Gagal Telpon keluar dengan asterisk, provider indihome


(Charlie Brinsley) #1

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(Andreas Sugiharto) #2

gan,
minta tolong di share :

  1. log verbose dari asterisk CLI nya
  2. dari asterisk CLI: --> ‘sip show registry’ dan ‘sip show peers’

(Charlie Brinsley) #3

log ketika mencoba panggilan keluar

Audio is at 16696
Adding codec ulaw to SDP
Adding codec codec2 to SDP
Adding codec g723 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding codec silk to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.10:5060:
INVITE sip:+6282387786234@10.0.0.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK0d297408
Max-Forwards: 70
From: sip:+6277840XXXXX@10.0.0.10;tag=as6e841e90
To: sip:+6282387786234@10.0.0.10;user=phone
Contact: sip:+6277840XXXXX@192.168.1.100:5060
Call-ID: 02f7bcf67b28ac1e20a9f04a6d011758@10.0.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.1.3
Date: Thu, 22 Mar 2018 02:18:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1094

v=0
o=root 1536103603 1536103603 IN IP4 192.168.1.100
s=Asterisk PBX 15.1.3
c=IN IP4 192.168.1.100
t=0 0
m=audio 16696 RTP/AVP 0 4 8 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 84 119 97 9 102 115 116 120 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:122 L16/12000
a=rtpmap:118 L16/16000
a=rtpmap:123 L16/24000
a=rtpmap:124 L16/32000
a=rtpmap:125 L16/44000
a=rtpmap:126 L16/48000
a=rtpmap:127 L16/96000
a=rtpmap:96 L16/192000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:84 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:120 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv


<— SIP read from UDP:10.0.0.10:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK0d297408;received=10.16.75.16
Call-ID: 02f7bcf67b28ac1e20a9f04a6d011758@10.0.0.10
From: sip:+6277840XXXXX@10.0.0.10;tag=as6e841e90
To: sip:+6282387786234@10.0.0.10;user=phone
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘asbcss7gree78eg9e47ff47q2g248fge82ia@19500.0.ATS.sb1m-ats01.telkom.net.id.27’ Method: NOTIFY
Really destroying SIP dialog ‘asbcda4f2si7bb7dd7a2b8as7qbqr8a84bs8@19500.0.ATS.sb1m-ats01.telkom.net.id.27’ Method: NOTIFY
Reliably Transmitting (no NAT) to 10.0.0.10:5060:
OPTIONS sip:10.0.0.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK4cd4ce60
Max-Forwards: 70
From: “asterisk” sip:+6277840XXXXX@192.168.1.100;tag=as2e15ade2
To: sip:10.0.0.10;user=phone
Contact: sip:+6277840XXXXX@192.168.1.100:5060
Call-ID: 4161a4e13f9a53eb205005ce671f5218@192.168.1.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.1.3
Date: Thu, 22 Mar 2018 02:19:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.0.0.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK4cd4ce60
Call-ID: 4161a4e13f9a53eb205005ce671f5218@192.168.1.100:5060
From: "asterisk"sip:+6277840XXXXX@192.168.1.100:5060;tag=as2e15ade2
To: sip:10.0.0.10;user=phone;tag=sbc0805myzhlfdh
CSeq: 102 OPTIONS
Allow: OPTIONS,NOTIFY,SUBSCRIBE,INFO,REGISTER,MESSAGE,REFER,UPDATE,PRACK,BYE,CANCEL,ACK,INVITE
Supported: privacy,precondition,100rel
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘4161a4e13f9a53eb205005ce671f5218@192.168.1.100:5060’ Method: OPTIONS
Scheduling destruction of SIP dialog ‘02f7bcf67b28ac1e20a9f04a6d011758@10.0.0.10’ in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.0.0.10:5060:
CANCEL sip:+6282387786234@10.0.0.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK0d297408
Max-Forwards: 70
From: sip:+6277840XXXXX@10.0.0.10;tag=as6e841e90
To: sip:+6282387786234@10.0.0.10;user=phone
Call-ID: 02f7bcf67b28ac1e20a9f04a6d011758@10.0.0.10
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 15.1.3
Content-Length: 0


Scheduling destruction of SIP dialog ‘02f7bcf67b28ac1e20a9f04a6d011758@10.0.0.10’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.0.0.10:5060 —>
SIP/2.0 481 Call leg or Transaction not exist
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK0d297408
Call-ID: 02f7bcf67b28ac1e20a9f04a6d011758@10.0.0.10
From: sip:+6277840XXXXX@10.0.0.10;tag=as6e841e90
To: sip:+6282387786234@10.0.0.10;user=phone;tag=sbc0805tr0x1yxp
CSeq: 102 CANCEL
Content-Length: 0

<------------->
— (7 headers 0 lines) —
[Mar 22 09:19:45] WARNING[2015][C-00000001]: chan_sip.c:25051 handle_response: Remote host can’t match request CANCEL to call ‘02f7bcf67b28ac1e20a9f04a6d011758@10.0.0.10’. Giving up.