Can't Send calls to Pbx.(Error 404,not found)

1. Brekeke Product Name and version:bpbx 2.4.5.5/286

2. Java version:1.5.0_16

3. OS type and the version:windows 2003

4. UA (phone), gateway or other hardware/software involved:Asterisk UA,Sip softdialer,Voip Switch etc

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

6. Your problem:I upgraded to version 2.4.5.5 last week and have updated the Sip Server default dial plan as recommended.I've tried many dial plans but haven't been successful in sending calls to pbx instead I'm getting error 404 (Not found)

Here's the dial plan I have on the Sip Server:

[matching pattern]
$request=^INVITE
$addr=my_DID_provider's ip

[Deploy pattern]
$target=localhost:15060
$auth=false

And here's the dial plan I have on the pbx [pattern-in]

[matching patter]
To=sip:my_DID_number@

[Deploy pattern]
To=pbx_extension

I'm not sure what I'm missing out.I'd appreciate if someone could point that out for me.

Thanks for your anticipated help,
Kennedy Joseph.

Which version of Brekeke PBX were you using before you upgraded ?

Do you have any other DialPlan rules??

Hi James,

Thanks for your response and here's my answer to your questions:

(1) Which version of Brekeke PBX were you using before you upgraded ?

I WAS USING VERSION 2.1.6.6

(2) Do you have any other DialPlan rules??

YES I DO.HERE'S ANOTHER ONE THAT I'VE ALSO TRIED THAT DIDN'T WORK.

SIP SERVER DIAL PLAN
------------------------------------------
[Matching Patterns]
$request = ^INVITE
$addr = My_DID_Provider's_IP
To = sip:(.+)@

[Deploy Patterns]
$auth = false
To = sip:%1@127.0.0.1:15060


PBX DIAL PLAN
------------------------------------------
ARS Pattern - IN
------------------------------------------
[Matching Patterns]
From:sip:(.+)@My_DID_Provider's_IP
To: sip:(.+)@

[Deploy Patterns]
To: PBX_user

With the above dial plan I get error 486 busy response.

I really don't know what to do,I've been on this for a week,please help.

Thanks,
Kennedy Joseph.

[matching pattern]
$request=^INVITE
$addr=my_DID_provider's ip

[Deploy pattern]
$target=localhost:15060
$auth=false

try with putting above dial plan on top of all other dial plans

Hello Hope,

Thanks for the dial plan.However,I've tried it before but it didn't work.

I don't think the issue is dial plan.I ran a "netstat" on the server today and it seems port:15060 is not listening. Any ideas on what I should do?

Regards,
Kennedy Joseph.

can you see udp port 15060 with command "netstat -ano" ?
can you make calls between pbx extensions?
did you click "Apply Rules" button when changing dial plan order?
what dialing number can you see in the To header in the INVITE the packets sent from DID provider? is it your DID number?

Hello Hope,here are my answers to your questions:

(1)can you see udp port 15060 with command "netstat -ano" ?

YES I SAW THE udp port 15060 LISTED UNDER LOCAL ADDRESS.
IT’S LISTED THIS WAY: udp:joseph:15060 (joseph’s my server name)

(2) can you make calls between pbx extensions?

NO I CAN’T MAKE ANY CALL TO PBX EXT.ALL CALLS TO PBX RETURNS
ERROR 404 (NOT FOUND)

(3) did you click "Apply Rules" button when changing dial plan order?

YES I DID AND I OFTEN RESTART BPBX AFTER APPLYING ANY NEW DIAL PLAN RULE.

(4) what dialing number can you see in the To header in the INVITE the packets sent from DID provider? is it your DID number?

YES,IT’S MY DID NUMBER THA’S IN THE TO HEADER.

I have two questions?

(1) why is it that non of the udp port is listening or established?

(2)I was thinking that port 15060 should be listed this way
127.0.0.1:15060 instead of joseph:15060. It’s seems to me as
If “joseph” is replacing the localhost ip.

waiting on you.
Thanks.

not sure what cause the display of netstat -ano like that.
it may not be the cause of the problem.

Can you check from task manager with the 15060 pid show from netstat -ano to see if the it is used by brekeke pbx(java) or some other application?

is pbx started successful?
Is there any error shown from first page after log into brekeke pbx gui from browser?

if pbx, media server and sip server are showing active without error,
Then disable all non-default dial plans from brekeke sip server,
register two phones(xlite) to brekeke and create pbx users for these two phones,
can make call between phones with default dial plans?

I decided to reinstall pbx.I installed the newest version (2.4.6.7)last night but still can't route calls in/out throug the pbx.

Here's what my network environment and call flow pattern looks like.

I have 3 sip Server clients (a,b,c ) on port 55060.Calls coming from my DID are received by the sip server and sent to (a,b,c)@localhost:55060.(This works)

From the localhost:55060 I want to route these calls out through the pbx ARS to an ITSP. I haven't been able to do this.I've tried registering my DID_provider's URI at the pbx ARS pattern in like this:

Register URi:sip:&v1@my_DID_provider's ip
Proxy Address:DID_ip
Realm:DID_ip

[Matching patterns]
To:sip:&v1@ (This is the incoming DID number )

[Deploy patterns]
To:&v3 (This is my sip client "a")

The above dial plan didn't work.I've also tried binding the incoming DID number to localhost 127.0.0.1 like this:

Register URi:sip:&v1@127.0.0.1
Proxy Address:127.0.0.1
Realm:127.0.0.1

But it didn't work either.

I've tried using this dial plan to send calls to the pbx but it didn't work.

[matching patterns]
$request=^INVITE
To=sip:(.+)

[Deploy patterns]
To=sip:%1@127.0.0.1

I've also tried to send calls from pbx to the sip server with this dial plan but it didn't work:

[matching patterns]
$request=^INVITE
To=sip:(.*)

[Deploy patterns]
$auth=false
To=sip:%1@200.21.567.222 (My sip server ip)

This is the only dial plan that works but can only send calls out through the Sip Server.

[matching patterns]
$request=^INVITE
$addr=my_DID_provider's_ip
To=sip:my_DID_number@

[Deploy patterns]
$auth=false
To=sip:a@localhost:55060

It seems there's something I'm missing out but not sure what that is.I'll appreciate any idea that could help me unlock this puzzle.

Thanks,
KJ

is there pbx user created for sip client a?
if register another sip phone and create pbx user, can it call to sip client a?

maybe you'd better contact brekeke support for help; support@brekeke.com