Pbx unable to accept incoming calls from public Softswitch

1. Brekeke Product Name and version: Pbx v 3

2. Java version:

3. OS type and the version: Windows 7

4. UA (phone), gateway or other hardware/software involved: Cisco SPA 301, 303, Xlite

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : 8

6. Your problem: PBX V3 is unable to accept incoming calls outside the LAN.
Work done:
----------
Type of connection: ADSL

I have installed PBX (trial version) V3 (in a local network and allocated the IP: 192.168.1.2), it works perfectly: registered users can send/receive call from each other.

I managed to build an ARS to call Sip users registered in another SIP Server (SIP provider) with a public IP address (212.X.X.X).

I configured the port forwarding in my router to accept incoming calls from the public IP with UDP protocol on Port 5060. And I also deactivated the firewall of the server where the Pbx V3 is running (very dangerous). Tried various dial plans gathered from the forum, none of them work in my case.

My local Pbx V3 still not accepting incoming calls from the Sip provider (with public IP)

I thought the issue was from my SIP provider (212.X.X.X) but he confirmed that its outbound Sip server is working perfectly after verification.

Please help on the proper Dial plan/ARS required to accept incoming calls from public Sip provider.

- Have you set router global IP at brekeke PBX / sip server/configuration/sip network interface address field?
- do you need register to SIP provider by setting registration in ARS rule?
- what ARS and dial plan rule have you created?
- if running wireshark at pbx server, can you see the invite sent from SIP provider? and what are From and To header like?

- Have you set router global IP at brekeke PBX / sip
server/configuration/sip network interface address field?

" Yes "

- do you need register to SIP provider by setting registration in ARS rule?

" No "

- what ARS and dial plan rule have you created?

ARS
=======
Pattern IN

From : sip:(.+)@109.yyy.zzz.xxx
In : sip:(.+)@

Deploy
To : $1

DIAL PLAN
=======
Matching Patterns:
$request= ^INVITE
To : sip:8(.+)@

Deploy Patterns:
$auth = false
To: sip:%1@
&register.contact.remote = true
$continue = false

- if running wireshark at pbx server, can you see the invite sent from SIP provider? and what are From and To header like?

"Will revert to you"

change dial plan rule as below and put it on top of all other dial plan rules
change sip provider IP address to real IP

Matching Patterns:
$request= ^INVITE
$addr = sip provider IP address
$pbx.inport = (.+)

Deploy Patterns:
$target = 127.0.0.1:%1
$transport = udp
$replaceuri = false
$b2bua = true
&net.sip.replace.callid = false
$auth = false
&net.sip.fixed.addrport.uas = true
&net.sip.fixed.addrport.uac = true


and also remove From setting in ARS pattern-IN matching patterns for testing and if work, set IP back and check if 109.yyy.zzz.xxx is the IP the same as that in INIVTE From header

ARS
=======
Pattern IN

From :
To : sip:(.+)@

Thank Hope.

Unfortunately I have changed the topology to try to sort out the issue. I change the environnement, PBX has a public address (270.yyy.zzz.xxx).

Both SIp provider (109.yyy.zzz.xxx) and PBX have public IP and Cisco IP phones are running behind adsl router but:

In the PBX side(270.yyy.zzz.xxx)
--------------------------------------
2, IP phones are registered with the adsl global IP address

sip: 1001@X.82.Y.150:5060
sip: 1002@X.82.9Y.150:5060

In the Sip provider side (109.yyy.zzz.xxx)
------------------------------------------------
2 Ip phones are registered with the adsl global Ip address

1300@X.82.Y.150:5060
1500@X.82.9Y.150:5060

Current issue in the Pbx with public IP adress (270.yyy.zzz.xxx):
--------------------------------------------------------------------------
When you can extension within the Pbx, phone ring, are accepted but no voice/audio and after 15s, call is ended.

Incoming calls from Sip provider (109.yyy.zzz.xxx) not going thru and ring Busyyyy.


Pbx V3 is running in Windows 2008 Server R2.

Thank for your help.

Incoming calls from Sip provider (109.yyy.zzz.xxx) not going thru and ring Busyyyy.

add the dial plan rule in last post and
create an auto attendant extension, such as 400
change ARS rule
ARS
=======
Pattern IN
matching:
From : sip:(.+)@
To : sip:[0-9]{7,}@

Deploy
To : 400

check if incoming call from provider can hear auto attendant greeting
if work, set From head with the IP shown in provider INVITE From header in ARS matching From

When you can extension within the Pbx, phone ring, are accepted but no voice/audio and after 15s, call is ended.

- is this call between pbx extensions?
- what call status shown in sip server side/active sessions?
- disable any stun server setting at client phone and disable SIP alg setting at client side router

Currently no call working between extentions
--------------------------------------------------------
- is this call between pbx extensions?

Yes, no audio in call between extensions

- what call status shown in sip server side/active sessions?

ADSL Ip address = XXX.82.70.149

EX-SID 35
From-uri sip:8002@270.yyy.zzz.xxx
From-ip XXX.82.70.149:5060 (UDP)
From-if 270.yyy.zzz.xxx:5060
To-uri sip:8001@270.yyy.zzz.xxx
To-ip 127.0.0.1:5052 (UDP)
To-if 127.0.0.1:5060
Call-ID 1f98e4d6-9d949392@192.168.1.21
rule To PBX
plug-in InviteSession
sip-packet-total 8
listen-port 5060
session-status Accepted
time-inviting Sun Aug 19 15:42:21 BST 2012
rtp-relay on
rtp-srcdst
media audio
transport RTP/AVP
payload 0 (PCMU/8000)
status active
listen-port 10050
send-port 10044
packet-count 539
packet/sec 49
current size 260
buffer size 260
rtpex plug-in RTPexJNI
rtp-dstsrc
media audio
transport RTP/AVP
payload 0 (PCMU/8000)
status active
listen-port 10044
send-port 10050
packet-count 5
packet/sec 0
current size 260
buffer size 260
rtpex plug-in RTPexJNI

- disable any stun server setting at client phone and disable SIP alg setting at client side router
Stun already disable at client side since beginning of test PBX.

If have disable SIp algo on client side router, I am still unable to make calls between extension.
I have changed the Dlink router with a Netgear, it worked just fine. I was abled to make call between extensions.

But incoming calls from SIP provider just don't go thru the PBX (therefore Auto Attendant not receiving any incoming call from Sip provider)


Thanks

in session details, session-status: Accepted
it seems the caller side ACK is not sent to Brekeke server
and call is not established
in Brekeke SIP server admin guide section "3.2.2.
Session Details", there is list of session status
http://www.brekeke-sip.com/download/bss ... min_en.pdf

is caller 8002 registered at Brekeke?
if yes, set it to a shorter registration time or enable default dial plan rule "Register Behind NAT" to adjust 8002 registration time by setting &net.registrar.adjust.expires = 10, remove the _ in the setting.
and set SIP server side / RTP / RTP relay (UA on this machine): off

I have tuned PBX as advised in your post :

1) GREAT, I am now able to reach the Auto Attendant created in the PBX from an extension of the Sip Provider (xxx.yyy.11.106), see session below.
The Sip provider sends "1111" (4 digits) with the called extension= 11118006, 8006 being the extension in the Pbx.

EX-SID 2617

From-uri sip:1103@uuu.vvv.27.114
From-ip xxx.yyy.11.106:5060 (UDP)

From-if uuu.vvv.27.114:5060
To-uri sip:11118006@uuu.vvv.27.114
To-ip 127.0.0.1:5052 (UDP)
To-if 127.0.0.1:5060
Call-ID 13318175389740374249904010920011
rule IncomingMiz01
plug-in InviteSession
sip-packet-total 6
listen-port 5060
session-status Talking
time-inviting Wed Aug 22 01:03:22 BST 2012
time-talking Wed Aug 22 01:03:22 BST 2012
length-talking 00:00:05
rtp-relay off

My ARS
Pattern -IN Priority : 100
Matching patterns
------------
From : sip:(.+)@
To : sip:[0-9]{7,}@

Deploy patterns
To: 8000 <------ Auto Attendant in the Pbx

Dial Plan : IncomingMiz01

Matching Patterns:
$request= ^INVITE
$addr = xxx.yyy.11.106
$pbx.inport = (.+)

Deploy Patterns:
$target = 127.0.0.1:%1
$transport = udp
$replaceuri = false
$b2bua = true
&net.sip.replace.callid = false
$auth = false
&net.sip.fixed.addrport.uas = true
&net.sip.fixed.addrport.uac = true

Question:
1) I cannot dial any extension from the Auto Attendant menu.

2) How to call an extension directly without the 4 digits (1111) that the Sip provider sends to the PBX.

3) Outgoing calls from the Pbx to Sip server no longer working

Patterns - OUT Priority :900
Matching patterns

To: sip:10([0-9]{4,})@

Deploy patterns
From : sip:9000@ xxx.yyy.11.106
To : sip:$1@ xxx.yyy.11.106

4) Prioties explanation regarding "mediaserver_prefix", why is set to 9999.

regards

2) How to call an extension directly without the 4 digits (1111) that the Sip provider sends to the PBX.

change ARS pattern-in as
Pattern IN
matching:
From : sip:(.+)@
To : sip:1111(.+)@

Deploy
To : $1

3) Outgoing calls from the Pbx to Sip server no longer working

with above changes the outgoing call will work

1) I cannot dial any extension from the Auto Attendant menu.

if call from other extension, can dial other ext from auto attendant?

4) Prioties explanation regarding "mediaserver_prefix", why is set to 9999.

it will be applied later than the other ARS rule, smaller number will have higher priority

It worked just as you mentionned it. I am able to receive calls from the SIP provider with the public IP address. Many thanks

But I have the current issues ;

1)- Outgoing calls from Pbx to the SIP provider no longer working with the following ARS: Ringing Busy

PATTERN OUT - Priority 100
Matching patterns
To : sip:1111([0-9]{4,})@

Deploy patterns
From : sip:9000@Sip_Provider IP
To : sip:$1@Sip_Provider IP

2)- Cannot dial any extension when I call the Auto Attendant

3) Can I put both the Outgoing and Incoming ARS in the same ARS

4)- I am about to bought couple of Did from didx.net to allow external users to access to a conference room. How to configure the did numbers in the Pbx.

5)- How to force the PBX to use a codec from the configuration options?

Best rgds[/b]

1) for outgoing calls, please set different prefix other than 1111 in pattern-out,
when dial number with prefix 1111, the call will apply pattern-in first which is also defined with prefix 1111

2) for auto attendant, please check auto attendant setting for "Max input digits" and "speed dial"
if any conflict with you dialed number

3) yes, ars pattern in and out can be in the same ARS rule

4) you can use default ARS template named "gw1" for setting up DID numbers, put DID numbers in v1 column one on each row
and put pbx extension numbers in v3 column for each row with DID number

5) at pbx side, codecs can be defined in each user ext, ARS rule and [Options]
default codec is g711u
http://wiki.brekeke.com/wiki/Restrict-o ... cs-to-G729

Thanks. I'll do the testing and get back to yu asap.

1) for outgoing calls, please set different prefix other than 1111 in pattern-out,
when dial number with prefix 1111, the call will apply pattern-in first which is also defined with prefix 1111

I have put a different prefix but I still have the same problem. If both active, no incoming call thru the Pbx. If I disable the ARS OUT, the ARS IN works and vice versa.

2) for auto attendant, please check auto attendant setting for "Max input digits" and "speed dial" if any conflict with you dialed number

I am unabled to dial any extension after having reach the Auto Attendant.

5) at pbx side, codecs can be defined in each user ext, ARS rule and [Options] default codec is g711u

Do I have to key the numeric code of the codec or the G711u

Regards

Sort out the Outgoing call issues. It was due to disable services in Windows Server 2008 : RRA service.

Now both incoming and outgoing from/to SIP PROVIDER are working well.

But the voicemail on the Pbx extension is not working when the call is coming from the SIP PROVIDER with public IP address.

Regards,

- does voicemail work when call between pbx extensions?
- how you dial to pbx extension voicemail from external users? transfer from auto attendant?

- does voicemail work when call between pbx extensions?

No.

- how you dial to pbx extension voicemail from external users? transfer from auto attendant?

Both options have the same results. I am unable to dial any extension or conference room.

from a phone registered at pbx and assigned to a pbx user
dial 07*ext_number, can you hear voicemail greeting?

for dial voicemail from auto attendant, set
Transfer to unregistered users: enable
Max input digits: 10
remove any speed dial setting when testing

call to auto attendant and dial 07*ext_number to leave voice message this extension
or dial 08*ext_number, to retrieve voicemail of this extension