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martin
06-09-2006, 12:48 PM
Hi,

Could you please post your current problems with VoXaLot and any requests.

Thanks

affinity
06-09-2006, 01:22 PM
Okay, first request:

- allow forward to mobile or fixed line number via one of our providers.

Second request:
- give us feedback as to the status of registrations.

Third request:
- fix speed dials for voxalot.com.au

Is that enough, too much?

Thanks very much for all your efforts Martin and team.

AndrewM

richard
06-09-2006, 02:12 PM
Hi Martin

Voxalot support for vBuzzer ;-)

Currently one-way audio (Voxalot registered ATA can hear caller but caller can't hear sound coming from Voxalot registered ATA).

Don't know if this info helps:

http://www.vbuzzer.com/forum/viewtopic.php?t=513&highlight=asterisk


Richard

BBD
06-09-2006, 03:47 PM
Second request:
- give us feedback as to the status of registrations.


i must agree with this one. also, it would be great if we could get BBP Global to work. i know your trying and i am very appreciative.

hrmm apart from those two, i cant think of anything else just yet, but if i do, ill make sure i let you know.

ataboy
06-09-2006, 04:43 PM
1. Voipfone works in dialplan for outgoing calls. Although it appears to be registered for incoming calls, it isn't working (and never has) for me.

2. I get no Caller ID at all on incoming calls to any of my registered VSPs.

ataboy

Ron
06-09-2006, 08:43 PM
Voxalot support for vBuzzer ;-)

Currently one-way audio (Voxalot registered ATA can hear caller but caller can't hear sound coming from Voxalot registered ATA).


Richard,

I have Vbuzzer set up on VoXaLot as a SIP registered provider and have no audio problems on outgoing calls (I don't have a Vbuzzer incoming number so I can't test that direction).

One-way audio problems are usually related to NAT issues on the user's router. You might want to confirm this by temporarily connecting your ATA directly to your modem so that no router is involved. If the problem goes away, you will know the problem is on your end. NAT issues can usually be solved with appropiate settings in the router and/or ATA.

Ron

Ron
06-09-2006, 08:47 PM
2. I get no Caller ID at all on incoming calls to any of my registered VSPs.

I can confirm this problem. CallerID is displayed properly on calls forwarded to VoXaLot (i.e sip:xxxxxx@voxalot.com), but CallerID is lost on incoming calls via SIP registered providers.

Ron

richard
06-09-2006, 10:20 PM
Thanks for your suggestions Ron.

Yes, my OUTBOUND calls via vBuzzer through Voxalot (SIP registered Provider) works fine too. No problem with audio. Its the incoming calls on the vBuzzer PSTN that doesn't work.

I tried putting the ATA into the DMZ and still the same one-way audio problem and with your suggestion I connected the ATA directly to my cable modem (eliminated the router) and I got the same problem. One-way communications. This is the reason why I think its a Voxalot/Asterisk issue.

I'm not sure that making any other changes to my router will help, but if you can think of any, please send it my way. All other VSPs work fine.

Take care

Richard

Richard,

I have Vbuzzer set up on VoXaLot as a SIP registered provider and have no audio problems on outgoing calls (I don't have a Vbuzzer incoming number so I can't test that direction).

One-way audio problems are usually related to NAT issues on the user's router. You might want to confirm this by temporarily connecting your ATA directly to your modem so that no router is involved. If the problem goes away, you will know the problem is on your end. NAT issues can usually be solved with appropiate settings in the router and/or ATA.

Ron

xcaliba
06-09-2006, 11:01 PM
I don't know if it is possible but can we have multiple extensions register to one account.

Those of us with a spa2000 (or other) can then use both lines incomming so when an incomming call comes in both phones ring and if one is busy then the other phone will still ring.

Ron
06-09-2006, 11:30 PM
with your suggestion I connected the ATA directly to my cable modem (eliminated the router) and I got the same problem. One-way communications. This is the reason why I think its a Voxalot/Asterisk issue.

Richard,

That certainly eliminates your router (and any NAT issues it might have).

I wish I had an incoming DID at Vbuzzer that I could test for you also. Given that I don't, I'm not coming up with any additional thoughts at the moment.

Ron

Ron
06-09-2006, 11:38 PM
I don't know if it is possible but can we have multiple extensions register to one account.

Those of us with a spa2000 (or other) can then use both lines incomming so when an incomming call comes in both phones ring and if one is busy then the other phone will still ring.

Multiple VoXaLot SIP registrations are permitted and appear to work exactly as you (and I) hope they would.

I've had both channels of a PAP2 registered to the same VoXaLot number for some time. Incoming calls alternately ring both extensions. The first one picked up gets the call. As far as I can tell, the two extensions can be used in any combination for incoming and outgoing calls.

Ron

ctylor
06-10-2006, 12:22 AM
Multiple VoXaLot SIP registrations are permitted and appear to work exactly as you (and I) hope they would.

I've had both channels of a PAP2 registered to the same VoXaLot number for some time. Incoming calls alternately ring both extensions. The first one picked up gets the call. As far as I can tell, the two extensions can be used in any combination for incoming and outgoing calls.

Ron

Thanks Ron, I didn't know that and that is good information to know!

I would like to report to Martin that three-way calling thru Voxalot is still completely broke and doesn't work.

Ron
06-10-2006, 12:55 AM
I would like to report to Martin that three-way calling thru Voxalot is still completely broke and doesn't work.

Three-way calling works fine for me. It has since the beginning and I just checked to make sure it still does. Martin may have to correct me here, but I believe three-way calling has more to do with the ATA than it does with VoXaLot. Also, for three-way calling, if you're using the same provider for both legs, that provider would have to support two simultaneous outbound calls.

Ron

ctylor
06-10-2006, 01:15 AM
My three-way calling problem uses the same provider--Voxee--for both legs of the call (the provider permits two simultaneous outbound calls), and when my ATA is registered to Voxalot, the second channel call will experience a long pause after the dial string is sent and then eventually a fast busy. When my ATA is registered directly with the provider, three-way calling works fine and absolutely normally. Changing the provider in my Voxalot dial plan from Voxee to Teliax doesn't help, Teliax can't complete the second leg of the call either when done through Voxalot. By itself it can do three-way calling just fine. I am perfectly willing to entertain speculation that it is the way my ATA (a Sipura) is configured that is creating a conflict with Voxalot, but I tend to see it as more of a Voxalot bug since I would never even have known I had a 'three-way call' problem if I never had used Voxalot.

I'd be happy to send you Ron, or Martin, a copy of my Sipura admin voip settings so you can help troubleshoot. Just let me know by private message.

BBD
06-10-2006, 02:03 AM
how do you use three-way using voxalot/voip? my provider allows multiple outgoing calls and i have a SPA3000

Ron
06-10-2006, 04:18 AM
how do you use three-way using voxalot/voip? my provider allows multiple outgoing calls and i have a SPA3000

While talking to the first party, press the FLASH button on your phone. The first party will be placed on hold and you will get a new dialtone (often slightly different from the normal one). Place your call to the second party. Once they answer, press the FLASH button again and all three of you should be connected.

Ron

BJReplay
06-10-2006, 06:32 AM
I used to be able to have incoming PSTN calls handled by voxalot voice mail, but no longer can.

I can confirm that this still works with voxalot.com, but not with voxalot.com.au.

My Setup:

SPA-3000
Line 1 Registered to Voxalot.com.
Line 1 Voice Mail Server: $USER@voxalot.com
User 1 Cfwd No Answer Dest: $USER
User 1 Cfwd No Answer Delay: 12 (seconds).
PSTN Ring Thru Line 1: Yes


If I call my PSTN, the SPA-3000 rings Line 1 as a PSTN Ring Thru. After 12 seconds, Line 1 forwards the call to my voxalot account, and Voice Mail answers immediately. All Good.

If I change my registration to voxalot.com.au (and my voice mail server to $USER@voxalot.com.au - but of course this isn't necessary to test this - rather this is required for MWI & VWMI if calls ever got through to VM in the first place), then after 12 seconds, the phones connected to the FXS port (Line 1) stop ringing for a moment, but start ringing again.

I can swap back to Voxalot.com and voice mail answers.

Looks like voxalot.com.au doesn't detect that the call is being forwarded by the same account as the destination, and I guess that after the brief pause, the phone starts ringing because I get a SIP invite from voxalot. I can trace this if required to debug.

ciao

BJ

fozzie
06-10-2006, 09:46 AM
I can confirm this problem. CallerID is displayed properly on calls forwarded to VoXaLot (i.e sip:xxxxxx@voxalot.com), but CallerID is lost on incoming calls via SIP registered providers.

RonI do get CallerID from my Sipgate DID, with Sipgate registered with Voxalot.

fozzie
06-10-2006, 09:49 AM
As well as supporting the request for some sort of SIP registration status, I have a couple of other requests.

1. User definable Voicemail answer delay.
2. Call logs.

Many thanks.

moollar
06-10-2006, 10:58 AM
The only suggestions I can think of at the moment are the ones I mentioned in regards to voicemail here: http://forum.voxalot.com/showthread.php?t=204

I would probably have more, but I have the flu at the moment and can't quite think straight. :confused:

affinity
06-10-2006, 12:25 PM
Yes, call logs would be an excellent addition too, keeps the VSPs honest.

JohnnyGolden
06-10-2006, 02:27 PM
The only issue I have had is with the site itself - often need to enter my e-mail and password twice to successfully log on to the control panel. Not a biggie though...

As far as feature requests go, I'll throw my hat in with the registration status crowd. Also would be great to be able to disable voicemail, and to specify our own voicemail extension number (instead of the default 500).

Other more glamorous ideas:

- Call forwarding via dial plan, i.e. ability to set up a call forward to any number (including PSTN) which would be passed to the dialplan to process.
- Similar functionality for Web Callback - option to route call legs via dialplan defaults rather than specific providers

richard
06-10-2006, 06:50 PM
Hi Ron

You can use my vBuzzer account with a PSTN number if you want. It would be helpful. I sent you a Private Message via Voxlot Forums with details.

Richard

Richard,

That certainly eliminates your router (and any NAT issues it might have).

I wish I had an incoming DID at Vbuzzer that I could test for you also. Given that I don't, I'm not coming up with any additional thoughts at the moment.

Ron

fozzie
06-11-2006, 08:22 AM
The only issue I have had is with the site itself - often need to enter my e-mail and password twice to successfully log on to the control panel.That'll be an issue with your browser/cookie handling. No problems here.

Jorge
06-11-2006, 06:45 PM
I don't know if something like this could be done:

For configuring DDIs in easypbax.com (similar to registered providers in Voxalot) there is a field named "PeerString" which is for passing some extra settings to a provider:
# PeerString: If you type somthing in this field it will be added at the end of the SIP peer configuration in sip.conf ex: dtmfmode=info - you can add several lines here.
For what I see from this forum, some providers can't be registered in Voxalot because they have unusual requirements, such as caller ID and outbound proxy.

Maybe some providers which seem to be Voxalot incompatible, like Voicestick, could work if there was a way to add some lines like this. I could really use a local number in Los Angeles from them.

(Unlike Voxalot, I could never get easypabx to work, thanks for this great service!)

reader
06-11-2006, 11:15 PM
My problem is I 'always' get a busy tone (beeping once per sec.) on first attempt at dialing. Mentioned here -
http://forum.voxalot.com/showpost.php?p=1430&postcount=1
I tried getting rid of all dial plans except one that let everything through (_.)
Tried different phones, cold reboot of modem/ATA. Rechecked voxalot registration settings.

Strangely, yesterday when the voxalot website went down this problem disappeared and dialing worked first try. I would have dialed over 20 times through the day so it was no fluke. The problem returned at close to the same time the website returned (maybe exactly). [edit: This was after a week of consistent time with the problem]

I guess the best thing my end would be to try 'set to factory default' of ATA or try another ATA if I can get hold of one. Seeing the problem resolve for a while yesterday doesn't fit with that though.

Ron
06-12-2006, 12:07 AM
Strangely, yesterday when the voxalot website went down this problem disappeared and dialing worked first try. I would have dialed over 20 times through the day so it was no fluke. The problem returned at close to the same time the website returned (maybe exactly). [edit: This was after a week of consistent time with the problem]

Do you have voxalot.com or voxalot.com.au set as your prefered server on the VoXaLot Member Details page? You should have your ATA registered with whichever one you have selected there. When www.voxalot.com was down yesterday, so was the voxalot.com SIP server (but voxalot.com.au was not). It seems pretty clear that you're tangled up between these two servers somehow if operation is normal when voxalot.com is down.

Ron

Oigle05
06-12-2006, 01:21 AM
I can't dial 1800 numbers via voxalot. When I do the test it shows that it will be directed via the chosen vsp but when a real call is made, I only get busy tone. Tried 4 different vsp's and all were the same. 1300 numbers are in the same dial plan as 1800's and they work ok.

martin
06-12-2006, 01:36 AM
Hi,

It is now on the to-do list. We will look into this soon.

affinity
06-12-2006, 05:25 AM
I can't dial 1800 numbers via voxalot.
Try prefixing the numbers with 61 or stopping ENUM from resolving to a US number... using advanced dial plans.

reader
06-12-2006, 06:17 AM
It seems pretty clear that you're tangled up between these two servers somehow if operation is normal when voxalot.com is down.
Ron
Thanks Ron. That does sound like the right track but both the member details and the sip registration in the ATA were/are both set to voxalot.com.au

SIP Proxy: voxalot.com.au
SIP Proxy port: 5060
SIP Proxy Domain: voxalot.com.au
Register Expire Time: 3600
VoIP Phone Number: 523004
Auth. ID: 523004
Auth. Password: ******

Member details -
Preferred Server (for SIP registrations): voxalot.com.au
I'm using Pennytel and Sipme and have tried registering my voxalot account to Pennytel but it makes no difference if registered or not.
Also made up a new voxalot account (my sister will use web callback with it) and tried that with same result.

We can survive of course registered direct to the VSP without dial plans but only just.. now I've seen what dial plans can do!

Oigle05
06-12-2006, 06:32 AM
Try prefixing the numbers with 61 or stopping ENUM from resolving to a US number... using advanced dial plans.
Tried 61 before number - same result. The ENUM lookup has 61 before the number and if I remove it the test shows it being sent to an American number, so the 61 is replaced, 1300's ok, 1800's busy!!:confused:

Ron
06-12-2006, 07:09 AM
That does sound like the right track but both the member details and the sip registration in the ATA were/are both set to voxalot.com.au

Reader,

Do you have the same problem if your prefered server and ATA are set to voxalot.com? Do you have access to another ATA (especially another model) or a softphone that you could try (there are free ones available for download)? I don't believe anyone else has reported this problem, so we need to find another clue.

Ron

xcaliba
06-12-2006, 09:25 AM
Multiple VoXaLot SIP registrations are permitted and appear to work exactly as you (and I) hope they would.

I've had both channels of a PAP2 registered to the same VoXaLot number for some time. Incoming calls alternately ring both extensions. The first one picked up gets the call. As far as I can tell, the two extensions can be used in any combination for incoming and outgoing calls.

Ron

I have this set up now but only one phone rings, the one registered to line 1. I can pick up either extn and talk if the phone on line 1 rings and I can make calls from both successfully. Also if I am on an outgoing call on line 1 line 2 will actually ring for an incomming call. Any ideas? I am using a spa2100 with a real internet address so no nat issues.

reader
06-12-2006, 10:31 AM
Do you have the same problem if your prefered server and ATA are set to voxalot.com? Do you have access to another ATA (especially another model) or a softphone..

I switched back to voxalot.com (including in 'member details') but can't get that to ring at all now, just the busy tones. (I used to use it before the .au one, no problem.)

I don't have access to another ATA but the softphone is a good idea I hadn't thought of. Can't do it before the weekend but you have given me something to work on thanks Ron. Hope to post back with results.

Cheers, Jeff

dontknowalot
06-12-2006, 10:58 AM
I'm not sure if this is currently available with Dial Plans;
If I set up two plans that are identical except they connect to two different Service Providers will the lower priority plan be executed if the connection to the service provider in the initially executed plan fails ?
I hope I have explained this clearly, it would be a very nice way of providing a graceful failover service.

e.g.
Assume Vbuzzer is the my main provider but if they are off-line I would route my calls through Wengo.

Ron
06-12-2006, 11:13 AM
I have this set up now but only one phone rings, the one registered to line 1. I can pick up either extn and talk if the phone on line 1 rings and I can make calls from both successfully. Also if I am on an outgoing call on line 1 line 2 will actually ring for an incomming call. Any ideas? I am using a spa2100 with a real internet address so no nat issues.

My guess is that it's a setting in your ATA somewhere. In searching for a clue in my PAP2, I discovered a 'Synchronized Ring' setting which was set to No. When I changed it to Yes, both lines ring in unison now (I actually like that a bit better). Since you can answer either line, both must be being called. My only experience is with PAP2's, so I don't know how much an SPA-2100 might differ. Look over all the possible settings and see if anything jumps out as a candidate to change.

Ron

Ron
06-12-2006, 11:17 AM
I switched back to voxalot.com (including in 'member details') but can't get that to ring at all now, just the busy tones. (I used to use it before the .au one, no problem.)

Jeff,

Keep in mind that changing servers may take an hour or so for everything to work properly again. Existing SIP registrations (which are typically one hour) have to have time to expire on their own.

Ron

reader
06-12-2006, 11:56 AM
Jeff,
Keep in mind that changing servers may take an hour or so for everything to work properly again.

Thanks, I think that may be why my test of voxalot.com failed. Will try again and allow generous time.

ctylor
06-13-2006, 03:16 AM
For what I see from this forum, some providers can't be registered in Voxalot because they have unusual requirements, such as caller ID and outbound proxy.

Maybe some providers which seem to be Voxalot incompatible, like Voicestick, could work if there was a way to add some lines like this. I could really use a local number in Los Angeles from them.



That's a really great proposal, if Voxalot can handle that depth of individualization per account (and per registration!).

If that is implemented, then we could add tips and tricks (and the suggested exact Asterisk-type string needed) on the tutorial pages for getting specific SIP providers--like Voicestick--to work through Voxalot. The advantage for that is saving Martin and the other owners/admins of Voxalot from figuring out how to get all the SIP servers to work with behind the scene tweaks and letting the forum community to do it for them.

xcaliba
06-13-2006, 09:09 AM
Another request:

When you edit your dial plan you can change the priority dynamicly. EG: you want to change your dial plan so a new item is priority 2, everything 2 and dowm automaticly 'makes room' for your new priority 2 item.

Ron
06-13-2006, 09:26 AM
When you edit your dial plan you can change the priority dynamicly. EG: you want to change your dial plan so a new item is priority 2, everything 2 and dowm automaticly 'makes room' for your new priority 2 item.

The ability to reorganize and move entries around has been discussed and is planned for a future enhancement.

Ron

affinity
06-13-2006, 10:50 AM
That takes me back to learning BASIC [beginners all purpose symbolic instruction code] on a Tandy TRS-80 microcomputer...

I've found that dial plan numbering can cover a huge range, so the trick is to have gaps built-in; the old BASIC code:
10 PRINT "HELLO WORLD"
20 GOTO 10

....
15 PRINT "- HAVE A GREAT DAY!"

New code:
10 ...
15 ...
20 ...

So, if you leave gaps, then you can always add entries between them.

xcaliba
06-13-2006, 11:51 AM
That takes me back to learning BASIC [beginners all purpose symbolic instruction code] on a Tandy TRS-80 microcomputer...

I've found that dial plan numbering can cover a huge range, so the trick is to have gaps built-in; the old BASIC code:
10 PRINT "HELLO WORLD"
20 GOTO 10

....
15 PRINT "- HAVE A GREAT DAY!"

New code:
10 ...
15 ...
20 ...

So, if you leave gaps, then you can always add entries between them.

Simple ideas are often so easy :)
Solves my problem completely. Thanks.

affinity
06-13-2006, 12:00 PM
A better option would be to have no numbering, like QBASIC today.... and have a facility to move the rules up/down as required.

alfredwesterveld
06-13-2006, 03:43 PM
I can't get Voipbuster to work. I can't receive incoming calls even when i enable checkbox register.

Grtz,
Alfred Westerveld

hisybr
06-14-2006, 12:46 PM
Hi,

Have just set up and still getting some kinks worked out. Took a while to figure everything out. The concepts are all quite elegantly simple in a way, but like many 'simple' paradigm shifts, they are a bit mind boggling. Great work guys!

I am reasonably tech saavy, as is my father, who turned me on to VOIP. We've been running with SipPhone for about a year, making lots of calls to the US mostly. My dad has been going even longer, first with Vonage and later SipPhone and now GizmoProject. The tutorials seem to be reasonably well written, but it was still hard for the two of us to work things out. If the concept is to cater to techies, may be OK, but if there is an intention to spread to the masses, something probably needs to be done to simplify the website documentation -- an "idiot's guide" version...

Seem to have my system running OK, except AstraTel giving me intermittent problems. Could be them; new service. Sometimes get "cannot dial that number messages". Today my wife got disconnected repeatedly after a few minutes of a call. As it works sometimes and not others, I assume my settings are OK and it is on their end. Going to do more tests this weekend with my dad. Haven't been able to get web call to work with AstraTel so far. Both lines ring, but no sound either way. Will try other providers this weekend.

Suggestion: Would be good to publish a list of VMP that are compatible with VoXaLot (maybe a bit of promotion of them, a links page, would encourage them to seek compatibility).

My "dual registration" title comes from my dad's experience. He wasn't getting any calls after setting up. I changed my ATA to voxalot.com.au, but he was reluctant to change settings. I finally figured out what he had done. He listed all of his providers on VoXaLot and set them to Sip registered. So, he had one of his numbers registered through both VoXaLot and his ATA. Hadn't set up any forwarding. VoXaLot was intercepting his calls, but couldn't pass them on (got VoXaLot voicemail).

Suggestion: VoXaLot should look to see if a provider is registered elsewhere and should warn user before saving settings or advise by email if dual registration is detected.

Sorry for the long-windedness. The whole concept you've established is absolutely brilliant!

Brian

reader
06-15-2006, 02:13 AM
something probably needs to be done to simplify the website documentation -- an "idiot's guide" version...

I have to thank the writer of the following page -
http://voip.wikispaces.com/VoXaLot
Combined with the voxalot tutorial page it is well on the way to being an idiots guide. It enabled this idiot to get up and running!

martin
06-17-2006, 12:17 AM
If I change my registration to voxalot.com.au (and my voice mail server to $USER@voxalot.com.au - but of course this isn't necessary to test this - rather this is required for MWI & VWMI if calls ever got through to VM in the first place), then after 12 seconds, the phones connected to the FXS port (Line 1) stop ringing for a moment, but start ringing again.

Hi BJ,

What happens if you register with voxalot.com.au and point your voicemail server to voxalot.com?

FYI, all voicemail is handled by voxalot.com even if you are registered with voxalot.com.au

martin
06-17-2006, 12:29 AM
@reader,

Are you still having the registration / calling problems you mentioned in this thread?

martin
06-17-2006, 12:44 AM
See this thread people:

http://forum.voxalot.com/showthread.php?t=247

reader
06-17-2006, 03:42 AM
@reader, Are you still having the registration / calling problems you mentioned in this thread?

Yes I am... well last night it was still the same.

As I seem to be the only one with this problem (having to dial twice to get switched to the callee) and there are more tests I can do (try a softphone) the ball is in my court. Just haven't had time yet.

Tslam
06-17-2006, 04:33 AM
I want to send calls to voxalot's voicemail server, as per in my recent posts on the Support forum. That thread has stalled, so I'm writing to ask a few pertinent questions here:

1) Is it ok in principle for registered users to send calls to voxalot's voice server? I would understand if you had a policy of not allowing that, if the cost of storage is an issue. But you could limit the no of calls queued for more than 2 days, for example. Or forward them directly to email!

2) Is the voxalot vmail server accessible from the internet? Do I understand correctly that it has its own IP address, and authentication protocol, just like a SIP server?

3) Does sending, or calling, the vmail server directly, avoid the diversion delay (30 odd secs) that occurs when calling mysip@voxalot.com:5061?

4) If the answer to 1)-3) is YES, then... what's the IP address, please? And what's the exact syntax for calling it?

Thanks again.

ps All hail Voxalot! It's a truly great innovation. I wonder though, whether its functionality will eventually become another generic feature of the internet - like DNS maybe. Or bittorrent. I'm not sure which of those is the best analogy. A question for another day, perhaps.

Tslam
06-17-2006, 07:20 AM
It seems that voxalot is registering me with Koala using my Voxalot AuthID instead of my Koala AuthID.

Koala now shows my account number equals my Voxalot AuthID. I have no Koala credit for that ID, and now I can't make any calls. Strange.

Please check and let me know what's going on. I thought it would be best to check with Voxalot before contacting Koala.

If required, I'll private you my account number(s).

BJReplay
06-17-2006, 09:18 AM
Hi BJ,

What happens if you register with voxalot.com.au and point your voicemail server to voxalot.com?

FYI, all voicemail is handled by voxalot.com even if you are registered with voxalot.com.au

Hi Martin,

It seems to be all working with voxalot.com.au - I swapped to .au last weekend when voxalot was off the air for a while (I think it went down on Sunday around the middle of the day), and callers were getting through to voice mail on dialling my PSTN and being forwarded. It has been working since and I've stayed with .au.

I've got my voice mail server set as .au as well, and when I pick up the phone, it has a warm line to 500 - again working just fine. MWI & VWMI are working.

If the VM server is the .com server, I assume that I'm still getting MWI & VWMI based on notifies from the .au server.

Anyway, it's all working, even if I don't understand exactly how.

martin
06-17-2006, 10:29 AM
Glad to here it BJ.

Tslam
06-17-2006, 10:40 AM
I've got my voice mail server set as .au as well, and when I pick up the phone, it has a warm line to 500 - again working just fine. MWI & VWMI are working.

If the VM server is the .com server, I assume that I'm still getting MWI & VWMI based on notifies from the .au server.

Hi again BJ,

Would you please confirm:

. Line1->VM Voicemail Server = @voxalot.com.au ; does it use the @ prefix?
. PSTNLine->Cfwd No Ans Dest = 500
. registered proxy = voxalot.com.au, not the US server

If I warmline to 500, it answers immediately but I'm asked for a password. How do I configure it so that PSTN callers hear my greeting??

BJReplay
06-17-2006, 11:32 PM
Hi again BJ,

Would you please confirm:

. Line1->VM Voicemail Server = @voxalot.com.au ; does it use the @ prefix?
No, Line 1->Voice Mail Server = $USER@voxalot.com.au - which is equivalent to 123456@voxalot.com.au: Line 1 is registered to voxalot.com.au, and my UserID is would be 123456.
. PSTNLine->Cfwd No Ans Dest = 500
No, User 1->Cfwd No Ans Dest = 123456 (I assume you meant User 1, not Line 1 :)
. registered proxy = voxalot.com.au, not the US server
Yes :)

If I warmline to 500, it answers immediately but I'm asked for a password. How do I configure it so that PSTN callers hear my greeting??
My warmline is for me to pick up the phone attached to the FXS port, to retrieve VMs. I'm not sure if what you're doing can be done.

I've attached my config. Have a look see. Feel free to attach yours, if you want me to have a peek. I've changed IP addresses, account numbers, and so on to protect the guilty, but other than that, what you see is what I run.

Tslam
06-19-2006, 09:28 AM
For the record, this problem has been solved. It was specific to Koala, when reg'd by some other SIP servers including voxalot. See:

http://forums.whirlpool.net.au/forum-replies.cfm?t=542020#r14

BJReplay
06-19-2006, 09:57 AM
For the record, this problem has been solved. It was specific to Koala, when reg'd by some other SIP servers including voxalot. See:

Was it resolved by a change at Koala or a change at Voxalot? Do you have any more details?

vpsaini
06-19-2006, 10:28 AM
Till last day I was making calls through some service providers, registered with Voxalot in my account and today I am not able to make calls, through web call back as well as through my ATA... Any reply please..
Thanks and Regards
Vishnu

Tslam
06-19-2006, 10:39 AM
Was it resolved by a change at Koala or a change at Voxalot? Do you have any more details?

A change at Koala. The problem also occurred between Koala and some other SIP registrars. Martin has indicated that voxalot shouldn't affect external SIP id's. See: http://forums.whirlpool.net.au/forum-replies.cfm?t=542020&p=2#r26

ps I'll get back to the other matter we discussed when I have some free time (I've been spending too much time on the SIP rego thing).

renanmm
06-19-2006, 01:57 PM
Hi!

I have noticed that my calls get a slight higher delay when using my voip provider routed via VoXaLot. Is there any settings to reduce this latency? Codecs, specific changes, etc? I'm using a Sipura SPA-2100 ATA with 300 Kbps TBF QoS activated, Voipcheap account and I'm behind a 4 MBs (d/l) and 600 Kbps (u/p) in a cable modem. Thanks.