search for: ast_rtp_raw_write

...format slin [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 160 sample intervals [Jan 12 23:13:55] DEBUG[8717]: channel.c:5297 ast_channel_start_silence_generator: Started silence generator on 'SIP/xtreme-00000639' [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2796 ast_rtp_raw_write: Difference is 1736, ms is 237 [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 12 23:13:55] DEBUG[8717]: channel.c:5310 ast_channel_stop_silence_generator: Stopped silence generator on 'SIP/xtreme-00000639' [Jan 12 23:13:55] DEBUG[87...

...point (note: NoOp required for GotoIf at priority 1) exten => s,6,NoOp Output when the originating side ends the call first (unexpectedly stops at priority 1 in macro): (normal call setup and progress not shown - I show everything after hangup) *CLI> DEBUG[30737]: File rtp.c, Line 739 (ast_rtp_raw_write): Difference is 10888, ms is 1381 DEBUG[30737]: File channel.c, Line 2015 (ast_channel_bridge): Didn't get a frame from channel: SIP/2203-751b DEBUG[30737]: File channel.c, Line 2083 (ast_channel_bridge): Bridge stops bridging channels SIP/2203-751b and SIP/2205-12f1 == Spawn extension (inter...

I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the

.... 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

...exten => 112,n,Playback(demo-congrats) exten => 112,n,Hangup I see this executing on the CLI. However I have no audio. Enabling RTP debug I see the Got RTP packet but there are no send RTP packets going out. I edited the source and put logging messages first in main/rtp.c and I saw the ast_rtp_raw_write() getting called 1 time. so I backed up the tree. Got into channels/chan_sip.c sip_write() and it only gets called 1 time. I have had a couple of times where I heard audio. Hangup up and tried again. And NO audio for bunch more times... What can be causing my RTP issue and no audio? Jerry

...om the console - they ring, and I can answer.. --Executing Dial("OSS/dsp", "SIP/2001|15|t") in new stack --called 2001 --SIP/2001-c7b1 is ringing --SIP/2001-c7b1 answered OSS/dsp <<Console call has been answered>> Dec 17 12:26:26 NOTICE[7078] : rtp.c:1193 ast_rtp_raw_write: RTP Transmission error to <IPADDR>:23658: Network in unreachable (plus another 12 messages the same) ==Spawn extension (local, 2001, 1) exited non-zero on 'OSS/dsp' <<Hangup on console>> Anyone got any ideas? Since it's my first setup, it's probably somethi...

...e_interpret_helper: Feature detected: fname=Blind Transfer sname=blindxfer exten=#8 -- Music class default requested but no musiconhold loaded. [Oct 2 11:09:21] DEBUG[29533]: channel.c:3616 set_format: Set channel SIP/1000-0a292360 to write format gsm [Oct 2 11:09:21] DEBUG[29533]: rtp.c:3099 ast_rtp_raw_write: Difference is 1952, ms is 264 [Oct 2 11:09:21] DEBUG[29533]: channel.c:2362 ast_settimeout: Scheduling timer at 160 sample intervals -- <SIP/1000-0a292360> Playing 'pbx-transfer.gsm' (language 'en') [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Ev...

...e_interpret_helper: Feature detected: fname=Blind Transfer sname=blindxfer exten=#8 -- Music class default requested but no musiconhold loaded. [Oct 2 11:09:21] DEBUG[29533]: channel.c:3616 set_format: Set channel SIP/1000-0a292360 to write format gsm [Oct 2 11:09:21] DEBUG[29533]: rtp.c:3099 ast_rtp_raw_write: Difference is 1952, ms is 264 [Oct 2 11:09:21] DEBUG[29533]: channel.c:2362 ast_settimeout: Scheduling timer at 160 sample intervals -- <SIP/1000-0a292360> Playing 'pbx-transfer.gsm' (language 'en') [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Ev...

...e_interpret_helper: Feature detected: fname=Blind Transfer sname=blindxfer exten=#8 -- Music class default requested but no musiconhold loaded. [Oct 2 11:09:21] DEBUG[29533]: channel.c:3616 set_format: Set channel SIP/1000-0a292360 to write format gsm [Oct 2 11:09:21] DEBUG[29533]: rtp.c:3099 ast_rtp_raw_write: Difference is 1952, ms is 264 [Oct 2 11:09:21] DEBUG[29533]: channel.c:2362 ast_settimeout: Scheduling timer at 160 sample intervals -- <SIP/1000-0a292360> Playing 'pbx-transfer.gsm' (language 'en') [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Ev...

...e_interpret_helper: Feature detected: fname=Blind Transfer sname=blindxfer exten=#8 -- Music class default requested but no musiconhold loaded. [Oct 2 11:09:21] DEBUG[29533]: channel.c:3616 set_format: Set channel SIP/1000-0a292360 to write format gsm [Oct 2 11:09:21] DEBUG[29533]: rtp.c:3099 ast_rtp_raw_write: Difference is 1952, ms is 264 [Oct 2 11:09:21] DEBUG[29533]: channel.c:2362 ast_settimeout: Scheduling timer at 160 sample intervals -- <SIP/1000-0a292360> Playing 'pbx-transfer.gsm' (language 'en') [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Ev...