this
Yep, the backend here is a bit different than the standard asterisk dial plan but yep .. I'll have a look at handling it a bit more cleanly and also give an option to manage the timeout in the portal.
Quick question on codecs...if I want HD calls between extensions should I put G722 as my first priority codec and G711 subsequently?
Yep, the backend here is a bit different than the standard asterisk dial plan but yep .. I'll have a look at handling it a bit more cleanly and also give an option to manage the timeout in the portal
appreciate the effort but please prioritise to change the default timeout immediately to something like 2 mins at least. We are still losing about 50% of the parked calls.
The most recent one was rather comical, she took 44 seconds between parking the call, and then dialling the bosses extenstion, seems she couldnt find her phone list. Then by the time the boss dialled 501 to pick up the call, the customer had already called back and was talking to someone else already.
sending the parked call somewhere useful after timeout is a less of an issue if the timeout is long enough.
appreciate the effort but please prioritise to change the default timeout immediately to something like 2 mins at least.
I've set the timeout to 120 seconds now
The most recent one was rather comical, she took 44 seconds between parking the call, and then dialling the bosses extenstion, seems she couldnt find her phone list. Then by the time the boss dialled 501 to pick up the call, the customer had already called back and was talking to someone else already.
Don't think that's Telecube fault. If customers are hanging up within 45secs they are certainly going to be hanging up within 2min.
I've set the timeout to 120 seconds now
Thanks
Don't think that's Telecube fault. If customers are hanging up within 45secs they are certainly going to be hanging up within 2min.
No customers are hanging up, You have read it wrong. /forum-replies.cfm?t=2450974&p=45#r887
Also pretty small size, can do conference calls, and shows codec and bandwidth in use live.
Worked quite well.
FWIW. I think this Telecube special offer for WP members is one of the greatest things since sliced bread!
The "try before you buy" feature was the clincher for me. I've now added some money and got a "local" DID.
As an aside, a friend that I told about this got a poor/useless result because he is stuck with that horrible interim NBN satellite internet. Not a criticism of Telecube, just a fact of life, interim NBN satellite internet has too much latency for usable VoIP!!!
interim NBN satellite internet
Yep it's quite normal that voip over a satellite connection just doesn't work.
A 3G/4G mobile connection would probably offer a reasonable solution if he has access to it, we have a number of customers connected like that.
Gotta look at the telecube website.
Best to look at the Telecube Whirlpool offer. You won't find this on the Telecube website.
Also, remember that Telecube will give you up to $10 of your existing Siptalk credit, or up to $30 if you port a DID. See here. Take a screenshot of the Siptalk portal page showing your balance.
Seriously considering switching. Gotta look at the telecube website.
Look at the rates at the start of this thread.
There's some news coming .. watch this space ;-)
There's some news coming .. watch this space ;-)
Woohoo.... we have another entrepreneur :P
So, there have been a few customer (existing Telecube too) that have emailed, please email in again with your account balances and we'll honour the losses as originally offered.
Even if you're an existing Telecube customer and no DID to port, email again please and we'll add up to $10 credit to your account.
I'm one of these. I've had an account with SipTalk and another with Telecube for the last couple of years. I've made 99% of my calls through SipTalk (configured my phone this way purely because SipTalk was a "locally based" business) but the support I've had from John for the paltry few dollars I've spent with Telecube has been absolutely outstanding.
I'll be writing off the credit I have currently with SipTalk and transferring my business to Telecube without any incentives, simply because they have earned it. I'd encourage others to do the same.
News
Telecube has just executed an agreement with Summit to purchase the Siptalk business.
We'll be maintaining the pricing and honouring the credit balances.
An email to customers will be going out very soon.
Great news for everyone concerned :)
Telecube has just executed an agreement with Summit to purchase the Siptalk business.
We'll be maintaining the pricing and honouring the credit balances.
An email to customers will be going out very soon.
Thanks to John for working so swiftly on this transaction.
A fantastic outcome for all.
This just goes to show the power of Businesses working together for an outstanding outcome for the customers.
:)
Greg
Thanks Greg, cheers.
Great outcome, well done John!
How many more customers will you be picking up with the SipTalk purchase?
Is your platform and hardware easily scalable, or do you think major infrastructure upgrades would be required to integrate the additional customers?
upgrades would be required to integrate the additional customers?
It's a small percentage of what we are already carrying, there'll be no need to upgrade to cater for them.
Telecube has just executed an agreement with Summit to purchase the Siptalk business.
Great news John. So does Summit end up buying anything?? Just the stock?? Or are you simply buying the 'budget' list as referred to by Greg on the SipTalk thread??
We're taking over the VoIP business. Siptalk and OnPBX
I can't speak for Summit but I believe their primary interest was the PC Range assets.
Telecube has just executed an agreement with Summit to purchase the Siptalk business.
So apart from picking up a few extra customers, do you get any technology or infrastructure that can be incorporated in your existing system to make it better?
We don't use the Fax very often. But when we do we really need it, any idea when it will be available?
** NOTICE **
We apologise for the inconvenience caused, however the FAX SENDING service is currently not available.
A restoration date for the service is not known at this stage unfortunately.
You will be notified once the service resumes.
It is recommended that you use an alternate means of sending faxes at this time.
Apologies again for the inconvenience.
Regards,
Telecube Support
Telecube has just executed an agreement with Summit to purchase the Siptalk business.
Great news John.
So apart from picking up a few extra customers, do you get any technology or infrastructure that can be incorporated in your existing system to make it better?
It's a bit early to say at this stage, we're still feeling our way through the systems
Thanks so much John, I'll be transferring my credit from SipTalk ASAP. =)
Thanks to John for working so swiftly on this transaction.
A fantastic outcome for all.
This just goes to show the power of Businesses working together for an outstanding outcome for the customers.
It may be a fantastic outcome for Siptalk customers who now don't loose their credit.
I don't see how it is of benefit to existing customers of Telecube who will now have to fund the Siptalk losses.
Either rates will need to go up or rate reductions will be postponed. The money has to come from somewhere.
I may be getting cynical in my old age but it now seems rather common for Telcos to provide benefits for new customers that are simply not available to existing customers.
Such is life.
S.
I have mine setup for whitelist only.
Essentially, if somebody calls with a caller id not in my address book, or no caller id, the call goes straight to VM.
All other calls (numbers in my address book) will ring the phone.
This was a little bit of work to setup initially but it works really well now.
Obviously, this would be no good for a business, but for my house number it works brilliantly.
I know not all ATAs or VoiP phones are capable of this so perhaps this is something that could be implemented on the TC side.
So essentially a whitelist of known numbers, and all other calls get recorded message to enter a pin. Failure to enter pin the call would then be rejected. Sounds like something I would use, sign me up!
Yep exactly that.
Essentially, if somebody calls with a caller id not in my address book, or no caller id, the call goes straight to VM.
I know not all ATAs or VoiP phones are capable of this so perhaps this is something that could be implemented on the TC side.
Yep it's something we could add as a feature.
Even without ports forwarded the calls can get through. The modem retains routes for the phones, usually on specific ports and if an incoming invite hits the right port the modem allows the traffic through to the local device.
Decent routers use stateful firewalls which stop this happening.
What you need is caller number recognition and a prompt to enter a pin for unrecognised numbers. Failed pin entry sends the calls to vm.
For those who run a PBX, FusionPBX already provides many options to deal with such situations:
- Call screening
'Anonymous' callers will be asked to say their name which will be played to the callee. Callee then needs to press 1 to accept the call or the call will be sent to voicemail. - Call screening using a whitelist
Please refer: whrl.pl/RewRAa
Note this can be implemented on any FusionPBX installation be it on arm,intel/AMD.
/archive/2454187 - Inbound route
You can add a 'condition' to filter calls using the network address (IP address) of your provider. Calls coming in from other addresses would go to the next route. You then add a final route for all calls to hang up. This is done in the GUI using a few mouse clicks. - Internal and external SIP profiles
External profile runs on port 5080 and by default doesn't accept registrations. If you have valid credentials including the internal domain, it is possible to register. Internal profile will only accept LAN clients.
im using Siemens Gigaset A510. im not hearing any ringer or ringtone whenever i dial out.
i tried calling a mobile number and even international number, no ringer sound just suddenly the other side picks up.
when I was with iiNet Netphone though it was fine. is this a setting that needs to be set?
im not hearing any ringer or ringtone whenever i dial out.
I just made a call to WA and another to Vic and on both occasions I heard part of the telstra double ring tone then a couple of seconds silence with empty rtp before the called party answered. What you describe can happen when the vsp doesnt bridge you to the telstra network until the call is answered. TC use 183 session progress which enables you to hear the telstra tones as soon as connection is made to the pstn. Ring your own mobile and dont answer and see if you hear the telstra ring tone.
Anyone having call failures today? I have a ticket lodged (764-555-932), and Toran is looking into it, but I thought I'd see if anyone else is having issues.
It happens at random times. Any number called from my Yealink handset says "Forbidden" call failed. And it doesn't connect. Sometimes rebooting the phone helps...sometimes not. I have two extensions on my handset and it's only doing them on one of them.
One of my other users in Sydney just reported the same issue and she is on a totally separate network and the only user down there.
Anyone?
Hi John,
Is it just me or is there a problem updating call queues. I click on the extensions I want to add and it comes up "There is an error" when I try to submit it. I deleted my call queue and tried again and was able to enter the extensions the first time and then thereafter it comes up with the error if I try to edit it. From what I can see it looks like it putting a blank line in at the bottom so when you try to update the queue it is getting an invalid extension or something. It doesn't happen when you first add new extensions to a new call queue.
Is it just me or is there a problem updating call queues. I click on the extensions I want to add and it comes up "There is an error" when I try to submit it.
Are you using Chrome? I had this issue as well.
Ahhh....that's exactly what it would be. Thanks!
I am just getting back into VOIP, waiting for ATA to arrive and signed up with Telecube. One thing that confuses me though... the website says:
"No Emergency 000 Calls
Telecube voip service is not a replacement for your fixed line telephone."
But the Telecube page on this site says:
"000 Support: Yes. See /forum-replies.cfm?t=2323366&r=47110200#r47110200"
I'll be looking to use this as my only phone line after we get the NBN so it needs to definitely have emergency number support.
I'll be looking to use this as my only phone line after we get the NBN so it needs to definitely have emergency number support.
000 calling is supported but we can't guarantee that your address will be automatically known to the operator. Being a voip service you may need to confirm or tell them your physical address.
To set it up properly you need to have a DID on your account that is used as caller id on outgoing calls and you need to advise us of your physical address so we can update IPND
Thanks John. Looking forward to trying out your service.
John, is IPND updated from the address that is contained on the accounts page. I just updated mine as I thought it would be good to have a physical address rather than a PO box for this purpose. We have multiple caller id's go out so would IPND update each and any caller ID from the account address? This is fine for my purposes but just wondering how it works. If you have multiple sites, would the best way be to have separate accounts for each different physical site for this purpose? Cheers
You won't be disappointed Delgesu :)
is IPND updated from the address that is contained on the accounts page
It's not, we are updating manually at the moment and need you to email support please. The address will be matched with a DID.
If you have multiple sites, would the best way be to have separate accounts for each different physical site for this purpose?
IPND details are matched to a DID and we will add functionality to the portal to allow you to specify addresses per DID, for the moment please email support with details and we'll handle it manually.
Anyone having call failures today? I have a ticket lodged (764-555-932), and Toran is looking into it, but I thought I'd see if anyone else is having issues.
Yep, had the same thing happen to me.
1 extension refused to makes calls
I upgraded the yealink firmware to the latest and it started working again.
Not sure it had anything to do with the upgrade, could have just been coincidental timing.
It happens at random times. Any number called from my Yealink handset says "Forbidden" call failed.
I'm getting that now between extensions on different accounts. Only happens one way from a gigaset to a yealink.
There should be Perth numbers available now. Sorry for the delay.
Kudos to Telecube !! Have just moved over from Adam Internet's AdamTalk, and John M has been very responsive to my questions, and Murray in Support spent the greater part of an hour on the phone with me this evening talking me through the setups. Now have my Gigaset configured, my mobiles connected and all working beautifully. Many thanks guys!!
Kudos to Telecube !!
Yes, I agree. I've just got my new ATA working and trying out Telecube. Excellent quality � no-one in the household has even noticed the changeover from landline to voip. Had a few teething problems getting my ata working and Telecube support were quick to respond via email.
This is possibly already a feature � but if not then a feature request =)
Is multi-tenancy supported? Reason being if I put a few businesses on there with 10 or 15 extensions each � it becomes quite a long list of extensions � expand that to 5 or 10 and it's a little bit out of control; so a way to either group or separate blocks of extensions to achieve a multi-tenancy would be great.
Also if a customer is given the internal extension numbers they could easily begin dialing other businesses accidentally/on purpose causing confusion. (I know people call wrong numbers a lot but this is a little bit different).
Cheers
Can you guys handle International Inbound Number?
We have 2 phones in the office on Maxo (the rest traditional landline) and have a NZ number redirect to our office in Melbourne for our NZ customers.
We are looking at moving it all over to Telecube, but the international redirect is pretty important.
I am only just learning about VoIP so sorry if this is a basic question.
Can you guys handle International Inbound Number?
Yep, NZ, UK or USA numbers is no problem we have direct carrier links.
Yep, NZ, UK or USA numbers is no problem we have direct carrier links.
Are you saying you provide DIDs in those countries?
Guys just a quick one. Have the numbers on the Wiki to dial into voicemail to change a greeting changed. I keep getting a disconnection message when I dial these numbers? Thanks
John, just in relation to the above DID's, what is the cost for these in each of the above countries. I sent an email to support regarding this also, but thought it might be useful information for the Whirlpool group also. Cheers
what is the cost for these in each of the above countries
US and UK geographic DIDs are AUD$2.95 per month plus AUD$0.0055/min routed into voip extensions.
NZ geographic DIDs are AUD$11 per month plus AUD$0.022/min routed to voip extensions.
Please note these prices may be subject to change
Standard voip pricing will apply for diversions to fixed or mobiles or international destinations on top of the per minute inward cost.
At this stage I don't have any special pricing for the WP offers but when I get a chance I will have a look into it.
Just curious how long it takes to get a phone delivered (to Perth). I ordered a Yealink T46Ga few days ago, it's still in 'processing'. I'm expecting about 2 weeks or so, would that be about right? ta
Just curious how long it takes to get a phone delivered (to Perth).
It will be best to please email support and ask. Murray will chase up and let you know what the status of the order is.
It will be best to please email support and ask. Murray will chase up and let you know what the status of the order is.
Will do John, thanks.
Hi John- I am looking to set our church up with your services following the generally good feedback and the availability of a reliable internet connection.
I tried to apply the WP link to the account and I got an error re access to a db server.
Am I doing something wrong?
Cheers,
Damien.
Am I doing something wrong?
No, that error does need attention.
Connection failed: SQLSTATE[HY000] [2005] Unknown MySQL server host 'db.billing.telecube.net.au' (2)
Thanks for the DID info John. Will contact you via email to arrange a couple of international DID's.
Just a quick one. I sent a fault into support this morning. As of about lunch time yesterday, I can't seem to manage my DID's in the portal. It just sends back a message that there is an error. It occurred yesterday around lunch time when I was doing some changes. It was working and then for the rest of the day and this morning, I keep getting the error. It occurs in Chrome, Firefox and IE, tablets and iPads etc. Also the services pages cuts off below the DID's and doesn't show the Line hunts, Queues etc etc. You may know about it but just wondered if it is just me or others are having this issue to.
but just wondered if it is just me or others are having this issue to.
There is a glitch somewhere...
I went in to the portal and all I get in services at 'VOIP Extension' is 'Jump to'.... and an up/down arrow, can't select an extension. None are visible/selectable.
Will be fixed soon I'm sure.
Edit: Just saw that all the log-ins are not listed either at the bottom of the page.
I got an error re access to a db server
This should have been resolved, please try again and let me know if you have the same problem,
I can't seem to manage my DID's in the portal
This should have been resolved too, part of the db connection issue reported above, please report back if not.
I went in to the portal and all I get in services at 'VOIP Extension' is 'Jump to'.... and an up/down arrow, can't select an extension. None are visible/selectable.
This too ..
Please let me know if they aren't resolved.
All working fine now John, thanks.
a man in his mid-forties
You're not being very kind to John. : ) I think he doesn't look a day over thirty.
Hey John, what's up with that Google + picture? If you Google "Telecube" it comes up with a strange picture of a man in his mid-forties, not sure what's up with that, just seems a bit unprofessional...
Same pic on https://au.linkedin.com/in/telecube
Have the numbers on the Wiki to dial into voicemail to change a greeting changed.
Yes, I have received an email from Telecube Support advising that the in-dial voicemail numbers are currently not being supported. You will need to record your voicemail locally & then upload it.
Just trying to verify my card for auto top up and it said that I need to confirm a check transaction of $1.xx or some such. When I went to my internet banking, this transaction did not take place. So now I cant confirm my card.
Worse thing is I just put 00 and that has cancelled the card and now I have to contact support. Ho hum...:(
When I went to my internet banking, this transaction did not take place. So now I cant confirm my card.
It can take 3 � 5 business days depending on your bank.
Worse thing is I just put 00
Why?
now I have to contact support
When you do see the transaction on your statement send an email to support with the amount and your customer id and we'll reset the allowed auth attempts and you can enter the correct amount.
OH I see. I can't recall reading that it can take 5 business days. I tried registering on Friday and thought it would be instant. I only looked today and assumed there would be no transaction.
Anyway, my bad.
I can't recall reading that it can take 5 business days.
Actually depends on your bank. If you use a major bank (with one notable exception) it only takes a few hours for the "magic" $1.xx DR to show in your account. "Which" bank you use will determine the additional delay and some of the pseudo-banks take ages. This is not a fault of Telecube or their bank. G.
No not blaming anyone. I just assumed it will be there but wasn't.
In the Telecube My Account area how can we set our own password for a SIP account.
We can see the option to generate a new password but not to be able to select our own password.
Probably safer to not let users enter a password manually so users don't enter basic passwords.
We can see the option to generate a new password but not to be able to select our own password.
Sorry, we don't allow customers to set extension passwords. The only reasons I can think of for wanting to set your own extension password would be so they are memorable or less complex, both are really bad.
If you can give me a good reason I'll consider it.
If you can give me a good reason I'll consider it.
Accidentally set an extension to inactive but then needed it enabled again but when I did the password reset to another password so want to change it back to the exact previous password that was generated as these users VoIP phone is in another location and they aren't the most tech savvy so is a major task to have them reconfigure the phone.
want to change it back to the exact previous password
If you set that password as label for the extension in the portal and email the details to support I'll change it back for you.
Don't email the password please, just add it to the extension label and I'll go get it from there.
I signed up with DidLogic a few days ago, topped up my account and had the top-up refunded without explanation. Sunday afternoon I came across Telecube when I was reading a thread about DL's terrible customer service here on whirlpool, and I have to say, it's bloody fantastic! We're using it for our business � in minutes I've got 20 DIDs going, voicemail, call queuing and other handy features. Now I'm recording our IVR and that's pretty much it.
I've found the call quality to be fantastic too. Very happy with the service, and a huge thank you for the big wp discount.
Apologies in advance if this is a daft question (as I've read over the Telecube wiki and suspect I know the answers but just want to double check).
The Telecube WP offer looks fantastic......we're just about to move house....my plan is to get NBN FW via Skymesh and use Telecube VoIP for all phone calls.
For hardware I'd be using a TP-Link-WR841N router and an Android mobile phone as the dedicated 'home phone' to use Telecube through (with a DiD as well).
Is that viable & likely to be a reasonable solution (assuming all unknown variables are ok)?
If I've missed anything please let me know as the Telecube offering looks a LOT better than the bolt-on VoIP offers any NBN FW providers are supplying.
I'm a lil unsure if the older Android phone would be an issue.
Thanks in advance for your feedback. :-)
Is that viable & likely to be a reasonable solution (assuming all unknown variables are ok)?
It should be fine and you can easily use different handsets at a later date anyway if there are any issues with the older Android phone.
We can provide the NBN connection for you too if you are interested, we're keen to do new business and will waive install fees and contracts at the moment.
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