Can't make and receive call externally(PSTN)

michelle · Brekeke Junior Member Joined: 29 Apr 2007 Posts: 8

1. Brekeke Product Name and version: Brekeke PBX, Version 2.0.7.2 Pro Evaluation

2. Java version: Java SE

3. OS type and the version: Server 2003

4. UA (phone), gateway or other hardware/software involved:
USB Phone and My Net Fone provider - gateway

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

6. Your problem: Can't make and receive call externally(PSTN)

I don't know if it just my setting thats why i can't make call and receive call.

Gateway - sip01.mynetphone.com.au
SIP Proxy - 127.0.0.1

My ARS setting:

Register URI: username@sip01.mynetphone.com.au
SIP Proxy: 127.0.0.1
Username: XXX - this is from my gateway provider
Password: XXX - from my gateway provider

Pattern In
Matching Patterns
To: user@

Deploy Patterns
To: PBX user - extension number

Pattern Out
Matching Patterns
To: sip:9(.+)@

Deploy Patterns
From: "user" <sip:user@sip01@mynetfone.com.au>
To: sip:$1@sip01.mynetfone.com.au

Do i still need to configure something else except ARS???

Thank you in advanced

Michelle

voipwell.com · Posted: Tue May 01, 2007 6:35 am Post subject:

Hi,

Your ARS looks correct. Go to the sip server and click on the logs tab and display all the logs for a day and post what the error is all the way to the right for the calls that fail. That should give a good indication of what the problem is.

Nick

michelle · Brekeke Junior Member Joined: 29 Apr 2007 Posts: 8

Hi Nick,

Thank you for your reply...

Here's the error that im getting in if i dial out:

6190 sip:1002@192.168.0.25 sip:989761346@192.168.0.25 00:00:00 Fri May 04 09:15:56 EST 2007 Fri May 04 09:15:57 EST 2007 Failure 400

thanks again

michelle

voipwell.com · Posted: Fri May 04, 2007 3:23 pm Post subject:

Michelle,

It looks like you have 2 @ in the deploy.

From: "user" <sip:user@sip01@mynetfone.com.au>

change to

From: "user" <sip:user@sip01.mynetfone.com.au>
^

That should do it.

Nick

michelle · Brekeke Junior Member Joined: 29 Apr 2007 Posts: 8

Nick,

You asked me to change the deploy pattern but the change that you asked me to do is the same from the one it was before...

And for the sip proxy i put 127.0.0.1 - this one is the IP address of the NIC of the computer. I tried to change it to the IP address of my voip provider, I can able to call external but can't able to call internal(extension that i create on Brekeke PBX)

If you can able to help me what to I need to do or change...

Thank you in advance...

Michelle

michelle · Brekeke Junior Member Joined: 29 Apr 2007 Posts: 8

Nick,

Im using Brekeke PBX with SIP Server, how can I make it to work both because at the moment only internal(extension that i created in SIP Server) line works. I'm trying to work out the PBX part, i got all the details from my provider so I put all the details in ARS but still not working but once I change the SIP Proxy address of my provider the external call will work but my internal call won't work.

Thank you

Michelle

voipwell.com · Posted: Fri May 18, 2007 6:39 pm Post subject:

Michelle,

When one extension dials another extension it doesn't go thru the pbx so adding an ars with the correct proxy address shouldn't stop it from working. Your proxy address should be sip01.mynetfone.com.au and your deploy string should be "user" <sip:user@sip01.mynetfone.com.au> . You should not have changed anything in the dial plan of the sip server. Make sure your sip status page is showing both your nat address and your external ip address.

mexsip · Posted: Fri May 18, 2007 7:20 pm Post subject:

Hello Michelle,

Did you create the ext under pbx user?
Did you create the ext under authentication of sip-server?

Hello Nick,

default dialplan on sipserver makes sure all calls are going thru pbx. unless a dialplan rule was added before these default diaplan rules and there was a succesfull match for this dialplan rule.

good luck,

Pablo

voipwell.com · Posted: Sat May 19, 2007 12:39 pm Post subject:

Hello Mexsip,

Yes when the pbx is installed the "to pbx" and "from pbx" dial plans are used to check for how to handle user preferences like call forwarding, busy/no answer etc and ars. But, in a case where you have one sip user calling another, the pbx does nothing. You don't even need the users set up in the pbx for you to be able to call from one sip user to another. So in a call from one sip user to another the pbx does nothing.

I was trying to help michelle regarding

mexsip · Posted: Sat May 19, 2007 7:10 pm Post subject:

Hello Nick,

What you are saying is that when sip user calls other sip user, they can't use pbx funcionality, like transfer call to other user?

Or this is the case when the corresponding pbx user doesn't exist?

thanks, Pablo.

voipwell.com · Posted: Sat May 19, 2007 7:23 pm Post subject:

Hi Pablo,

What I am saying is that one sip user can call another sip user without having to set up the users in the pbx. After you set up the users in the pbx then you can use all the pbx functions for that user. If you only want to set up the users in the sip server that is ok and you can still connect to each other, but of course they can't use any pbx features but they may be able to use call transfer since most ua's can peform the function without pbx.

I only am mentioning this to assure Michelle that there is no reason that by putting in the correct proxy address it should stop user to user calls.

Ok, Michelle got a lot of advice on this one. Smile

Nick

michelle · Brekeke Junior Member Joined: 29 Apr 2007 Posts: 8

Hi Nick,

On my SIP Server Status I can only see the Interface IP Address(I believe this is the LAN IP Address)...

I don't know where to put the WAN IP Address that you mention...

Thank you

Michelle

michelle · Brekeke Junior Member Joined: 29 Apr 2007 Posts: 8

I figure out where do I need to put the WAN(Interface 1) and I change my sip proxy address to my provider address but still not working...

Only one side(external or internal) works not both...

Thank you

Michelle

voipwell.com · Posted: Mon May 21, 2007 8:14 am Post subject:

Michelle,

What are the extension numbers you assigned to your users and when the call fails what does your sip server log say. The Sip log is in the log tab in the sip server. All the way to the right of the screen is the result code. -1 means successful and 404 would mean number not found.

Also, when you make changes to the ip address interface you should reboot the computer.

Nick