Brekeke Forum - Search : Brekeke PBX, Brekeke SIP Server, Multi-Tenant Hosted PBX

Thanks for the information.if i enable ICE the packet size goes beyond MTU limit size 1500 bytes.so call landing issues coming in low bandwith network.so to avoid that i have disabled that option.can ...

i am using imsdroid client. i have disabled stun and ice. but sip server behive correctly and works if private and public ip address like this

10.35.xxx.xxx
112.79.xxx.xxx

bu ...

i have tried with the dial plan,but still its using the private ip address for RTP and INVITE.

1. Brekeke Product Name and Version:
brekeke sip server 3.1.7.8/348.2
2. Java version:
1.7.0_21
3. OS type and the version:
Windows
4. UA (phone), g ...

Hi,
Thanks.its working now.

Regards
Bensly

1. Brekeke Product Name and Version:

2. Java version:

3. OS type and the version:

4. UA (phone), gateway or other hardware/software involved:

5. Your problem:

i have deployed two dial ...

if i use dial plan &net.rtp.ifsrc = 59.90.246.89 ,&net.rtp.ifdst = 59.90.246.89 like this, still it does't replace to global ip address.

For RTP/AVPF problem already i have pasted the SIP ...

i have checked the same as you suggested instead of advance page,if i put dial plan like this

$request = ^INVITE $ifsrc = 59.90.246.89
...

I have tried as you suggested below in advance page

net.rtp.ifsrc=59.90.246.89
net.rtp.ifdst=59.90.246.89

but still it does't replace to public IP address.

>> Are SIP Server and SIP clients located in the same LAN behind NAT?

yes, sip clients and sip server are located in same LAN behind NAT.
we are registering the client using public IP. b ...

The IP address as you said is correct. and in Interface also listed my local and public IP. for more information i have enabled the sip server logs. please find the log below to get identify the probl ...

Sip server behind NAT,and i have configured local and public IP in interface 1 and 2. and RTP replay is ON , RTP Relay ( UA) is auto.port mapping is source port, and send UA remote address is yes.

...

Hi,
[Port Mapping] settings selected as source port only.and RTP relay mode is on.still same problem.client using local IP address in the SDP,but sip server not replacing the public IP address. i ...

Hi,
i am using imsdorid client in android mobile.as you said in SDP local ip address only sending by the client.but after enabled Send UA's remote address= yes also the problem is same.

Regards ...

1. Brekeke Product Name and Version: v3.1.7.8

2. Java version:

3. OS type and the version: Windows 2003

4. UA (phone), gateway or other hardware/software involved:

5. Your problem:
...

Hi,
i have upgraded to v3.1.7.8. but still the issue there.i am creating new post for this.please replay on the new post.

Regards
Bensly

Hi,
Thanks for the solutions and its worked for me.but now i am facing another problem is both side no audio.but call successfully established. i am able to see the RTP packet goes to sip server f ...

Hi laurie.
Thanks for your suggestion. i will check the setting as you said.and let you know the status shortly.

Regards
Bensly

1. Brekeke Product Name and Version:

2. Java version: 1.5.0_06

3. OS type and the version: Linux 2.6.9-89.ELsmp

4. UA (phone), gateway or other hardware/software involved:

5. Your problem: ...

Thanks for your great help, and its works well now.

yes i want to make call between registered users.

and as per the log it says "Packet Too Big"

log is:
7, sip:666@192.168.1.237, sip:125@192.168.1.237, 0, 1357543153468, , 135754315345 ...

After deployed the dial plan also the call declined. if mobile in 3G network also the call declined.if i enable ICE, all the scenario the call was declined.

i am using imsdroid app in android mobile.

we are using static ip address only.
we are developed mobile based app and given to the client.so client does't want to know and change the IP address and NAT info.so we are used static public IP ...

i have added the public ip address in the interface address.but still it decline the request.previously it was works.