Sending traffic to VoIP provider does not work. Help needed.

[Shrek]

1. Brekeke Product Name and version: SIP Server 60 Days Evaluation

2. Java version: 6

3. OS type and the version: Windows 2003

4. UA (phone), gateway or other hardware/software involved: Eyebeam

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : I guess 3...

6. Your problem:

Hello,

I am a kind of newbie to VoIP stuff, need your help please.
On my remote server I installed SIP Server product. All calls from this server must be routed to wholesale VoIP termination provider which requires authentication by login and password. On my home PC I start Eyebeam softphone (I created user for it on server) which connects to that remote server. The problem is that when I dial some number, for example 12125556677 it is not routed to VoIP wholesale provider so that I could reach a US landline number, but Eyebeam immediately tells me that "Call failed: Not Found", so I suppose it is looking for this number on SIP server and does not try to send traffic to VoIP provider at all. Pangolin softphone tells me "Not Found 404"

Below is my dialplan:

Matching Patterns

$request = ^INVITE $registered = true

Deploy Patterns

To = sip:222222222:33333@sip.myvoipprovider.com

(222222222 is login and 33333 is password for VoIP provider).


Please help me to fix the problem.

Thank you.

[hope]

it is better to use Brekeke pbx, whose ARS feature can be used to set authentication info.
sample setting: http://wiki.brekeke.com/wiki/ITSP

[janP]

>> To = sip:222222222:33333@sip.myvoipprovider.com
>> (222222222 is login and 33333 is password for VoIP provider).

Why did you set like that?
Is it the format your VoIP provider suggested?

[Shrek]

Provider requires authentication by login and password. I looked for the info and found that the following syntax can be used to pass login and password in SIP URI:

sip:user:password@host:port;uri-parameters?headers

But seems like SIP server does not support such.

Also I have another question. If I want to enable logging, do I need to put such records in Deploy Pattern:

$net.registrar.loglevel.file = 255
$net.sip.loglevel.file = 255

[janP]

which VoIP provider are you using?

You can use Brekeke SIP Server..
but use Brekeke PBX's ARS. It will be best in your case.

[Shrek]

I was using voicetrading.com
Yes unfortunately I found some post in PBX forum that SIP server cannot normally authorize by login/password. So it is not possible to use providers who authorize using account credentials . PBX is limited by users quantity and concurrent calls that is what not good for me.

[janP]

Are there any other users you want to share this provider account?
If yes, you need to use Brekeke PBX.
If no, you can use Brekeke SIP Server.