<------------>
    -- Executing [260326@outcoming:1] Dial("SIP/100-00000000", "SIP/multifon-out/74212260326,30,r") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 15224
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:74212260326@sbc.megafon.ru:5060 SIP/2.0
Via: SIP/2.0/TCP 79.122.224.42:5060;branch=z9hG4bK531bff27;rport
Max-Forwards: 70
From: "100" <sip:79294190050@multifon.ru>;tag=as77135e47
To: <sip:74212260326@sbc.megafon.ru:5060>
Contact: <sip:79294190050@79.122.224.42:5060;transport=TCP>
Call-ID: 081f3619586ce78301fc1dad36b251f5@multifon.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Date: Thu, 06 Apr 2017 07:25:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 156132901 156132901 IN IP4 79.122.224.42
s=Asterisk PBX 11.13.1~dfsg-2+deb8u2
c=IN IP4 79.122.224.42
t=0 0
m=audio 15224 RTP/AVP 18 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
    -- Called SIP/multifon-out/74212260326

<--- Transmitting (NAT) to 192.168.1.36:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.36:5060;branch=z9hG4bK2465919378470525507;received=192.168.1.36;rport=5060
From: 100 <sip:100@192.168.1.33:5090>;tag=865330312
To: "260326" <sip:260326@192.168.1.33;user=phone>;tag=as3088a42c
Call-ID: 297262468328011-26258271891572@192.168.1.36
CSeq: 2 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:260326@192.168.1.33:5090>
Content-Length: 0


<------------>

<--- SIP read from TCP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 79.122.224.42:5060;received=79.122.224.42;branch=z9hG4bK531bff27;rport=37594
From: "100" <sip:79294190050@multifon.ru>;tag=as77135e47
To: <sip:74212260326@multifon.ru:5060>
Call-ID: 081f3619586ce78301fc1dad36b251f5@multifon.ru
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from TCP:193.201.229.35:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP 79.122.224.42:5060;received=79.122.224.42;branch=z9hG4bK531bff27;rport=37594
From: "100" <sip:79294190050@multifon.ru>;tag=as77135e47
To: <sip:74212260326@multifon.ru:5060>;tag=aprqngfrt-gu2t7u20000c6
Call-ID: 081f3619586ce78301fc1dad36b251f5@multifon.ru
CSeq: 102 INVITE
Reason: Q.850;cause=55;text="Call Terminated"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:74212260326@sbc.megafon.ru:5060 SIP/2.0
Via: SIP/2.0/TCP 79.122.224.42:5060;branch=z9hG4bK531bff27;rport
Max-Forwards: 70
From: "100" <sip:79294190050@multifon.ru>;tag=as77135e47
To: <sip:74212260326@sbc.megafon.ru:5060>;tag=aprqngfrt-gu2t7u20000c6
Contact: <sip:79294190050@79.122.224.42:5060;transport=TCP>
Call-ID: 081f3619586ce78301fc1dad36b251f5@multifon.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Content-Length: 0


---
[Apr  6 17:25:27] WARNING[9694][C-00000000]: chan_sip.c:23045 handle_response_invite: Received response: "Forbidden" from '"100" <sip:79294190050@multifon.ru>;tag=as77135e47'
Scheduling destruction of SIP dialog '081f3619586ce78301fc1dad36b251f5@multifon.ru' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/100-00000000' status is 'CHANUNAVAIL'

<--- Reliably Transmitting (NAT) to 192.168.1.36:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.36:5060;branch=z9hG4bK2465919378470525507;received=192.168.1.36;rport=5060
From: 100 <sip:100@192.168.1.33:5090>;tag=865330312
To: "260326" <sip:260326@192.168.1.33;user=phone>;tag=as3088a42c
Call-ID: 297262468328011-26258271891572@192.168.1.36
CSeq: 2 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0