search for: ast_rtp_writ

...s exactly this invite back as incoming call. The answer is g722,g711 in the ok sdp. Now, Asterisk can't decide, which codec to use. It frequently changes the codec just as it likes to apparently without any visible reason. [2017-05-11 17:28:03] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:03] DEBUG[5113][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:04] DEBUG[5123][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-1...

...ted (ie, no ringing). It was kind of confusing because sometimes `SESSION PROGRESS` takes longer than others, during this time we are in MOH. The call does eventually connect and the MOH stops. When debugging I saw the following debug message: [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote address on RTP instance '0xb6e00b20' so dropping frame [Sep 10 10:56:12] DEBUG[7930]: res_rtp_asterisk.c:1228 ast_rtp_write: No remote a...

Hi, try to setup asterisk as an ISDN2H323-Gateway. The only problem i have after establishing a call is, that Audio works only from IP to ISDN-Phone but not from ISDN to IP-Phone. A known problem ??? Thanks in advance Michael i am using asterisk-cvs, pwlib V1.6.6 (janus), openh323 V1.13.5 (janus) and oh323-0.6.0 Here are my config's ############## # modem.conf # ##############

...[Aug 4 17:50:11] -- Executing [s at connect-to-me:2] SayDigits("SIP/gafachi1a-00000000", "'1'") in new stack [Aug 4 17:50:11] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a-00000000 to write format slin [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3904 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 4 17:50:11] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 4 17:50:11]...

...EBUG[1099414448]: chan_sip.c:825 __sip_ack: Stopping retransmission on 'ea418cc2-2c42-49a2-9a8a-b068a8795e32@192.168.1.199' of Response 1: Found Sep 23 11:49:33 DEBUG[1090837424]: chan_iax2.c:5208 socket_read: Ooh, video format changed to 262144 Sep 23 11:49:33 DEBUG[1112759216]: rtp.c:1166 ast_rtp_write: Ooh, format changed from UNKN to H261 Sep 23 11:49:33 DEBUG[1090837424]: chan_iax2.c:5179 socket_read: Ooh, voice format changed to 2 Sep 23 11:49:33 DEBUG[1112759216]: rtp.c:1166 ast_rtp_write: Ooh, format changed from UNKN to GSM Sep 23 11:50:01 DEBUG[1090837424]: chan_iax2.c:5208 socket_read:...

...ne 952 (find_user): Call from user 'xirak' is 1 out of 0 DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: Contact hop : <sip:192.168.7.3> -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format changed from U NKN to ULAW DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling timer at 16 0 sample intervals -- Playing 'vm-login' DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping retransmission on '6E5D898E-492D-400B-A42B-8B25FE25F2E...

...253.1069: P 155:361(206) ack 270 win 6500 <nop,nop,timestamp 154312894 19012> (DF) 16:36:32.681821 192.168.101.253.1069 > lpr-2.pelagiris.org.1720: . ack 155 win 8042 16:36:32.740325 192.168.101.253.1069 > lpr-2.pelagiris.org.1720: . ack 361 win 7990 DEBUG[442389]: File rtp.c, Line 787 (ast_rtp_write): Ooh, format changed from 0 to 2 16:36:33.622651 lpr-2.pelagiris.org.1720 > 192.168.101.253.1069: P 361:546(185) ack 270 win 6500 <nop,nop,timestamp 154312994 19012> (DF) 16:36:33.624943 lpr-2.pelagiris.org.61886 > 192.168.101.253.30006: udp 45 (DF) 16:36:33.651466 lpr-2.pelagiris.org...

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

...8]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> -- Executing Wait("SIP/sip3-768a", "1") in new stack -- Executing AgentLogin("SIP/sip3-768a", "") in new stack Aug 12 16:31:37 DEBUG[1127562160]: rtp.c:1156 ast_rtp_write: Ooh, format changed from UNKN to ULAW Aug 12 16:31:37 DEBUG[1127562160]: channel.c:1101 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'agent-user' (language 'en') Aug 12 16:31:37 DEBUG[1103408048]: chan_sip.c:817 __sip_ack: Stopping retransmission on '...

...144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP (4)? >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP (4)? >Oct 5 11:11:20 DEBUG[25104]: rtp.c:1193 ast_rtp_write: Ooh, format changed from >ulaw to ilbc >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP (4)? >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that is...

...0 o=root 586 586 IN IP4 10.1.1.2 s=session c=IN IP4 10.1.1.2 t=0 0 m=audio 10524 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.1.11:5060 Jul 8 16:47:24 DEBUG[262159]: rtp.c:1123 ast_rtp_write: Ooh, format changed from UNKN to ULAW Jul 8 16:47:24 DEBUG[262159]: chan_sip.c:1976 sip_rtp_read: Oooh, format changed to 2 Jul 8 16:47:24 DEBUG[262159]: rtp.c:1123 ast_rtp_write: Ooh, format changed from ULAW to GSM 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:10.1.1.11 SIP/2.0 Vi...

...e(9) on channel 11 (index 0) [Mar 26 14:36:41] DEBUG[12577]: chan_dahdi.c:1784 dahdi_enable_ec: Enabled echo cancellation on channel 11 [Mar 26 14:36:41] DEBUG[12577]: chan_sip.c:7241 transmit_response_with_sdp: Setting framing from config on incoming call [Mar 26 14:36:41] DEBUG[12577]: rtp.c:2901 ast_rtp_write: Ooh, format changed from unknown to alaw [Mar 26 14:36:41] DEBUG[12577]: rtp.c:2918 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 [Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:5048 __dahdi_exception: Exception on 25, channel 11 [Mar 26 14:36:57] DEBUG[12577]: chan_dahdi.c:4142 dahdi_h...

...systems (or any others I've tested - all outside of my home nat). The calls are being made with ulaw as the only codec allowed. The sip debug indicates that the call setup has worked and agreed upon ulaw as the codec. CLI provides multiple repeats of the following two errors: rtp.c:1215 ast_rtp_write: Not sure about sending format SLINR packets rtp.c:1058 ast_rtp_raw_write: Not sure about timestamp format for codec Any thoughts, comments, suggestions? Google has been less than helpful given any keywords I came up with to avoice all the NAT/Firewall one way audio posts. -Chris

Hi, We observed the packet size used in asterisk is about 20 ms. We would like to know if is possible to change this value to 10 or 30 ms for example. If so, how could I change it? Thanks in advance and best regards __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

...duling timer at 160 sample intervals Urgent handler Jan 25 17:34:04 DEBUG[10020]: channel.c:1379 ast_read: Generator got voice, switching to phase locked mode Jan 25 17:34:04 DEBUG[10020]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 25 17:34:04 DEBUG[10020]: rtp.c:1188 ast_rtp_write: Ooh, format changed from unknown to alaw Radovan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/d7f5bcb1/attachment.htm

Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase

...568543197@192.168.1.103:5060> -- Executing SetVar("SIP/6568543197-86c2", "sip_codec=g729") in new stack -- Executing Dial("SIP/6568543197-86c2", "Zap/g1/91596323") in new stack -- Called g1/91596323 Oct 4 00:53:41 DEBUG[1146877376]: rtp.c:1162 ast_rtp_write: Ooh, format changed from UNKN to G729A Oct 4 00:53:41 DEBUG[1146877376]: rtp.c:438 ast_rtp_read: RTP NAT: Using address 68.2.178.157:16410 Oct 4 00:53:46 DEBUG[1089555136]: chan_zap.c:1179 zt_enable_ec: Enabled echo cancellation on channel 1 -- Zap/1-1 is ringing Oct 4 00:53:46 WARNING[114...

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

...-- externalIpAddress: 192.168.1.201 -- externalPort: 15508 -- SessionID: 1 -- Direction: IsReceiver -- Started logical channel: receiving G.711-ALaw-64k{sw} -- channelsOpen = 2 Aug 30 11:53:36 DEBUG[114731952]: rtp.c:1156 ast_rtp_write: Ooh, format changed from UNKN to ALAW Any idea about this "H245 Read error: Interrupted system call" that appears in the debug messages ??? Thanks, Damien. BTW, the H.323 channel has been compiled with the recommended PWLib 1.5.2 and OpenH323 1.12.2. -------------- next part...

...12 22:47:09] DEBUG[22725]: channel.c:2548 ast_read_generator_actions: Generator got voice, switching to phase locked mode [May 12 22:47:09] DEBUG[22725]: channel.c:2434 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [May 12 22:47:09] DEBUG[22725]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to alaw [May 12 22:47:09] DEBUG[22725]: rtp.c:3904 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 [May 12 22:47:13] DEBUG[22725]: rtp.c:1240 ast_rtcp_read: Got RTCP report of 88 bytes [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Proc...