Hardware Version     C1
Firmware Version    GE_1.01
DSP Version    1.33-PP
IPv4 / IPv6 Setting    IPv4 Only

<--- Transmitting (no NAT) to 192.168.26.117:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.26.117:5060;branch=z9hG4bK_00265AE2C041_T3DCDC6E2;received=192.168.26.117;rport=5060
From: <sip:107@192.168.26.6>;tag=00265AE2C041_T261530068
To: <sip:107@192.168.26.6>;tag=as58e11986
Call-ID: REGISTER_00265AE2C041_T1441971383@192.168.26.117
CSeq: 71 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 60
Contact: <sip:107@192.168.26.117:5060>;expires=60
Date: Sun, 18 Apr 2010 21:50:17 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'REGISTER_00265AE2C041_T1441971383@192.168.26.117' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.26.117:5060 --->

<------------->
    -- Executing [xxxxxxx0239@incoming:1] Set("H323/ip$213.59.41.1:51220/6833", "CALLERID(num)=xxxxxxx4579") in new stack
    -- Executing [xxxxxxx0239@incoming:2] Dial("H323/ip$213.59.41.1:51220/6833", "SIP/107|120") in new stack
Audio is at 192.168.26.6 port 15322
Adding codec 0x100 (g729) to SDP
Reliably Transmitting (no NAT) to 192.168.26.117:5060:
INVITE sip:107@192.168.26.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.26.6:5060;branch=z9hG4bK283cdb95;rport
From: "xxxxxxx4579" <sip:xxxxxxx4579@192.168.26.6>;tag=as3cb0b571
To: <sip:107@192.168.26.117:5060>
Contact: <sip:xxxxxxx4579@192.168.26.6>
Call-ID: 14982dae56a7b5fa14e6216424ade021@192.168.26.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 18 Apr 2010 21:50:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 73332 73332 IN IP4 192.168.26.6
s=session
c=IN IP4 192.168.26.6
t=0 0
m=audio 15322 RTP/AVP 18
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 107
aster*CLI>
<--- SIP read from 192.168.26.117:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.26.6:5060;branch=z9hG4bK283cdb95;rport
From: "xxxxxxx4579" <sip:xxxxxxx4579@192.168.26.6>;tag=as3cb0b571
To: <sip:107@192.168.26.117>
Call-ID: 14982dae56a7b5fa14e6216424ade021@192.168.26.6
Contact: <sip:107@192.168.26.117:5060>
CSeq: 102 INVITE
User-Agent: DPH-150SE 01.01
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
aster*CLI>
<--- SIP read from 192.168.26.117:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.26.6:5060;branch=z9hG4bK283cdb95;rport
From: "xxxxxxx4579" <sip:xxxxxxx4579@192.168.26.6>;tag=as3cb0b571
To: <sip:107@192.168.26.117>;tag=00265AE2C041_T2022084191
Call-ID: 14982dae56a7b5fa14e6216424ade021@192.168.26.6
Contact: <sip:107@192.168.26.117:5060>
CSeq: 102 INVITE
User-Agent: DPH-150SE 01.01
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/107-29e7c000 is ringing
aster*CLI>
<--- SIP read from 192.168.26.117:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.26.6:5060;branch=z9hG4bK283cdb95;rport
From: "xxxxxxx4579" <sip:xxxxxxx4579@192.168.26.6>;tag=as3cb0b571
To: <sip:107@192.168.26.117>;tag=00265AE2C041_T2022084191
Call-ID: 14982dae56a7b5fa14e6216424ade021@192.168.26.6
Contact: <sip:107@192.168.26.117:5060>
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Supported: 100rel,timer,replaces
Content-Type: application/sdp
Content-Length: 185

v=0
o=- 13398300 13398300 IN IP4 192.168.26.117
s=DPH-150SE 01.01
c=IN IP4 192.168.26.117
t=0 0
m=audio 41000 RTP/AVP 18
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 18
Peer audio RTP is at port 192.168.26.117:41000
Found audio description format G729 for ID 18
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.26.117:41000
list_route: hop: <sip:107@192.168.26.117:5060>
set_destination: Parsing <sip:107@192.168.26.117:5060> for address/port to send to
set_destination: set destination to 192.168.26.117, port 5060
Transmitting (no NAT) to 192.168.26.117:5060:
ACK sip:107@192.168.26.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.26.6:5060;branch=z9hG4bK6fab58fc;rport
From: "xxxxxxx4579" <sip:xxxxxxx4579@192.168.26.6>;tag=as3cb0b571
To: <sip:107@192.168.26.117:5060>;tag=00265AE2C041_T2022084191
Contact: <sip:xxxxxxx4579@192.168.26.6>
Call-ID: 14982dae56a7b5fa14e6216424ade021@192.168.26.6
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/107-29e7c000 answered H323/ip$213.59.41.1:51220/6833