************************* 200287 Playback no answer - OK *************************
[Nov 9 17:35:25] DEBUG[12056]: pbx.c:4006 pbx_extension_helper: Launching 'Playback'
[Nov 9 17:35:25] -- Executing [_ZXXX@from-cucm:39] Playback("SIP/cucm-091819f0", "vlastni/nedostupny,noanswer") in new stack
[Nov 9 17:35:25] DEBUG[12056]: channel.c:3885 set_format: Set channel SIP/cucm-091819f0 to write format slin
[Nov 9 17:35:25] DEBUG[12056]: chan_sip.c:10017 transmit_response_with_sdp: Setting framing from config on incoming call
[Nov 9 17:35:25] DEBUG[12056]: chan_sip.c:9687 add_sdp: ** Our capability: 0x100 (g729) Video flag: True Text flag: True
[Nov 9 17:35:25] DEBUG[12056]: chan_sip.c:9688 add_sdp: ** Our prefcodec: 0x0 (nothing)
[Nov 9 17:35:25] Audio is at 78.31.26.174 port 17036
[Nov 9 17:35:25] Adding codec 0x100 (g729) to SDP
[Nov 9 17:35:25] DEBUG[12056]: rtp_engine.c:609 ast_rtp_codecs_payload_code: Found code 256 at payload 18 on 0x91c1d18
[Nov 9 17:35:25] Adding non-codec 0x1 (telephone-event) to SDP
[Nov 9 17:35:25] DEBUG[12056]: rtp_engine.c:609 ast_rtp_codecs_payload_code: Found code 1 at payload 101 on 0x91c1d18
[Nov 9 17:35:25] DEBUG[12056]: chan_sip.c:9828 add_sdp: -- Done with adding codecs to SDP
[Nov 9 17:35:25] DEBUG[12056]: channel.c:3295 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=1048576 chan->timingfd=-1)
[Nov 9 17:35:25] DEBUG[12056]: chan_sip.c:9950 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729)
[Nov 9 17:35:25]
<--- Transmitting (no NAT) to 10.0.156.33:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.156.33:5060;branch=z9hG4bKfed5713be87;received=10.0.156.33
From: "Michal Gust" <sip:1322@10.0.156.33>;tag=0635268a-0640-41a8-aac3-9f9685a0ff80-44433214
To: <sip:1322@78.31.26.174>;tag=as742c1524
Call-ID: deaa0680-af8144ca-f79-219c000a@10.0.156.33
CSeq: 102 INVITE
Server: SIP PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:1322@78.31.26.174>
Content-Type: application/sdp
Content-Length: 274

v=0
o=ipbx 1774319921 1774319921 IN IP4 78.31.26.174
s=SIP Call
c=IN IP4 78.31.26.174
t=0 0
m=audio 17036 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Nov 9 17:35:25] DEBUG[12056]: chan_sip.c:3417 __sip_xmit: Trying to put 'SIP/2.0 18' onto UDP socket destined for 10.0.156.33:5060
[Nov 9 17:35:25] -- <SIP/cucm-091819f0> Playing 'vlastni/nedostupny.slin' (language 'cz')
[Nov 9 17:35:25] DEBUG[12056]: rtp_engine.c:609 ast_rtp_codecs_payload_code: Found code 256 at payload 18 on 0x91c1d18
[Nov 9 17:35:25] DEBUG[12056]: res_rtp_asterisk.c:1130 ast_rtp_write: Ooh, format changed from unknown to g729
[Nov 9 17:35:25] DEBUG[12056]: res_rtp_asterisk.c:1159 ast_rtp_write: Created smoother: format: 256 ms: 20 len: 20
[Nov 9 17:35:25] DEBUG[12056]: res_rtp_asterisk.c:1032 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x91c1cd8'
[Nov 9 17:35:25] DEBUG[12056]: res_rtp_asterisk.c:1551 ast_rtcp_read: Got RTCP report of 44 bytes
[Nov 9 17:35:25] DEBUG[12056]: rtp_engine.c:609 ast_rtp_codecs_payload_code: Found code 256 at payload 18 on 0x91c1d18
...