Outbound calls "All Circuits are Busy"

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 [[email protected]:1] Macro("SIP/301-0000002a", "user-callerid,LIMIT,EXTERNAL,") in new stack
        -- Executing [[email protected]:1] Set("SIP/301-0000002a", "TOUCH_MONITOR=1406678983.42") in new stack
        -- Executing [[email protected]:2] Set("SIP/301-0000002a", "AMPUSER=301") in new stack
        -- Executing [[email protected]:3] GotoIf("SIP/301-0000002a", "0?report") in new stack
        -- Executing [[email protected]:4] ExecIf("SIP/301-0000002a", "1?Set(REALCALLERIDNUM=301)") in new stack
        -- Executing [[email protected]:5] Set("SIP/301-0000002a", "AMPUSER=301") in new stack
        -- Executing [[email protected]:6] GotoIf("SIP/301-0000002a", "0?limit") in new stack
        -- Executing [[email protected]:7] Set("SIP/301-0000002a", "AMPUSERCIDNAME=aautin") in new stack
        -- Executing [[email protected]:8] GotoIf("SIP/301-0000002a", "0?report") in new stack
        -- Executing [[email protected]:9] Set("SIP/301-0000002a", "AMPUSERCID=301") in new stack
        -- Executing [[email protected]:10] Set("SIP/301-0000002a", "__DIAL_OPTIONS=Ttr") in new stack
        -- Executing [[email protected]:11] Set("SIP/301-0000002a", "CALLERID(all)="aautin" <301>") in new stack
        -- Executing [[email protected]:12] GotoIf("SIP/301-0000002a", "0?limit") in new stack
        -- Executing [[email protected]:13] ExecIf("SIP/301-0000002a", "1?Set(GROUP(concurrency_limit)=301)") in new stack
        -- Executing [[email protected]:14] GosubIf("SIP/301-0000002a", "7?sub-ccss,s,1(from-internal,1512809xxxx)") in new stack
        -- Executing [[email protected]:1] ExecIf("SIP/301-0000002a", "0?Return()") in new stack
        -- Executing [[email protected]:2] Set("SIP/301-0000002a", "CCSS_SETUP=TRUE") in new stack
        -- Executing [[email protected]:3] GosubIf("SIP/301-0000002a", "0?monitor_config,1(from-internal,1512809xxxx):monitor_default,1(from-internal,1512809xxxx)") in new stack
        -- Executing [[email protected]:1] GotoIf("SIP/301-0000002a", "0?is_exten") in new stack
        -- Executing [[email protected]:2] StackPop("SIP/301-0000002a", "") in new stack
        -- Executing [[email protected]:3] Return("SIP/301-0000002a", "FALSE") in new stack
        -- Executing [[email protected]:15] GotoIf("SIP/301-0000002a", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,28)
        -- Executing [[email protected]:28] Set("SIP/301-0000002a", "CALLERID(number)=301") in new stack
        -- Executing [[email protected]:29] Set("SIP/301-0000002a", "CALLERID(name)=aautin") in new stack
        -- Executing [[email protected]:30] Set("SIP/301-0000002a", "CDR(cnum)=301") in new stack
        -- Executing [[email protected]:31] Set("SIP/301-0000002a", "CDR(cnam)=aautin") in new stack
        -- Executing [[email protected]:32] Set("SIP/301-0000002a", "CHANNEL(language)=en") in new stack
        -- Executing [[email protected]:2] Set("SIP/301-0000002a", "MOHCLASS=default") in new stack
        -- Executing [[email protected]:3] Set("SIP/301-0000002a", "_NODEST=") in new stack
        -- Executing [[email protected]:4] Gosub("SIP/301-0000002a", "sub-record-check,s,1(out,1512809xxxx,)") in new stack
        -- Executing [[email protected]:1] Set("SIP/301-0000002a", "REC_POLICY_MODE_SAVE=") in new stack
        -- Executing [[email protected]:2] GotoIf("SIP/301-0000002a", "1?check") in new stack
        -- Goto (sub-record-check,s,7)
        -- Executing [[email protected]:7] Set("SIP/301-0000002a", "__MON_FMT=wav") in new stack
        -- Executing [[email protected]:8] GotoIf("SIP/301-0000002a", "1?next") in new stack
        -- Goto (sub-record-check,s,11)
        -- Executing [[email protected]:11] ExecIf("SIP/301-0000002a", "0?Return()") in new stack
        -- Executing [[email protected]:12] ExecIf("SIP/301-0000002a", "0?Set(__REC_POLICY_MODE=)") in new stack
        -- Executing [[email protected]:13] GotoIf("SIP/301-0000002a", "0?out,1") in new stack
        -- Executing [[email protected]:14] Set("SIP/301-0000002a", "__REC_STATUS=INITIALIZED") in new stack
        -- Executing [[email protected]:15] Set("SIP/301-0000002a", "NOW=1406678983") in new stack
        -- Executing [[email protected]:16] Set("SIP/301-0000002a", "__DAY=30") in new stack
        -- Executing [[email protected]:17] Set("SIP/301-0000002a", "__MONTH=07") in new stack
        -- Executing [[email protected]:18] Set("SIP/301-0000002a", "__YEAR=2014") in new stack
        -- Executing [[email protected]:19] Set("SIP/301-0000002a", "__TIMESTR=20140730-020943") in new stack
        -- Executing [[email protected]:20] Set("SIP/301-0000002a", "__FROMEXTEN=301") in new stack
        -- Executing [[email protected]:21] Set("SIP/301-0000002a", "__CALLFILENAME=out-1512809xxxx-301-20140730-020943-1406678983.42") in new stack
        -- Executing [[email protected]:22] Goto("SIP/301-0000002a", "out,1") in new stack
        -- Goto (sub-record-check,out,1)
        -- Executing [[email protected]:1] ExecIf("SIP/301-0000002a", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
        -- Executing [[email protected]:2] GosubIf("SIP/301-0000002a", "0?record,1(exten,1512809xxxx,301)") in new stack
        -- Executing [[email protected]:3] Return("SIP/301-0000002a", "") in new stack
        -- Executing [[email protected]:5] Macro("SIP/301-0000002a", "dialout-trunk,5,1512809xxxx,,off") in new stack
        -- Executing [[email protected]:1] Set("SIP/301-0000002a", "DIAL_TRUNK=5") in new stack
        -- Executing [[email protected]:2] GosubIf("SIP/301-0000002a", "0?sub-pincheck,s,1()") in new stack
        -- Executing [[email protected]:3] GotoIf("SIP/301-0000002a", "0?disabletrunk,1") in new stack
        -- Executing [[email protected]:4] Set("SIP/301-0000002a", "DIAL_NUMBER=1512809xxxx") in new stack
        -- Executing [[email protected]:5] Set("SIP/301-0000002a", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
        -- Executing [[email protected]:6] Set("SIP/301-0000002a", "OUTBOUND_GROUP=OUT_5") in new stack
        -- Executing [[email protected]:7] GotoIf("SIP/301-0000002a", "1?nomax") in new stack
        -- Goto (macro-dialout-trunk,s,9)
        -- Executing [[email protected]:9] GotoIf("SIP/301-0000002a", "0?skipoutcid") in new stack
        -- Executing [[email protected]:10] Set("SIP/301-0000002a", "DIAL_TRUNK_OPTIONS=Tt") in new stack
        -- Executing [[email protected]:11] Macro("SIP/301-0000002a", "outbound-callerid,5") in new stack
        -- Executing [[email protected]:1] ExecIf("SIP/301-0000002a", "0?Set(CALLERPRES()=)") in new stack
        -- Executing [[email protected]:2] ExecIf("SIP/301-0000002a", "0?Set(REALCALLERIDNUM=301)") in new stack
        -- Executing [[email protected]:3] GotoIf("SIP/301-0000002a", "1?normcid") in new stack
        -- Goto (macro-outbound-callerid,s,6)
        -- Executing [[email protected]:6] Set("SIP/301-0000002a", "USEROUTCID=") in new stack
        -- Executing [[email protected]:7] Set("SIP/301-0000002a", "EMERGENCYCID=") in new stack
        -- Executing [[email protected]:8] Set("SIP/301-0000002a", "TRUNKOUTCID=") in new stack
        -- Executing [[email protected]:9] GotoIf("SIP/301-0000002a", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,14)
        -- Executing [[email protected]:14] ExecIf("SIP/301-0000002a", "0?Set(CALLERID(all)=)") in new stack
        -- Executing [[email protected]:15] ExecIf("SIP/301-0000002a", "0?Set(CALLERID(all)=)") in new stack
        -- Executing [[email protected]:16] ExecIf("SIP/301-0000002a", "0?Set(CALLERID(all)=)") in new stack
        -- Executing [[email protected]:17] ExecIf("SIP/301-0000002a", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
        -- Executing [[email protected]:18] Set("SIP/301-0000002a", "CDR(outbound_cnum)=301") in new stack
        -- Executing [[email protected]:19] Set("SIP/301-0000002a", "CDR(outbound_cnam)=aautin") in new stack
        -- Executing [[email protected]:12] GosubIf("SIP/301-0000002a", "0?sub-flp-5,s,1()") in new stack
        -- Executing [[email protected]:13] Set("SIP/301-0000002a", "OUTNUM=1512809xxxx") in new stack
        -- Executing [[email protected]:14] Set("SIP/301-0000002a", "custom=SIP/outbound") in new stack
        -- Executing [[email protected]:15] ExecIf("SIP/301-0000002a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
        -- Executing [[email protected]:16] ExecIf("SIP/301-0000002a", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
        -- Executing [[email protected]:17] Macro("SIP/301-0000002a", "dialout-trunk-predial-hook,") in new stack
        -- Executing [[email protected]:1] MacroExit("SIP/301-0000002a", "") in new stack
        -- Executing [[email protected]:18] GotoIf("SIP/301-0000002a", "0?bypass,1") in new stack
        -- Executing [[email protected]:19] ExecIf("SIP/301-0000002a", "1?Set(CONNECTEDLINE(num,i)=1512809xxxx)") in new stack
        -- Executing [[email protected]:20] ExecIf("SIP/301-0000002a", "1?Set(CONNECTEDLINE(name,i)=CID:301)") in new stack
        -- Executing [[email protected]:21] GotoIf("SIP/301-0000002a", "0?customtrunk") in new stack
        -- Executing [[email protected]: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 [[email protected]:23] NoOp("SIP/301-0000002a", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
        -- Executing [[email protected]:24] GotoIf("SIP/301-0000002a", "0?continue,1:s-CHANUNAVAIL,1") in new stack
        -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
        -- Executing [[email protected]:1] Set("SIP/301-0000002a", "RC=21") in new stack
        -- Executing [[email protected]:2] Goto("SIP/301-0000002a", "21,1") in new stack
        -- Goto (macro-dialout-trunk,21,1)
        -- Executing [[email protected]:1] Goto("SIP/301-0000002a", "continue,1") in new stack
        -- Goto (macro-dialout-trunk,continue,1)
        -- Executing [[email protected]:1] NoOp("SIP/301-0000002a", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
        -- Executing [[email protected]:2] Set("SIP/301-0000002a", "CALLERID(number)=301") in new stack
        -- Executing [[email protected]:6] Macro("SIP/301-0000002a", "outisbusy,") in new stack
        -- Executing [[email protected]:1] Progress("SIP/301-0000002a", "") in new stack
        -- Executing [[email protected]:2] GotoIf("SIP/301-0000002a", "0?emergency,1") in new stack
        -- Executing [[email protected]:3] GotoIf("SIP/301-0000002a", "0?intracompany,1") in new stack
        -- Executing [[email protected]-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
[[email protected] ~]# 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`▒[email protected]▒▒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!!!

just a curious question, for that voip provider did you used the “register string”? because IMHO your problem is the register string

cheers!

rfernandez_net,

Thank you for your reply. I am currently not using a “Register String”, but it was not required on the previous host this was setup on. Just in case, I have contacted twilio (after failed attempted to google the string) and asked them directly if a register string is required. I’ll update you once I get an answer.

Regards,
Albert

any answer on getting twilio sip trunk to work?

I had this problem endlessly and I found that using qualify on the trunk and increasing the UDP timeout on my firewall did the trick. FYI Im using a Sonicwall device as my router / firewall.

Hey use these settings with twilio trunk configuration.

qualify=yes
qualifyfreq=600

the trunks are showing unmonitored because they are checked by the Asterisk.
after entering qualify asterisk will send packets to check whether the trunk is enabled or not and shows the status.