Problem making in/outbound with 2.6.0.5
Submitted by cyberglobe on Thu, 04/03/2008 - 2:34pm.
Hello,
Using Trixbox 2.6.0.5, I have copied the exact trunk settings from another system and I can not seem to be able to make the connections work.
My current config is as follows:
eth0: Internal Lan 192.168.222.1 255.255.255.0
eth1: DMZ for DSL PPPoE connection
ppp0: PPPoE Connection with a Static IP address x.x.x.129
default gateway is set to ppp0 with 0.0.0.0
I have the following trunks setup:
Outgoing provider
allow=ulaw
canreinvite=no
disallow=all
dtmf=rfc2833
dtmfmode=rfc2833
host=x.x.x.138
insecure=very
nat=no
qualify=10000
sendrpid=yes
trustrpid=yes
type=peer
Incoming provider-in
allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmf=rfc2833
dtmfmode=rfc2833
host=x.x.x.138
insecure=very
nat=no
qualify=10000
sendrpid=yes
trustrpid=yes
type=peer
Problem I am having is that when I try to dial out, it does not even try to connect to the remote server. Yet the Ping and Trace shows they are active.
I will have the logs on here shortly.
Here are the Logs of the connection.
Thu, 04/03/2008 - 3:29pm
Asterisk 1.4.18-3 RPM by vc-rpms@voipconsulting.nl, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
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.
=========================================================================
Connected to Asterisk 1.4.18-3 RPM by vc-rpms@voipconsulting.nl currently running on PBX (pid = 2671)
Scheduling destruction of SIP dialog '03b96dc3603c70374bc709ae30d1f176@192.168.1.252' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.102:5060:
NOTIFY sip:1101@192.168.1.102:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK06c20d08;rport
From: "Unknown"
To:
Contact:
Call-ID: 03b96dc3603c70374bc709ae30d1f176@192.168.1.252
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.252
Voice-Message: 0/0 (0/0)
---
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK06c20d08;rport=5060;received=192.168.1.252
From: "Unknown"
To:
Call-ID: 03b96dc3603c70374bc709ae30d1f176@192.168.1.252
CSeq: 102 NOTIFY
Contact:
Server: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '03b96dc3603c70374bc709ae30d1f176@192.168.1.252' Method: NOTIFY
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Connect attempt from '127.0.0.1' unable to authenticate
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
INVITE sip:5145555555@192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd439d84ba74c53c5d
Max-Forwards: 70
From: "Phone 1"
To: "5145555555"
Call-ID: 3c773c73ed2390ef
CSeq: 31610 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Phone 1
Supported: timer, 100rel, replaces
User-Agent: Aastra 51i/2.2.0.166
Content-Type: application/sdp
Content-Length: 596
v=0
o=MxSIP 0 0 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 25 lines) ---
Sending to 192.168.1.102 : 5060 (no NAT)
Using INVITE request as basis request - 3c773c73ed2390ef
<--- Reliably Transmitting (NAT) to 192.168.1.102:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd439d84ba74c53c5d;received=192.168.1.102
From: "Phone 1"
To: "5145555555"
Call-ID: 3c773c73ed2390ef
CSeq: 31610 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66d55813"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '3c773c73ed2390ef' in 32000 ms (Method: INVITE)
Found user '1101'
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
ACK sip:5145555555@192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd439d84ba74c53c5d
Max-Forwards: 70
From: "Phone 1"
To: "5145555555"
Call-ID: 3c773c73ed2390ef
CSeq: 31610 ACK
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
INVITE sip:5145555555@192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a
Proxy-Authorization: Digest username="1101",realm="asterisk",nonce="66d55813",uri="sip:5145555555@192.168.1.252:5060",response="99c68deebd8213900efede80d033f4fd",algorithm=MD5
Max-Forwards: 70
From: "Phone 1"
To: "5145555555"
Call-ID: 3c773c73ed2390ef
CSeq: 31611 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Phone 1
Supported: timer, 100rel, replaces
User-Agent: Aastra 51i/2.2.0.166
Content-Type: application/sdp
Content-Length: 596
v=0
o=MxSIP 0 0 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 25 lines) ---
Sending to 192.168.1.102 : 5060 (NAT)
Using INVITE request as basis request - 3c773c73ed2390ef
Found user '1101'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 106
Found RTP audio format 107
Found RTP audio format 113
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 115
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.102:3000
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found unknown media description format BV16 for ID 106
Found unknown media description format BV32 for ID 107
Found audio description format L16 for ID 113
Found audio description format PCMU for ID 110
Found audio description format PCMA for ID 111
Found audio description format L16 for ID 112
Found unknown media description format G726-16 for ID 98
Found unknown media description format G726-24 for ID 97
Found audio description format G726-32 for ID 115
Found unknown media description format G726-40 for ID 96
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x194c (ulaw|alaw|g726|slin|g729|g722)/video=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 192.168.1.102:3000
Looking for 5145555555 in from-internal (domain 192.168.1.252)
list_route: hop:
<--- Transmitting (NAT) to 192.168.1.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a;received=192.168.1.102
From: "Phone 1"
To: "5145555555"
Call-ID: 3c773c73ed2390ef
CSeq: 31611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
-- Executing [5145555555@from-internal:1] Macro("SIP/1101-0991f8e8", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/1101-0991f8e8", "user-callerid: device 1101") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/1101-0991f8e8", "AMPUSER=1101") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/1101-0991f8e8", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/1101-0991f8e8", "1|Set|REALCALLERIDNUM=1101") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/1101-0991f8e8", "REALCALLERIDNUM is 1101") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/1101-0991f8e8", "AMPUSER=1101") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1101-0991f8e8", "AMPUSERCIDNAME=Phone 1") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/1101-0991f8e8", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/1101-0991f8e8", "AMPUSERCID=1101") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/1101-0991f8e8", "CALLERID(all)="Phone 1" <1101>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/1101-0991f8e8", "REALCALLERIDNUM=1101") in new stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/1101-0991f8e8", "1|Set|CHANNEL(language)=en") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/1101-0991f8e8", "TTL: ARG1: SKIPTTL") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/1101-0991f8e8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/1101-0991f8e8", "Using CallerID "Phone 1" <1101>") in new stack
-- Executing [5145555555@from-internal:2] Set("SIP/1101-0991f8e8", "_NODEST=") in new stack
-- Executing [5145555555@from-internal:3] Macro("SIP/1101-0991f8e8", "record-enable|1101|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1101-0991f8e8", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/1101-0991f8e8", "recordingcheck|20080403-182033|1207261233.6") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080403-182033|1207261233.6: Outbound recording enabled.
recordingcheck|20080403-182033|1207261233.6: CALLFILENAME=OUT1101-20080403-182033-1207261233.6
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:999] MixMonitor("SIP/1101-0991f8e8", "OUT1101-20080403-182033-1207261233.6.wav") in new stack
-- Executing [5145555555@from-internal:4] Macro("SIP/1101-0991f8e8", "dialout-trunk|3|5145555555||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1101-0991f8e8", "DIAL_TRUNK=3") in new stack
-- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/1101-0991f8e8", "0|Authenticate|") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1101-0991f8e8", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/1101-0991f8e8", "DIAL_NUMBER=5145555555") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/1101-0991f8e8", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/1101-0991f8e8", "GROUP()=OUT_3") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1101-0991f8e8", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1101-0991f8e8", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/1101-0991f8e8", "DIAL_TRUNK_OPTIONS=") in new stack
== Begin MixMonitor Recording SIP/1101-0991f8e8
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/1101-0991f8e8", "outbound-callerid|3") in new stack
-- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/1101-0991f8e8", "1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/1101-0991f8e8", "REALCALLERIDNUM is 1101") in new stack
-- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/1101-0991f8e8", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [s@macro-outbound-callerid:9] Set("SIP/1101-0991f8e8", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:10] Set("SIP/1101-0991f8e8", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:11] Set("SIP/1101-0991f8e8", "TRUNKOUTCID="MYDOTCOM.CA" <5145555555>") in new stack
-- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/1101-0991f8e8", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/1101-0991f8e8", "0?usercid") in new stack
-- Executing [s@macro-outbound-callerid:17] Set("SIP/1101-0991f8e8", "CALLERID(all)=MYDOTCOM.CA <5145555555>") in new stack
-- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/1101-0991f8e8", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [s@macro-outbound-callerid:22] NoOp("SIP/1101-0991f8e8", "CallerID set to "MYDOTCOM.CA" <5145555555>") in new stack
-- Executing [s@macro-dialout-trunk:12] AGI("SIP/1101-0991f8e8", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern .
== fixlocalprefix: Dialpattern . matched. 5145555555 -> 5145555555
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/1101-0991f8e8", "OUTNUM=5145555555") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/1101-0991f8e8", "custom=SIP/cgctemp") in new stack
-- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/1101-0991f8e8", "1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/1101-0991f8e8", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1101-0991f8e8", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/1101-0991f8e8", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:20] Dial("SIP/1101-0991f8e8", "SIP/cgctemp/5145555555|300|") in new stack
-- Couldn't call cgctemp/5145555555
Scheduling destruction of SIP dialog '2fb84a776a50d49000cfced0146a25ae@x.x.x.129' in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/1101-0991f8e8", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/1101-0991f8e8", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/1101-0991f8e8", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
-- Executing [5145555555@from-internal:5] Macro("SIP/1101-0991f8e8", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/1101-0991f8e8", "all-circuits-busy-now|noanswer") in new stack
Audio is at 192.168.1.252 port 15522
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
PBX*CLI>
<--- Transmitting (NAT) to 192.168.1.102:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a;received=192.168.1.102
From: "Phone 1"
To: "5145555555"
Call-ID: 3c773c73ed2390ef
CSeq: 31611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 2671 2671 IN IP4 192.168.1.252
s=session
c=IN IP4 192.168.1.252
t=0 0
m=audio 15522 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
--
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- Executing [s@macro-outisbusy:2] Playback("SIP/1101-0991f8e8", "pls-try-call-later|noanswer") in new stack
--
== Manager 'admin' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
-- Executing [s@macro-outisbusy:3] Macro("SIP/1101-0991f8e8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1101-0991f8e8", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/1101-0991f8e8", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/1101-0991f8e8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/1101-0991f8e8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/1101-0991f8e8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/1101-0991f8e8", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1101-0991f8e8'
Scheduling destruction of SIP dialog '3c773c73ed2390ef' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.1.102:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a;received=192.168.1.102
From: "Phone 1"
To: "5145555555"
Call-ID: 3c773c73ed2390ef
CSeq: 31611 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
== End MixMonitor Recording SIP/1101-0991f8e8
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
ACK sip:5145555555@192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8bd3da669e11de75a
Max-Forwards: 70
From: "Phone 1"
To: "5145555555"
Call-ID: 3c773c73ed2390ef
CSeq: 31611 ACK
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
== Connect attempt from '127.0.0.1' unable to authenticate
== Connect attempt from '127.0.0.1' unable to authenticate
Really destroying SIP dialog '2fb84a776a50d49000cfced0146a25ae@x.x.x.129' Method: INVITE
PBX*CLI>
<--- SIP read from x.x.x.138:5060 --->
OPTIONS sip:x.x.x.129 SIP/2.0
Via: SIP/2.0/UDP x.x.x.138:5060;branch=z9hG4bK14b334d2;rport
From: "asterisk"
To:
Contact:
Call-ID: 7cabe346625e9a8b7f0a239c1b2ec306@x.x.x.138
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:19:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain x.x.x.129)
<--- Transmitting (no NAT) to x.x.x.138:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.138:5060;branch=z9hG4bK14b334d2;received=x.x.x.138;rport=5060
From: "asterisk"
To:
Call-ID: 7cabe346625e9a8b7f0a239c1b2ec306@x.x.x.138
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7cabe346625e9a8b7f0a239c1b2ec306@x.x.x.138' in 32000 ms (Method: OPTIONS)
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
INVITE sip:1102@192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd70f66bdc2dbab600
Max-Forwards: 70
From: "Phone 1"
To: "1102"
Call-ID: 332cc09cdeab7e76
CSeq: 24618 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Phone 1
Supported: timer, 100rel, replaces
User-Agent: Aastra 51i/2.2.0.166
Content-Type: application/sdp
Content-Length: 596
v=0
o=MxSIP 0 0 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 25 lines) ---
Sending to 192.168.1.102 : 5060 (no NAT)
Using INVITE request as basis request - 332cc09cdeab7e76
<--- Reliably Transmitting (NAT) to 192.168.1.102:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd70f66bdc2dbab600;received=192.168.1.102
From: "Phone 1"
To: "1102"
Call-ID: 332cc09cdeab7e76
CSeq: 24618 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2c2d7e6b"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '332cc09cdeab7e76' in 32000 ms (Method: INVITE)
Found user '1101'
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
ACK sip:1102@192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKd70f66bdc2dbab600
Max-Forwards: 70
From: "Phone 1"
To: "1102"
Call-ID: 332cc09cdeab7e76
CSeq: 24618 ACK
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
INVITE sip:1102@192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK6a67d932b42493281
Proxy-Authorization: Digest username="1101",realm="asterisk",nonce="2c2d7e6b",uri="sip:1102@192.168.1.252:5060",response="d843e7060f3057c82d5853cd2ffb43f6",algorithm=MD5
Max-Forwards: 70
From: "Phone 1"
To: "1102"
Call-ID: 332cc09cdeab7e76
CSeq: 24619 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Contact: Phone 1
Supported: timer, 100rel, replaces
User-Agent: Aastra 51i/2.2.0.166
Content-Type: application/sdp
Content-Length: 596
v=0
o=MxSIP 0 0 IN IP4 192.168.1.102
s=SIP Call
c=IN IP4 192.168.1.102
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 25 lines) ---
Sending to 192.168.1.102 : 5060 (NAT)
Using INVITE request as basis request - 332cc09cdeab7e76
Found user '1101'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 106
Found RTP audio format 107
Found RTP audio format 113
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 115
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.102:3000
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found unknown media description format BV16 for ID 106
Found unknown media description format BV32 for ID 107
Found audio description format L16 for ID 113
Found audio description format PCMU for ID 110
Found audio description format PCMA for ID 111
Found audio description format L16 for ID 112
Found unknown media description format G726-16 for ID 98
Found unknown media description format G726-24 for ID 97
Found audio description format G726-32 for ID 115
Found unknown media description format G726-40 for ID 96
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x194c (ulaw|alaw|g726|slin|g729|g722)/video=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 192.168.1.102:3000
Looking for 1102 in from-internal (domain 192.168.1.252)
list_route: hop:
<--- Transmitting (NAT) to 192.168.1.102:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK6a67d932b42493281;received=192.168.1.102
From: "Phone 1"
To: "1102"
Call-ID: 332cc09cdeab7e76
CSeq: 24619 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
-- Executing [1102@from-internal:1] Macro("SIP/1101-09935dc8", "exten-vm|novm|1102") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/1101-09935dc8", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("SIP/1101-09935dc8", "user-callerid: device 1101") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/1101-09935dc8", "AMPUSER=1101") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/1101-09935dc8", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/1101-09935dc8", "1|Set|REALCALLERIDNUM=1101") in new stack
-- Executing [s@macro-user-callerid:5] NoOp("SIP/1101-09935dc8", "REALCALLERIDNUM is 1101") in new stack
-- Executing [s@macro-user-callerid:6] Set("SIP/1101-09935dc8", "AMPUSER=1101") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1101-09935dc8", "AMPUSERCIDNAME=Phone 1") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/1101-09935dc8", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/1101-09935dc8", "AMPUSERCID=1101") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/1101-09935dc8", "CALLERID(all)="Phone 1" <1101>") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/1101-09935dc8", "REALCALLERIDNUM=1101") in new stack
-- Executing [s@macro-user-callerid:12] ExecIf("SIP/1101-09935dc8", "1|Set|CHANNEL(language)=en") in new stack
-- Executing [s@macro-user-callerid:13] NoOp("SIP/1101-09935dc8", "TTL: ARG1: novm") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/1101-09935dc8", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("SIP/1101-09935dc8", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("SIP/1101-09935dc8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("SIP/1101-09935dc8", "Using CallerID "Phone 1" <1101>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/1101-09935dc8", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/1101-09935dc8", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/1101-09935dc8", "EXTTOCALL=1102") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/1101-09935dc8", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/1101-09935dc8", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/1101-09935dc8", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/1101-09935dc8", "record-enable|1102|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/1101-09935dc8", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/1101-09935dc8", "recordingcheck|20080403-182044|1207261244.8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
== Manager 'admin' logged off from 127.0.0.1
recordingcheck|20080403-182044|1207261244.8: Inbound recording enabled.
recordingcheck|20080403-182044|1207261244.8: CALLFILENAME=20080403-182044-1207261244.8
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:999] MixMonitor("SIP/1101-09935dc8", "20080403-182044-1207261244.8.wav") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/1101-09935dc8", "dial||tr|1102") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/1101-09935dc8", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/1101-09935dc8", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
== Begin MixMonitor Recording SIP/1101-09935dc8
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Phone 1' number is '1101'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 1102 to extension map
-- dialparties.agi: Extension 1102 cf is disabled
-- dialparties.agi: Extension 1102 do not disturb is disabled
> dialparties.agi: extnum 1102 has: cw: 0; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 1102 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 1102
-- dialparties.agi: dbset CALLTRACE/1102 to 1101
-- dialparties.agi: Filtered ARG3: 1102
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/1101-09935dc8", "SIP/1102||tr") in new stack
Audio is at 192.168.1.252 port 15432
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.107:5060:
INVITE sip:1102@192.168.1.107:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK18c90cf4;rport
From: "Phone 1"
To:
Contact:
Call-ID: 37c3dc34575f04ee55a5e475123636f4@192.168.1.252
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 2671 2671 IN IP4 192.168.1.252
s=session
c=IN IP4 192.168.1.252
t=0 0
m=audio 15432 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 1102
PBX*CLI>
<--- Transmitting (NAT) to 192.168.1.102:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK6a67d932b42493281;received=192.168.1.102
From: "Phone 1"
To: "1102"
Call-ID: 332cc09cdeab7e76
CSeq: 24619 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
PBX*CLI>
<--- SIP read from 192.168.1.107:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK18c90cf4;rport=5060;received=192.168.1.252
From: "Phone 1"
To:
Call-ID: 37c3dc34575f04ee55a5e475123636f4@192.168.1.252
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info:
Contact: Phone 2
Server: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
-- SIP/1102-09921a30 is ringing
== Connect attempt from '127.0.0.1' unable to authenticate
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
PBX*CLI>
<--- SIP read from 192.168.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK18c90cf4;rport=5060;received=192.168.1.252
From: "Phone 1"
To:
Call-ID: 37c3dc34575f04ee55a5e475123636f4@192.168.1.252
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info:
Contact: Phone 2
Server: Aastra 51i/2.2.0.166
Supported: timer, replaces
Content-Type: application/sdp
Content-Length: 259
v=0
o=MxSIP 0 0 IN IP4 192.168.1.107
s=SIP Call
c=IN IP4 192.168.1.107
t=0 0
m=audio 3000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.107:3000
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=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 192.168.1.107:3000
list_route: hop:
set_destination: Parsing
set_destination: set destination to 192.168.1.107, port 5060
Transmitting (NAT) to 192.168.1.107:5060:
ACK sip:1102@192.168.1.107:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK4b5d6115;rport
From: "Phone 1"
To:
Contact:
Call-ID: 37c3dc34575f04ee55a5e475123636f4@192.168.1.252
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/1102-09921a30 answered SIP/1101-09935dc8
Audio is at 192.168.1.252 port 13104
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.1.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK6a67d932b42493281;received=192.168.1.102
From: "Phone 1"
To: "1102"
Call-ID: 332cc09cdeab7e76
CSeq: 24619 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 264
v=0
o=root 2671 2671 IN IP4 192.168.1.252
s=session
c=IN IP4 192.168.1.252
t=0 0
m=audio 13104 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
ACK sip:1102@192.168.1.252 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK8aeb3c338b312514d
Proxy-Authorization: Digest username="1101",realm="asterisk",nonce="2c2d7e6b",uri="sip:1102@192.168.1.252",response="c7826eea5ef168f9eb61726fa20989ae",algorithm=MD5
Max-Forwards: 70
From: "Phone 1"
To: "1102"
Call-ID: 332cc09cdeab7e76
CSeq: 24619 ACK
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
== Connect attempt from '127.0.0.1' unable to authenticate
Reliably Transmitting (no NAT) to x.x.x.138:5060:
OPTIONS sip:x.x.x.138 SIP/2.0
Via: SIP/2.0/UDP x.x.x.129:5060;branch=z9hG4bK05c1909a;rport
From: "Unknown"
To:
Contact:
Call-ID: 67eb8ad619097be74f242366117ceeea@x.x.x.129
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
PBX*CLI>
<--- SIP read from x.x.x.138:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP x.x.x.129:5060;branch=z9hG4bK05c1909a;received=x.x.x.129;rport=5060
From: "Unknown"
To:
Call-ID: 67eb8ad619097be74f242366117ceeea@x.x.x.129
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '67eb8ad619097be74f242366117ceeea@x.x.x.129' Method: OPTIONS
PBX*CLI>
<--- SIP read from 192.168.1.108:5060 --->
REGISTER sip:192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKc75ec350d47c6dbc9
Max-Forwards: 70
From:
To:
Call-ID: de82517b006e9bbe
CSeq: 18270 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username="1103",realm="asterisk",nonce="7d1b5c9d",uri="sip:192.168.1.252:5060",response="4c14212ee901a561aafea7a2374b3cc3",algorithm=MD5
Contact: Phone 3
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.108 : 5060 (no NAT)
<--- Transmitting (NAT) to 192.168.1.108:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKc75ec350d47c6dbc9;received=192.168.1.108
From:
To:
Call-ID: de82517b006e9bbe
CSeq: 18270 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 192.168.1.108:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKc75ec350d47c6dbc9;received=192.168.1.108
From:
To:
Call-ID: de82517b006e9bbe
CSeq: 18270 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="44824fc1"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'de82517b006e9bbe' in 32000 ms (Method: REGISTER)
PBX*CLI>
<--- SIP read from 192.168.1.108:5060 --->
REGISTER sip:192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKa2eaa7146bf8e7fe5
Max-Forwards: 70
From:
To:
Call-ID: de82517b006e9bbe
CSeq: 18271 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username="1103",realm="asterisk",nonce="44824fc1",uri="sip:192.168.1.252:5060",response="89ca3c7e669277db0d9cab507f3aa0f3",algorithm=MD5
Contact: Phone 3
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.108 : 5060 (NAT)
<--- Transmitting (NAT) to 192.168.1.108:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKa2eaa7146bf8e7fe5;received=192.168.1.108
From:
To:
Call-ID: de82517b006e9bbe
CSeq: 18271 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
PBX*CLI>
<--- Transmitting (NAT) to 192.168.1.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bKa2eaa7146bf8e7fe5;received=192.168.1.108
From:
To:
Call-ID: de82517b006e9bbe
CSeq: 18271 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact:
Date: Thu, 03 Apr 2008 22:20:49 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'de82517b006e9bbe' in 32000 ms (Method: REGISTER)
Reliably Transmitting (NAT) to 192.168.1.108:5060:
OPTIONS sip:1103@192.168.1.108:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK1e2eecac;rport
From: "Unknown"
To:
Contact:
Call-ID: 0b1a752a30193c2345e5add07123d59a@192.168.1.252
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
PBX*CLI>
<--- SIP read from 192.168.1.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK1e2eecac;rport=5060;received=192.168.1.252
From: "Unknown"
To:
Call-ID: 0b1a752a30193c2345e5add07123d59a@192.168.1.252
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '0b1a752a30193c2345e5add07123d59a@192.168.1.252' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.107:5060:
OPTIONS sip:1102@192.168.1.107:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK61584f7b;rport
From: "Unknown"
To:
Contact:
Call-ID: 278e74cf3430f97c07401df45e64c572@192.168.1.252
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
PBX*CLI>
<--- SIP read from 192.168.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK61584f7b;rport=5060;received=192.168.1.252
From: "Unknown"
To:
Call-ID: 278e74cf3430f97c07401df45e64c572@192.168.1.252
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '278e74cf3430f97c07401df45e64c572@192.168.1.252' Method: OPTIONS
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
Reliably Transmitting (NAT) to 192.168.1.102:5060:
OPTIONS sip:1101@192.168.1.102:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK7a1d9178;rport
From: "Unknown"
To:
Contact:
Call-ID: 6f8b4c693ff86d1c7cfa75fe5c59e51f@192.168.1.252
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK7a1d9178;rport=5060;received=192.168.1.252
From: "Unknown"
To:
Call-ID: 6f8b4c693ff86d1c7cfa75fe5c59e51f@192.168.1.252
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '6f8b4c693ff86d1c7cfa75fe5c59e51f@192.168.1.252' Method: OPTIONS
PBX*CLI>
<--- SIP read from 192.168.1.107:5060 --->
REGISTER sip:192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK71492ec264386e24d
Max-Forwards: 70
From:
To:
Call-ID: 1849ba88b3e18754
CSeq: 28266 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username="1102",realm="asterisk",nonce="6e59c487",uri="sip:192.168.1.252:5060",response="9d67232af3cbb5912434ca784d5e3288",algorithm=MD5
Contact: Phone 2
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.107 : 5060 (no NAT)
<--- Transmitting (NAT) to 192.168.1.107:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK71492ec264386e24d;received=192.168.1.107
From:
To:
Call-ID: 1849ba88b3e18754
CSeq: 28266 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 192.168.1.107:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK71492ec264386e24d;received=192.168.1.107
From:
To:
Call-ID: 1849ba88b3e18754
CSeq: 28266 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f42c431"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1849ba88b3e18754' in 32000 ms (Method: REGISTER)
PBX*CLI>
<--- SIP read from 192.168.1.107:5060 --->
REGISTER sip:192.168.1.252:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK2e9f04a300659dcd3
Max-Forwards: 70
From:
To:
Call-ID: 1849ba88b3e18754
CSeq: 28267 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Authorization: Digest username="1102",realm="asterisk",nonce="6f42c431",uri="sip:192.168.1.252:5060",response="6941ed63ff9199c6f30c18d6e00abce8",algorithm=MD5
Contact: Phone 2
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.107 : 5060 (NAT)
<--- Transmitting (NAT) to 192.168.1.107:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK2e9f04a300659dcd3;received=192.168.1.107
From:
To:
Call-ID: 1849ba88b3e18754
CSeq: 28267 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
PBX*CLI>
<--- Transmitting (NAT) to 192.168.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK2e9f04a300659dcd3;received=192.168.1.107
From:
To:
Call-ID: 1849ba88b3e18754
CSeq: 28267 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact:
Date: Thu, 03 Apr 2008 22:20:50 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1849ba88b3e18754' in 32000 ms (Method: REGISTER)
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged off from 127.0.0.1
Reliably Transmitting (no NAT) to x.x.x.138:5060:
OPTIONS sip:x.x.x.138 SIP/2.0
Via: SIP/2.0/UDP x.x.x.129:5060;branch=z9hG4bK782c2204;rport
From: "Unknown"
To:
Contact:
Call-ID: 1a61cf3a75b13a2d57c2b22870661eb8@x.x.x.129
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 03 Apr 2008 22:20:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
PBX*CLI>
<--- SIP read from x.x.x.138:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP x.x.x.129:5060;branch=z9hG4bK782c2204;received=x.x.x.129;rport=5060
From: "Unknown"
To:
Call-ID: 1a61cf3a75b13a2d57c2b22870661eb8@x.x.x.129
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '1a61cf3a75b13a2d57c2b22870661eb8@x.x.x.129' Method: OPTIONS
== Connect attempt from '127.0.0.1' unable to authenticate
PBX*CLI>
<--- SIP read from 192.168.1.107:5060 --->
BYE sip:1101@192.168.1.252 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bKd635ca7e5015b3314
Max-Forwards: 70
From:
To: "Phone 1"
Call-ID: 37c3dc34575f04ee55a5e475123636f4@192.168.1.252
CSeq: 1230 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference
Call-Info:
Supported: timer
User-Agent: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.107 : 5060 (NAT)
<--- Transmitting (NAT) to 192.168.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bKd635ca7e5015b3314;received=192.168.1.107
From:
To: "Phone 1"
Call-ID: 37c3dc34575f04ee55a5e475123636f4@192.168.1.252
CSeq: 1230 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
<------------>
== Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/1101-09935dc8'
-- Executing [h@macro-dial:1] Macro("SIP/1101-09935dc8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1101-09935dc8", "w") in new stack
Really destroying SIP dialog '37c3dc34575f04ee55a5e475123636f4@192.168.1.252' Method: BYE
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/1101-09935dc8", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/1101-09935dc8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/1101-09935dc8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/1101-09935dc8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/1101-09935dc8", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1101-09935dc8'
Scheduling destruction of SIP dialog '332cc09cdeab7e76' in 32000 ms (Method: ACK)
set_destination: Parsing
set_destination: set destination to 192.168.1.102, port 5060
Reliably Transmitting (NAT) to 192.168.1.102:5060:
BYE sip:1101@192.168.1.102:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK17d00dcf;rport
From: "1102"
To: "Phone 1"
Call-ID: 332cc09cdeab7e76
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== End MixMonitor Recording SIP/1101-09935dc8
PBX*CLI>
<--- SIP read from 192.168.1.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK17d00dcf;rport=5060;received=192.168.1.252
From: "1102"
To: "Phone 1"
Call-ID: 332cc09cdeab7e76
CSeq: 102 BYE
Server: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '332cc09cdeab7e76' Method: ACK
Scheduling destruction of SIP dialog '06f618956ee1a36b07eb1f53570f42b9@192.168.1.252' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.108:5060:
NOTIFY sip:1103@192.168.1.108:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK6f06f743;rport
From: "Unknown"
To:
Contact:
Call-ID: 06f618956ee1a36b07eb1f53570f42b9@192.168.1.252
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.252
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '52b538a51515d9af6dbddbc223711477@192.168.1.252' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.107:5060:
NOTIFY sip:1102@192.168.1.107:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK688c31a3;rport
From: "Unknown"
To:
Contact:
Call-ID: 52b538a51515d9af6dbddbc223711477@192.168.1.252
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:*97@192.168.1.252
Voice-Message: 0/0 (0/0)
---
PBX*CLI>
<--- SIP read from 192.168.1.108:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK6f06f743;rport=5060;received=192.168.1.252
From: "Unknown"
To:
Call-ID: 06f618956ee1a36b07eb1f53570f42b9@192.168.1.252
CSeq: 102 NOTIFY
Contact:
Server: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '06f618956ee1a36b07eb1f53570f42b9@192.168.1.252' Method: NOTIFY
PBX*CLI>
<--- SIP read from 192.168.1.107:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK688c31a3;rport=5060;received=192.168.1.252
From: "Unknown"
To:
Call-ID: 52b538a51515d9af6dbddbc223711477@192.168.1.252
CSeq: 102 NOTIFY
Contact:
Server: Aastra 51i/2.2.0.166
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '52b538a51515d9af6dbddbc223711477@192.168.1.252' Method: NOTIFY
PBX*CLI> exit
No, the statements work
Sun, 09/07/2008 - 8:53am
No, the statements work however they are not evaluated much like an access list.
So if your intent is to restrict calls in this trunk to only the uLaw CODEC you would do the following:
disallow=all allow=ulaw
As the peer is parsed it encounters the disallow=all statements which exludes all CODEC's. Then it finds the allow=ulaw.
The previous versions where not clear in how they processed these directives, now it works as it should.
--
Scott
aka "Skyking"
Member Since:
2007-02-25