Author Topic: Calls released / dropped in SIP server  (Read 5829 times)

Offline krish_kk123

  • Newbie
  • *
  • Posts: 23
  • Karma: 0
Calls released / dropped in SIP server
« on: March 24, 2011, 12:07:36 AM »
Hi,

We are having issues with the SIP server, where URS issues a route request to SIP server for routing the call to 0800 number which is an external site. The problem is...SIP server is showing session progress and then busy here....no event ringing,because the destination is busy, and it does not generate any event error to URS. in normal scenarios, SIP server generates EventError set to the 231 code "DN is Busy".

Here below are the SIP snippets with order...Can anyone of you please tell me how do we handle this BUSY message to control the call from dropping.

INVITE sip:0800xxxxxx@x.218.237.147:5060 SIP/2.0
From: <sip:xxx@x.218.237.134:5060>;tag=002AC54C-325C-1D3E-93FB-84EDDA0AAA77-17014978
To: <sip:vvvv@x.218.237.134:5060>
Call-ID: 002AC52E-325C-1D3E-93FB-84EDDA0AAA77-12082807@10.218.237.134
CSeq: 1 INVITE

SIP/2.0 100 Trying

SIP/2.0 183 Session Progress

SIP/2.0 183 Session Progress

SIP/2.0 183 Session Progress

SIP/2.0 183 Session Progress

SIP/2.0 486 Busy here


Regards
Karthik


Offline René

  • Administrator
  • Hero Member
  • *****
  • Posts: 1800
  • Karma: 59
Re: Calls released / dropped in SIP server
« Reply #1 on: March 24, 2011, 02:58:49 AM »
Hi Karthik,

I met this issue last year and the root cause is that SIP Server generates EventRouteUsed message when it receives 183 Session Progress message. And this is signal for URS to stop the routing strategy. There are two ways how to solve it

1/ Disable support for provisional message at application level by setting the 'sip-enable-100rel' option to 'false'.
2/ Disable provisional messages from particular call from the routing strategy by attaching 'sip-enable-100rel' key with value 'false' to the Extensions attribute of RequestRouteCall (supported starting SIP Server release 8.0.300.44).

R.

Offline krish_kk123

  • Newbie
  • *
  • Posts: 23
  • Karma: 0
Re: Calls released / dropped in SIP server
« Reply #2 on: March 24, 2011, 06:07:43 PM »
Hey Rene,

thank you so much for your quick turn around. I will try implement the options the you have given. Will it be possible for you to explain a bit more on the provisional message that you are talking about ? am sorry, am really new to this and got no clarity on what it is...so...

Regards
Karthik

Offline René

  • Administrator
  • Hero Member
  • *****
  • Posts: 1800
  • Karma: 59
Re: Calls released / dropped in SIP server
« Reply #3 on: March 29, 2011, 03:03:43 AM »
Hi Karthik,

Look at the Introduction chapter of the RFC 3262 - http://www.ietf.org/rfc/rfc3262.txt - that defines the provisional message.

R.

Offline krish_kk123

  • Newbie
  • *
  • Posts: 23
  • Karma: 0
Re: Calls released / dropped in SIP server
« Reply #4 on: March 30, 2011, 07:26:21 PM »
Thank you for the link...very useful.

FYI...the problem was identified, and it is due to "divert-on-ringing" set as true. We had set it to True due to some RONA issue and now it looks like we have to set it to False and find out a different way of handling rona issue.

Thank for your help again!

Regards
Karthik

Offline krish_kk123

  • Newbie
  • *
  • Posts: 23
  • Karma: 0
Re: Calls released / dropped in SIP server
« Reply #5 on: July 14, 2011, 06:27:09 PM »
Hello Guys,


Am again stuck in the same issue.

Last time we had an issue of losing calls when it was routed to an extenal partner number. So we decided to Turn Divert-on-ringing=false and the issue was resolved. but this time, it has caused an reporting issue in infomart, Mediation_Segment_Fact data is missing: the call segment which shows that the call is handled by an agent cannot be found. The only data in Mediation_Segment_Fact is the RONA segment.

any suggestions ?

Regards
Karthik

Offline René

  • Administrator
  • Hero Member
  • *****
  • Posts: 1800
  • Karma: 59
Re: Calls released / dropped in SIP server
« Reply #6 on: July 21, 2011, 05:20:25 PM »
Hi Karthik,

It's really strange and I would doubt that change of divert-on-ringing attribute caused this issue. Have you checked Interaction Concentrator and Info Mart log?

R.

Offline krish_kk123

  • Newbie
  • *
  • Posts: 23
  • Karma: 0
Re: Calls released / dropped in SIP server
« Reply #7 on: January 09, 2012, 10:21:28 PM »
Hey Rene,

Sorry for responding so late on this post.
yes when divert on ringing was set to false, it is designed in such a way that it will not create interactions after any diverts and hence no data was written for calls which diverted from segment a to b to c.....