Call problem
Moderator: Brekeke Support Team
Call problem
1. Brekeke Product Name and version: Brekeke SIP Server 2.2.6.2 Academic
2. Java version:
3. OS type and the version: Win2000 Professional SP4
4. UA (phone), gateway or other hardware/software involved: X-Lite, Cisco 2621
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
6. Your problem:
Hi all, i am doing some test on the SIP Server and all working well but one thing. I explain my network schema:
- Brekeke SIP Server with private IP 192.168.40.x and a static nat from Cisco 2621 with public static IP;
- I can registrar telephone (Linksys, Cisco) and softphone (X-Lite) and others from internet (public IP) and from intranet (private IP, from class 10.x.x.x) with no problem.
So, this is the story.
Now, when I call from indise (private) to indise (private) all is working well.
From indise (private) to outside (public) all is working well.
From outside (public) to outside (public) all is working well.
From outside (public) to indise (private) NOT work.
When I try, the remote doesn't ring and my X-Lite rest in "calling..."
The same thing with other phone (Cisco, other UA).
I have noticed on the SIP Server that when I call, the "active session" page show me the call from "my_number@192.168.40.x:port" to "remote@public_IP" and it doesn't work.
When I try outside-outside call, for example, I see "my_number@192.168.40.x:port" to "remote@192.168.40.x:port" and it work well.
I think the problem is there.
Hope make me clear.
Tnx in advance
2. Java version:
3. OS type and the version: Win2000 Professional SP4
4. UA (phone), gateway or other hardware/software involved: X-Lite, Cisco 2621
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
6. Your problem:
Hi all, i am doing some test on the SIP Server and all working well but one thing. I explain my network schema:
- Brekeke SIP Server with private IP 192.168.40.x and a static nat from Cisco 2621 with public static IP;
- I can registrar telephone (Linksys, Cisco) and softphone (X-Lite) and others from internet (public IP) and from intranet (private IP, from class 10.x.x.x) with no problem.
So, this is the story.
Now, when I call from indise (private) to indise (private) all is working well.
From indise (private) to outside (public) all is working well.
From outside (public) to outside (public) all is working well.
From outside (public) to indise (private) NOT work.
When I try, the remote doesn't ring and my X-Lite rest in "calling..."
The same thing with other phone (Cisco, other UA).
I have noticed on the SIP Server that when I call, the "active session" page show me the call from "my_number@192.168.40.x:port" to "remote@public_IP" and it doesn't work.
When I try outside-outside call, for example, I see "my_number@192.168.40.x:port" to "remote@192.168.40.x:port" and it work well.
I think the problem is there.
Hope make me clear.
Tnx in advance
>> From outside (public) to indise (private) NOT work.
>>
>> I have noticed on the SIP Server that when I call, the "active
>> session" page show me the call
>> from "my_number@192.168.40.x:port" to "remote@public_IP"
>> and it doesn't work.
At that time,what kind of "Status" can you see in the [Active Sessions] page?
Is it "Inviting"? or Is it "Ringing"?
>>
>> I have noticed on the SIP Server that when I call, the "active
>> session" page show me the call
>> from "my_number@192.168.40.x:port" to "remote@public_IP"
>> and it doesn't work.
At that time,what kind of "Status" can you see in the [Active Sessions] page?
Is it "Inviting"? or Is it "Ringing"?
Probably I was not clear in my previous post.
When I said that I see "my_number@192.168.40.x:port" (and I was nat behind a public IP address) to "remote@public_IP", this remote UA is a Linksys SPA941 registred to the SIP server with 103@10.0.1.2:5060 because its local with the server.
The "remote@public_IP" that I see is the static public IP address that I have assigned to the SIP Server.
For best comprehension I put the info like apper to me:
< Registrered Clients >
103 sip:103@10.0.1.2:5060 Expires : 3600
Priority : 1000
User Agent : Sipura/SPA941-4.1.8
Requester : 10.0.1.2:5060
(the "remote" Linksys SPA941 for me)
12345 sip:12345@217.133.47.15:10728 Expires : 3600
Priority : 1000
User Agent : X-Lite release 1100l stamp 47546
Requester : 217.133.47.15:10728
(my X-lite phone)
< Active sessions >
EX-SID 4224
From-uri sip:12345@192.168.40.100
From-ip 217.133.47.15:10728 (UDP)
From-if 192.168.40.100:5060 ***private IP of the SIP server
To-uri sip:103@static_public_ip_address_of_the_SIP_server
To-ip static_public_ip_address_of_the_SIP_server (UDP)
To-if 192.168.40.100:5060
Call-ID ZmI2NTg1NjQ1ODg2ODMxNjY0MWNiNmYyYTNhNjg2ZmI.
rule outbound
plug-in InviteSession
sip-packet-total 1
listen-port 5060
session-status Inviting
rtp-relay off
Brekeke SIP Server , Version 2.2.6.2 Academic
ID: 00039920
Copyright © 2002-2008, Brekeke Software, Inc.
I did not understand how can, the SIP server, address the "To-uri" field with the public static IP that I assigned to instead to address it with the private one the 101 user registered itself (10.0.1.1)....
Hope that its clear now
Tnx you
When I said that I see "my_number@192.168.40.x:port" (and I was nat behind a public IP address) to "remote@public_IP", this remote UA is a Linksys SPA941 registred to the SIP server with 103@10.0.1.2:5060 because its local with the server.
The "remote@public_IP" that I see is the static public IP address that I have assigned to the SIP Server.
For best comprehension I put the info like apper to me:
< Registrered Clients >
103 sip:103@10.0.1.2:5060 Expires : 3600
Priority : 1000
User Agent : Sipura/SPA941-4.1.8
Requester : 10.0.1.2:5060
(the "remote" Linksys SPA941 for me)
12345 sip:12345@217.133.47.15:10728 Expires : 3600
Priority : 1000
User Agent : X-Lite release 1100l stamp 47546
Requester : 217.133.47.15:10728
(my X-lite phone)
< Active sessions >
EX-SID 4224
From-uri sip:12345@192.168.40.100
From-ip 217.133.47.15:10728 (UDP)
From-if 192.168.40.100:5060 ***private IP of the SIP server
To-uri sip:103@static_public_ip_address_of_the_SIP_server
To-ip static_public_ip_address_of_the_SIP_server (UDP)
To-if 192.168.40.100:5060
Call-ID ZmI2NTg1NjQ1ODg2ODMxNjY0MWNiNmYyYTNhNjg2ZmI.
rule outbound
plug-in InviteSession
sip-packet-total 1
listen-port 5060
session-status Inviting
rtp-relay off
Brekeke SIP Server , Version 2.2.6.2 Academic
ID: 00039920
Copyright © 2002-2008, Brekeke Software, Inc.
I did not understand how can, the SIP server, address the "To-uri" field with the public static IP that I assigned to instead to address it with the private one the 101 user registered itself (10.0.1.1)....
Hope that its clear now
Tnx you
james wrote:Hi
Does your SIP server have two private IP addresses?
One is 192.168.40.x. Another is 10.0.1.x. Is it correct?
What kind of interface IP addresses can you see in the [Server Status] page?
Really not....
SIP server has only one private address: 192.168.40.100.
UA/Phone on 10.0.1.x reach it via routing and exactly via one cisco router with two FE interface: one with IP address on net 10.x.x.x/8 and the other on net 192.168.40.x/24
The <Status> page look like this:
<Server Status>
Status ACTIVE
server-product Brekeke SIP Server
server-ver 2.2.6.2/276
server-name TEST_SIP_SERVER
machine-name SIP-Server
listen-port 5060
transport UDP
interface 192.168.40.100, 192.168.40.101
startup-user SYSTEM
work-directory C:\Programmi\Brekeke\proxy
session-active 0
session-total 2
session-peak 2
registered-record 2
os-name Windows 2000
os-ver 5.0
java-ver 1.4.2_06
admin-sip
admin-mail
Database Status
registered-database Status : Connected
Error : 0
userdir-database Status : Connected
Error : 0
alias-database For Advanced Edition Only
Brekeke SIP Server , Version 2.2.6.2 Academic
ID: 00039920
Copyright © 2002-2008, Brekeke Software, Inc.
The "interface" line report two IP because I configure two IP address on the "network tool" of Windows 2000 Professional server where run Brekeke SIP Server software; we need to reach it from remote via VNC.
Hope is not the problem....