List Info

Thread: Tracking down the ast_translator_build_path warnings with 1.2.12.1
Tracking down the ast_translator_build_path warnings with 1.2.12.1
user name
 2006-10-12 16:31:00 
Thomas Sandford wrote:

 > 1) Whether they see these warnings or not.

Yes, I do see the warnings.

-rw-r--r--  1 root  wheel  1207279590 Oct 12 21:28 messages

ouch!

 > 2) The version of FreeBSD installed

FreeBSD 6.1 RELEASE

 > 3) Whether they are running the GENERIC, or a patched
kernel (and if
 > patched, in what way).

Systems with GENERIC and systems with kernels from the
'freebsd-update' 
project.

 > 4) Any options used in building the asterisk port

WITHOUT_H323=yes
WITHOUT_ODBC=yes

on several machines, while no extra options on others.

 > 5) The versions of libpri, zaptel and any other
asterisk related 
ports > eg asterisk-addons

asterisk-addons-1.2.3_1
zaptel-1.0_1
libpri-1.2.3

 > 6) Any zaptel (or similar, eg Sangoma) hardware
installed

# kldstat
Id Refs Address    Size     Name
  1    5 0xc0400000 6a2ae0   kernel
  2    1 0xc0aa3000 58554    acpi.ko
  3    2 0xc4ff2000 31000    zaptel.ko
  4    1 0xc5048000 2000     ztdummy.ko

 > 7) The codecs they believe are actually in use.

The problem happens on all of my * boxes

disallow = all
allow = ulaw

on one machine, while another box having:

disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=h263
allow=h263p

 > 8) Anything else that would help determine the
circumstances, eg

IAX2 to IAX2 and SIP to IAX2 calls are not affected.  Looks
like this is 
related to the '183 Session Progress' status sent, as if
there is no 
such status generated during SIP dialog, those messages do
not appear.

During SIP to SIP calls, here is what I see:

     -- Executing Dial("SIP/a225-08931000", 
"SIP/10.21.32.3/78361146|120") in new stack
     -- Called 10.21.32.3/78361146
     -- SIP/10.21.32.3-08963000 is ringing
     -- SIP/10.21.32.3-08963000 is making progress passing
it to 
SIP/a225-08931000
     -- SIP/10.21.32.3-08963000 answered SIP/a225-08931000
     -- Attempting native bridge of SIP/a225-08931000 and 
SIP/10.21.32.3-08963000
Oct 12 21:09:16 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:09:16 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:09:16 WARNING[13815]: translate.c:133
[ * REPEATED 17 MORE TIMES * ]
   == Spawn extension (macro-CallPSTN, s, 8) exited non-zero
on 
'SIP/a225-08931000' in macro 'CallPSTN'
   == Spawn extension (macro-CallPSTN, s, 8) exited non-zero
on 
'SIP/a225-08931000'

This happens on a single device that has both FXO and FXS
ports. Same 
thing happens for calls spanning 2 different devices. For
example this 
is between SPA-3000 and Addpac FXO gateway:

     -- Executing Dial("SIP/10027-09aba000", 
"SIP/XX1552435119X.25X.XX.13X|60") in new stack
     -- Called XX1552435119X.25X.XX.13X
     -- SIP/195.250.74.131-09ac6000 is making progress
passing it to 
SIP/10027-09aba000
Oct 12 21:13:17 WARNING[591]: translate.c:88 powerof:
Powerof 0: No power??
Oct 12 21:13:17 WARNING[591]: translate.c:88 powerof:
Powerof 0: No power??
Oct 12 21:13:17 WARNING[591]: translate.c:133
ast_translator_build_path: 
No translator path from alaw to unknown
[ * REPEATED ABOUT 100 TIMES * ]
     -- SIP/19X.25X.XX.13X-09ac6000 answered
SIP/10027-09aba000
     -- Attempting native bridge of SIP/10027-09aba000 and 
SIP/19X.25X.XX.13X-09ac6000



Here is the detailed sip debug:

*CLI> sip debug
SIP Debugging enabledI>
*CLI>
<-- SIP read from 10.21.32.3:5060:
INVITE sip:223main SIP/2.0
Via: SIP/2.0/UDP
10.21.32.3:5060;rport;branch=z9hG4bKiq72t00lr989uqo6hinx
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>
Contact: <sip:a23310.21.32.3:5060>
Call-ID: 3261510.21.32.3
CSeq: 19691 INVITE
MAX-Forwards: 70
Content-Type: application/sdp
Content-Length: 237

v=0
o=a233 59834399 59834399 IN IP4 10.21.32.3
s=RTP Audio
c=IN IP4 10.21.32.3
t=0 0
m=audio 2088 RTP/AVP 0 4 108 18 8
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:108 FAX/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000

--- (10 headers 11 lines)---
Using INVITE request as basis request - 3261510.21.32.3
Sending to 10.21.32.3 : 5060 (NAT)
Reliably Transmitting (no NAT) to 10.21.32.3:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
10.21.32.3:5060;rport;branch=z9hG4bKiq72t00lr989uqo6hinx;rec
eived=10.21.32.3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as25455b19
Call-ID: 3261510.21.32.3
CSeq: 19691 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact: <sip:22310.21.32.1>
Proxy-Authenticate: Digest algorithm=MD5,
realm="main", nonce="016b42ff"
Content-Length: 0


---
Scheduling destruction of call '3261510.21.32.3' in 15000 ms
Found user 'a233'
*CLI>
<-- SIP read from 10.21.32.3:5060:
INVITE sip:223main SIP/2.0
Via: SIP/2.0/UDP
10.21.32.3:5060;rport;branch=z9hG4bKn5tg12xj2c6354yf49d3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>
Contact: <sip:a23310.21.32.3:5060>
Call-ID: 3261510.21.32.3
CSeq: 19692 INVITE
MAX-Forwards: 70
Proxy-Authorization: Digest 
username="a233",realm="main",nonce="
;016b42ff",response="a802e12e98991e0c3046177818ec7
002",uri="sip:223main",algorithm=MD5
Content-Type: application/sdp
Content-Length: 237

v=0
o=a233 59834707 59834707 IN IP4 10.21.32.3
s=RTP Audio
c=IN IP4 10.21.32.3
t=0 0
m=audio 2088 RTP/AVP 0 4 108 18 8
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:108 FAX/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000

--- (11 headers 11 lines)---
Using INVITE request as basis request - 3261510.21.32.3
Sending to 10.21.32.3 : 5060 (NAT)
Found user 'a233'
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 108
Found RTP audio format 18
Found RTP audio format 8
Peer audio RTP is at port 10.21.32.3:2088
Found description format PCMU
Found description format G723
Found description format FAX
Found description format G729
Found description format PCMA
Capabilities: us - 0x4 (ulaw), peer - audio=0x10d 
(g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer -
0x0 
(nothing), combined - 0x0 (nothing)
Looking for 223 in main (domain main)
list_route: hop: <sip:a23310.21.32.3:5060>
Transmitting (no NAT) to 10.21.32.3:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
10.21.32.3:5060;rport;branch=z9hG4bKn5tg12xj2c6354yf49d3;rec
eived=10.21.32.3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>
Call-ID: 3261510.21.32.3
CSeq: 19692 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact: <sip:22310.21.32.1>
Content-Length: 0


---
     -- Executing Macro("SIP/a233-08931000",
"Call|223") in new stack
     -- Executing Dial("SIP/a233-08931000",
"SIP/a223|60|t") in new stack
We're at 10.21.32.1 port 13314
Adding codec 0x4 (ulaw) to SDP
13 headers, 8 lines
Reliably Transmitting (no NAT) to 10.21.32.4:5060:
INVITE sip:a22310.21.32.4 SIP/2.0
Via: SIP/2.0/UDP
10.21.32.1:5060;branch=z9hG4bK1b36cf9d;rport
From: "233" <sip:23310.21.32.1>;tag=as4a4a99aa
To: <sip:a22310.21.32.4>
Contact: <sip:23310.21.32.1>
Call-ID: 18ec7f2e4e91b1ca3f2f6d1242a1d92610.21.32.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 12 Oct 2006 16:12:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Content-Type: application/sdp
Content-Length: 156

v=0
o=root 13815 13815 IN IP4 10.21.32.1
s=session
c=IN IP4 10.21.32.1
t=0 0
m=audio 13314 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
     -- Called a223
*CLI>
<-- SIP read from 10.21.32.4:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.21.32.1:5060;branch=z9hG4bK1b36cf9d;rport
From: "233"<sip:23310.21.32.1>;tag=as4a4a99aa
To: <sip:a22310.21.32.4>;tag=56057972
Call-ID: 18ec7f2e4e91b1ca3f2f6d1242a1d92610.21.32.1
CSeq: 102 INVITE
Contact: <sip:a22310.21.32.4:5060>
Content-Type: application/sdp
Content-Length: 136

v=0
o=a223 54881576 54881576 IN IP4 10.21.32.4
s=RTP Audio
c=IN IP4 10.21.32.4
t=0 0
m=audio 2072 RTP/AVP 0
a=rtpmap:0 PCMU/8000

--- (9 headers 7 lines)---
Found RTP audio format 0
Peer audio RTP is at port 10.21.32.4:2072
Found description format PCMU
Capabilities: us - 0x4 (ulaw), peer - audio=0x4
(ulaw)/video=0x0 
(nothing), combined - (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer -
0x0 
(nothing), combined - 0x0 (nothing)
     -- SIP/a223-08963000 is ringing
Transmitting (no NAT) to 10.21.32.3:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
10.21.32.3:5060;rport;branch=z9hG4bKn5tg12xj2c6354yf49d3;rec
eived=10.21.32.3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as28fb7413
Call-ID: 3261510.21.32.3
CSeq: 19692 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact: <sip:22310.21.32.1>
Content-Length: 0


---
     -- SIP/a223-08963000 is making progress passing it to
SIP/a233-08931000
We're at 10.21.32.1 port 10960
Adding codec 0x4 (ulaw) to SDP
Transmitting (no NAT) to 10.21.32.3:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
10.21.32.3:5060;rport;branch=z9hG4bKn5tg12xj2c6354yf49d3;rec
eived=10.21.32.3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as28fb7413
Call-ID: 3261510.21.32.3
CSeq: 19692 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact: <sip:22310.21.32.1>
Content-Type: application/sdp
Content-Length: 156

v=0
o=root 13815 13815 IN IP4 10.21.32.1
s=session
c=IN IP4 10.21.32.1
t=0 0
m=audio 10960 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
[snip]
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
Retransmitting #1 (no NAT) to 10.21.32.3:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
10.21.32.3:5060;rport;branch=z9hG4bKiq72t00lr989uqo6hinx;rec
eived=10.21.32.3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as25455b19
Call-ID: 3261510.21.32.3
CSeq: 19691 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact: <sip:22310.21.32.1>
Proxy-Authenticate: Digest algorithm=MD5,
realm="main", nonce="016b42ff"
Content-Length: 0


---
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
[snip]
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
*CLI>
<-- SIP read from 10.21.32.3:5060:
ACK sip:22310.21.32.1 SIP/2.0
Via: SIP/2.0/UDP
10.21.32.3:5060;rport;branch=z9hG4bKn5tg12xj2c6354yf49d3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as25455b19
Call-ID: 3261510.21.32.3
CSeq: 19691 ACK
MAX-Forwards: 70
Content-Length: 0


--- (8 headers 0 lines)---
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:38 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
[snip]
Oct 12 21:12:39 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:39 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:39 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
Retransmitting #2 (no NAT) to 10.21.32.3:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
10.21.32.3:5060;rport;branch=z9hG4bKiq72t00lr989uqo6hinx;rec
eived=10.21.32.3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as25455b19
Call-ID: 3261510.21.32.3
CSeq: 19691 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact: <sip:22310.21.32.1>
Proxy-Authenticate: Digest algorithm=MD5,
realm="main", nonce="016b42ff"
Content-Length: 0


---
Oct 12 21:12:39 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:39 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:39 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
[snip]
Oct 12 21:12:39 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:39 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:39 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
*CLI>
<-- SIP read from 10.21.32.3:5060:
ACK sip:22310.21.32.1 SIP/2.0
Via: SIP/2.0/UDP
10.21.32.3:5060;rport;branch=z9hG4bKn5tg12xj2c6354yf49d3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as25455b19
Call-ID: 3261510.21.32.3
CSeq: 19691 ACK
MAX-Forwards: 70
Content-Length: 0


--- (8 headers 0 lines)---
Oct 12 21:12:39 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:39 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:39 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
[snip]
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
*CLI>
<-- SIP read from 10.21.32.4:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.21.32.1:5060;branch=z9hG4bK1b36cf9d;rport
From: "233"<sip:23310.21.32.1>;tag=as4a4a99aa
To: <sip:a22310.21.32.4>;tag=56057972
Call-ID: 18ec7f2e4e91b1ca3f2f6d1242a1d92610.21.32.1
CSeq: 102 INVITE
Contact: <sip:a22310.21.32.4:5060>
Content-Type: application/sdp
Content-Length: 136

v=0
o=a223 56059032 56059032 IN IP4 10.21.32.4
s=RTP Audio
c=IN IP4 10.21.32.4
t=0 0
m=audio 2072 RTP/AVP 0
a=rtpmap:0 PCMU/8000

--- (9 headers 7 lines)---
Found RTP audio format 0
Peer audio RTP is at port 10.21.32.4:2072
Found description format PCMU
Capabilities: us - 0x4 (ulaw), peer - audio=0x4
(ulaw)/video=0x0 
(nothing), combined - (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer -
0x0 
(nothing), combined - 0x0 (nothing)
list_route: hop: <sip:a22310.21.32.4:5060>
set_destination: Parsing <sip:a22310.21.32.4:5060> for
address/port to 
send to
set_destination: set destination to 10.21.32.4, port 5060
Transmitting (no NAT) to 10.21.32.4:5060:
ACK sip:a22310.21.32.4:5060 SIP/2.0
Via: SIP/2.0/UDP
10.21.32.1:5060;branch=z9hG4bK4f2106ad;rport
From: "233" <sip:23310.21.32.1>;tag=as4a4a99aa
To: <sip:a22310.21.32.4>;tag=56057972
Contact: <sip:23310.21.32.1>
Call-ID: 18ec7f2e4e91b1ca3f2f6d1242a1d92610.21.32.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
     -- SIP/a223-08963000 answered SIP/a233-08931000
We're at 10.21.32.1 port 10960
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 10.21.32.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.21.32.3:5060;rport;branch=z9hG4bKn5tg12xj2c6354yf49d3;rec
eived=10.21.32.3
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as28fb7413
Call-ID: 3261510.21.32.3
CSeq: 19692 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact: <sip:22310.21.32.1>
Content-Type: application/sdp
Content-Length: 156

v=0
o=root 13815 13816 IN IP4 10.21.32.1
s=session
c=IN IP4 10.21.32.1
t=0 0
m=audio 10960 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
[snip]
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
*CLI>
<-- SIP read from 10.21.32.3:5060:
ACK sip:22310.21.32.1 SIP/2.0
Via: SIP/2.0/UDP
10.21.32.3:5060;rport;branch=z9hG4bK6csl5518y6l1vf56l83e
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as28fb7413
Call-ID: 3261510.21.32.3
CSeq: 19692 ACK
MAX-Forwards: 70
Content-Length: 0


--- (8 headers 0 lines)---
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
[snip]
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:88 powerof:
Powerof 0: No 
power??
Oct 12 21:12:40 WARNING[13815]: translate.c:133 
ast_translator_build_path: No translator path from gsm to
unknown
<-- SIP read from 10.21.32.3:5060:
BYE sip:22310.21.32.1 SIP/2.0
Via: SIP/2.0/UDP
10.21.32.3:5060;rport;branch=z9hG4bK138g2591a31xatz375m9
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as25455b19
Call-ID: 3261510.21.32.3
CSeq: 19693 BYE
MAX-Forwards: 70
Content-Length: 0


--- (8 headers 0 lines)---
Sending to 10.21.32.3 : 5060 (NAT)
Transmitting (NAT) to 10.21.32.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
10.21.32.3:5060;branch=z9hG4bK138g2591a31xatz375m9;received=
10.21.32.3;rport=5060
From: <sip:a233main>;tag=434nm5276w38x4y461x4
To: <sip:223main>;tag=as25455b19
Call-ID: 3261510.21.32.3
CSeq: 19693 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact: <sip:22310.21.32.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
Scheduling destruction of call 
'18ec7f2e4e91b1ca3f2f6d1242a1d92610.21.32.1' in 32000 ms
set_destination: Parsing <sip:a22310.21.32.4:5060> for
address/port to 
send to
set_destination: set destination to 10.21.32.4, port 5060
Reliably Transmitting (no NAT) to 10.21.32.4:5060:
BYE sip:a22310.21.32.4:5060 SIP/2.0
Via: SIP/2.0/UDP
10.21.32.1:5060;branch=z9hG4bK2ab89fdc;rport
From: "233" <sip:23310.21.32.1>;tag=as4a4a99aa
To: <sip:a22310.21.32.4>;tag=56057972
Contact: <sip:23310.21.32.1>
Call-ID: 18ec7f2e4e91b1ca3f2f6d1242a1d92610.21.32.1
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
   == Spawn extension (macro-Call, s, 7) exited non-zero on 
'SIP/a233-08931000' in macro 'Call'
   == Spawn extension (macro-Call, s, 7) exited non-zero on 
'SIP/a233-08931000'
*CLI>


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