Asterisk NAT unable to hear voice (outgoing call)


#1

Dear All,

My asterisk (asterisk 1.8.6) is located behind firewall
I have natted the following ports:


in my /etc/sysconfig/iptables

-A INPUT -p tcp -m tcp --dport 5060 -j ACCEPT
-A INPUT -p udp -m udp --dport 5060 -j ACCEPT
-A INPUT -p udp -m udp --dport 4569 -j ACCEPT
-A INPUT -p udp -m udp --dport 10001:20000 -j ACCEPT
-A INPUT -p tcp -m tcp --dport 5038 -j ACCEPT

-A NAT -p tcp -m tcp -d 202.a.b.c/32 -i eth3 --dport 5060 -j DNAT --to-destination 172.a.b.c:5060
-A NAT -p udp -m udp -d 202.a.b.c/32 -i eth3 --dport 5060 -j DNAT --to-destination 172.a.b.c:5060
-A NAT -p udp -m udp -d 202.a.b.c/32 -i eth3 --dport 10001:20000 -j DNAT --to-destination 172.a.b.c:10001-20000
-A NAT -p udp -m udp -d 202.a.b.c/32 -i eth3 --dport 4569 -j DNAT --to-destination 172.a.b.c:4569
-A NAT -p tcp -m tcp -d 202.a.b.c/32 -i eth3 --dport 5038 -j DNAT --to-destination 172.a.b.c:5038

in my /etc/asterisk/rtp.conf
rtpstart=10001
rtpend=20000

and my /etc/asterisk/sip_general_additional.conf is:

callevents=no
jbenable=no
rtptimeout=30
maxexpiry=3600
allowguest=yes
defaultexpiry=120
minexpiry=60
srvlookup=no
registerattempts=0
registertimeout=20
g726nonstandard=no
t38pt_udptl=no
videosupport=no
maxcallbitrate=384
canreinvite=no
notifyhold=yes
notifyringing=yes
checkmwi=10
rtpkeepalive=0
rtpholdtimeout=300
nat=yes
externip=202.a.b.c
localnet=172.a.b.0/255.255.255.0
[root@asterisk asterisk]# netstat -tunap |grep asterisk
tcp        0      0 0.0.0.0:5038                0.0.0.0:*                   LISTEN      2786/asterisk
tcp        0      0 127.0.0.1:5038              127.0.0.1:50243             ESTABLISHED 2786/asterisk
udp        0      0 0.0.0.0:5060                0.0.0.0:*                               2786/asterisk
udp        0      0 0.0.0.0:4569                0.0.0.0:*                               2786/asterisk
[root@asterisk asterisk]#

is there anything else I missed?

why, I am unable to hear the voice of someone I called

please help

thanks & Regards
Winanjaya


#2

Please start the asterisk CLI by typing “asterisk -rvvvvvv” without the quotes. Then try “sip set debug on”. Perform the call and then please copy and paste the results into the reply :smiley:


#3

Forwarding ports, setting nat/localnet/externip is a wasted effort(most of the time). Your asterisk installation is fully capable of calling an external provider without any of these.


#4

Forwarding ports is always necessary in a NAT situation, except if the router peaks inside the SIP packets (and routers that do this tend to get things wrong).

Using net=yes is wrong, because it misunderstands the purpose.

Without externip, etc., the requests will still still reach the service provider, but the responses won’t get back unless they have enabled the equivalent of nat=yes at their end. It is definitely good manners to use these.


#5

You must have not heard about comedia a.k.a symmetric rtp. :laughing:

Check also viewtopic.php?t=80028


#6

below what I got:

I called 90817145779 from ext. 250 …but the person I called unable to hear my voice.
please help.

many thanks in advance

Regards

login as: root
root@172.16.1.214's password:
Last login: Tue Oct 11 06:25:33 2011 from 172.16.1.31

======================================
    Welcome to The FreePBX Distro
======================================

[root@asterisk ~]# asterisk -rvvvvvv
Asterisk 1.8.6.0, Copyright (C) 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
Connected to Asterisk 1.8.6.0 currently running on asterisk (pid = 2801)
Verbosity is at least 6
asterisk*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:110.136.235.237:26913 --->


<------------->

<--- SIP read from UDP:110.136.235.237:26913 --->
INVITE sip:90817145779@202.137.27.230 SIP/2.0
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-a007750c1068c751-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:250@110.136.235.237:26913>
To: "90817145779"<sip:90817145779@202.137.27.230>
From: "250"<sip:250@202.137.27.230>;tag=5748b070
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 473

v=0
o=- 7 2 IN IP4 110.136.235.237
s=CounterPath X-Lite 3.0
c=IN IP4 110.136.235.237
t=0 0
m=audio 27902 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 2 : NIB91Ozo TyE7ODJ8 10.0.128.18 4244
a=alt:2 1 : CytQg3Wn 6aFJS0fF 192.168.1.14 4244
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 17 lines) ---
Sending to 110.136.235.237:26913 (NAT)
Using INVITE request as basis request - NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
Found peer '250' for '250' from 110.136.235.237:26913

<--- Reliably Transmitting (NAT) to 110.136.235.237:26913 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-a007750c1068c751-1--d87543-;received=110.136.235.237;rport=26913
From: "250"<sip:250@202.137.27.230>;tag=5748b070
To: "90817145779"<sip:90817145779@202.137.27.230>;tag=as01f6b114
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 1 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0f6bfff8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.' in 10880 ms (Method: INVITE)

<--- SIP read from UDP:110.136.235.237:26913 --->
ACK sip:90817145779@202.137.27.230 SIP/2.0
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-a007750c1068c751-1--d87543-;rport
To: "90817145779"<sip:90817145779@202.137.27.230>;tag=as01f6b114
From: "250"<sip:250@202.137.27.230>;tag=5748b070
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:110.136.235.237:26913 --->
INVITE sip:90817145779@202.137.27.230 SIP/2.0
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-2c2c7e7a04755f4a-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:250@110.136.235.237:26913>
To: "90817145779"<sip:90817145779@202.137.27.230>
From: "250"<sip:250@202.137.27.230>;tag=5748b070
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Authorization: Digest username="250",realm="asterisk",nonce="0f6bfff8",uri="sip:90817145779@202.137.27.230",response="442f9c0cc31d5592975caf3b1d869351",algorithm=MD5
Content-Length: 473

v=0
o=- 7 2 IN IP4 110.136.235.237
s=CounterPath X-Lite 3.0
c=IN IP4 110.136.235.237
t=0 0
m=audio 27902 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 2 : NIB91Ozo TyE7ODJ8 10.0.128.18 4244
a=alt:2 1 : CytQg3Wn 6aFJS0fF 192.168.1.14 4244
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (13 headers 17 lines) ---
Sending to 110.136.235.237:26913 (NAT)
Using INVITE request as basis request - NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
Found peer '250' for '250' from 110.136.235.237:26913
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x20000040c (ulaw|alaw|speex16|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 110.136.235.237:27902
Looking for 90817145779 in from-internal (domain 202.137.27.230)
list_route: hop: <sip:250@110.136.235.237:26913>

<--- Transmitting (NAT) to 110.136.235.237:26913 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-2c2c7e7a04755f4a-1--d87543-;received=110.136.235.237;rport=26913
From: "250"<sip:250@202.137.27.230>;tag=5748b070
To: "90817145779"<sip:90817145779@202.137.27.230>
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:90817145779@202.137.27.230:5060>
Content-Length: 0


<------------>
    -- Executing [90817145779@from-internal:1] Macro("SIP/250-000000c1", "user-callerid,LIMIT,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/250-000000c1", "AMPUSER=250") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/250-000000c1", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/250-000000c1", "1?Set(REALCALLERIDNUM=250)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/250-000000c1", "AMPUSER=250") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/250-000000c1", "AMPUSERCIDNAME=250") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/250-000000c1", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/250-000000c1", "AMPUSERCID=250") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/250-000000c1", "CALLERID(all)="250" <250>") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/250-000000c1", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/250-000000c1", "1?Set(GROUP(concurrency_limit)=250)") in new stack
    -- Executing [s@macro-user-callerid:11] GosubIf("SIP/250-000000c1", "7?sub-ccss,s,1(from-internal,90817145779)") in new stack
    -- Executing [s@sub-ccss:1] ExecIf("SIP/250-000000c1", "0?Return()") in new stack
    -- Executing [s@sub-ccss:2] Set("SIP/250-000000c1", "CCSS_SETUP=TRUE") in new stack
    -- Executing [s@sub-ccss:3] GosubIf("SIP/250-000000c1", "0?monitor_config,1(from-internal,90817145779):monitor_default,1(from-internal,90817145779)") in new stack
    -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/250-000000c1", "0?is_exten") in new stack
    -- Executing [monitor_default@sub-ccss:2] StackPop("SIP/250-000000c1", "") in new stack
    -- Executing [monitor_default@sub-ccss:3] Return("SIP/250-000000c1", "FALSE") in new stack
    -- Executing [s@macro-user-callerid:12] ExecIf("SIP/250-000000c1", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/250-000000c1", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,26)
    -- Executing [s@macro-user-callerid:26] Set("SIP/250-000000c1", "CALLERID(number)=250") in new stack
    -- Executing [s@macro-user-callerid:27] Set("SIP/250-000000c1", "CALLERID(name)=250") in new stack
    -- Executing [s@macro-user-callerid:28] Set("SIP/250-000000c1", "CHANNEL(language)=en") in new stack
    -- Executing [90817145779@from-internal:2] Set("SIP/250-000000c1", "MOHCLASS=default") in new stack
    -- Executing [90817145779@from-internal:3] Set("SIP/250-000000c1", "_NODEST=") in new stack
    -- Executing [90817145779@from-internal:4] Macro("SIP/250-000000c1", "record-enable,250,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/250-000000c1", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/250-000000c1", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/250-000000c1", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,14)
    -- Executing [s@macro-record-enable:14] GotoIf("SIP/250-000000c1", "0?IN") in new stack
    -- Executing [s@macro-record-enable:15] ExecIf("SIP/250-000000c1", "1?MacroExit()") in new stack
    -- Executing [90817145779@from-internal:5] Macro("SIP/250-000000c1", "dialout-trunk,1,0817145779,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/250-000000c1", "DIAL_TRUNK=1") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/250-000000c1", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/250-000000c1", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/250-000000c1", "DIAL_NUMBER=0817145779") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/250-000000c1", "DIAL_TRUNK_OPTIONS=trwW") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/250-000000c1", "OUTBOUND_GROUP=OUT_1") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/250-000000c1", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/250-000000c1", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/250-000000c1", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/250-000000c1", "outbound-callerid,1") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/250-000000c1", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/250-000000c1", "0?Set(REALCALLERIDNUM=250)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/250-000000c1", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/250-000000c1", "USEROUTCID=250") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/250-000000c1", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/250-000000c1", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/250-000000c1", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/250-000000c1", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/250-000000c1", "1?Set(CALLERID(all)=250)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/250-000000c1", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/250-000000c1", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/250-000000c1", "0?sub-flp-1,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/250-000000c1", "OUTNUM=0817145779") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/250-000000c1", "custom=DAHDI/1") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/250-000000c1", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/250-000000c1", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/250-000000c1", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/250-000000c1", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/250-000000c1", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/250-000000c1", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:20] Dial("SIP/250-000000c1", "DAHDI/1/0817145779,300,") in new stack
    -- Called DAHDI/1/0817145779
    -- DAHDI/1-1 answered SIP/250-000000c1
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 110.136.235.237:26913 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-2c2c7e7a04755f4a-1--d87543-;received=110.136.235.237;rport=26913
From: "250"<sip:250@202.137.27.230>;tag=5748b070
To: "90817145779"<sip:90817145779@202.137.27.230>;tag=as7e32c019
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:90817145779@202.137.27.230:5060>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1621999769 1621999769 IN IP4 202.137.27.230
s=Asterisk PBX 1.8.6.0
c=IN IP4 202.137.27.230
t=0 0
m=audio 10528 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (NAT) to 110.136.235.237:26913:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-2c2c7e7a04755f4a-1--d87543-;received=110.136.235.237;rport=26913
From: "250"<sip:250@202.137.27.230>;tag=5748b070
To: "90817145779"<sip:90817145779@202.137.27.230>;tag=as7e32c019
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 2 INVITE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:90817145779@202.137.27.230:5060>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1621999769 1621999769 IN IP4 202.137.27.230
s=Asterisk PBX 1.8.6.0
c=IN IP4 202.137.27.230
t=0 0
m=audio 10528 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:110.136.235.237:26913 --->
ACK sip:90817145779@202.137.27.230:5060 SIP/2.0
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-4544197e76781b05-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:250@110.136.235.237:26913>
To: "90817145779"<sip:90817145779@202.137.27.230>;tag=as7e32c019
From: "250"<sip:250@202.137.27.230>;tag=5748b070
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 2 ACK
User-Agent: X-Lite release 1011s stamp 41150
Authorization: Digest username="250",realm="asterisk",nonce="0f6bfff8",uri="sip:90817145779@202.137.27.230",response="442f9c0cc31d5592975caf3b1d869351",algorithm=MD5
Content-Length: 0

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

<--- SIP read from UDP:110.136.235.237:26913 --->
ACK sip:90817145779@202.137.27.230:5060 SIP/2.0
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-4544197e76781b05-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:250@110.136.235.237:26913>
To: "90817145779"<sip:90817145779@202.137.27.230>;tag=as7e32c019
From: "250"<sip:250@202.137.27.230>;tag=5748b070
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 2 ACK
User-Agent: X-Lite release 1011s stamp 41150
Authorization: Digest username="250",realm="asterisk",nonce="0f6bfff8",uri="sip:90817145779@202.137.27.230",response="442f9c0cc31d5592975caf3b1d869351",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (NAT) to 61.247.36.147:49853:
OPTIONS sip:252@172.16.250.124:5062 SIP/2.0
Via: SIP/2.0/UDP 202.137.27.230:5060;branch=z9hG4bK3309102f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@202.137.27.230>;tag=as31db32b4
To: <sip:252@172.16.250.124:5062>
Contact: <sip:Unknown@202.137.27.230:5060>
Call-ID: 5d2cc46b504c22d8344304ff113b6f64@202.137.27.230:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.6.0)
Date: Tue, 11 Oct 2011 00:07:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:61.247.36.147:49853 --->


<------------->

<--- SIP read from UDP:61.247.36.147:49853 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.137.27.230:5060;branch=z9hG4bK3309102f;rport
From: "Unknown" <sip:Unknown@202.137.27.230>;tag=as31db32b4
To: <sip:252@172.16.250.124:5062>;tag=1642307063
Call-ID: 5d2cc46b504c22d8344304ff113b6f64@202.137.27.230:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T20P 9.60.0.140
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '5d2cc46b504c22d8344304ff113b6f64@202.137.27.230:5060' Method: OPTIONS

<--- SIP read from UDP:110.136.235.237:26913 --->
BYE sip:90817145779@202.137.27.230:5060 SIP/2.0
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-7c612b6ff220f623-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:250@110.136.235.237:26913>
To: "90817145779"<sip:90817145779@202.137.27.230>;tag=as7e32c019
From: "250"<sip:250@202.137.27.230>;tag=5748b070
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 3 BYE
User-Agent: X-Lite release 1011s stamp 41150
Authorization: Digest username="250",realm="asterisk",nonce="0f6bfff8",uri="sip:90817145779@202.137.27.230:5060",response="290ec8b84fa605927efb2af3dc5a0898",algorithm=MD5
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 110.136.235.237:26913 (NAT)
Scheduling destruction of SIP dialog 'NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.' in 10880 ms (Method: BYE)

<--- Transmitting (NAT) to 110.136.235.237:26913 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 110.136.235.237:26913;branch=z9hG4bK-d87543-7c612b6ff220f623-1--d87543-;received=110.136.235.237;rport=26913
From: "250"<sip:250@202.137.27.230>;tag=5748b070
To: "90817145779"<sip:90817145779@202.137.27.230>;tag=as7e32c019
Call-ID: NWUwMGUxYTE0MGQ5NDRkZTk4ODk5OGNmYTg4MTZhNzk.
CSeq: 3 BYE
Server: FPBX-2.9.0(1.8.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/250-000000c1", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/250-000000c1", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] Hangup("SIP/250-000000c1", "") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/250-000000c1' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/250-000000c1'
    -- Hanging up on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
  == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/250-000000c1' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 90817145779, 5) exited non-zero on 'SIP/250-000000c1'

<--- SIP read from UDP:110.136.235.237:26913 --->


<------------->
asterisk*CLI> sip set debug off
SIP Debugging Disabled