Author |
Message |
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Tue Aug 07, 2012 6:08 pm Post subject: 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/networkpatterns.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. _________________ Nomad |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Wed Aug 08, 2012 11:21 am Post subject: |
|
|
- 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? |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Sun Aug 12, 2012 3:47 am Post subject: |
|
|
- 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@
®ister.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" _________________ Nomad |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Mon Aug 13, 2012 11:07 am Post subject: |
|
|
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:(.+)@ |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Mon Aug 13, 2012 12:40 pm Post subject: |
|
|
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. _________________ Nomad |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Tue Aug 14, 2012 11:36 am Post subject: |
|
|
Quote: |
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
Quote: |
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 |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Tue Aug 21, 2012 8:16 am Post subject: |
|
|
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 _________________ Nomad |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Tue Aug 21, 2012 10:20 am Post subject: |
|
|
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/v3_x/bssv3_admin_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 |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Tue Aug 21, 2012 5:47 pm Post subject: |
|
|
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 _________________ Nomad |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Wed Aug 22, 2012 10:18 am Post subject: |
|
|
Quote: |
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
Quote: |
3) Outgoing calls from the Pbx to Sip server no longer working |
with above changes the outgoing call will work
Quote: |
1) I cannot dial any extension from the Auto Attendant menu. |
if call from other extension, can dial other ext from auto attendant?
Quote: |
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 |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Thu Aug 23, 2012 5:41 pm Post subject: |
|
|
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] _________________ Nomad |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Fri Aug 24, 2012 8:58 am Post subject: |
|
|
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-outbound-call-codecs-to-G729 |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Fri Aug 24, 2012 1:17 pm Post subject: |
|
|
Thanks. I'll do the testing and get back to yu asap. _________________ Nomad |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Sat Aug 25, 2012 9:14 am Post subject: |
|
|
Quote: |
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.
Quote: |
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.
Quote: |
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 _________________ Nomad |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Sun Aug 26, 2012 6:04 pm Post subject: |
|
|
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, _________________ Nomad |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Mon Aug 27, 2012 11:15 am Post subject: |
|
|
- does voicemail work when call between pbx extensions?
- how you dial to pbx extension voicemail from external users? transfer from auto attendant? |
|
Back to top |
|
nomad Brekeke Member
Joined: 05 Aug 2012 Posts: 10
Location: SSA
|
Posted: Tue Aug 28, 2012 5:58 am Post subject: |
|
|
Quote: |
- does voicemail work when call between pbx extensions? |
No.
Quote: |
- 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. _________________ Nomad |
|
Back to top |
|
hope Brekeke Master Guru
Joined: 15 Jan 2008 Posts: 862
|
Posted: Fri Aug 31, 2012 9:34 am Post subject: |
|
|
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 |
|
Back to top |
|
|