terpsichore*CLI> sip set debug ip ph0002
SIP Debugging Enabled for IP: 172.16.1.202
terpsichore*CLI> 
<--- SIP read from 172.16.1.202:5062 --->
INVITE sip:99@sip.serviceplanet.nl SIP/2.0
Via: SIP/2.0/UDP 172.16.1.202:5062;branch=z9hG4bKb231f9980abd2373
From: <sip:sipsma@sip.serviceplanet.nl>;tag=a953cb4e625ee201
To: <sip:99@sip.serviceplanet.nl>
Contact: <sip:sipsma@172.16.1.202:5062;transport=udp>
Supported: replaces, timer, path
Call-ID: dd64a664a296b591@172.16.1.202
CSeq: 14056 INVITE
User-Agent: Grandstream GXP2020 1.1.5.15
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 407

v=0
o=sipsma 8000 8000 IN IP4 172.16.1.202
s=SIP Call
c=IN IP4 172.16.1.202
t=0 0
m=audio 5004 RTP/AVP 8 0 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
--- (13 headers 19 lines) ---
Sending to 172.16.1.202 : 5062 (no NAT)
Using INVITE request as basis request - dd64a664a296b591@172.16.1.202

<--- Reliably Transmitting (no NAT) to 172.16.1.202:5062 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.16.1.202:5062;branch=z9hG4bKb231f9980abd2373;received=172.16.1.202
From: <sip:sipsma@sip.serviceplanet.nl>;tag=a953cb4e625ee201
To: <sip:99@sip.serviceplanet.nl>;tag=as75f3bfcc
Call-ID: dd64a664a296b591@172.16.1.202
CSeq: 14056 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="<<Removed to protect the guilty>>"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'dd64a664a296b591@172.16.1.202' in 32000 ms (Method: INVITE)
Found user 'sipsma'
terpsichore*CLI> 
<--- SIP read from 172.16.1.202:5062 --->
ACK sip:99@sip.serviceplanet.nl SIP/2.0
Via: SIP/2.0/UDP 172.16.1.202:5062;branch=z9hG4bKb231f9980abd2373
From: <sip:sipsma@sip.serviceplanet.nl>;tag=a953cb4e625ee201
To: <sip:99@sip.serviceplanet.nl>;tag=as75f3bfcc
Contact: <sip:sipsma@172.16.1.202:5062;transport=udp>
Supported: path
Call-ID: dd64a664a296b591@172.16.1.202
CSeq: 14056 ACK
User-Agent: Grandstream GXP2020 1.1.5.15
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
terpsichore*CLI> 
<--- SIP read from 172.16.1.202:5062 --->
INVITE sip:99@sip.serviceplanet.nl SIP/2.0
Via: SIP/2.0/UDP 172.16.1.202:5062;branch=z9hG4bK368163d1c9e8aad3
From: <sip:sipsma@sip.serviceplanet.nl>;tag=a953cb4e625ee201
To: <sip:99@sip.serviceplanet.nl>
Contact: <sip:sipsma@172.16.1.202:5062;transport=udp>
Supported: replaces, timer, path
Proxy-Authorization: Digest username="sipsma", realm="asterisk", algorithm=MD5, uri="sip:99@sip.serviceplanet.nl", nonce="<<Removed to protect the guilty>>", response="c66b10f667d71442f44b4874f1758b4a"
Call-ID: dd64a664a296b591@172.16.1.202
CSeq: 14057 INVITE
User-Agent: Grandstream GXP2020 1.1.5.15
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 407

v=0
o=sipsma 8000 8001 IN IP4 172.16.1.202
s=SIP Call
c=IN IP4 172.16.1.202
t=0 0
m=audio 5004 RTP/AVP 8 0 4 18 2 97 9 3 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/16000
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<------------->
--- (14 headers 19 lines) ---
Sending to 172.16.1.202 : 5062 (no NAT)
Using INVITE request as basis request - dd64a664a296b591@172.16.1.202
Found user 'sipsma'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 172.16.1.202:5004
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.1.202:5004
Looking for 99 in phones (domain sip.serviceplanet.nl)
list_route: hop: <sip:sipsma@172.16.1.202:5062;transport=udp>

<--- Transmitting (no NAT) to 172.16.1.202:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.202:5062;branch=z9hG4bK368163d1c9e8aad3;received=172.16.1.202
From: <sip:sipsma@sip.serviceplanet.nl>;tag=a953cb4e625ee201
To: <sip:99@sip.serviceplanet.nl>
Call-ID: dd64a664a296b591@172.16.1.202
CSeq: 14057 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:99@172.16.1.25>
Content-Length: 0


<------------>
P[ 0]  --> Group Call group: kpn_isdn2
P[ 1] Port Down L2:0 L1:0
P[ 2] Port Down L2:0 L1:0
P[ 3] Port Down L2:0 L1:0
P[ 4] Port Down L2:0 L1:0
[Jan 23 12:04:08] WARNING[24623]: chan_misdn.c:3226 misdn_request: Could not Dial out on group 'kpn_isdn2'.
        Either the L2 and L1 on all of these ports where DOWN (see 'show application misdn_check_l2l1')
        Or there was no free channel on none of the ports

[Jan 23 12:04:08] WARNING[24623]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'mISDN' (cause 0 - Unknown)

<--- Transmitting (no NAT) to 172.16.1.202:5062 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.16.1.202:5062;branch=z9hG4bK368163d1c9e8aad3;received=172.16.1.202
From: <sip:sipsma@sip.serviceplanet.nl>;tag=a953cb4e625ee201
To: <sip:99@sip.serviceplanet.nl>;tag=as41eb18f9
Call-ID: dd64a664a296b591@172.16.1.202
CSeq: 14057 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:99@172.16.1.25>
Content-Length: 0


<------------>
Really destroying SIP dialog 'dd64a664a296b591@172.16.1.202' Method: INVITE
terpsichore*CLI> 
<--- SIP read from 172.16.1.202:5062 --->
ACK sip:99@sip.serviceplanet.nl SIP/2.0
Via: SIP/2.0/UDP 172.16.1.202:5062;branch=z9hG4bK368163d1c9e8aad3
From: <sip:sipsma@sip.serviceplanet.nl>;tag=a953cb4e625ee201
To: <sip:99@sip.serviceplanet.nl>;tag=as41eb18f9
Contact: <sip:sipsma@172.16.1.202:5062;transport=udp>
Supported: path
Proxy-Authorization: Digest username="sipsma", realm="asterisk", algorithm=MD5, uri="sip:99@sip.serviceplanet.nl", nonce="<<Removed to protect the guilty>>", response="c66b10f667d71442f44b4874f1758b4a"
Call-ID: dd64a664a296b591@172.16.1.202
CSeq: 14057 ACK
User-Agent: Grandstream GXP2020 1.1.5.15
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0


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