Summary:ASTERISK-09283: The call through SIP provider remains ringing after hangup
Reporter:gresko (gresko)Labels:
Date Opened:2007-04-19 08:16:56Date Closed:2011-06-07 14:07:28
Versions:Frequency of
Environment:Attachments:( 0) opbsipdebug
( 1) sipdebug
Description:When I make an external call through SIP provider and hangup before remote party picks up the called party is ringing forever.
Comments:By: Joshua C. Colp (jcolp) 2007-04-22 20:13:11

Without more information, like a sip debug and console output it is impossible to even begin to debug this. Please provide it.

By: gresko (gresko) 2007-04-23 04:17:29

The setup of my provider looks messed up by me. It listens for SIP call at port 5061 at IP1. Then it opens RTP stream from IP2. Could this be the source of problems?

I also found this setup messes up nf_conntrack_sip. I cannot hear ring (the RTP stream is filtered) until I open udp from IP2 manually - nf_conntrack_sip ignores this communication and i have added port 5061 to list of ports. However the call after pick up is OK whether I open the UDP from IP2 or not.

By: Joshua C. Colp (jcolp) 2007-05-17 10:34:59

It is perfectly normal to have the RTP at a different location. Everything seems normal though on the Asterisk side. Provider says the ringing audio is going to be provided over RTP and then when the call is hungup before being answered the provider replies back fine.

Where would you like to go from here? Is this still an issue on our side?

By: gresko (gresko) 2007-05-18 09:00:20

I tested the same setup and same provider with openpbx and everythink is working OK. So the problem should be on the asterisk side I think. The PSTN line remains ringing forever.

By: gresko (gresko) 2007-06-25 04:05:09

Problem persists in asterisk 1.4.5. What could be the cause?

By: Matt Riddell (zx81) 2007-06-25 21:59:51

Could you test with the latest SVN version of 1.4?

We had the same problem, and I updated a machine (last test was 2-3 weeks ago, latest today) and the problem is gone.

By: gresko (gresko) 2007-07-02 02:21:01

I tested it with version 1.4.6 and the problem still persists. I expect the 1.4.6 is newer version than the svn release you mentioned.

By: Joshua C. Colp (jcolp) 2007-07-05 09:47:22

Since this works with OpenPBX.org/Callweaver can you provide sip debug from there for comparison?

By: gresko (gresko) 2007-07-10 09:36:11

I added SIP debug from OpenPBX.

By: gresko (gresko) 2007-08-20 03:55:23

I have some new observations of the problem. The problem I described was through the Internet provider by optic fibre. I installed another machine with the same configuration a put it behind WiFi Internet provider - same results. When the WiFi connection is too buggy, openpbx have the same behavior. But, when I connect the same machine thorugh the ADSL Internet provider same as VoIP provider both asterisk and openpbx work as expected. Weird. Isn't it?

By: gresko (gresko) 2007-10-23 04:38:25

Solved. There should be nat=never instead of nat=no in the sip trunk configuration.

By: Joshua C. Colp (jcolp) 2007-10-23 08:11:44

Closed, configuration issue.

By: gresko (gresko) 2007-12-17 07:55:04.000-0600

I tried it once more and it has previous behaviour. The nat=never options does not help any more. Maybe it was working because of some temporary change on my providers side in the same time I tried nat=never. Now I use asterisk-1.4.15.

By: Joshua C. Colp (jcolp) 2008-02-11 09:56:31.000-0600

How about updated sip debug plus a packet capture using something like tcpdump or ethereal to confirm the packets are going out unaltered?

By: jmls (jmls) 2008-02-17 13:13:26.000-0600

gresko, any news on the updated debug as requested ?

By: Joshua C. Colp (jcolp) 2008-04-14 11:38:48

Suspending due to lack of response. If you can get that info I will gladly look at this.