search for: sip_rtp_read

Hi, I try to set up planet VIP-050 with asterisk. Everything works fine instead of the call transfer. When I press # console says something like this: >Oct 5 11:11:20 DEBUG[25104]: chan_sip.c:2222 sip_rtp_read: Oooh, format changed >to 1024 >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 isn't a...

...r att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help would be appreciate. This is debug from console. Feb 14 08:27:08 DEBUG[13349]: chan_sip.c:2969 sip_rtp_read: * Detected inband DTMF '*' Feb 14 08:27:08 DEBUG[13349]: channel.c:3253 ast_generic_bridge: Didn't get a frame from channel: Agent/401 Feb 14 08:27:08 DEBUG[13349]: channel.c:3525 ast_channel_bridge: Bridge stops bridging channels SIP/211-5396 and Agent/401 Feb 14 08:27:08 DEBUG[13349]...

...- timelimit = 0 -- - play_warning = 0 -- - warning_sound = (null) Nov 20 20:51:51 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '000f2300-08d000f6-4f620267-55399868 at 192.168.1.161' of Response 102: Match Found Nov 20 20:51:51 DEBUG[6042]: chan_sip.c:3051 sip_rtp_read: Oooh, format changed to 256 The Call doesn't go through --- Out put of `lspci` . . 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface . . . --- Output of `lsmod` Module Size Used by wctdm 37184 4 . . . ----- Output of /p...

I'm trying to use early-dial. Here, all hardware PBX have it. You dial numbers and, as soon as you have a matching dialplan entry, you get throught. I my Grandstream I enabled early-dial. And when I put exten=91,1,Milliwatt in my dialplan then it works as expected. Also, when I call other SIP or IAX phones it works. Hurray! But how can I get it working with external lines? In my

...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 Via: SIP/2.0/UDP 10.1.1.2:0;branch=z9hG4bK6eac0d88 From: "asterisk" <sip:asterisk@10.1.1.2...