Voxalot / SIP Broker Support Forums

Voxalot / SIP Broker Support Forums (https://forum.sipbroker.com/index.php)
-   Voxalot Support (https://forum.sipbroker.com/forumdisplay.php?f=4)
-   -   Web Callback with VOIP Buster (https://forum.sipbroker.com/showthread.php?t=52)

gwizz 03-30-2006 12:13 PM

Web Callback with VOIP Buster
 
Hi,

I've been trying to get Web Callback working with VOIP Buster but not much success.

I added the provider:
Description: VoipBuster
Username: username
Password: password
Codecs: ulaw;alaw;g729
Host: sip1.voipbuster.com
Port: 5060

And attempted a web callback:
Your Number: 00country code+my number
via: VoipBuster
Number To Dial: *393613
via: SIP Broker

Anyone had any luck with this?

You may be interested to know VOIP Buster, Voip Discount, Internet Calls, VoIPcheap, VoIPstunt, etc. are all the same company.

martin 03-30-2006 01:25 PM

Have not tried that particular provider. If they work on a GW slot of an ATA then they should work in VoXaLot.

Hopefully someone else that has tried them can help out.

blackbile 03-31-2006 05:40 AM

gwizz!

I tried my sparvoip account (which is the same company) and my phone has rang, I picked up and has rang the called number which was picked up too, but we could not hear anything ...

Ron 03-31-2006 06:52 AM

gwizz,

I had VoIPBuster (outbound only) until they took away my unused credit yesterday. The end-to-end delay with it was so horrendous I won't be giving them any more.

I'm not 100% sure of what you're expecting, but....

If you have a DID with VoiPBuster, it will not ring directly into VoXaLot even though you have set up a VoXaLot Provider entry for VoIPBuster . If it can be forwarded via SIP URI, you can point it to VoXaLot and you will receive those incoming calls on your VoXaLot phone.

From the VoXaLot Web Callback perspective, there are only outgoing calls. VoXaLot uses the "Your Number" provider to make the first outgoing call. Once it's completed, VoXaLot uses the "Number to Dial" provider to make the second call. It then bridges the two calls together (at least conceptually).

Ron

blackbile 03-31-2006 07:05 AM

Quote:

Originally Posted by anominous
gwizz,

It then bridges the two calls together (at least conceptually).

Ron

yes but unfortunatelly after the bridge is built I cant hear the other side...

Ron 03-31-2006 07:30 AM

Quote:

Originally Posted by blackbile
I tried my sparvoip account (which is the same company) and my phone has rang, I picked up and has rang the called number which was picked up too, but we could not hear anything ...

I've had a lot of experience with this situation over the past week or so. The problem is almost certainly with your router (its firmware, to be more exact). I found that DD-WRT v23 firmware works almost perfectly without any port forwarding. HyperWrt (+tofu13c) firmware without port forwarding exhibits the symptoms you describe. I had to forward both SIP and RTP port ranges to make Web Callback work (with sound) using HyperWRT firmware.

I also have to use a STUN server with my PAP2 with either firmware and must set 'Substitute VIA Addr' to Yes.

I've also experimented with D-Link (DI-524/DI-604) and Belkin (F5D7230-4) routers and come to the conclusion there is no reasonable way to make these work properly with all VoIP capabilities.

The Network Address Translation (NAT) used in most current home routers is just not VoIP friendly. Even the simplest configuration of a single provider with no expectation other than to be able to make and receive a call can be very problematic. If you also want to have multiple providers and things like 3-way calling, call forwarding, Web Call, etc. to work, you may find there's a fair amount of learning and experimentation required in order to be successful.

Ron

blackbile 03-31-2006 07:38 AM

you are wrong,

I dont use router, I try the Webcallback service which doesnt need any hardware device I think, isnt it?

gwizz 03-31-2006 07:47 AM

All I want to do is OUTGOING to the free countries listed for this provider.

I put in the details and get
"Your call has been initiated."

and no phones ring... ARG! :)

It would be nice if the system would give me some more information on the connection process. i.e. where it is going wrong.

blackbile 03-31-2006 07:53 AM

Quote:

Originally Posted by gwizz
All I want to do is OUTGOING to the free countries listed for this provider.

I put in the details and get
"Your call has been initiated."

and no phones ring... ARG! :)

It would be nice if the system would give me some more information on the connection process. i.e. where it is going wrong.

for me it was ringing...

I tried sip.sparvoip.de see here http://tf1999.uw.hu/

martin 03-31-2006 08:04 AM

Quote:

Originally Posted by gwizz
It would be nice if the system would give me some more information on the connection process. i.e. where it is going wrong.

We agree, however we decided to release a functional system for the trial. Over time we will add features such as those you suggest.

If you use Web Callback via 2 providers i.e. PSTN<-->PSTN there is no VoIP devices in the picture.

However you might want to check whether the numbers you are calling are in an ENUM registry because the Web Callback checks ENUM for a free call route before using the asociated provider.


All times are GMT. The time now is 09:46 PM.

Powered by vBulletin® Version 3.7.2
Copyright ©2000 - 2024, Jelsoft Enterprises Ltd.