search for: ast_rtp_instance_new

...8]: chan_sip.c:25057 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729) [Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for 4870fab953c16a611b9248584748fe59 at 127.0.0.1:0 - INVITE (No RTP) [Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x9ded230' [Mar 19 18:22:56] DEBUG[17418]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 19718 for RTP instance '0x9ded230' [Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:353 ast_rtp_instance_new: RTP instance '0x9ded230'...

...ng initreq for method INVITE - callid cf2990ba- 3f12-3d9e-adb6-52889c414ed3 Using INVITE request as basis request - cf2990ba-3f12-3d9e-adb6-52889c414ed3 Found peer 'vr1a882' for 'vr1a882' from 2.2.2.2:8558 [May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:421 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x9072064' [May 19 16:47:43] DEBUG[14160][C-00000007]: res_rtp_asterisk.c:2437 ast_rtp_new: Allocated port 17304 for RTP instance '0x9072064' [May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:430 ast_rtp_instance_new: RTP i...

...gt; func_callerid.so load => func_cut.so load => func_logic.so [global] Since Asterisk 16 (Debian Buster version) I have a dependency problem, where res_rtp_asterisk.so is dependent on res_pjproject.so: When I try to make a call: [Dec 22 22:00:55] ERROR[6093][C-00000001]: rtp_engine.c:474 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? mng*CLI> module load res_rtp_asterisk Unable to load module res_rtp_asterisk Command 'module load res_rtp_asterisk' failed. [Dec 22 22:03:39] ERROR[28261]: loader.c:170 module_load_error: res_rtp_asterisk loaded before dependency res_pjpr...

I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues with ARM and this problem but did not help me, not using gtalk or anything. Just call between two polycom phones on local network. Tried looking at the config.log for rtp anything and it looks ok. Anyone know what m...

...'. [Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting '192.168.2.172' into... [Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport: ...host '192.168.2.172' and port ''. [Jan 28 23:03:32] DEBUG[1654]: rtp_engine.c:345 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xb7507ae8' [Jan 28 23:03:32] DEBUG[1654]: res_rtp_asterisk.c:499 ast_rtp_new: Allocated port 18530 for RTP instance '0xb7507ae8' [Jan 28 23:03:32] DEBUG[1654]: rtp_engine.c:354 ast_rtp_instance_new: RTP instance '0xb7507ae8'...

I just upgraded to fedora 21. I'm running asterisk 11.6.0. All works with Fedora 20. -- Executing [s at DialOut:15] Dial("DAHDI/1-1", "motif/8447/+1212xxxyyyy at voice.google.com,,rTt") in new stack [Mar 1 21:24:06] ERROR[2477][C-00000000]: rtp_engine.c:259 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? [Mar 1 21:24:06] ERROR[2477][C-00000000]: chan_motif.c:1820 jingle_request: Unable to create Jingle session on endpoint '8447' ???? any help appreciated. sean