asterisk*CLI> core set verbose 99
Verbosity was 25 and is now 99
Here we go :
Connected to Asterisk 1.4.25.1 currently running on asterisk (pid = 12780)
Verbosity is at least 99
Core debug is at least 25
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK52feb2fe04930b5a
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50555 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581
v=0
o=grandstream 8000 8000 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ckes5wzRCfuIiZAGQM/8qCaQ3P8Xip0p8I96zlXD
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:sNVe0OMtQAp9kBzjgm3OO7aXtbWUlZwSTNsIwD6c
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
— (13 headers 21 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)
Using INVITE request as basis request - 49c8921ff0eebfbb@192.168.1.13
<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK52feb2fe04930b5a;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248;tag=as0d098212
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50555 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk-jocan”, nonce=“74a7ea42”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘49c8921ff0eebfbb@192.168.1.13’ in 32000 ms (Method: INVITE)
Found user ‘grandstream’
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
ACK sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK52feb2fe04930b5a
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248;tag=as0d098212
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: path
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50555 ACK
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
— (12 headers 0 lines) —
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
INVITE sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKb918a0ee9bde174a
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: replaces, timer, path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:20@192.168.1.248", nonce=“74a7ea42”, response=“73c382dfef046527bb6b26b888d3f1ca”
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50556 INVITE
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 581
v=0
o=grandstream 8000 8001 IN IP4 192.168.1.13
s=SIP Call
c=IN IP4 192.168.1.13
t=0 0
m=audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ckes5wzRCfuIiZAGQM/8qCaQ3P8Xip0p8I96zlXD
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:sNVe0OMtQAp9kBzjgm3OO7aXtbWUlZwSTNsIwD6c
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:9 G722/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
<------------->
— (14 headers 21 lines) —
Sending to 192.168.1.13 : 5060 (no NAT)
Using INVITE request as basis request - 49c8921ff0eebfbb@192.168.1.13
Found user ‘grandstream’
[Jun 23 16:58:42] WARNING[17504]: chan_sip.c:5270 process_sdp: Unsupported SDP media type in offer: audio 5004 RTP/SAVP 8 0 3 18 2 4 9 97 101
<— Reliably Transmitting (no NAT) to 192.168.1.13:5060 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKb918a0ee9bde174a;received=192.168.1.13
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248;tag=as0d098212
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50556 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘49c8921ff0eebfbb@192.168.1.13’ in 32000 ms (Method: INVITE)
asterisk*CLI>
<— SIP read from 192.168.1.13:5060 —>
ACK sip:20@192.168.1.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bKb918a0ee9bde174a
From: sip:grandstream@192.168.1.248;tag=71360aaf46d50083
To: sip:20@192.168.1.248;tag=as0d098212
Contact: sip:grandstream@192.168.1.13:5060;transport=udp
Supported: path
Proxy-Authorization: Digest username=“grandstream”, realm=“asterisk-jocan”, algorithm=MD5, uri="sip:20@192.168.1.248", nonce=“74a7ea42”, response=“73c382dfef046527bb6b26b888d3f1ca”
Call-ID: 49c8921ff0eebfbb@192.168.1.13
CSeq: 50556 ACK
User-Agent: Grandstream GXP2010 1.1.6.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
— (13 headers 0 lines) —
asterisk*CLI> exit