Hi guys,
I have a problem that suck the life out of me
and i will apreciate any help with it.
this is a NAT issue in my opnion.
i got petnuim 4 with PBX in a flash 2.0.6.2
Cent OS 6.2
FreePBX 2.9.0.9
Asterisk 1.8.8.0
the server is connected to a TP-Link WR941ND router (UDP 5004-5082,10001-20000 Ports Opnes)
the router connected to Sagemcom F@st 3184 that acts as a Cable Modem (bridge)
the TP-link router update my NO-IP DDNS host.
i got an remote extension (Iphone runing Bria software) that got static IP.
the asterisk in configure like so:
sip_nat.conf:
nat=yes
externhost=XXXXX.no-ip.org
externrefresh=120
localnet=192.168.5.0/255.255.255.0
Qualify=yes
The Remote Extansion Config
deny=0.0.0.0/0.0.0.0
disallow=all
secret=dDhe653gg34
dtmfmode=rfc2833
canreinvite=yes
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
allow=g729,h263,ulaw,gsm
dial=SIP/702
mailbox=702@default
permit=156.18.121.135/255.255.255.255
callerid=device <702>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
THE problem is that that remote extension hangs up after 20-30 secs when i call other extension and when i use any trunk
I tried to disable the firewall in the TP-link router but it didn’t fixed the probelm it’s just make the hangs up after 60sec
here is the Sip debug and the CLI debug:
[Quote] [2012-02-27 19:32:15] VERBOSE[1788] chan_sip.c: – Registered SIP ‘702’ at 156.18.121.135:50546
[2012-02-27 19:32:15] NOTICE[1788] chan_sip.c: Peer ‘702’ is now Reachable. (255ms / 2000ms)
[2012-02-27 19:32:24] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:156.18.121.135:50546 —>
<------------->
[2012-02-27 19:32:26] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:156.18.121.135:50546 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 156.18.121.135:50546;rport;branch=z9hG4bKPjT930sn4NEC1WDXX4ghmjzYIr3ah0VFJv
Max-Forwards: 70
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected]
Contact: “702” sip:[email protected]:50546
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22812 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Bria iOS 2.0.1
Content-Type: application/sdp
Content-Length: 406
v=0
o=- 3539352749 3539352749 IN IP4 156.18.121.135
s=cpc_med
c=IN IP4 156.18.121.135
t=0 0
m=audio 4000 RTP/AVP 18 112 0 8 105 3 101
c=IN IP4 156.18.121.135
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 SILK/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[2012-02-27 19:32:26] VERBOSE[1788] chan_sip.c: — (15 headers 18 lines) —
[2012-02-27 19:32:26] VERBOSE[1788] chan_sip.c: Sending to 156.18.121.135:50546 (NAT)
[2012-02-27 19:32:26] VERBOSE[1788] chan_sip.c: Using INVITE request as basis request - DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
[2012-02-27 19:32:26] VERBOSE[1788] chan_sip.c: Found peer ‘702’ for ‘702’ from 156.18.121.135:50546
[2012-02-27 19:32:26] VERBOSE[1788] chan_sip.c:
<— Reliably Transmitting (NAT) to 156.18.121.135:50546 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPjT930sn4NEC1WDXX4ghmjzYIr3ah0VFJv;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as59a220f7
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22812 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="26c8fdd9"
Content-Length: 0
<------------>
[2012-02-27 19:32:26] VERBOSE[1788] chan_sip.c: Scheduling destruction of SIP dialog ‘DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg’ in 16320 ms (Method: INVITE)
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Retransmitting #1 (NAT) to 156.18.121.135:50546:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPjT930sn4NEC1WDXX4ghmjzYIr3ah0VFJv;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as59a220f7
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22812 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="26c8fdd9"
Content-Length: 0
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:156.18.121.135:50546 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 156.18.121.135:50546;rport;branch=z9hG4bKPjT930sn4NEC1WDXX4ghmjzYIr3ah0VFJv
Max-Forwards: 70
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected]
Contact: “702” sip:[email protected]:50546
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22812 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Bria iOS 2.0.1
Content-Type: application/sdp
Content-Length: 406
v=0
o=- 3539352749 3539352749 IN IP4 156.18.121.135
s=cpc_med
c=IN IP4 156.18.121.135
t=0 0
m=audio 4000 RTP/AVP 18 112 0 8 105 3 101
c=IN IP4 156.18.121.135
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 SILK/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: — (15 headers 18 lines) —
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Ignoring this INVITE request
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:156.18.121.135:50546 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 156.18.121.135:50546;rport;branch=z9hG4bKPjT930sn4NEC1WDXX4ghmjzYIr3ah0VFJv
Max-Forwards: 70
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as59a220f7
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22812 ACK
Content-Length: 0
<------------->
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: — (8 headers 0 lines) —
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:156.18.121.135:50546 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 156.18.121.135:50546;rport;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3
Max-Forwards: 70
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected]
Contact: “702” sip:[email protected]:50546
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Bria iOS 2.0.1
Authorization: Digest username=“702”, realm=“asterisk”, nonce=“26c8fdd9”, uri="sip:[email protected]", response=“cec903e81f0a84513e9cdc90d1ff212d”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 406
v=0
o=- 3539352749 3539352749 IN IP4 156.18.121.135
s=cpc_med
c=IN IP4 156.18.121.135
t=0 0
m=audio 4000 RTP/AVP 18 112 0 8 105 3 101
c=IN IP4 156.18.121.135
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 SILK/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: — (16 headers 18 lines) —
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Sending to 156.18.121.135:50546 (NAT)
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Using INVITE request as basis request - DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found peer ‘702’ for ‘702’ from 156.18.121.135:50546
[2012-02-27 19:32:27] VERBOSE[1788] netsock2.c: == Using SIP RTP TOS bits 184
[2012-02-27 19:32:27] VERBOSE[1788] netsock2.c: == Using SIP RTP CoS mark 5
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found RTP audio format 18
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found RTP audio format 112
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found RTP audio format 0
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found RTP audio format 8
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found RTP audio format 105
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found RTP audio format 3
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found RTP audio format 101
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found audio description format G729 for ID 18
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found unknown media description format SILK for ID 112
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found audio description format PCMU for ID 0
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found audio description format PCMA for ID 8
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found audio description format iLBC for ID 105
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found audio description format GSM for ID 3
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Capabilities: us - 0x80106 (gsm|ulaw|g729|h263), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x106 (gsm|ulaw|g729)
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Peer audio RTP is at port 156.18.121.135:4000
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Peer doesn’t provide video
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: Looking for 701 in from-internal (domain XXXXX.no-ip.org)
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: list_route: hop: sip:[email protected]:50546
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c:
<— Transmitting (NAT) to 156.18.121.135:50546 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected]
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:156.18.121.135:50546 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 156.18.121.135:50546;rport;branch=z9hG4bKPjT930sn4NEC1WDXX4ghmjzYIr3ah0VFJv
Max-Forwards: 70
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as59a220f7
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22812 ACK
Content-Length: 0
<------------->
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: — (8 headers 0 lines) —
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [701@from-internal:1] ExecIf(“SIP/702-0000000d”, “0?Set(__RINGTIMER=0)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [701@from-internal:2] Macro(“SIP/702-0000000d”, “exten-vm,701,701,0,0,0”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:1] Macro(“SIP/702-0000000d”, “user-callerid,”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:1] Set(“SIP/702-0000000d”, “AMPUSER=702”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:2] GotoIf(“SIP/702-0000000d”, “0?report”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:3] ExecIf(“SIP/702-0000000d”, “1?Set(REALCALLERIDNUM=702)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:4] Set(“SIP/702-0000000d”, “AMPUSER=702”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:5] Set(“SIP/702-0000000d”, “AMPUSERCIDNAME=702”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:6] GotoIf(“SIP/702-0000000d”, “0?report”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:7] Set(“SIP/702-0000000d”, “AMPUSERCID=702”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:8] Set(“SIP/702-0000000d”, “CALLERID(all)=“702” <702>”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:9] GotoIf(“SIP/702-0000000d”, “0?limit”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:10] ExecIf(“SIP/702-0000000d”, “0?Set(GROUP(concurrency_limit)=702)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:11] GosubIf(“SIP/702-0000000d”, “7?sub-ccss,s,1(macro-exten-vm,701)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@sub-ccss:1] ExecIf(“SIP/702-0000000d”, “0?Return()”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@sub-ccss:2] Set(“SIP/702-0000000d”, “CCSS_SETUP=TRUE”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@sub-ccss:3] GosubIf(“SIP/702-0000000d”, “0?monitor_config,1(macro-exten-vm,701):monitor_default,1(macro-exten-vm,701)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [monitor_default@sub-ccss:1] GotoIf(“SIP/702-0000000d”, “1?is_exten”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (sub-ccss,monitor_default,4)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [monitor_default@sub-ccss:4] Set(“SIP/702-0000000d”, “CALLCOMPLETION(cc_monitor_policy)=generic”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [monitor_default@sub-ccss:5] Set(“SIP/702-0000000d”, “CALLCOMPLETION(cc_max_monitors)=5”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [monitor_default@sub-ccss:6] Return(“SIP/702-0000000d”, “TRUE”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@sub-ccss:4] GosubIf(“SIP/702-0000000d”, “7?agent_config,1:agent_default,1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:1] Set(“SIP/702-0000000d”, “CALLCOMPLETION(cc_agent_policy)=generic”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:2] Set(“SIP/702-0000000d”, “CALLCOMPLETION(cc_offer_timer)=30”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:3] Set(“SIP/702-0000000d”, “CALLCOMPLETION(ccbs_available_timer)=”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:4] Set(“SIP/702-0000000d”, “CALLCOMPLETION(ccnr_available_timer)=”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:5] Set(“SIP/702-0000000d”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:6] ExecIf(“SIP/702-0000000d”, “1?Set(CALLCOMPLETION(cc_recall_timer)=)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:7] ExecIf(“SIP/702-0000000d”, “1?Set(CALLCOMPLETION(cc_max_agents)=)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:8] ExecIf(“SIP/702-0000000d”, “0?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/702_701@from-ccss-)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:9] Set(“SIP/702-0000000d”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [agent_config@sub-ccss:10] Return(“SIP/702-0000000d”, “”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@sub-ccss:5] Set(“SIP/702-0000000d”, “DB(AMPUSER/702/ccss/last_number)=701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@sub-ccss:6] Return(“SIP/702-0000000d”, “”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:12] ExecIf(“SIP/702-0000000d”, “0?Set(CHANNEL(language)=)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:13] GotoIf(“SIP/702-0000000d”, “0?continue”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:14] Set(“SIP/702-0000000d”, “__TTL=64”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:15] GotoIf(“SIP/702-0000000d”, “1?continue”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-user-callerid,s,26)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:26] Set(“SIP/702-0000000d”, “CALLERID(number)=702”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:27] Set(“SIP/702-0000000d”, “CALLERID(name)=702”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-user-callerid:28] Set(“SIP/702-0000000d”, “CHANNEL(language)=en”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:2] Set(“SIP/702-0000000d”, “RingGroupMethod=none”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:3] Set(“SIP/702-0000000d”, “__EXTTOCALL=701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:4] Set(“SIP/702-0000000d”, “__PICKUPMARK=701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:5] Set(“SIP/702-0000000d”, “RT=20”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:6] Macro(“SIP/702-0000000d”, “record-enable,701,IN”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-record-enable:1] GotoIf(“SIP/702-0000000d”, “1?check”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-record-enable,s,4)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-record-enable:4] ExecIf(“SIP/702-0000000d”, “0?MacroExit()”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-record-enable:5] GotoIf(“SIP/702-0000000d”, “0?Group:OUT”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-record-enable,s,14)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-record-enable:14] GotoIf(“SIP/702-0000000d”, “1?IN”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-record-enable,s,18)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-record-enable:18] ExecIf(“SIP/702-0000000d”, “1?MacroExit()”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:7] GotoIf(“SIP/702-0000000d”, “1?macrodial”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-exten-vm,s,13)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:13] GosubIf(“SIP/702-0000000d”, “0?clrheader,1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-exten-vm:14] Macro(“SIP/702-0000000d”, “dial-one,20,tr,701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:1] Set(“SIP/702-0000000d”, “DEXTEN=701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:2] Set(“SIP/702-0000000d”, “DIALSTATUS_CW=”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:3] GosubIf(“SIP/702-0000000d”, “0?screen,1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:4] GosubIf(“SIP/702-0000000d”, “0?cf,1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:5] GotoIf(“SIP/702-0000000d”, “1?skip1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-dial-one,s,8)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:8] GotoIf(“SIP/702-0000000d”, “0?nodial”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:9] GotoIf(“SIP/702-0000000d”, “0?continue”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:10] Set(“SIP/702-0000000d”, “EXTHASCW=ENABLED”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:11] GotoIf(“SIP/702-0000000d”, “0?next1:cwinusebusy”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-dial-one,s,23)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:23] GotoIf(“SIP/702-0000000d”, “1?next3:continue”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-dial-one,s,24)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:24] ExecIf(“SIP/702-0000000d”, “0?Set(DIALSTATUS_CW=BUSY)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:25] GotoIf(“SIP/702-0000000d”, “0?nodial”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:26] GosubIf(“SIP/702-0000000d”, “1?dstring,1:dlocal,1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:1] Set(“SIP/702-0000000d”, “DSTRING=”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:2] Set(“SIP/702-0000000d”, “DEVICES=701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:3] ExecIf(“SIP/702-0000000d”, “0?Return()”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:4] ExecIf(“SIP/702-0000000d”, “0?Set(DEVICES=01)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:5] Set(“SIP/702-0000000d”, “LOOPCNT=1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:6] Set(“SIP/702-0000000d”, “ITER=1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:7] Set(“SIP/702-0000000d”, “THISDIAL=SIP/701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:8] GosubIf(“SIP/702-0000000d”, “1?zap2dahdi,1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/702-0000000d”, “0?Return()”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:2] Set(“SIP/702-0000000d”, “NEWDIAL=”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:3] Set(“SIP/702-0000000d”, “LOOPCNT2=1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:4] Set(“SIP/702-0000000d”, “ITER2=1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:5] Set(“SIP/702-0000000d”, “THISPART2=SIP/701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:6] ExecIf(“SIP/702-0000000d”, “0?Set(THISPART2=DAHDI/701)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/702-0000000d”, “NEWDIAL=SIP/701&”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/702-0000000d”, “ITER2=2”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/702-0000000d”, “0?begin2”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/702-0000000d”, “THISDIAL=SIP/701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/702-0000000d”, “”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:9] Set(“SIP/702-0000000d”, “DSTRING=SIP/701&”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:10] Set(“SIP/702-0000000d”, “ITER=2”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:11] GotoIf(“SIP/702-0000000d”, “0?begin”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:12] Set(“SIP/702-0000000d”, “DSTRING=SIP/701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [dstring@macro-dial-one:13] Return(“SIP/702-0000000d”, “”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:27] GotoIf(“SIP/702-0000000d”, “0?nodial”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:28] GotoIf(“SIP/702-0000000d”, “1?skiptrace”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Goto (macro-dial-one,s,30)
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:30] Set(“SIP/702-0000000d”, “D_OPTIONS=tr”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:31] ExecIf(“SIP/702-0000000d”, “0?SIPAddHeader(Alert-Info: )”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:32] ExecIf(“SIP/702-0000000d”, “0?SIPAddHeader()”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:33] ExecIf(“SIP/702-0000000d”, “0?Set(CHANNEL(musicclass)=)”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:34] GosubIf(“SIP/702-0000000d”, “0?qwait,1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:35] Set(“SIP/702-0000000d”, “__CWIGNORE=”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:36] Set(“SIP/702-0000000d”, “__KEEPCID=TRUE”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:37] GotoIf(“SIP/702-0000000d”, “0?usegoto,1”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:38] GotoIf(“SIP/702-0000000d”, “0?godial”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:39] Set(“SIP/702-0000000d”, “CONNECTEDLINE(name,i)=701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:40] Set(“SIP/702-0000000d”, “CONNECTEDLINE(num)=701”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:41] Set(“SIP/702-0000000d”, “D_OPTIONS=trI”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] pbx.c: – Executing [s@macro-dial-one:42] Dial(“SIP/702-0000000d”, “SIP/701,20,trI”) in new stack
[2012-02-27 19:32:27] VERBOSE[8135] netsock2.c: == Using SIP RTP TOS bits 184
[2012-02-27 19:32:27] VERBOSE[8135] netsock2.c: == Using SIP RTP CoS mark 5
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c: Audio is at 5060
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c: Video is at 192.168.5.101:5060
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c: Adding video codec 0x80000 (h263) to SDP
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c: Reliably Transmitting (NAT) to 192.168.5.100:28473:
INVITE sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d SIP/2.0
Via: SIP/2.0/UDP 192.168.5.101:5060;branch=z9hG4bK7c2fc0c3;rport
Max-Forwards: 70
From: “702” sip:[email protected];tag=as4b190a31
To: sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.8.0)
Date: Mon, 27 Feb 2012 17:32:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 380
v=0
o=root 1183107675 1183107675 IN IP4 192.168.5.101
s=Asterisk PBX 1.8.8.0
c=IN IP4 192.168.5.101
b=CT:384
t=0 0
m=audio 15284 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 16980 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
[2012-02-27 19:32:27] VERBOSE[8135] app_dial.c: – Called SIP/701
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c:
<— Transmitting (NAT) to 156.18.121.135:50546 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
[2012-02-27 19:32:27] VERBOSE[8135] app_dial.c: – Connected line update to SIP/702-0000000d prevented.
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:192.168.5.100:28473 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.5.101:5060;branch=z9hG4bK7c2fc0c3;rport=5060
Contact: sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d
To: sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d;tag=93cd713d
From: "702"sip:[email protected];tag=as4b190a31
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0
<------------->
[2012-02-27 19:32:27] VERBOSE[1788] chan_sip.c: — (9 headers 0 lines) —
[2012-02-27 19:32:27] VERBOSE[8135] app_dial.c: – SIP/701-0000000e is ringing
[2012-02-27 19:32:27] VERBOSE[8135] chan_sip.c:
<— Transmitting (NAT) to 156.18.121.135:50546 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:192.168.5.100:28473 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.101:5060;branch=z9hG4bK7c2fc0c3;rport=5060
Contact: sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d
To: sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d;tag=93cd713d
From: "702"sip:[email protected];tag=as4b190a31
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 295
v=0
o=- 12974837548376845 1 IN IP4 192.168.5.100
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.5.100
t=0 0
m=audio 58080 RTP/AVP 0 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 58002 RTP/AVP 34
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=2;CIF=2;VGA=2
a=inactive
<------------->
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: — (12 headers 13 lines) —
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Found RTP audio format 0
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Found RTP audio format 3
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Found RTP audio format 101
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Found audio description format telephone-event for ID 101
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Found RTP video format 34
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Found video description format H263 for ID 34
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Capabilities: us - 0x80106 (gsm|ulaw|g729|h263), peer - audio=0x6 (gsm|ulaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80006 (gsm|ulaw|h263)
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Peer audio RTP is at port 192.168.5.100:58080
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Peer video RTP is at port 192.168.5.100:58002
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: list_route: hop: sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: set_destination: Parsing sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d for address/port to send to
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: set_destination: set destination to 192.168.5.100:28473
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Transmitting (NAT) to 192.168.5.100:28473:
ACK sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d SIP/2.0
Via: SIP/2.0/UDP 192.168.5.101:5060;branch=z9hG4bK04de6e93;rport
Max-Forwards: 70
From: “702” sip:[email protected];tag=as4b190a31
To: sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d;tag=93cd713d
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.8.0)
Content-Length: 0
[2012-02-27 19:32:29] VERBOSE[8135] app_dial.c: – Connected line update to SIP/702-0000000d prevented.
[2012-02-27 19:32:29] VERBOSE[8135] app_dial.c: – SIP/701-0000000e answered SIP/702-0000000d
[2012-02-27 19:32:29] VERBOSE[8135] chan_sip.c: Audio is at 5060
[2012-02-27 19:32:29] VERBOSE[8135] chan_sip.c: Adding codec 0x100 (g729) to SDP
[2012-02-27 19:32:29] VERBOSE[8135] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2012-02-27 19:32:29] VERBOSE[8135] chan_sip.c: Adding codec 0x2 (gsm) to SDP
[2012-02-27 19:32:29] VERBOSE[8135] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2012-02-27 19:32:29] VERBOSE[8135] chan_sip.c:
<— Reliably Transmitting (NAT) to 156.18.121.135:50546 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1975884091 1975884091 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10184 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
[2012-02-27 19:32:29] VERBOSE[1788] chan_sip.c: Retransmitting #1 (NAT) to 156.18.121.135:50546:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1975884091 1975884091 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10184 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-02-27 19:32:30] VERBOSE[1788] chan_sip.c: Retransmitting #2 (NAT) to 156.18.121.135:50546:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1975884091 1975884091 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10184 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-02-27 19:32:30] NOTICE[8135] res_rtp_asterisk.c: Unknown RTP codec 126 received from ‘192.168.5.100:58002’
[2012-02-27 19:32:30] NOTICE[8135] res_rtp_asterisk.c: Unknown RTP codec 126 received from ‘192.168.5.100:58002’
[2012-02-27 19:32:30] NOTICE[8135] res_rtp_asterisk.c: Unknown RTP codec 126 received from ‘192.168.5.100:58002’
[2012-02-27 19:32:31] VERBOSE[1788] chan_sip.c: Retransmitting #3 (NAT) to 156.18.121.135:50546:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1975884091 1975884091 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10184 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:192.168.5.100:28473 —>
<------------->
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:192.168.5.100:28473 —>
SUBSCRIBE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:28473;branch=z9hG4bK-d8754z-62472787c7cff835-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:28473
To: "701"sip:[email protected];tag=as772d6705
From: "701"sip:[email protected];tag=e45e996c
Call-ID: YThmNGYxMGRkZGM3NzJjNTJhZjY3NjkxNjQ4MGMxYzg.
CSeq: 191 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username=“701”,realm=“asterisk”,nonce=“1d6f8d81”,uri=“sip:[email protected]:5060”,response=“4df86206a9d364ca7df9469387c543c2”,algorithm=MD5
Event: message-summary
Content-Length: 0
<------------->
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c: — (14 headers 0 lines) —
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c: Found peer ‘701’ for ‘701’ from 192.168.5.100:28473
[2012-02-27 19:32:32] NOTICE[1788] chan_sip.c: Correct auth, but based on stale nonce received from ‘"701"sip:[email protected];tag=e45e996c’
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c:
<— Transmitting (NAT) to 192.168.5.100:28473 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.5.100:28473;branch=z9hG4bK-d8754z-62472787c7cff835-1—d8754z-;received=192.168.5.100;rport=28473
From: "701"sip:[email protected];tag=e45e996c
To: "701"sip:[email protected];tag=as772d6705
Call-ID: YThmNGYxMGRkZGM3NzJjNTJhZjY3NjkxNjQ4MGMxYzg.
CSeq: 191 SUBSCRIBE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“38e8f3c8”, stale=true
Content-Length: 0
<------------>
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c: Scheduling destruction of SIP dialog ‘YThmNGYxMGRkZGM3NzJjNTJhZjY3NjkxNjQ4MGMxYzg.’ in 6400 ms (Method: SUBSCRIBE)
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:192.168.5.100:28473 —>
SUBSCRIBE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.100:28473;branch=z9hG4bK-d8754z-0363651e196e1a73-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:28473
To: "701"sip:[email protected];tag=as772d6705
From: "701"sip:[email protected];tag=e45e996c
Call-ID: YThmNGYxMGRkZGM3NzJjNTJhZjY3NjkxNjQ4MGMxYzg.
CSeq: 192 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username=“701”,realm=“asterisk”,nonce=“38e8f3c8”,uri=“sip:[email protected]:5060”,response=“cfa856c249729af0bf237df89e35c050”,algorithm=MD5
Event: message-summary
Content-Length: 0
<------------->
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c: — (14 headers 0 lines) —
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c: Found peer ‘701’ for ‘701’ from 192.168.5.100:28473
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c: Scheduling destruction of SIP dialog ‘YThmNGYxMGRkZGM3NzJjNTJhZjY3NjkxNjQ4MGMxYzg.’ in 310000 ms (Method: SUBSCRIBE)
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c:
<— Transmitting (NAT) to 192.168.5.100:28473 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.100:28473;branch=z9hG4bK-d8754z-0363651e196e1a73-1—d8754z-;received=192.168.5.100;rport=28473
From: "701"sip:[email protected];tag=e45e996c
To: "701"sip:[email protected];tag=as772d6705
Call-ID: YThmNGYxMGRkZGM3NzJjNTJhZjY3NjkxNjQ4MGMxYzg.
CSeq: 192 SUBSCRIBE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: sip:[email protected]:5060;expires=300
Content-Length: 0
<------------>
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c: Reliably Transmitting (NAT) to 192.168.5.100:28473:
NOTIFY sip:[email protected]:28473 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.101:5060;branch=z9hG4bK51d4a46e;rport
Max-Forwards: 70
Route: sip:[email protected]:28473
From: “Unknown” sip:[email protected];tag=as772d6705
To: sip:[email protected]:28473;tag=e45e996c
Contact: sip:[email protected]:5060
Call-ID: YThmNGYxMGRkZGM3NzJjNTJhZjY3NjkxNjQ4MGMxYzg.
CSeq: 197 NOTIFY
User-Agent: FPBX-2.9.0(1.8.8.0)
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 93
Messages-Waiting: no
Message-Account: sip:*[email protected]:5060
Voice-Message: 0/0 (0/0)
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:192.168.5.100:28473 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.101:5060;branch=z9hG4bK51d4a46e;rport=5060
Contact: sip:[email protected]:28473
To: sip:[email protected]:28473;tag=e45e996c
From: "Unknown"sip:[email protected];tag=as772d6705
Call-ID: YThmNGYxMGRkZGM3NzJjNTJhZjY3NjkxNjQ4MGMxYzg.
CSeq: 197 NOTIFY
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0
<------------->
[2012-02-27 19:32:32] VERBOSE[1788] chan_sip.c: — (9 headers 0 lines) —
[2012-02-27 19:32:33] VERBOSE[1788] chan_sip.c: Retransmitting #4 (NAT) to 156.18.121.135:50546:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1975884091 1975884091 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10184 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-02-27 19:32:33] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:156.18.121.135:50546 —>
<------------->
[2012-02-27 19:32:37] VERBOSE[1788] chan_sip.c: Retransmitting #5 (NAT) to 156.18.121.135:50546:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1975884091 1975884091 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10184 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-02-27 19:32:40] NOTICE[8135] res_rtp_asterisk.c: Unknown RTP codec 126 received from ‘192.168.5.100:58002’
[2012-02-27 19:32:41] VERBOSE[1788] chan_sip.c: Retransmitting #6 (NAT) to 156.18.121.135:50546:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1975884091 1975884091 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10184 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-02-27 19:32:42] VERBOSE[1788] chan_sip.c:
<— SIP read from UDP:156.18.121.135:50546 —>
<------------->
[2012-02-27 19:32:45] VERBOSE[1788] chan_sip.c: Retransmitting #7 (NAT) to 156.18.121.135:50546:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 156.18.121.135:50546;branch=z9hG4bKPj3ioSWUVsVvM1kmOjxmDa74YSbPEd7XJ3;received=156.18.121.135;rport=50546
From: “702” sip:[email protected];tag=JTrnIXKSPTb6rk3QxNe3J16LID02XX1G
To: sip:[email protected];tag=as055b3eed
Call-ID: DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg
CSeq: 22813 INVITE
Server: FPBX-2.9.0(1.8.8.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1975884091 1975884091 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 10184 RTP/AVP 18 0 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2012-02-27 19:32:45] WARNING[1788] chan_sip.c: Retransmission timeout reached on transmission DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg for seqno 22813 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 16319ms with no response
[2012-02-27 19:32:45] WARNING[1788] chan_sip.c: Hanging up call DIefmgBmx1H9OzEEi1jc6Wy11w036Tyg - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2012-02-27 19:32:45] VERBOSE[8135] pbx.c: – Executing [h@macro-dial-one:1] Macro(“SIP/702-0000000d”, “hangupcall,”) in new stack
[2012-02-27 19:32:45] VERBOSE[8135] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/702-0000000d”, “1?theend”) in new stack
[2012-02-27 19:32:45] VERBOSE[8135] pbx.c: – Goto (macro-hangupcall,s,3)
[2012-02-27 19:32:45] VERBOSE[8135] pbx.c: – Executing [s@macro-hangupcall:3] Hangup(“SIP/702-0000000d”, “”) in new stack
[2012-02-27 19:32:45] VERBOSE[8135] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/702-0000000d’ in macro ‘hangupcall’
[2012-02-27 19:32:45] VERBOSE[8135] features.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/702-0000000d’
[2012-02-27 19:32:46] VERBOSE[8135] chan_sip.c: Scheduling destruction of SIP dialog ‘37e8b7290dfc4c63[email protected]:5060’ in 6400 ms (Method: INVITE)
[2012-02-27 19:32:46] VERBOSE[8135] chan_sip.c: set_destination: Parsing sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d for address/port to send to
[2012-02-27 19:32:46] VERBOSE[8135] chan_sip.c: set_destination: set destination to 192.168.5.100:28473
[2012-02-27 19:32:46] VERBOSE[8135] chan_sip.c: Reliably Transmitting (NAT) to 192.168.5.100:28473:
BYE sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d SIP/2.0
Via: SIP/2.0/UDP 192.168.5.101:5060;branch=z9hG4bK2717d591;rport
Max-Forwards: 70
From: “702” sip:[email protected];tag=as4b190a31
To: sip:[email protected]:28473;rinstance=cbfa55e95f77cb8d;tag=93cd713d
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: FPBX-2.9.0(1.8.8.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
[Quote/]
thanks for the helpers.