Script started on Thu Aug  4 11:33:48 2005
obelix:~# gdb /usr/sbin/asterisk 7798obelix:~# asterisk -r
Asterisk CVS-NHEAD-08/04/05-11:17:06, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <markster@digium.com>
=========================================================================
Connected to Asterisk CVS-NHEAD-08/04/05-11:17:06 currently running on obelix (pid = 7879)
obelix*CLI> quitsip debug level 4set verbose 4
obelix*CLI> Verbosity was 0 and is now 4
obelix*CLI> set verbose 4quitsip debug level 4
obelix*CLI> Debugging level set to 4, file '<any>'
obelix*CLI> debug level 4set verbose 4quitsip debug
obelix*CLI> SIP Debugging enabled
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
PUBLISH sip:u8@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK0.b61c9346.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-gzdg7128fgp4;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=6idlsapsga
To: "u8" <sip:u8@voip.sysfrog.org>
Call-ID: 3c26701cef42-r373exygo3mg@snom190
CSeq: 2 PUBLISH
Max-Forwards: 16
Event: proxy-config
Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e1598c321e92f90e8f98b565a6e4fb2c413d",uri="sip:u8@voip.sysfrog.org",response="7fb37115656148668f151e2e2d260e02",algorithm=md5
Content-Type: application/text
Content-Length: 0

--- (12 headers 0 lines)---
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK0.b61c9346.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-gzdg7128fgp4
From: "u8" <sip:u8@voip.sysfrog.org>;tag=6idlsapsga
To: "u8" <sip:u8@voip.sysfrog.org>;tag=as67fa5591
Call-ID: 3c26701cef42-r373exygo3mg@snom190
CSeq: 2 PUBLISH
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: 
Accept: application/sdp
Content-Length: 0


---
Aug  4 11:34:24 NOTICE[7896]: chan_sip.c:9714 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX'
Destroying call '3c26701cef42-r373exygo3mg@snom190'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
PUBLISH sip:u8@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK60ce.8d032ae3.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-2rrmk8hw28da;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=uzvy8w9osj
To: "u8" <sip:u8@voip.sysfrog.org>
Call-ID: 3c26701d7530-dfh1jr3schcp@snom190
CSeq: 2 PUBLISH
Max-Forwards: 16
Event: number-guessing
Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e15aac0f3700863099e5323019b367fbace3",uri="sip:u8@voip.sysfrog.org",response="44bd48607cbaadb9391be051484e99cd",algorithm=md5
Content-Type: application/text
Content-Length: 25

Number: 86
Max-Hits: 3
--- (12 headers 2 lines)---
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK60ce.8d032ae3.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-2rrmk8hw28da
From: "u8" <sip:u8@voip.sysfrog.org>;tag=uzvy8w9osj
To: "u8" <sip:u8@voip.sysfrog.org>;tag=as26f10ec2
Call-ID: 3c26701d7530-dfh1jr3schcp@snom190
CSeq: 2 PUBLISH
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: 
Accept: application/sdp
Content-Length: 0


---
Aug  4 11:34:25 NOTICE[7896]: chan_sip.c:9714 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX'
Destroying call '3c26701d7530-dfh1jr3schcp@snom190'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:86@voip.sysfrog.org;user=phone SIP/2.0
Record-Route: <sip:XX.XX.XX.XX;ftag=m7dxxmv5er;lr=on>
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKcf31.07e24b.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-83phpcmzq2mz;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 2 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e15aac0f3700863099e5323019b367fbace3",uri="sip:86@voip.sysfrog.org;user=phone",response="9b878c89a72b8d16d2909817cdb7019b",algorithm=md5
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 604740038 604740038 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (20 headers 17 lines)---
Using INVITE request as basis request - 3c26701d7ef4-avnd7zuk8gvc@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Found peer 'sipsepp'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7026
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 86 in incoming-sip
list_route: hop: <sip:XX.XX.XX.XX;ftag=m7dxxmv5er;lr=on>
list_route: hop: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKcf31.07e24b.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-83phpcmzq2mz
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI>     -- Executing Dial("SIP/voip.sysfrog.org-0823b5c0", "SIP/u6@voip.sysfrog.org|20|r") in new stack
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 12604
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 12 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u6@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK512440b0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Thu, 04 Aug 2005 09:34:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7915 7915 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 12604 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI>     -- Called u6@voip.sysfrog.org
obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKcf31.07e24b.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-83phpcmzq2mz
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK512440b0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3724 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@voip.sysfrog.org out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 180 Ringing
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK512440b0
Record-Route: <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 0

--- (9 headers 0 lines)---
obelix*CLI>     -- SIP/voip.sysfrog.org-643c is ringing
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK512440b0
Record-Route: <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on>
Contact: u6 <sip:u6@YY.YY.YY.YY:6386>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 238
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 170602628 170602628 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7030 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (13 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7030
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on>
list_route: hop: <sip:u6@YY.YY.YY.YY:6386>
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK71c19916
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI>     -- SIP/voip.sysfrog.org-643c answered SIP/voip.sysfrog.org-0823b5c0
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 15188
obelix*CLI> Answering with capability 0x8 (alaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKcf31.07e24b.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-83phpcmzq2mz
Record-Route: <sip:XX.XX.XX.XX;ftag=m7dxxmv5er;lr=on>
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 7915 7915 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 15188 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI>     -- Attempting native bridge of SIP/voip.sysfrog.org-0823b5c0 and SIP/voip.sysfrog.org-643c
obelix*CLI> set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on> for address/port to send to
obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 12604
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x1 (g723)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with capability 0x10 (g726)
obelix*CLI> Answering with capability 0x100 (g729)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 15 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5d4f945d
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 341

v=0
o=root 7915 7916 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 8 4 3 0 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5d4f945d
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@YY.YY.YY.YY:6386 out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5d4f945d
Contact: u6 <sip:u6@YY.YY.YY.YY:6386>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 238
Content-Type: application/sdp

v=0
o=- 170602831 170602831 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7030 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (10 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7030
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK4ca9d9a8
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 103 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
ACK sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-l4rgw428goso;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 2 ACK
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Content-Length: 0

--- (10 headers 0 lines)---
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=m7dxxmv5er;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
We're at ZZ.ZZ.ZZ.ZZ port 15188
Answering with capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport
Route: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7915 7916 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:6554 out_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 604740038 604740039 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7026
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug  4 11:34:27 NOTICE[7896]: chan_sip.c:3353 process_sdp: No compatible codecs!
list_route: hop: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK771827bc;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 604740038 604740039 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7026
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug  4 11:34:28 NOTICE[7896]: chan_sip.c:3353 process_sdp: No compatible codecs!
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2a4119ff;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 604740038 604740039 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7026
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Aug  4 11:34:29 NOTICE[7896]: chan_sip.c:3353 process_sdp: No compatible codecs!
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6e116267;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKdf31.d75a5361.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-p8axl569u7i7;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 3 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 604740038 604740040 IN IP4 10.122.32.84
s=call
c=IN IP4 0.0.0.0
t=0 0
m=audio 10996 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c26701d7ef4-avnd7zuk8gvc@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:10996
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    -- Started music on hold, class 'default', on SIP/voip.sysfrog.org-643c
We're at ZZ.ZZ.ZZ.ZZ port 15188
Answering with capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKdf31.d75a5361.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-p8axl569u7i7
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 3 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7915 7917 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> Aug  4 11:34:31 NOTICE[7915]: res_musiconhold.c:216 ast_moh_files_next: SIP/voip.sysfrog.org-643c Opened file 0 '/var/lib/asterisk/moh-native/busstrafik'
obelix*CLI> set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on> for address/port to send to
obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 12604
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 12 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK71bb393a
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 104 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7915 7917 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 12604 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK71bb393a
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 104 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@YY.YY.YY.YY:6386 out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 104 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK71bb393a
Contact: u6 <sip:u6@YY.YY.YY.YY:6386>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 238
Content-Type: application/sdp

v=0
o=- 170603194 170603194 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7030 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (10 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7030
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6dacd40a
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 104 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 604740038 604740039 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Response message INVITE arrived
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0550e83d;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
ACK sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-t2hw81oupfum;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 3 ACK
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Content-Length: 0

--- (10 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
PUBLISH sip:u8@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9708.5817b935.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-9039x9hwm3x0;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=p51it5v9zl
To: "u8" <sip:u8@voip.sysfrog.org>
Call-ID: 3c2670249c40-styma3r0wm8f@snom190
CSeq: 2 PUBLISH
Max-Forwards: 16
Event: proxy-config
Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e161f4c4775ff0cfbb65d0f7938f464b9651",uri="sip:u8@voip.sysfrog.org",response="3fac0588c5d6ed7b25e8ba91d29aeef8",algorithm=md5
Content-Type: application/text
Content-Length: 0

--- (12 headers 0 lines)---
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9708.5817b935.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-9039x9hwm3x0
From: "u8" <sip:u8@voip.sysfrog.org>;tag=p51it5v9zl
To: "u8" <sip:u8@voip.sysfrog.org>;tag=as26f4bb0e
Call-ID: 3c2670249c40-styma3r0wm8f@snom190
CSeq: 2 PUBLISH
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: 
Accept: application/sdp
Content-Length: 0


---
Aug  4 11:34:32 NOTICE[7896]: chan_sip.c:9714 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX'
Destroying call '3c2670249c40-styma3r0wm8f@snom190'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
PUBLISH sip:u8@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK850b.25b7e293.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-w5tblh4ai3eo;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=pmcedhhmls
To: "u8" <sip:u8@voip.sysfrog.org>
Call-ID: 3c267025222e-k6gqu5vvg4t5@snom190
CSeq: 2 PUBLISH
Max-Forwards: 16
Event: number-guessing
Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e1625e45c56e4ebb391e81b776acb4140a3c",uri="sip:u8@voip.sysfrog.org",response="5b5fc85d7546e426bc123b6c7ee619fe",algorithm=md5
Content-Type: application/text
Content-Length: 25

Number: 85
Max-Hits: 3
--- (12 headers 2 lines)---
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK850b.25b7e293.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-w5tblh4ai3eo
From: "u8" <sip:u8@voip.sysfrog.org>;tag=pmcedhhmls
To: "u8" <sip:u8@voip.sysfrog.org>;tag=as20ebbbe0
Call-ID: 3c267025222e-k6gqu5vvg4t5@snom190
CSeq: 2 PUBLISH
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: 
Accept: application/sdp
Content-Length: 0


---
Aug  4 11:34:33 NOTICE[7896]: chan_sip.c:9714 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX'
Destroying call '3c267025222e-k6gqu5vvg4t5@snom190'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:85@voip.sysfrog.org;user=phone SIP/2.0
Record-Route: <sip:XX.XX.XX.XX;ftag=elvnjnhzkz;lr=on>
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKe05d.accc3ec7.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-ct010485ri6v;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 2 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e1625e45c56e4ebb391e81b776acb4140a3c",uri="sip:85@voip.sysfrog.org;user=phone",response="85e3a8cc5a6167b2eee77dd7998d2185",algorithm=md5
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 150881502 150881502 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (20 headers 17 lines)---
Using INVITE request as basis request - 3c26702546cd-0z66kq5n4653@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Found peer 'sipsepp'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7034
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 85 in incoming-sip
list_route: hop: <sip:XX.XX.XX.XX;ftag=elvnjnhzkz;lr=on>
list_route: hop: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKe05d.accc3ec7.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-ct010485ri6v
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI>     -- Executing Dial("SIP/voip.sysfrog.org-08270198", "SIP/u5@voip.sysfrog.org|20|r") in new stack
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18362
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 12 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u5@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2ac7dd81
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Thu, 04 Aug 2005 09:34:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7918 7918 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 18362 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI>     -- Called u5@voip.sysfrog.org
obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKe05d.accc3ec7.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-ct010485ri6v
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2ac7dd81
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@voip.sysfrog.org out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 180 Ringing
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2ac7dd81
Record-Route: <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 0

--- (9 headers 0 lines)---
obelix*CLI>     -- SIP/voip.sysfrog.org-79a3 is ringing
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2ac7dd81
Record-Route: <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on>
Contact: u5 <sip:u5@YY.YY.YY.YY:6393>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 209
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 15797877 15797877 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7038 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (12 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7038
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on>
list_route: hop: <sip:u5@YY.YY.YY.YY:6393>
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK561da4fa
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI>     -- SIP/voip.sysfrog.org-79a3 answered SIP/voip.sysfrog.org-08270198
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 17206
obelix*CLI> Answering with capability 0x8 (alaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKe05d.accc3ec7.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-ct010485ri6v
Record-Route: <sip:XX.XX.XX.XX;ftag=elvnjnhzkz;lr=on>
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 7918 7918 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 17206 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI>     -- Attempting native bridge of SIP/voip.sysfrog.org-08270198 and SIP/voip.sysfrog.org-79a3
obelix*CLI> set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on> for address/port to send to
obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18362
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x1 (g723)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with capability 0x10 (g726)
obelix*CLI> Answering with capability 0x100 (g729)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 15 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51c151c1
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 341

v=0
o=root 7918 7919 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 8 4 3 0 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51c151c1
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:6393 out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51c151c1
Contact: u5 <sip:u5@YY.YY.YY.YY:6393>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 209
Content-Type: application/sdp

v=0
o=- 15797942 15797942 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7038 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (10 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7038
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7970b89c
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 103 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
ACK sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-tijwvebsqu0l;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 2 ACK
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Content-Length: 0

--- (10 headers 0 lines)---
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=elvnjnhzkz;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
We're at ZZ.ZZ.ZZ.ZZ port 17206
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 10 lines
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport
Route: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7918 7919 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7038 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:6554 out_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 150881502 150881503 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7034
Found description format pcma
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7fdeabd4;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 150881502 150881503 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7034
Found description format pcma
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3c446648;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 604740038 604740039 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Response message INVITE arrived
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK017953ca;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 150881502 150881503 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7034
Found description format pcma
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK26575cec;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf05d.4e2de167.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-0pp6ey27i376;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 3 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 150881502 150881504 IN IP4 10.122.32.84
s=call
c=IN IP4 0.0.0.0
t=0 0
m=audio 10704 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c26702546cd-0z66kq5n4653@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:10704
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    -- Started music on hold, class 'default', on SIP/voip.sysfrog.org-79a3
We're at ZZ.ZZ.ZZ.ZZ port 17206
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf05d.4e2de167.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-0pp6ey27i376
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 3 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7918 7920 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7038 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> Aug  4 11:34:35 NOTICE[7918]: res_musiconhold.c:216 ast_moh_files_next: SIP/voip.sysfrog.org-79a3 Opened file 0 '/var/lib/asterisk/moh-native/busstrafik'
obelix*CLI> set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on> for address/port to send to
obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18362
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 12 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7cd5bf6c
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 104 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7918 7920 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 18362 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7cd5bf6c
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 104 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:6393 out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 104 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7cd5bf6c
Contact: u5 <sip:u5@YY.YY.YY.YY:6393>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 209
Content-Type: application/sdp

v=0
o=- 15798137 15798137 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7038 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (10 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7038
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK59c0c8bb
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 104 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
ACK sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-bcye3dzwvidq;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 3 ACK
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Content-Length: 0

--- (10 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 150881502 150881503 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Response message INVITE arrived
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0812b162;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
Transfer to 85 in incoming-sip
Transfer from u8 in incoming-sip
    -- Stopped music on hold on SIP/voip.sysfrog.org-643c
    -- Stopped music on hold on SIP/voip.sysfrog.org-79a3
Transmitting (no NAT) to YY.YY.YY.YY:6554:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0
X-Asterisk-HangupCause:: Normal Clearing


---
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 103 NOTIFY
User-Agent: Asterisk
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 104 BYE
User-Agent: Asterisk
X-Asterisk-HangupCause:: Normal Clearing
Content-Length: 0


---
obelix*CLI> set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
obelix*CLI> set_destination: set destination to YY.YY.YY.YY, port 6554
obelix*CLI> Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on> for address/port to send to
obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18362
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 12 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5c6e92d4
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 105 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7918 7921 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 18362 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5c6e92d4
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 105 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:6393 out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 105 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5c6e92d4
Contact: u5 <sip:u5@YY.YY.YY.YY:6393>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 209
Content-Type: application/sdp

v=0
o=- 15798344 15798344 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7038 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (10 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7038
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as7e51302d;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc463a1
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
To: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 105 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
Transfer to 85 in incoming-sip
Transfer from u8 in incoming-sip
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
BYE sip:asterisk@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK6877.862a0721.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-536332af
From: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
To: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 101 BYE
Max-Forwards: 16
User-Agent: Sipura/SPA841-3.1.2(d)
Content-Length: 0

--- (10 headers 0 lines)---
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK6877.862a0721.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-536332af
From: <sip:u5@voip.sysfrog.org>;tag=9e2ffb6524c174d3i0
To: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as7e51302d
Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ
CSeq: 101 BYE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0
X-Asterisk-HangupCause:: Normal Clearing


---
obelix*CLI> set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as2382a1f1;lr=on> for address/port to send to
obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
BYE sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK448feee4
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 105 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Destroying call '0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u6@voip.sysfrog.org>;tag=a075727231d1e08i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as2382a1f1
Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ
CSeq: 105 BYE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK448feee4
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 0

--- (8 headers 0 lines)---
Destroying call '18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ'
obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 103 NOTIFY
User-Agent: Asterisk
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 104 BYE
User-Agent: Asterisk
X-Asterisk-HangupCause:: Normal Clearing
Content-Length: 0


---
obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
Transfer to 85 in incoming-sip
Transfer from u8 in incoming-sip
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 150881502 150881503 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Response message INVITE arrived
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6c275e5a;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 103 NOTIFY
User-Agent: Asterisk
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 104 BYE
User-Agent: Asterisk
X-Asterisk-HangupCause:: Normal Clearing
Content-Length: 0


---
obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 604740038 604740039 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Response message INVITE arrived
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK78879740;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 103 NOTIFY
User-Agent: Asterisk
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 104 BYE
User-Agent: Asterisk
X-Asterisk-HangupCause:: Normal Clearing
Content-Length: 0


---
obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
Transfer to 85 in incoming-sip
Transfer from u8 in incoming-sip
obelix*CLI> Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 103 NOTIFY
User-Agent: Asterisk
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 104 BYE
User-Agent: Asterisk
X-Asterisk-HangupCause:: Normal Clearing
Content-Length: 0


---
obelix*CLI> Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #5 (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 103 NOTIFY
User-Agent: Asterisk
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Retransmitting #5 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 104 BYE
User-Agent: Asterisk
X-Asterisk-HangupCause:: Normal Clearing
Content-Length: 0


---
obelix*CLI> Retransmitting #5 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKc05d.770c3a73.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 4 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 150881502 150881505 IN IP4 10.122.32.84
s=call
c=IN IP4 0.0.0.0
t=0 0
m=audio 10704 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c26702546cd-0z66kq5n4653@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 5 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 604740038 604740041 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c26701d7ef4-avnd7zuk8gvc@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKc05d.770c3a73.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 4 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 150881502 150881505 IN IP4 10.122.32.84
s=call
c=IN IP4 0.0.0.0
t=0 0
m=audio 10704 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
--- (18 headers 17 lines)---
Ignoring this INVITE request

<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 5 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 604740038 604740041 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (18 headers 17 lines)---
Ignoring this INVITE request

<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 150881502 150881503 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Response message INVITE arrived
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7764249d;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Aug  4 11:34:49 WARNING[7896]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c26701d7ef4-avnd7zuk8gvc@snom190 for seqno 103 (Non-critical Request)
Aug  4 11:34:49 WARNING[7896]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c26701d7ef4-avnd7zuk8gvc@snom190 for seqno 104 (Non-critical Request)
Aug  4 11:34:49 WARNING[7896]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c26702546cd-0z66kq5n4653@snom190 for seqno 103 (Non-critical Request)
Destroying call '3c26702546cd-0z66kq5n4653@snom190'
Destroying call '3c26701d7ef4-avnd7zuk8gvc@snom190'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKc05d.770c3a73.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 4 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 150881502 150881505 IN IP4 10.122.32.84
s=call
c=IN IP4 0.0.0.0
t=0 0
m=audio 10704 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c26702546cd-0z66kq5n4653@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Found peer 'sipsepp'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:10704
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 85 in incoming-sip
list_route: hop: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKc05d.770c3a73.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 4 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI>     -- Executing Dial("SIP/voip.sysfrog.org-08242c08", "SIP/u5@voip.sysfrog.org|20|r") in new stack
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 5 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 604740038 604740041 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c26701d7ef4-avnd7zuk8gvc@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Found peer 'sipsepp'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7026
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 86 in incoming-sip
list_route: hop: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 5 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI>     -- Executing Dial("SIP/voip.sysfrog.org-0824f240", "SIP/u6@voip.sysfrog.org|20|r") in new stack
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 11240
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 12 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u5@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK34d79d6c
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
To: <sip:u5@voip.sysfrog.org>
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Thu, 04 Aug 2005 09:34:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7925 7925 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 11240 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI>     -- Called u5@voip.sysfrog.org
obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKc05d.770c3a73.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 4 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK34d79d6c
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
To: <sip:u5@voip.sysfrog.org>
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@voip.sysfrog.org out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18816
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 12 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u6@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK68cb5e8e
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
To: <sip:u6@voip.sysfrog.org>
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Thu, 04 Aug 2005 09:34:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7926 7926 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 18816 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI>     -- Called u6@voip.sysfrog.org
obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 5 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK68cb5e8e
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
To: <sip:u6@voip.sysfrog.org>
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@voip.sysfrog.org out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 180 Ringing
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK68cb5e8e
Record-Route: <sip:XX.XX.XX.XX;ftag=as6b06de5b;lr=on>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 0

--- (9 headers 0 lines)---

<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 180 Ringing
To: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK34d79d6c
Record-Route: <sip:XX.XX.XX.XX;ftag=as4276f4bc;lr=on>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 0

--- (9 headers 0 lines)---
obelix*CLI>     -- SIP/voip.sysfrog.org-2407 is ringing
obelix*CLI>     -- SIP/voip.sysfrog.org-a963 is ringing
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK34d79d6c
Record-Route: <sip:XX.XX.XX.XX;ftag=as4276f4bc;lr=on>
Contact: u5 <sip:u5@YY.YY.YY.YY:6393>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 209
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 15799626 15799626 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (12 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7040
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:XX.XX.XX.XX;ftag=as4276f4bc;lr=on>
list_route: hop: <sip:u5@YY.YY.YY.YY:6393>
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as4276f4bc;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK09f63201
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
To: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
    -- SIP/voip.sysfrog.org-2407 answered SIP/voip.sysfrog.org-08242c08
We're at ZZ.ZZ.ZZ.ZZ port 19156
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKc05d.770c3a73.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 4 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 7925 7925 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 19156 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/voip.sysfrog.org-08242c08 and SIP/voip.sysfrog.org-2407
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as4276f4bc;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
We're at ZZ.ZZ.ZZ.ZZ port 11240
Answering/Requesting with root capability 0x8 (alaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK15d2ebd4
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
To: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7925 7926 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 11240 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
aobelix*CLI> =fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK15d2ebd4
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
To: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:6393 out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
ACK sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-wi073saxgi79;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 4 ACK
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Content-Length: 0

--- (10 headers 0 lines)---
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
We're at ZZ.ZZ.ZZ.ZZ port 19156
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
11 headers, 10 lines
Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7925 7926 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK15d2ebd4
Contact: u5 <sip:u5@YY.YY.YY.YY:6393>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 209
Content-Type: application/sdp

v=0
o=- 15800261 15800261 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (10 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7040
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as4276f4bc;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u5@YY.YY.YY.YY:6393 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK68a03ff2
Route: <sip:u5@YY.YY.YY.YY:6393>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
To: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 103 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7925 7926 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 604740038 604740039 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK68cb5e8e
Record-Route: <sip:XX.XX.XX.XX;ftag=as6b06de5b;lr=on>
Contact: u6 <sip:u6@YY.YY.YY.YY:6386>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 238
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 170605152 170605152 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (13 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7042
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:XX.XX.XX.XX;ftag=as6b06de5b;lr=on>
list_route: hop: <sip:u6@YY.YY.YY.YY:6386>
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as6b06de5b;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7b3b896d
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
    -- SIP/voip.sysfrog.org-a963 answered SIP/voip.sysfrog.org-0824f240
We're at ZZ.ZZ.ZZ.ZZ port 14658
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 5 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 7926 7926 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 14658 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/voip.sysfrog.org-0824f240 and SIP/voip.sysfrog.org-a963
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as6b06de5b;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
We're at ZZ.ZZ.ZZ.ZZ port 18816
Answering/Requesting with root capability 0x8 (alaw)
Answering with capability 0x1 (g723)
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x10 (g726)
Answering with capability 0x100 (g729)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 15 lines
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a8ae2ba
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 341

v=0
o=root 7926 7927 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7026 RTP/AVP 8 4 3 0 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a8ae2ba
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@YY.YY.YY.YY:6386 out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1"

--- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a8ae2ba
Contact: u6 <sip:u6@YY.YY.YY.YY:6386>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 238
Content-Type: application/sdp

v=0
o=- 170606152 170606152 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (10 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7042
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as6b06de5b;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK4d5dd139
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 103 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
ACK sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-10fz744gtbti;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 5 ACK
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Content-Length: 0

--- (10 headers 0 lines)---
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
We're at ZZ.ZZ.ZZ.ZZ port 14658
Answering with capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
11 headers, 10 lines
Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7926 7927 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 OK
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 103 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a8ae2ba
Contact: u6 <sip:u6@YY.YY.YY.YY:6386>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 238
Content-Type: application/sdp

v=0
o=- 170606152 170606152 IN IP4 YY.YY.YY.YY
s=-
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (10 headers 12 lines)---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7042
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:XX.XX.XX.XX;ftag=as6b06de5b;lr=on> for address/port to send to
set_destination: set destination to XX.XX.XX.XX, port 5060
Transmitting (no NAT) to XX.XX.XX.XX:5060:
ACK sip:u6@YY.YY.YY.YY:6386 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2e69a5b3
Route: <sip:u6@YY.YY.YY.YY:6386>
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
To: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 103 ACK
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7925 7926 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7926 7927 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
BYE sip:asterisk@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8ef6.edf59345.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6386;branch=z9hG4bK-1bc36695
From: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
To: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 101 BYE
Max-Forwards: 16
User-Agent: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 0

--- (10 headers 0 lines)---
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8ef6.edf59345.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6386;branch=z9hG4bK-1bc36695
From: <sip:u6@voip.sysfrog.org>;tag=787170188283fba7i0
To: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as6b06de5b
Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ
CSeq: 101 BYE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0
Xobelix*CLI> -Asterisk-HangupCause:: Normal Clearing


---
obelix*CLI> Destroying call '2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ'
obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7925 7926 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7926 7927 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 4 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
obelix*CLI> Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7925 7926 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
BYE sip:asterisk@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK401f.c19573f6.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-5dc97018
From: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
To: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 101 BYE
Max-Forwards: 16
User-Agent: Sipura/SPA841-3.1.2(d)
Content-Length: 0

--- (10 headers 0 lines)---
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK401f.c19573f6.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-5dc97018
From: <sip:u5@voip.sysfrog.org>;tag=a9d0f338a3ab2df1i0
To: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as4276f4bc
Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ
CSeq: 101 BYE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0
X-Asterisk-HangupCause:: Normal Clearing


---
obelix*CLI> Destroying call '6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 200 Ok
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 INVITE
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Session-Expires: 3600
User-Agent: snom190/3.60k
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Content-Type: application/sdp
Content-Length: 217

v=0
o=root 150881502 150881503 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (14 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port YY.YY.YY.YY:7034
Found description format pcma
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Transmitting (no NAT) to YY.YY.YY.YY:6554:
ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7aa3156c;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7926 7927 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 5 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26701d7ef4-avnd7zuk8gvc%40snom190%3Bto-tag%3Das6d3df7db%3Bfrom-tag%3Dm7dxxmv5er
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
Transfer to 86 in incoming-sip
Transfer from u8 in incoming-sip
Aug  4 11:35:08 WARNING[7896]: chan_sip.c:8933 attempt_transfer: Transfer attempted without dual ownership?
Transmitting (no NAT) to YY.YY.YY.YY:6554:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 5 REFER
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 104 NOTIFY
User-Agent: Asterisk
Event: refer;id=5
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
set_destination: Parsing <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv> for address/port to send to
set_destination: set destination to YY.YY.YY.YY, port 6554
Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 105 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7926 7927 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 5 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26701d7ef4-avnd7zuk8gvc%40snom190%3Bto-tag%3Das6d3df7db%3Bfrom-tag%3Dm7dxxmv5er
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
Transfer to 86 in incoming-sip
Transfer from u8 in incoming-sip
obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 104 NOTIFY
User-Agent: Asterisk
Event: refer;id=5
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 105 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #5 (no NAT) to YY.YY.YY.YY:6554:
INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport
From: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
To: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 7926 7927 IN IP4 YY.YY.YY.YY
s=session
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7042 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 5 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26701d7ef4-avnd7zuk8gvc%40snom190%3Bto-tag%3Das6d3df7db%3Bfrom-tag%3Dm7dxxmv5er
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
Transfer to 86 in incoming-sip
Transfer from u8 in incoming-sip
obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 104 NOTIFY
User-Agent: Asterisk
Event: refer;id=5
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 105 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> Aug  4 11:35:12 WARNING[7896]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c26701d7ef4-avnd7zuk8gvc@snom190 for seqno 102 (Non-critical Request)
Destroying call '3c26701d7ef4-avnd7zuk8gvc@snom190'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8f31.d4457802.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-jm4qpgkrrysj;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 6 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 604740038 604740042 IN IP4 10.122.32.84
s=call
c=IN IP4 0.0.0.0
t=0 0
m=audio 10996 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendonly
--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c26701d7ef4-avnd7zuk8gvc@snom190
obelix*CLI> Sending to XX.XX.XX.XX : 5060 (non-NAT)
obelix*CLI> Found peer 'sipsepp'
obelix*CLI> Found RTP audio format 0
obelix*CLI> Found RTP audio format 8
obelix*CLI> Found RTP audio format 9
obelix*CLI> Found RTP audio format 2
obelix*CLI> Found RTP audio format 3
obelix*CLI> Found RTP audio format 18
obelix*CLI> Found RTP audio format 4
obelix*CLI> Found RTP audio format 101
obelix*CLI> Peer audio RTP is at port 0.0.0.0:10996
obelix*CLI> Found description format pcmu
obelix*CLI> Found description format pcma
obelix*CLI> Found description format g722
obelix*CLI> Found description format g726-32
obelix*CLI> Found description format gsm
obelix*CLI> Found description format g729
obelix*CLI> Found description format g723
obelix*CLI> Found description format telephone-event
obelix*CLI> Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
obelix*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
obelix*CLI> Looking for 86 in incoming-sip
obelix*CLI> list_route: hop: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8f31.d4457802.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-jm4qpgkrrysj
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 6 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI>     -- Executing Dial("SIP/voip.sysfrog.org-0824d6b0", "SIP/u6@voip.sysfrog.org|20|r") in new stack
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKa05d.e51aacd.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-nx2g5ux53lzs;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 6 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
obelix*CLI> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 150881502 150881506 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
obelix*CLI> --- (18 headers 17 lines)---
obelix*CLI> Using INVITE request as basis request - 3c26702546cd-0z66kq5n4653@snom190
obelix*CLI> Sending to XX.XX.XX.XX : 5060 (non-NAT)
obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 11728
obelix*CLI> Answering/Requesting with root capability 0x8 (alaw)
obelix*CLI> Answering with capability 0x2 (gsm)
obelix*CLI> Answering with capability 0x4 (ulaw)
obelix*CLI> Answering with non-codec capability 0x1 (telephone-event)
obelix*CLI> 12 headers, 12 lines
obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
INVITE sip:u6@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6c3d9c1b
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as510099ca
To: <sip:u6@voip.sysfrog.org>
Contact: <sip:asterisk@ZZ.ZZ.ZZ.ZZ>
Call-ID: 2035685651fcc9842b9d68995d5717c3@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Thu, 04 Aug 2005 09:35:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 7935 7935 IN IP4 ZZ.ZZ.ZZ.ZZ
s=session
c=IN IP4 ZZ.ZZ.ZZ.ZZ
t=0 0
m=audio 11728 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
obelix*CLI>     -- Called u6@voip.sysfrog.org
obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8f31.d4457802.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-jm4qpgkrrysj
From: "u8" <sip:u8@voip.sysfrog.org>;tag=m7dxxmv5er
To: <sip:86@voip.sysfrog.org;user=phone>;tag=as6d3df7db
Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190
CSeq: 6 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:86@ZZ.ZZ.ZZ.ZZ>
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6c3d9c1b
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as510099ca
To: <sip:u6@voip.sysfrog.org>
Call-ID: 2035685651fcc9842b9d68995d5717c3@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3724 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@voip.sysfrog.org out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1"

obelix*CLI> --- (9 headers 0 lines)---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 180 Ringing
To: <sip:u6@voip.sysfrog.org>;tag=592c53bf95be9d09i0
From: "u8" <sip:asterisk@ZZ.ZZ.ZZ.ZZ>;tag=as510099ca
Call-ID: 2035685651fcc9842b9d68995d5717c3@ZZ.ZZ.ZZ.ZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6c3d9c1b
Record-Route: <sip:XX.XX.XX.XX;ftag=as510099ca;lr=on>
Server: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 0

--- (9 headers 0 lines)---
obelix*CLI>     -- SIP/voip.sysfrog.org-a359 is ringing
obelix*CLI> Retransmitting #4 (no NAT) to XX.XX.XX.XX:5060:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 480 User Not Registered
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 103 BYE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3723 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv out_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv via_cnt==1"

--- (9 headers 0 lines)---
Response message BYE arrived
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKa05d.e51aacd.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-nx2g5ux53lzs;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 6 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
obelix*CLI> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 150881502 150881506 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (18 headers 17 lines)---
Ignoring this INVITE request
obelix*CLI> Retransmitting #3 (no NAT) to XX.XX.XX.XX:5060:
NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 104 NOTIFY
User-Agent: Asterisk
Event: refer;id=5
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Retransmitting #3 (no NAT) to XX.XX.XX.XX:5060:
BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Contact: <sip:85@ZZ.ZZ.ZZ.ZZ>
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 105 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 480 User Not Registered
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 105 BYE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv out_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv via_cnt==1"

--- (9 headers 0 lines)---
Response message BYE arrived

<-- SIP read from XX.XX.XX.XX:5060: 
SIP/2.0 480 User Not Registered
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport=5060
From: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
To: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 104 NOTIFY
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3724 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv out_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv via_cnt==1"

--- (9 headers 0 lines)---
Response message NOTIFY arrived
Destroying call '3c26702546cd-0z66kq5n4653@snom190'
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 5 REFER
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26701d7ef4-avnd7zuk8gvc%40snom190%3Bto-tag%3Das6d3df7db%3Bfrom-tag%3Dm7dxxmv5er
Referred-By: sip:u8@voip.sysfrog.org
User-Agent: snom190/3.60k
Content-Length: 0

--- (13 headers 0 lines)---
Transfer to 86 in incoming-sip-default
Transfer from u8 in incoming-sip-default
Aug  4 11:35:14 WARNING[7896]: chan_sip.c:8933 attempt_transfer: Transfer attempted without dual ownership?
Transmitting (no NAT) to XX.XX.XX.XX:5060:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 5 REFER
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: 
Content-Length: 0


---
Aug  4 11:35:14 NOTICE[7896]: chan_sip.c:3518 copy_header: No field 'Call-ID' present to copy
11 headers, 1 lines
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
NOTIFY  SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK485b4484;rport
From: ;tag=as02127b3c
To: 
Contact: 
CSeq: 102 NOTIFY
User-Agent: Asterisk
Event: refer;id=5
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
Aug  4 11:35:14 NOTICE[7896]: chan_sip.c:3518 copy_header: No field 'Call-ID' present to copy
Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060:
BYE  SIP/2.0
Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7afa5be8;rport
From: ;tag=as02127b3c
To: ;tag=as02127b3c
Contact: 
CSeq: 103 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI> 
<-- SIP read from XX.XX.XX.XX:5060: 
INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKa05d.e51aacd.0
Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-nx2g5ux53lzs;rport=6554
From: "u8" <sip:u8@voip.sysfrog.org>;tag=elvnjnhzkz
To: <sip:85@voip.sysfrog.org;user=phone>;tag=as67271bcd
Call-ID: 3c26702546cd-0z66kq5n4653@snom190
CSeq: 6 INVITE
Max-Forwards: 16
Contact: <sip:u8@YY.YY.YY.YY:6554;line=c371zoiv>
P-Key-Flags: keys="3"
User-Agent: snom190/3.60k
Accept: application/sdp
obelix*CLI> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 365

v=0
o=root 150881502 150881506 IN IP4 YY.YY.YY.YY
s=call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 7034 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c26702546cd-0z66kq5n4653@snom190
Sending to XX.XX.XX.XX : 5060 (non-NAT)
Found peer 'sipsepp'
obelix*CLI>   == Primary D-Channel on span 1 up
  == Primary D-Channel on span 1 up
obelix*CLI>   == Primary D-Channel on span 1 up
obelix*CLI> quit
obelix:~# 
Script done on Thu Aug  4 11:35:35 2005