|Summary:||ASTERISK-15773: incoming INVITE received no progress, just 200 OK, causing Sipra pstn to go off-hook|
|Date Opened:||2010-03-08 22:12:05.000-0600||Date Closed:||2011-06-07 14:00:59|
|Description:||With a Sipura 3102 using asterisk as its proxy (pstn->voip). the Sipura sends asterisk an INVITE, then asterisk responds with a "100 Trying" almost immediately, but then as soon as asterisk starts dialing extensions to handle the incoming call it sends "200 OK" to the Sipura. This causes the Sipura to prematurely go off-hook.|
Shouldn't asterisk be sending a "180 Ringing" to the Sipura until one of the extensions answers the call at which point the channels would be bridged?
****** ADDITIONAL INFORMATION ******
From memory, the previous version of asterisk I used (v1.2.13), did not cause the Sipura to go off-hook until one of the dialed extensions answered.
|Comments:||By: Leif Madsen (lmadsen) 2010-03-10 10:33:35.000-0600|
You'll need to provide the dialplan you're using along with the SIP trace of the call. Both of those things are at a minimum required when filing SIP issues.
This page may also be a useful read: http://www.asterisk.org/developers/bug-guidelines
Item 6 under Opening An Issue In The Issue Tracker - A Checklist deals with filing SIP related issues.
By: jw-asterisk (jw-asterisk) 2010-03-10 18:17:12.000-0600
Well there is absolutely no way I am going to be able to do that. Because asterisk peppers all of its output with a myriad of ^M's (as though it was running on MS Windows), and it also reveals juicy details like my IP addresses, port numbers, extension names, and more.
Having recently survived a hackers prolonged attack on my machine which consumed much of my available bandwidth (you cannot stop UDP packets arriving on ones doorstep), I am not going to publish handy details about my configuration. Sorry.
By: Leif Madsen (lmadsen) 2010-03-15 13:11:59
Well you can easily obfuscate the revealing information with a search and replace.
By: Paul Belanger (pabelanger) 2010-04-28 15:57:56
Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.
Further information can be found at http://www.asterisk.org/developers/bug-guidelines