search for: ast_rtp_bridg

...from. It is behind NAT. It appears that the app_prepaid is not taking this into consideration since I see: Jun 18 17:46:25 DEBUG[1133742896]: chan_sip.c:4130 build_route: build_route: Contact hop: <sip:7708183799@192.168.1.101:5060;line=jet7pbic> Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1406 ast_rtp_bridge: Oooh, 'SIP/7708183799-8d6d' changed end address to 192.168.1.101:10094 (format 6) Jun 18 17:46:25 DEBUG[1192491824]: rtp.c:1408 ast_rtp_bridge: Oooh, 'SIP/7708183799-8d6d' was 65.202.115.115:10094/(format 6) Any help would be greatly appreciated. Thanks, Brian

Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ........... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to be me, never do anything right !) Thanks -- Clive Email : clive.carter@sbcs.co.uk Alt : clivecarter@orange.net Tel...

...le of minutes this problem was gone, without me doing anything..Has anyone observed this thing before... Called 810 -- SIP/810-b6dc is ringing -- SIP/810-b6dc answered SIP/910-6c4e -- Attempting native bridge of SIP/910-6c4e and SIP/810-b6dc WARNING[1227879616]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 = 524302 is not codec1 = 524302, can't do reinvite -- Got SIP response 482 "Loop Detected" back from 129.82.44.226 WARNING[1142106560]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries exceeded on call 3c2706bad222-n2s56u19hj1l@129-82-44-226 for seqno 1 (Response...

...ecuting Dial("SIP/andrew-89e3", "SIP/jeremy|20") in new stack -- Called jeremy -- SIP/jeremy-b7a9 is ringing -- SIP/jeremy-b7a9 answered SIP/andrew-89e3 -- Attempting native bridge of SIP/andrew-89e3 and SIP/jeremy-b7a9 Jul 25 16:49:36 WARNING[851980]: rtp.c:1392 ast_rtp_bridge: codec0 = 12 is not codec1 = 256, cannot native bridge. == Spawn extension (default, 801, 1) exited non-zero on 'SIP/andrew-89e3' -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 192.168.200.226 Jul 25 16:49:42 WARNING[114695]: chan_sip.c:2820 process...

...I got such error messages [Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose I got nothing error during installation of asterisk-addons-1.4.2 after I had change the Make file on the chan_ooh323.s...

...79 process_sdp: Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123') Sep 16 08:27:47 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123') Sep 16 08:27:47 WARNING[360465]: rtp.c:1382 ast_rtp_bridge: codec0 = 268 is not codec1 = 0, cannot native bridge. == Spawn extension (sip, 88888888, 1) exited non-zero on 'SIP/105-1559' (123.123.123.123 is the IP of our VoIP-provider, 88888888 is my cell phone, and 105 is the asterisk-connected phone). Regards, Evert

...and SIP/3013-5f1c -- Started music on hold, class 'default', on SIP/3013-5f1c -- Stopped music on hold on Zap/1-1 -- Stopped music on hold on SIP/3013-5f1c -- Attempting native bridge of SIP/3000-1368<ZOMBIE> and SIP/3000-4b9e Feb 22 16:13:39 WARNING[16172]: rtp.c:1365 ast_rtp_bridge: Can't find native functions for channel 'SIP/3000-1368<ZOMBIE>' Feb 22 16:13:39 WARNING[16172]: channel.c:2634 ast_channel_bridge: Private bridge between SIP/3000-1368<ZOMBIE> and SIP/3000-4b9e failed well, duh it failed. how can a channel bridge with its zombified self?...

...nstalled, and two phones - Cisco 7960 and Cisco 7905. If g729 is the only available codec for 7905's configuration, then call from 7960 to 7905 goes without any problem and both phones use g729. But if I call from 7905 to 7960 the following is displayed on * console: WARNING[5220]: rtp.c:1545 ast_rtp_bridge: codec0 = 256 is not codec1 = 4, cannot native bridge. And * does transcoding from g729 to g711. Both phones have reinvite turned on. Why everything works only way and does not work other way? Question #2: What approach should be used to have an * as a MoH server? For example, I want to have...

Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk "just" as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone

...modules, can some one help on those issues? Error during loading the modules; Basically, chan_ooh323.so, and res_config_mysql.so "" [Jun 23 12:10:01] WARNING[30257] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 23 12:10:01] WARNING[30257] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose "" If I manually disable the modules in the modules.conf then my Asterisk 1.4.5 will run. I am using be...

...I got such error messages [Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose I got nothing error during installation of asterisk-addons-1.4.2 after I had change the Make file on the chan_ooh323.s...

been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan?

Hi list, I'm trying to get channel gtalk working in asterisk 1.4.5 I have it built and configured as follows: *jabber.conf:* [general] debug=yes autoprune=no autoregister=no [myaccount] type=client serverhost=talk.google.com username=myaccount at gmail.com/Talk secret=mypassword port=5222 usetls=yes usesasl=yes statusmessage="Talk to me" timeout=100 *gtalk.conf:* [general]