Hello,
I was able to succesfully configure FreePBX before using these same trunks and routes on a different server, but for some reason it’s not working now. I’m using FreePBX 2.11 with Asterisk 11.11.0 running on an OVH VPS with CentOS x32.
I hope someone can help me here, I am stumped and have tried reinstalling the system, changing nat, setting externip/static and public ip, making sure firewall is completely open, and a number of things. I can receive incoming calls just fine, and can dial myself aswell.
When attempting to make an outbound call, I hear a recording that says “All circuits are busy right now, please try again later.”. The /var/log/asterisk/full doesn’t say much except "
[2014-07-30 02:09:43] WARNING[1917][C-00000017] chan_sip.c: Received response: "Forbidden" from '"aautin" <sip:[email protected]>;tag=as45d774cc'"
Which doesn’t really mean anything to me unfortunately.
One strange thing is that it shows 0 SIP Registrations, is that normal?
asterisk -rx 'sip show registry'
Host dnsmgr Username Refresh State
Reg.Time
0 SIP registrations.
The outbound trunk is setup with twilio, and has worked in the past.
Here are the trunk details:
host=klenhost.sip.twilio.com
type=peer
qualify=no
And some more technical details below, I have censored the following items for security:
Server Hostname will show as: XXXXXX.klenhost.com
Server IP Address will show as: 192.99.xx.xx
The SIP Phone (Using X-Lite) will show as: 68.203.xx.xx
And my cell phone will show as: 1512809xxxx
Here is some very verbose output of asterisk when attempting to make the call:
asterisk -rvvvvvvvvvvvvvvv
Asterisk 11.11.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
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 11.11.0 currently running on vps25113 (pid = 1890)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [1512809xxxx@from-internal:1] Macro("SIP/301-0000002a", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/301-0000002a", "TOUCH_MONITOR=1406678983.42") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/301-0000002a", "AMPUSER=301") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/301-0000002a", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/301-0000002a", "1?Set(REALCALLERIDNUM=301)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/301-0000002a", "AMPUSER=301") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/301-0000002a", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/301-0000002a", "AMPUSERCIDNAME=aautin") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/301-0000002a", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/301-0000002a", "AMPUSERCID=301") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/301-0000002a", "__DIAL_OPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/301-0000002a", "CALLERID(all)="aautin" <301>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/301-0000002a", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/301-0000002a", "1?Set(GROUP(concurrency_limit)=301)") in new stack
-- Executing [s@macro-user-callerid:14] GosubIf("SIP/301-0000002a", "7?sub-ccss,s,1(from-internal,1512809xxxx)") in new stack
-- Executing [s@sub-ccss:1] ExecIf("SIP/301-0000002a", "0?Return()") in new stack
-- Executing [s@sub-ccss:2] Set("SIP/301-0000002a", "CCSS_SETUP=TRUE") in new stack
-- Executing [s@sub-ccss:3] GosubIf("SIP/301-0000002a", "0?monitor_config,1(from-internal,1512809xxxx):monitor_default,1(from-internal,1512809xxxx)") in new stack
-- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/301-0000002a", "0?is_exten") in new stack
-- Executing [monitor_default@sub-ccss:2] StackPop("SIP/301-0000002a", "") in new stack
-- Executing [monitor_default@sub-ccss:3] Return("SIP/301-0000002a", "FALSE") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/301-0000002a", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/301-0000002a", "CALLERID(number)=301") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/301-0000002a", "CALLERID(name)=aautin") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/301-0000002a", "CDR(cnum)=301") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/301-0000002a", "CDR(cnam)=aautin") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/301-0000002a", "CHANNEL(language)=en") in new stack
-- Executing [1512809xxxx@from-internal:2] Set("SIP/301-0000002a", "MOHCLASS=default") in new stack
-- Executing [1512809xxxx@from-internal:3] Set("SIP/301-0000002a", "_NODEST=") in new stack
-- Executing [1512809xxxx@from-internal:4] Gosub("SIP/301-0000002a", "sub-record-check,s,1(out,1512809xxxx,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/301-0000002a", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/301-0000002a", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/301-0000002a", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/301-0000002a", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/301-0000002a", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/301-0000002a", "0?Set(__REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/301-0000002a", "0?out,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/301-0000002a", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/301-0000002a", "NOW=1406678983") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/301-0000002a", "__DAY=30") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/301-0000002a", "__MONTH=07") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/301-0000002a", "__YEAR=2014") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/301-0000002a", "__TIMESTR=20140730-020943") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/301-0000002a", "__FROMEXTEN=301") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/301-0000002a", "__CALLFILENAME=out-1512809xxxx-301-20140730-020943-1406678983.42") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/301-0000002a", "out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] ExecIf("SIP/301-0000002a", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
-- Executing [out@sub-record-check:2] GosubIf("SIP/301-0000002a", "0?record,1(exten,1512809xxxx,301)") in new stack
-- Executing [out@sub-record-check:3] Return("SIP/301-0000002a", "") in new stack
-- Executing [1512809xxxx@from-internal:5] Macro("SIP/301-0000002a", "dialout-trunk,5,1512809xxxx,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/301-0000002a", "DIAL_TRUNK=5") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/301-0000002a", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/301-0000002a", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/301-0000002a", "DIAL_NUMBER=1512809xxxx") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/301-0000002a", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/301-0000002a", "OUTBOUND_GROUP=OUT_5") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/301-0000002a", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/301-0000002a", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/301-0000002a", "DIAL_TRUNK_OPTIONS=Tt") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/301-0000002a", "outbound-callerid,5") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/301-0000002a", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/301-0000002a", "0?Set(REALCALLERIDNUM=301)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/301-0000002a", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/301-0000002a", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/301-0000002a", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/301-0000002a", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/301-0000002a", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,14)
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/301-0000002a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/301-0000002a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/301-0000002a", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/301-0000002a", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:18] Set("SIP/301-0000002a", "CDR(outbound_cnum)=301") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/301-0000002a", "CDR(outbound_cnam)=aautin") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/301-0000002a", "0?sub-flp-5,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/301-0000002a", "OUTNUM=1512809xxxx") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/301-0000002a", "custom=SIP/outbound") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/301-0000002a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/301-0000002a", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/301-0000002a", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/301-0000002a", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/301-0000002a", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/301-0000002a", "1?Set(CONNECTEDLINE(num,i)=1512809xxxx)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/301-0000002a", "1?Set(CONNECTEDLINE(name,i)=CID:301)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/301-0000002a", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/301-0000002a", "SIP/outbound/1512809xxxx,300,Tt") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/outbound/1512809xxxx
[2014-07-30 02:09:43] WARNING[1917][C-00000017]: chan_sip.c:23167 handle_response_invite: Received response: "Forbidden" from '"aautin" <sip:[email protected]>;tag=as45d774cc'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/301-0000002a", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/301-0000002a", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/301-0000002a", "RC=21") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/301-0000002a", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/301-0000002a", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/301-0000002a", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("SIP/301-0000002a", "CALLERID(number)=301") in new stack
-- Executing [1512809xxxx@from-internal:6] Macro("SIP/301-0000002a", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/301-0000002a", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/301-0000002a", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/301-0000002a", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/301-0000002a", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/301-0000002a> Playing 'all-circuits-busy-now.gsm' (language 'en')
> 0x98768c8 -- Probation passed - setting RTP source address to 68.203.xx.xx:57256
-- <SIP/301-0000002a> Playing 'pls-try-call-later.gsm' (language 'en')
[2014-07-30 02:09:45] NOTICE[2648][C-00000017]: res_rtp_asterisk.c:4100 `ast_rtp_read: Unknown RTP codec 126 received from '68.203.xx.xx:57256'`
Show channels:
asterisk -rx 'sip show channels'
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
0 active SIP dialogs
Show peers:
asterisk -rx 'sip show peers'
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
301/301 68.203.xx.xx D No No A 57004 OK (101 ms)
INCOMING2 107.21.222.153 Yes Yes 5060 Unmonitored
INCOMING3 107.21.211.20 Yes Yes 5060 Unmonitored
outbound/301 107.21.231.147 Yes Yes 5060 Unmonitored
twilioip-1 107.21.222.153 Yes Yes 5060 Unmonitored
twilioip-10 54.232.85.84 Yes Yes 5060 Unmonitored
twilioip-11 54.232.85.85 Yes Yes 5060 Unmonitored
twilioip-12 54.228.219.168 Yes Yes 5060 Unmonitored
twilioip-13 54.228.233.229 Yes Yes 5060 Unmonitored
twilioip-14 176.34.236.224 Yes Yes 5060 Unmonitored
twilioip-15 176.34.236.247 Yes Yes 5060 Unmonitored
twilioip-16 46.137.219.1 Yes Yes 5060 Unmonitored
twilioip-17 46.137.219.3 Yes Yes 5060 Unmonitored
twilioip-18 46.137.219.35 Yes Yes 5060 Unmonitored
twilioip-19 46.137.219.135 Yes Yes 5060 Unmonitored
twilioip-2 107.21.211.20 Yes Yes 5060 Unmonitored
twilioip-20 54.249.244.21 Yes Yes 5060 Unmonitored
twilioip-21 54.249.244.24 Yes Yes 5060 Unmonitored
twilioip-22 54.249.244.27 Yes Yes 5060 Unmonitored
twilioip-23 54.249.244.28 Yes Yes 5060 Unmonitored
twilioip-3 107.21.231.147 Yes Yes 5060 Unmonitored
twilioip-4 54.236.81.101 Yes Yes 5060 Unmonitored
twilioip-5 54.236.96.128 Yes Yes 5060 Unmonitored
twilioip-6 54.236.97.29 Yes Yes 5060 Unmonitored
twilioip-7 54.236.97.135 Yes Yes 5060 Unmonitored
twilioip-8 54.232.85.81 Yes Yes 5060 Unmonitored
twilioip-9 54.232.85.82 Yes Yes 5060 Unmonitored
yes 107.21.231.147 Yes Yes 5060 Unmonitored
28 sip peers [Monitored: 1 online, 0 offline Unmonitored: 27 online, 0 offline]
TCPDUMP:
tcpdump -s0 -w/tmp/capture.pcap -C50 udp and port 5060
tcpdump: listening on venet0, link-type LINUX_SLL (Linux cooked), capture size 65535 bytes
^C15 packets captured
15 packets received by filter
0 packets dropped by kernel
[root@vps25113 ~]# cat /tmp/capture.pcap
▒ò▒▒▒q:▒SV▒▒▒E▒6 mw▒D▒▒▒c▒▒ެ▒▒B▒INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-d796004e89697f59-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:57004>
To: <sip:[email protected]>
From: "aautin"<sip:[email protected]>;tag=3772225a
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4.7.0 73589-d1af294b-W6.1
Content-Length: 301
v=0
o=- 13051153162800986 1 IN IP4 68.203.xx.xx
s=X-Lite release 4.7.0 stamp 73589
c=IN IP4 68.203.xx.xx
t=0 0
m=audio 51062 RTP/AVP 125 100 0 8 9 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
:▒SXyy▒E`i▒▒@ܨ▒c▒▒D▒▒▒ެU▒▒SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-d796004e89697f59-1---d8754z-;received=68.203.xx.xx;rport=57004
From: "aautin"<sip:[email protected]>;tag=3772225a
To: <sip:[email protected]>;tag=as1cb4e566
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="654744a1"
Content-Length: 0
:▒Sl▒▒▒▒E▒6
my▒D▒▒▒c▒▒ެ▒▒▒ ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-d796004e89697f59-1---d8754z-;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as1cb4e566
From: "aautin"<sip:[email protected]>;tag=3772225a
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 1 ACK
Content-Length: 0
:▒S▒▒WW▒EG6
mv▒D▒▒▒c▒▒ެ▒3+▒INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-11d12d3bb3fb4314-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:57004>
To: <sip:[email protected]>
From: "aautin"<sip:[email protected]>;tag=3772225a
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4.7.0 73589-d1af294b-W6.1
Authorization: Digest username="301",realm="asterisk",nonce="654744a1",uri="sip:[email protected]",response="81eac3801c6985174ccbe66bcff5e1e8",algorithm=MD5
Content-Length: 301
v=0
o=- 13051153162800986 1 IN IP4 68.203.xx.xx
s=X-Lite release 4.7.0 stamp 73589
c=IN IP4 68.203.xx.xx
t=0 0
m=audio 51062 RTP/AVP 125 100 0 8 9 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
:▒S9▒HH▒E`8▒▒@▒▒▒c▒▒D▒▒▒ެ$▒SIP/2.0 100 Trying
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-11d12d3bb3fb4314-1---d8754z-;received=68.203.xx.xx;rport=57004
From: "aautin"<sip:[email protected]>;tag=3772225a
To: <sip:[email protected]>
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
:▒S▒zz▒E`jm,@eQ▒c▒▒k▒▒▒V▒INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.99.xx.xx:5060;branch=z9hG4bK2078ac53;rport
Max-Forwards: 70
From: "aautin" <sip:[email protected]>;tag=as467caae1
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.11.0)
Date: Wed, 30 Jul 2014 00:19:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 58071594 58071594 IN IP4 192.99.xx.xx
s=Asterisk PBX 11.11.0
c=IN IP4 192.99.xx.xx
t=0 0
m=audio 18706 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
:▒SO▒▒▒E{@*▒▒k▒▒c▒▒▒▒g▒▒SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.99.xx.xx:5060;received=192.99.xx.xx;branch=z9hG4bK2078ac53;rport=5060
From: "aautin" <sip:[email protected]>;tag=as467caae1
To: <sip:[email protected]>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Server: Twilio Gateway
Content-Length: 0
:▒S
▒▒▒▒E▒@*▒▒k▒▒c▒▒▒▒▒x▒SIP/2.0 403 Forbidden
To: <sip:[email protected]>;tag=62601438_6772d868_fa4ca5df-c6dd-4217-afab-5ac533493272
Via: SIP/2.0/UDP 192.99.xx.xx:5060;received=192.99.xx.xx;branch=z9hG4bK2078ac53;rport=5060
CSeq: 102 INVITE
Call-ID: [email protected]:5060
From: "aautin" <sip:[email protected]>;tag=as467caae1
Contact: <sip:10.115.130.163:5060>
Content-Length: 0
:▒S٭▒E`▒m-@f▒▒c▒▒k▒▒▒▒"▒ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.99.xx.xx:5060;branch=z9hG4bK2078ac53;rport
Max-Forwards: 70
From: "aautin" <sip:[email protected]>;tag=as467caae1
To: <sip:[email protected]>;tag=62601438_6772d868_fa4ca5df-c6dd-4217-afab-5ac533493272
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.11.0)
Content-Length: 0
:▒S ▒rr▒E`b▒▒@ۭ▒c▒▒D▒▒▒ެNWxSIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-11d12d3bb3fb4314-1---d8754z-;received=68.203.xx.xx;rport=57004
From: "aautin"<sip:[email protected]>;tag=3772225a
To: <sip:[email protected]>;tag=as75160b54
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 1494715253 1494715253 IN IP4 192.99.xx.xx
s=Asterisk PBX 11.11.0
c=IN IP4 192.99.xx.xx
t=0 0
m=audio 10180 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
:▒S5▒00▒E 67mz▒D▒▒▒c▒▒ެ▒
V▒
:▒S▒▒
zz▒Ej6KmxrD▒▒▒c▒▒ެ▒V▒▒CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-11d12d3bb3fb4314-1---d8754z-;rport
Max-Forwards: 70
To: <sip:[email protected]>
From: "aautin"<sip:[email protected]>;tag=3772225a
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 2 CANCEL
User-Agent: X-Lite 4.7.0 73589-d1af294b-W6.1
Authorization: Digest username="301",realm="asterisk",nonce="654744a1",uri="sip:[email protected]",response="a7fb6d27d2fee9db790bc0232f2e42c5",algorithm=MD5
Content-Length: 0
:▒S▒▒
33▒E`#▒▒@▒▒▒c▒▒D▒▒▒ެE▒SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-11d12d3bb3fb4314-1---d8754z-;received=68.203.xx.xx;rport=57004
From: "aautin"<sip:[email protected]>;tag=3772225a
To: <sip:[email protected]>;tag=as75160b54
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
:▒S▒▒
##▒E`▒▒@▒▒▒c▒▒D▒▒▒ެ▒▒gSIP/2.0 200 OK
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-11d12d3bb3fb4314-1---d8754z-;received=68.203.xx.xx;rport=57004
From: "aautin"<sip:[email protected]>;tag=3772225a
To: <sip:[email protected]>;tag=as75160b54
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 2 CANCEL
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
:▒S▒`
▒▒▒E▒6LmyFD▒▒▒c▒▒ެ▒▒▒ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 68.203.xx.xx:57004;branch=z9hG4bK-d8754z-11d12d3bb3fb4314-1---d8754z-;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as75160b54
From: "aautin"<sip:[email protected]>;tag=3772225a
Call-ID: ZjcyNGUxYzA5MTVhNzYyMzIyY2JiOTQ2NWQ1ZjYzYmI
CSeq: 2 ACK
Content-Length: 0
Please help!!!