rendered paste body<--- Transmitting (no NAT) to 192.168.2.237:8160 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;received=192.168.2.237;rport=8160
From: "Phillip 45"<sip:46@192.168.2.6>;tag=394d9618
To: <sip:2299614@192.168.2.6>
Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2299614@192.168.2.6>
Content-Length: 0
<------------>
-- Executing [2299614@numberplan-custom-1:1] Macro("SIP/46-09022000", "trunkdial|Zap/g1/2299614") in new stack
-- Executing [s@macro-trunkdial:1] Dial("SIP/46-09022000", "Zap/g1/2299614") in new stack
[May 30 18:57:22] WARNING[34627]: channel.c:3084 ast_request: No translator path exists for channel type Zap (native 76) to 256
[May 30 18:57:22] WARNING[34627]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-trunkdial:2] Goto("SIP/46-09022000", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-trunkdial,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-trunkdial:1] NoOp("SIP/46-09022000", "No pude relaizar la llamada: CHANUNAVAIL| 0") in new stack
== Auto fallthrough, channel 'SIP/46-09022000' status is 'CHANUNAVAIL'
<--- Transmitting (no NAT) to 192.168.2.237:8160 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;received=192.168.2.237;rport=8160
From: "Phillip 45"<sip:46@192.168.2.6>;tag=394d9618
To: <sip:2299614@192.168.2.6>;tag=as024ce994
Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2299614@192.168.2.6>
Content-Length: 0
<------------>
pbx*CLI>
<--- SIP read from 192.168.2.237:8160 --->
INVITE sip:2299614@192.168.2.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:46@192.168.2.237:8160>
To: <sip:2299614@192.168.2.6>
From: "Phillip 45"<sip:46@192.168.2.6>;tag=394d9618
Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="46",realm="asterisk",nonce="29d290c4",uri="sip:2299614@192.168.2.6",response="d0944b2b218082dd5b5f37cc4c0142d9",algorithm=MD5
Supported: eventlist
User-Agent: eyeBeam release 3010n stamp 19039
Content-Length: 535
v=0
o=- 26720725 26720742 IN IP4 192.168.2.237
s=eyeBeam
c=IN IP4 192.168.2.237
t=0 0
m=audio 8144 RTP/AVP 18 3 0 100 6 8 99 98 97 5 102 101
a=alt:1 1 : E681ABC9 0000007D 192.168.2.237 8144
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:99 evrc/8000
a=rtpmap:98 ilbc/8000
a=rtpmap:97 speex/8000
a=rtpmap:102 l16/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
m=video 41934 RTP/AVP 105 34
a=alt:1 1 : 98EEFBCB 00000002 192.168.2.237 41934
a=fmtp:105 profile=3;level=20
a=rtpmap:105 h263-2000/90000
a=sendrecv
<------------->
--- (14 headers 20 lines) ---
Ignoring this INVITE request
[May 30 18:57:22] NOTICE[34627]: chan_sip.c:13671 handle_request_invite: Unable to create/find SIP channel for this INVITE
<--- Transmitting (no NAT) to 192.168.2.237:8160 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;received=192.168.2.237;rport=8160
From: "Phillip 45"<sip:46@192.168.2.6>;tag=394d9618
To: <sip:2299614@192.168.2.6>;tag=as024ce994
Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2299614@192.168.2.6>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'a9646d79260ae028@VG9zaGliYVBuZXVtYW5u' in 32000 ms (Method: INVITE)
Really destroying SIP dialog 'a9646d79260ae028@VG9zaGliYVBuZXVtYW5u' Method: INVITE
pbx*CLI>
<--- SIP read from 192.168.2.237:8160 --->
ACK sip:2299614@192.168.2.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;rport
To: <sip:2299614@192.168.2.6>;tag=as024ce994
From: "Phillip 45"<sip:46@192.168.2.6>;tag=394d9618
Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 192.168.2.237:8160 --->
ACK sip:2299614@192.168.2.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;rport
To: <sip:2299614@192.168.2.6>;tag=as024ce994
From: "Phillip 45"<sip:46@192.168.2.6>;tag=394d9618
Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 192.168.2.237:8160 --->