Summary:	ASTERISK-24848: INVITE not send by asterisk on chan_pjsip when endpoint has TCP Transport
Reporter:	Javier Riveros (goseeped)	Labels:
Date Opened:	2015-03-04 16:14:32.000-0600	Date Closed:	2015-03-06 11:47:47.000-0600
Priority:	Major	Regression?	Yes
Status:	Closed/Complete	Components:	Channels/chan_pjsip
Versions:	12.8.1 13.0.0 13.1.1 13.2.0	Frequency of Occurrence	Constant
Related Issues:	duplicates ASTERISK-22658 PJSIP: If a transport is set on an endpoint, Asterisk will not reuse established connections for that endpoint
Environment:	Asterisk 13.2 OS : Ubuntu 14.04 Standart softphone	Attachments:	( 0) Full_Ast_Debug_INVITE_TCP.txt ( 1) pjsip_conf.txt ( 2) sip_capture.txt
Description:	Asterisk not send INVITE to TCP transport endpoint , captures reflect that and also after some seconds it raise the following error. {code} [Mar  4 21:48:25] ERROR[14964]: pjsip:0 <?>:  tcpc0xb48588 TCP connect() error: Connection timed out [code=120110] {code} The result of this is that you can't make a SIP call to an endpoint that has TCP transport enable on chan_pjsip. To reproduce this use a one or two SIP standart client and set this up with TCP transport. Full Debug , sip capture and pjsip_conf is attached
Comments:	By: Joshua C. Colp (jcolp) 2015-03-06 11:06:14.090-0600 Same here, except since it's an endpoint that is dynamic remove the transport= line and set rewrite_contact=yes. This will cause the established TCP connection to get used instead of creating a new one. By: Javier Riveros (goseeped) 2015-03-06 11:41:44.877-0600 Thank you Joshua , i failed to find information about it in wiki , so since it not being generating the INVITE i thought it was a really bug , thanks for not closing it as tech support , hope this help to others , please close this and sorry for the noise.