for provider 1


INVITE sip:8615883555343@phone.plivo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 103.21.168.42:5060;branch=z9hG4bK48e585ca;rport
Max-Forwards: 70
From: "8613161226065" <sip:VoipCallAmeyo170725134327@phone.plivo.com>;tag=as6a1ce0eb
To: <sip:8615883555343@phone.plivo.com:5060>
Contact: <sip:VoipCallAmeyo170725134327@103.21.168.42>
Call-ID: 40eac1e45f2869876d8189bf04c177f7@phone.plivo.com
CSeq: 102 INVITE ////////////////////////--------------------------------------//////////////////////
User-Agent: Asterisk PBX 1.6.2
Date: Wed, 17 Jan 2018 11:51:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 180079596 180079596 IN IP4 103.21.168.42
s=Asterisk PBX 1.6.2
c=IN IP4 103.21.168.42
t=0 0
m=audio 28496 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000

INVITE sip:8615883555343@phone.plivo.com:5060 SIP/2.0
Via: SIP/2.0/UDP 103.21.168.42:5060;branch=z9hG4bK48e585ca;rport
Max-Forwards: 70
From: "8613161226065" <sip:VoipCallAmeyo170725134327@phone.plivo.com>;tag=as6a1ce0eb
To: <sip:8615883555343@phone.plivo.com:5060>
Contact: <sip:VoipCallAmeyo170725134327@103.21.168.42>
Call-ID: 40eac1e45f2869876d8189bf04c177f7@phone.plivo.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2
Date: Wed, 17 Jan 2018 11:51:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 180079596 180079596 IN IP4 103.21.168.42
s=Asterisk PBX 1.6.2
c=IN IP4 103.21.168.42
t=0 0
m=audio 28496 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv



for 2nd provider


<------------->
--- (13 headers 0 lines) ---
^MESC[KAmeyoallinone*CLI> ^MESC[0KScheduling destruction of SIP dialog '67c841217aebcedc1c4756046ce042df@127.0.0.1' in 32000 ms (Method: REGISTER)
[Jan 17 20:18:25] ESC[1;33mNOTICEESC[0m[7552]: ESC[1;37mchan_sip.cESC[0m:ESC[1;37m18705ESC[0m ESC[1;37mhandle_response_registerESC[0m: Outbound Registration: Expiry for
 207.198.106.236 is 120 sec (Scheduling reregistration in 105 s)
^MESC[KAmeyoallinone*CLI> ^MESC[0KReally destroying SIP dialog '3ec8aaa5-d4c14ec-78d8751@10.2.5.10' Method: OPTIONS
^MESC[KAmeyoallinone*CLI> ^MESC[0KRetransmitting #6 (NAT) to 207.198.106.236:5060:
INVITE sip:639176623471@207.198.106.236:5060 SIP/2.0
Via: SIP/2.0/UDP 103.21.168.42:5060;branch=z9hG4bK350ebfab;rport
Max-Forwards: 70
From: "8613161226065" <sip:8613161226065@207.198.106.236>;tag=as58d1965c
To: <sip:639176623471@207.198.106.236:5060>
Contact: <sip:8613161226065@103.21.168.42>
Call-ID: 7a6af54c1a9371662c680955577feb7c@207.198.106.236
CSeq: 102 INVITE                  /////////---------------------------------////////
User-Agent: Asterisk PBX 1.6.2
Date: Wed, 17 Jan 2018 12:18:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2002897467 2002897467 IN IP4 103.21.168.42
s=Asterisk PBX 1.6.2
c=IN IP4 103.21.168.42
t=0 0
m=audio 24818 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Scheduling destruction of SIP dialog '7a6af54c1a9371662c680955577feb7c@207.198.106.236' in 32000 ms (Method: INVITE)
    -- SIP/VERT_NEW-00000015 is circuit-busy
Scheduling destruction of SIP dialog '7a6af54c1a9371662c680955577feb7c@207.198.106.236' in 32000 ms (Method: INVITE)
^MESC[KAmeyoallinone*CLI> ^MESC[0K  == Everyone is busy/congested at this time (1:0/1/0)
^MESC[KAmeyoallinone*CLI> ^MESC[0K    -- Executing [dial@from-manager-core:2] ESC[1;36mGotoESC[0m("ESC[1;35mSIP/webrtc_CRM3-00000014ESC[0m", "ESC[1;35mcalllegwait,1ESC[
0m") in new stack
    -- Goto (from-manager-core,calllegwait,1)
    -- Executing [calllegwait@from-manager-core:1] ESC[1;36mWaitESC[0m("ESC[1;35mSIP/webrtc_CRM3-00000014ESC[0m", "ESC[1;35m3600ESC[0m") in new stack
^MESC[KAmeyoallinone*CLI> ^MESC[0K  == Spawn extension (from-manager-core, calllegwait, 1) exited non-zero on 'SIP/webrtc_CRM3-00000014'
    -- Executing [h@from-manager-core:1] ESC[1;36mHangupESC[0m("ESC[1;35mSIP/webrtc_CRM3-00000014ESC[0m", "ESC[1;35mESC[0m") in new stack
  == Spawn extension (from-manager-core, h, 1) exited non-zero on 'SIP/webrtc_CRM3-00000014'
Scheduling destruction of SIP dialog '0980315e3a124e6f5397247b33208a48@10.10.12.41' in 32000 ms (Method: INVITE)
^MESC[KAmeyoallinone*CLI> ^MESC[0Kset_destination: Parsing <sip:webrtc_CRM3@10.10.12.41:8800> for address/port to send to
set_destination: set destination to 10.10.12.41, port 8800
^MESC[KAmeyoallinone*CLI> ^MESC[0KReliably Transmitting (no NAT) to 10.10.12.41:8800:
BYE sip:webrtc_CRM3@10.10.12.41:8800 SIP/2.0
Via: SIP/2.0/UDP 10.10.12.41:5060;branch=z9hG4bK6ced47c4;rport
Max-Forwards: 70
From: "639176623471" <sip:639176623471@10.10.12.41>;tag=as1c0339eb
To: <sip:webrtc_CRM3@10.10.12.41:8800>;tag=as2d2570e2
Call-ID: 0980315e3a124e6f5397247b33208a48@10.10.12.41
CSeq: 103 BYE                               ////////////--------------------------------//////////
User-Agent: Asterisk PBX 1.6.2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0