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-   -   Free DID Numbers, Free Voip Calls, & more Voip Info (https://forum.sipbroker.com/showthread.php?t=1776)

emoci 01-08-2008 03:31 AM

Quote:

Originally Posted by addo62 (Post 14997)
Just wondering whether this combo actually works in Australia or not?

Most of the user setup guides do not mention Australia??

I have been trying this since yesterday & followed the user guides exactly but I have had no joy in getting a call through.

As suggested I have tried my home # in different formats i.e. with & without dialing codes, but still nothing.

PhoneGnome shows calling records but no calls actually make it through to my home phone.

I even tried the combination using MySipSwitch as well with no luck.

Any advice appreciated:(

Australian numbers supported as below:
Australia 61 All Except 1, 4, 5, 71, 89100, 89101
See: Qualifying International Numbers

That said, there has been an issue for the last few days getting PhoneGnome to accomplish the forwarding....SipBroker times out at about 30sec whereas it seems PhoneGnome is taking 40-42 secs before setting up the call..... (I can confirm this is affecting the MySipSwitch and PBXes setup, have not tested with VoXalot Call Forwarding, but Using VoXalot to call with PhoneGnome works fine, even though it takes about 40 sec before you hear any ringing)

At this point, it is a waiting game, waiting for PhoneGnome to resolve their issues/speed up their call setup process

addo62 01-08-2008 10:11 AM

The number I am dialing does fit into the supported number range.

I tried it again tonight using just Voxalot & PhoneGnome, used the #'s in all formats with no joy, just an engaged tone with the phone not ringing.

Tried the MySipSwitch combo & this time I actually get the recorded voice saying free call & dialing the local # but then I get the engaged tone & the phone again not ringing..

emoci 01-19-2008 03:56 AM

A little bit of news, or at least something to try:

Receiving your SIP Calls in Skype: Sip2Skype: Receiving SIP Calls in Skype

affinity 01-19-2008 07:37 AM

Interesting reading, but I don't use Skpe and am fundamentally against Skype in my views for a number of reasons. However, if there was no risk of lacking enough channels, I'm sure it could be a worthwhile option for many.

satphoneguy 01-19-2008 03:43 PM

Quote:

Originally Posted by affinity (Post 15185)
Interesting reading, but I don't use Skpe and am fundamentally against Skype in my views for a number of reasons. However, if there was no risk of lacking enough channels, I'm sure it could be a worthwhile option for many.

now you can call people who use skype without ever having to join skype yourself!!

Jorge 03-15-2008 10:20 PM

Camundanet is not in the list.

It provides break-in numbers, as an alternative to SIP Broker, for example in Mexico where numbers listed in the SIP Broker site are totally useless.

I have never tried, but they also allow to call toll free numbers in Argentina.

voipnewbie 03-16-2008 08:59 PM

Gizmo to mysipswitch to Phonegnome?
 
Hi all,

I was able to set up mysipswitch to forward gizmo coming call to phonegnome. The next day I try a gain but the call did not go through. Look like the gizmo incoming is fine but when it forward to phonegnome there is a problem (I'm not sure what it is but look at the call log I see the status is 487 and the response is RequestTerminated). Please give advice on how to fix this problem.

Thanks

emoci 03-16-2008 09:40 PM

Quote:

Originally Posted by voipnewbie (Post 15785)
Hi all,

I was able to set up mysipswitch to forward gizmo coming call to phonegnome. The next day I try a gain but the call did not go through. Look like the gizmo incoming is fine but when it forward to phonegnome there is a problem (I'm not sure what it is but look at the call log I see the status is 487 and the response is RequestTerminated). Please give advice on how to fix this problem.

Thanks

487 Request Terminated means the call hanged up...in any case just tested my setup and there seems to be an issue between MySipSwitch and PhoneGnome at the moment....hopefully this is temporary

There is a quick fix by getting a VoXalot Acct. to play middleman such that MySipSwitch uses VoXalot to forward rather than PG directly (and PG is setup within VoXalot)

Feel free to send me a PM if you are still having trouble with this...

voipnewbie 03-17-2008 12:10 AM

Quote:

Originally Posted by emoci (Post 15787)
487 Request Terminated means the call hanged up...in any case just tested my setup and there seems to be an issue between MySipSwitch and PhoneGnome at the moment....hopefully this is temporary

There is a quick fix by getting a VoXalot Acct. to play middleman such that MySipSwitch uses VoXalot to forward rather than PG directly (and PG is setup within VoXalot)

Feel free to send me a PM if you are still having trouble with this...

Thank you emoci,

I'll try VoXalot route. By the way, do you know if I can still connect my Asterisk server to PG?. I found a post back in 2006 but not sure if it is still workable. I'm just playing with VOIP for a month or so. I need to do a lot reading and searching.

emoci 03-17-2008 01:11 AM

Quote:

Originally Posted by voipnewbie (Post 15788)
Thank you emoci,

I'll try VoXalot route. By the way, do you know if I can still connect my Asterisk server to PG?. I found a post back in 2006 but not sure if it is still workable. I'm just playing with VOIP for a month or so. I need to do a lot reading and searching.

First, check PM

Secondly, I have had no issues with PG registrations in general (ATA, VoXalot, MySipSwitch, PBXes) so I don't see why asterisk wouldn't work.....the settings are the same as the one's for the SoftPhone.........in general their model is pretty neat (if just their box was priced a little lower)


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