Call drops @ :58 seconds

1. Brekeke Product Name and version:
Brekeke OnDO SIP Server for OnDO PBX
Version 1.5.3.0

2. Java version:
J2SE 1.5.0

3. OS type and the version:
XP Pro SP2

4. UA (phone), gateway or other hardware/software involved:
X-Lite & Linksys SPA941

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
Pattern 3

6. Your problem:
Hello, I'm having a problem that I can't seem to pin down. We have a server running both the OnDo SIP/PBX services. In addition, we also have an additional SIP proxy server. The 2nd SIP proxy is more restrictive so that users that connect can only dial internal #s. We also have a Multitech gateway in place so that our OnDo PBX can communicate with our Mer!d!@n PBX.

Before I go any further, if I setup a sofphone/IP phone & register with the 1st SIP/PBX server, I have no issues whatsover, it works great. However, we have certain users that will need to be restricted to internal calling, therefore I point them to the 2nd SIP server with a restricted Dial Plan. When I point a user to the 2nd SIP server, I can call internally from my Mer!d!@n hard line phone to their softphone/IP phone without any problems. However, if they initiate the call from their softphone/IP phone & dial my Mer!d!@n extension, the call ALWAYS cuts off at exactly :58 seconds (according to the SIP log). The call shows as SUCCESS.

Could this be a problem with a dial plan on my 2nd SIP server? I have tried initiating calls from 2 different locations, both with the same results. I am a novice when it comes to dial plans.

If I may ask a 2nd question...do I need this 2nd SIP in order to "split" the 2 classes of users? Group A that connects to the 1st SIP/PBX server can go thru our Mer!d!@n PBX & make external calls as well as internal. Group B is restricted to calling internal 4 digit extensions only. What is the best way to implement this?

Many thanks for your assistance.

For your 2nd question, you can use only one SIP server and restrict group B users to call out by dial plan.

Matching Patterns:
$request=^INVITE
From=sip:(prefix.*)@ //group B users should have different prefix from A
To=sip:(....)@ //the number of digits for the external line

Deploy Patterns:
$action=603

Here are some links for Dial Plan rules:
http://www.brekeke-sip.com/download/bss ... lan_en.pdf
http://www.brekeke-sip.com/wiki/ (under "Brekeke SIP Server FAQ"/Software Detail, be patient a little bit slow for showing the content, but useful)

Hi I have had this problem, but it was due to ping issues, basically the called / calling gateway would ping my server IP to make sure I was still alive, so 90 seconds after connect it would ping and if it got no reply (regardless of call state RTP etc) the call would drop. As my server had ICMP turned off this would happen for every call! in the end I found the setting on the gateway and disabled it, otherwise you could turn on ICMP or this may not be the issue at all.

The other thing it could be is again polling ,the called / calling gateway may send a SIP message (can't remember which is maybe a notify message similar to the NAT keep alive message) and if you don't respond to it, or respond with error the gateway will drop the call.


Either way Wireshark/Ethereal trace will tell you who drops it and maybe why.

Hi kjyopilot,

Let me know what kind of softphone you are using.
Even if you try another softphone product, does the issue still happen?

Also, where is the gateway placed? Is it the same LAN? or behind NAT??
If there are NAT between gateway and softphone, you need to add some settings in the router.

Anyway, it is good idea to use a capture tools (Wireshark..) as Wayne mentioned.

I suppose the softphone didn't send ACK packet correctly...

Thank you all for your suggestions. I followed Hope's suggestion & enabled a new dial plan for the users that must have restricted access. This worked well.

Now I have time to examine why the other SIP drops the calls @ :58 seconds. I actually don't need this SIP, but I am curious to find out why it is dropping. The only difference I see now is that the SIP in question runs on XP Pro SP2, & our SIP/PBX runs on W2K3.

Wayne, I tried enabling ICMP, but to no avail. I will take your suggestion as well as Mohney's & run a sniffer on there.

Thank you all for your suggestions!

Hi..

After you made a call to gateway from a softphone and before 58 seconds is passed, what kind of "Status" can you see at [Active Sessions] page?
If it is "Talking", the SIP session is established correctly..