Summary:	ASTERISK-07341: Problems with parsing SIP URI Good ( 1.2.4 ) Bad ( 1.2.9.1 )
Reporter:	Enrique Martinez (enmaca)	Labels:
Date Opened:	2006-07-13 15:14:08	Date Closed:	2011-06-07 14:08:06
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/Interoperability
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) sip_trace_sonusuri.log
Description:	Hi all; Im testing a migration scenario with sonus and have experiencing some problems with version 1.2.9.1. Whit version 1.2.4 the call completed ok, but with 1.2.9.1 the call its dropped with a 404 Not Found Response. See the attach file for the sip traces ( 1.2.4 OK and 1.2.9.1 BAD ). Im adding four ast_log for debugging purposes. Im not to much familiar with the chan_sip.c
Comments:	By: Serge Vecher (serge-v) 2006-07-13 15:50:16 1. Please don't post code inline, but as patch-file only. 2. Please get a disclaimer on file and confirm when done with a note, so your code can get reviewed. 3. As per bug guidelines, we need to see a SIP debug of failing and successful transactions. 1) Prepare test environment (reduce the ammount of unrelated traffic on the server); 2) Make sure your logger.conf has the following line:   console => notice,warning,error,debug 3) restart Asterik. 4) Enable SIP transaction logging with the following CLI commands: set debug 4 set verbose 4 sip debug 5) Save complete console log to file and _attach_ said file to the bug for both "good" and "bad" case and post here as attachments. Thanks! By: Enrique Martinez (enmaca) 2006-07-13 16:36:36 | 1. Please don't post code inline, but as patch-file only. Sorry, Next Time :) | 2. Please get a disclaimer on file and confirm when done with a note, so your code can get reviewed. Its not a funciontality added or bugfix only its a debug option, so you car review the file attached and see the variables. If even you need a disclaimer its ok. | 3. As per bug guidelines, we need to see a SIP debug of failing and successful transactions. the file  sip_trace_sonusuri.log contains both of the traces If you need something else letme know By: mikma (mikma) 2006-07-15 03:08:08 Looking at the log it seems that 1.2.9.1 is acting correctly. The call is directed to the s extension since your uri contains a host name (8147779810), but no user name.    To: <sip:8147779810;phone-context=private@200.53.97.234:5060;user=phone> By: Serge Vecher (serge-v) 2006-07-17 08:23:26 enmaca: ok, I'm removing inline code since it's not disclaimed and is not a bug fix anyway. By: Serge Vecher (serge-v) 2006-07-28 13:58:54 ok, since file has committed r38420 to 1.2 branch, can you please test it and see if your problem is fixed now? By: Joshua C. Colp (jcolp) 2006-07-28 15:35:49 I'll just steal this from oej since I fixed it... By: Andrew Lindh (andrew) 2006-07-29 17:40:48 The chan_sip.c fix "r38420" broke calls from my polycom phones (SIP 1.6.7). I opened a bug for that new problem (may be I should not have?) ASTERISK-742117 By: Serge Vecher (serge-v) 2006-07-31 09:36:32 ok, let's see if 1.2 branch with > r35801 fixes this issue ... By: Olle Johansson (oej) 2006-08-07 10:31:17 What's the status with this? By: Olle Johansson (oej) 2006-08-07 10:37:46 Waiting for replies.