Can't make and receive call externally(PSTN)

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/ ... terns.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

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

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 wrote: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,

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

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

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

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.

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

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

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)

It's important that she understand that she doesn't need pbx users set up to call from one sip user to another. Unless an ars matched her user number the ars would have no bearing on her problem either so I don't think her pbx setup is related to her problem. I'm concerned she mentioned setting up the pbx user but didn't mention setting up the sip user.

Michelle,

you need to use the sip providers address as the sip proxy address. You also need to check your sip server status page and make sure the ip address of you lan (ie 192.168.1.100) and your wan (ie 24.229.11.5) are both apearing in the page. If not put them both in the sip config first page and reboot your computer. 127.0.0.1 is a local ip address only used by your computer, it is not a lan address. Brekeke needs to know your lan address and your public ip address. Also, make sure you set up the sip user/authentication for the two users in the sip server authentication tab. You can make a sip user to user call without setting up the user in the pbx. You ONLY need to set the user up in the pbx if you want to add features like call forwarding, busy/no answer, scheduling, voicemail and all the other pbx features available. A sip user to sip user call is not dependent on the pbx.

Thanks,

Nick

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.

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.

Nick

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

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

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