Inbound Call not work

I am having a two SIP account(022xxxxx1 and 022xxxxx2) from my ITSP. My FreePBX is under NAT. Outbound call is OK but I am not able to make Inbound call to work.
My configuration steps are as below:

  1. Create Extension (ex: 101, 102)
  2. Create SIP Trunk using 022xxxxx1 account
  3. Create an Inbound Routes as below
    Descrition: 022xxxxx1
    DID number: 022xxxxx1
    Fax Detect: NO
    Set Destination: Extentions->101
    Submit->Apply
  4. Other setting are in default setting
  5. Try to call from 022xxxxx2 to 022xxxxx1 to test but not work
  6. I try see the Asterisk Log Files but I can not see any call come into my FreePBX

Did i miss anything? What should goes wrong?

Thanks,

I had been working on this for a long way. Now, it is just one more step to complete. Please, help.

your point . .

  1. I try see the Asterisk Log Files but I can not see any call come into my FreePBX

suggests that either your firewall is not passing the call to astrisk or your vsp is not sending it to your firewall.

A little more detail as to how you set up your network (that includes routers) would help you get an answer.

I did DNAT on my router as below:

  1. Proto = UDP, Dst.Address= (mypublic_ip), Dst.port=5060 --> FreePBX IP (Private_IP)
  2. Proto = UDP, Dst.Address= (mypublic_ip), Dst.port=10001-20000 --> FreePBX IP (Private_IP)

and also config NAT Setting in Asterisk SIP Setting as below:

NAT = Yes
IP Configuration = Static
External IP = (mypublic_ip)
Internal IP = (my_private_LAN_ip)

FYI: sip show registry is Success state.

then from the asterisk cli (
rasterisk
#wait for connection . . .
sip set debug on

)

will show connections from your provider when you get an incoming call,

if you don’t see them as the call is made then you or the vsp did something wrong, if you do see them then what they say will “clue you”

Yes, I can see log when call coming in with “sip set debug on”. Below are the log:

[2012-07-21 12:04:37] VERBOSE[7185] chan_sip.c:
<— SIP read from UDP:203.176.131.8:5060 —>
INVITE sip:[email protected]:5060;cid=175 SIP/2.0
Via: SIP/2.0/UDP 203.176.131.8:5060;branch=z9hG4bK-d8754z-18fb574b438d7138-1—d8754z-;rport
Via: SIP/2.0/UDP 203.176.131.8:5061;branch=z9hG4bK73670ef937d067d1b709b083f1cdfc61;rport=5061
Max-Forwards: 69
Record-Route: sip:203.176.131.8;lr
Contact: "Anonymous"sip:203.176.131.8:5061
To: sip:[email protected]
From: sip:[email protected];tag=cd817637537ae295700b9eb8666de1c4
Call-ID: tcjE.xrVu.1qafd2qYsDW8UoBI1vYR05~o~o
CSeq: 200 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
Content-Length: 327
cisco-GUID: 1944352642-3538751969-3183476770-430351960
h323-conf-id: 1944352642-3538751969-3183476770-430351960

v=0
o=Sippy 140464876 0 IN IP4 203.176.131.7
s=cpc_med
t=0 0
m=audio 46302 RTP/AVP 112 0 8 105 3 101
c=IN IP4 203.176.131.7
a=sendrecv
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
<------------->
[2012-07-21 12:04:37] VERBOSE[7185] chan_sip.c: — (17 headers 15 lines) —
[2012-07-21 12:04:37] VERBOSE[7185] chan_sip.c: Sending to 203.176.131.8:5060 (NAT)
[2012-07-21 12:04:37] VERBOSE[7185] chan_sip.c: Using INVITE request as basis request - tcjE.xrVu.1qafd2qYsDW8UoBI1vYR05~o~o
[2012-07-21 12:04:37] VERBOSE[7185] chan_sip.c: Found peer ‘NextVoiz’ for ‘022611818’ from 203.176.131.8:5060
[2012-07-21 12:04:37] VERBOSE[7185] chan_sip.c:
<— Reliably Transmitting (NAT) to 203.176.131.8:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 203.176.131.8:5060;branch=z9hG4bK-d8754z-18fb574b438d7138-1—d8754z-;received=203.176.131.8;rport=5060
Via: SIP/2.0/UDP 203.176.131.8:5061;branch=z9hG4bK73670ef937d067d1b709b083f1cdfc61;rport=5061
From: sip:[email protected];tag=cd817637537ae295700b9eb8666de1c4
To: sip:[email protected];tag=as0ccc9f65
Call-ID: tcjE.xrVu.1qafd2qYsDW8UoBI1vYR05~o~o
CSeq: 200 INVITE
Server: FPBX-2.10.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="39e7d1df"
Content-Length: 0

FYI: 203.176.131.8 is my VSP. 172.16.31.125 is my FreePBX.

I see something like "Contact: “Anonymous"sip:203.176.131.8:5061” and “SIP/2.0 401 Unauthorized” . Does it mean anything?

I did config “Allow Anonymous Inbound SIP Calls? = Yes” in General Setting.

“SIP/2.0 401 Unauthorized” means that you do not authorize the call :slight_smile:

Either “register” against the VSP, or perhaps you will need to allow anonymous calls in your general settings and deploy a very restrictive firewall, especially as you are using a provider in Cambodia and that part of the world is notoriously full of voip hackers.

I already allowed anonymous calls in General Setting but I still don’t get incoming call. Any further check?

Here is another debug output:


[2012-07-24 00:26:33] VERBOSE[3493] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.31.124:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.16.31.125:5060;branch=z9hG4bK56d0e587
Max-Forwards: 70
From: “Unknown” (sip:[email protected]);tag=as653c1a58
To: (sip:[email protected]:5060;transport=udp)
Contact: (sip:[email protected]:5060)
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Mon, 23 Jul 2012 17:26:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

What i noticed it "From: “Unknown” (sip:[email protected]);tag=as653c1a58 ".

What does it mean?

That there is no CallerID attached to tha call.

I would start by allowing anonymous inbound connections (temporarily, and only if there is no “registration” option available fom your VSP).

Then create a “catch-all” inbound route, then you should see how asterisk /freepbx is processing the call more simple in a post mortem analysis of /vr/log/asterisk/full (providing your logger*.conf heirarchy is actually writing relevant detaials to that file)

OK, does it have anything to do Incoming Call? or does Incoming call fail because of “SIP/2.0 401 Unauthorized” only? How to fix this “SIP/2.0 401 Unauthorized” ? I already run through debug (-vvvvvvvvvvr) but cannot any clue.

I would also delete all your trunks because if you explicitly match the inbound with a wrong secret or some other auth function it will stop and not check any other peers.

Thanks for reply.

I did as below:

  1. config to allow anonymous call=yes, DID=any and forward the incoming call to 101 extension which is any IPphone using 172.16.31.123.
  2. I also tried to delete all information in User Context and User Detail. Still i can not see any weird log appear.
  3. My FreePBX is 172.16.31.125

Below is the log that I got from /var/log/asterisk/full when I called from outside:

[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c:
<— SIP read from UDP:172.16.31.123:5060 —>
REGISTER sip:172.16.31.125:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.31.123:5060;branch=z9hG4bK17858208872224111339;rport
From: 101 sip:[email protected]:5060;tag=160795492
To: 101 sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 5 REGISTER
Contact: sip:[email protected]:5060
Max-Forwards: 70
Expires: 60
User-Agent: huawei
Content-Length: 0

<------------->
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c: — (11 headers 0 lines) —
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c: Sending to 172.16.31.123:5060 (NAT)
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c:
<— Transmitting (no NAT) to 172.16.31.123:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.31.123:5060;branch=z9hG4bK17858208872224111339;received=172.16.31.123;rport=5060
From: 101 sip:[email protected]:5060;tag=160795492
To: 101 sip:[email protected]:5060;tag=as69d3b3af
Call-ID: [email protected]
CSeq: 5 REGISTER
Server: FPBX-2.10.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="72750ac2"
Content-Length: 0

<------------>
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c:
<— SIP read from UDP:172.16.31.123:5060 —>
REGISTER sip:172.16.31.125:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.31.123:5060;branch=z9hG4bK8238767927951840;rport
From: 101 sip:[email protected]:5060;tag=160795492
To: 101 sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 6 REGISTER
Contact: sip:[email protected]:5060
Authorization: Digest username=“101”, realm=“asterisk”, nonce=“72750ac2”, uri=“sip:172.16.31.125:5060”, response=“3bc4413f36685328b2cd4f7b655bf5ea”, algorithm=MD5
Max-Forwards: 70
Expires: 60
User-Agent: huawei
Content-Length: 0

<------------->
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c: — (12 headers 0 lines) —
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c: Sending to 172.16.31.123:5060 (no NAT)
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.31.123:5060:
OPTIONS sip:[email protected]16.31.123:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.31.125:5060;branch=z9hG4bK529481e6
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as0021ea29
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Tue, 24 Jul 2012 12:13:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c:
<— Transmitting (no NAT) to 172.16.31.123:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.31.123:5060;branch=z9hG4bK8238767927951840;received=172.16.31.123;rport=5060
From: 101 sip:[email protected]:5060;tag=160795492
To: 101 sip:[email protected]:5060;tag=as69d3b3af
Call-ID: [email protected]
CSeq: 6 REGISTER
Server: FPBX-2.10.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: sip:[email protected]:5060;expires=60
Date: Tue, 24 Jul 2012 12:13:27 GMT
Content-Length: 0

<------------>
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
[2012-07-24 19:13:27] VERBOSE[3375] chan_sip.c:
<— SIP read from UDP:172.16.31.123:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.31.125:5060;branch=z9hG4bK529481e6
From: Unknown sip:[email protected];tag=as0021ea29
To: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

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

?

anyone, please. It is just one more step!

I’ve the same problem.
Did you solved the issue?
Thanks in advance.

Dump from the SIP log using the Debug option:

<— SIP read from UDP:204.11.192.163:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40
f: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
t: sip:[email protected]
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m: sip:[email protected]:5060;transport=udp
Supported: timer
c: application/sdp
l: 339

v=0
o=NexTone-MSW 21269 5930 IN IP4 204.11.192.163
s=sip call
c=IN IP4 204.11.192.163
t=0 0
m=audio 51334 RTP/AVP 0 8 18 101
a=ptime:20
a=sendrecv
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=setup:actpass
<------------->
[2013-07-22 20:23:46] VERBOSE[1922] chan_sip.c: — (11 headers 16 lines) —
[2013-07-22 20:23:46] VERBOSE[1922] chan_sip.c: Sending to 204.11.192.163:5060 (NAT)
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Sending to 204.11.192.163:5060 (NAT)
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Using INVITE request as basis request - [email protected]
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: No matching peer for ‘19524356989’ from ‘204.11.192.163:5060’
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] netsock2.c: == Using SIP RTP TOS bits 184
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] netsock2.c: == Using SIP RTP CoS mark 5
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Found RTP audio format 0
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Found RTP audio format 8
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Found RTP audio format 18
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Found RTP audio format 101
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Found audio description format telephone-event for ID 101
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Found audio description format G729 for ID 18
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Found audio description format PCMA for ID 8
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Found audio description format PCMU for ID 0
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Peer audio RTP is at port 204.11.192.163:51334
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: Looking for s in from-sip-external (domain 71.82.125.24)
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: list_route: hop: sip:[email protected]:5060;transport=udp
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c:
<— Transmitting (NAT) to 204.11.192.163:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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

<------------>
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/66.193.176.35-00000065”, “0?checklang:noanonymous”) in new stack
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] pbx.c: – Goto (from-sip-external,s,5)
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] pbx.c: – Executing [[email protected]:5] Set(“SIP/66.193.176.35-00000065”, “TIMEOUT(absolute)=15”) in new stack
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] func_timeout.c: – Channel will hangup at 2013-07-22 20:24:01.579 CDT.
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] pbx.c: – Executing [[email protected]:6] Answer(“SIP/66.193.176.35-00000065”, “”) in new stack
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] chan_sip.c: Audio is at 18142
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2013-07-22 20:23:46] VERBOSE[31457][C-0000005a] chan_sip.c:
<— Reliably Transmitting (NAT) to 204.11.192.163:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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
Require: timer
Content-Length: 259

v=0
o=root 2027894801 2027894801 IN IP4 71.82.125.24
s=Asterisk PBX 11.4.0
c=IN IP4 71.82.125.24
t=0 0
m=audio 18142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[2013-07-22 20:23:47] VERBOSE[1922] chan_sip.c: Retransmitting #1 (NAT) to 204.11.192.163:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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
Require: timer
Content-Length: 259

v=0
o=root 2027894801 2027894801 IN IP4 71.82.125.24
s=Asterisk PBX 11.4.0
c=IN IP4 71.82.125.24
t=0 0
m=audio 18142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2013-07-22 20:23:47] VERBOSE[31457][C-0000005a] pbx.c: – Executing [[email protected]:7] Wait(“SIP/66.193.176.35-00000065”, “2”) in new stack
[2013-07-22 20:23:48] VERBOSE[1922] chan_sip.c: Retransmitting #2 (NAT) to 204.11.192.163:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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
Require: timer
Content-Length: 259

v=0
o=root 2027894801 2027894801 IN IP4 71.82.125.24
s=Asterisk PBX 11.4.0
c=IN IP4 71.82.125.24
t=0 0
m=audio 18142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2013-07-22 20:23:49] VERBOSE[31457][C-0000005a] pbx.c: – Executing [[email protected]:8] Playback(“SIP/66.193.176.35-00000065”, “ss-noservice”) in new stack
[2013-07-22 20:23:49] VERBOSE[31457][C-0000005a] file.c: – <SIP/66.193.176.35-00000065> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2013-07-22 20:23:50] VERBOSE[1922] chan_sip.c: Retransmitting #3 (NAT) to 204.11.192.163:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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
Require: timer
Content-Length: 259

v=0
o=root 2027894801 2027894801 IN IP4 71.82.125.24
s=Asterisk PBX 11.4.0
c=IN IP4 71.82.125.24
t=0 0
m=audio 18142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2013-07-22 20:23:54] VERBOSE[1922] chan_sip.c: Retransmitting #4 (NAT) to 204.11.192.163:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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
Require: timer
Content-Length: 259

v=0
o=root 2027894801 2027894801 IN IP4 71.82.125.24
s=Asterisk PBX 11.4.0
c=IN IP4 71.82.125.24
t=0 0
m=audio 18142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2013-07-22 20:23:54] VERBOSE[31457][C-0000005a] pbx.c: – Executing [[email protected]:9] PlayTones(“SIP/66.193.176.35-00000065”, “congestion”) in new stack
[2013-07-22 20:23:54] VERBOSE[31457][C-0000005a] pbx.c: – Executing [[email protected]:10] Congestion(“SIP/66.193.176.35-00000065”, “5”) in new stack
[2013-07-22 20:23:57] VERBOSE[1922] chan_sip.c: Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
[2013-07-22 20:23:58] VERBOSE[1922] chan_sip.c: Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
[2013-07-22 20:23:58] VERBOSE[1922] chan_sip.c: Retransmitting #5 (NAT) to 204.11.192.163:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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
Require: timer
Content-Length: 259

v=0
o=root 2027894801 2027894801 IN IP4 71.82.125.24
s=Asterisk PBX 11.4.0
c=IN IP4 71.82.125.24
t=0 0
m=audio 18142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2013-07-22 20:23:59] VERBOSE[31457][C-0000005a] pbx.c: == Spawn extension (from-sip-external, s, 10) exited non-zero on ‘SIP/66.193.176.35-00000065’
[2013-07-22 20:23:59] VERBOSE[31457][C-0000005a] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/66.193.176.35-00000065”, “”) in new stack
[2013-07-22 20:23:59] VERBOSE[31457][C-0000005a] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/66.193.176.35-00000065’
[2013-07-22 20:23:59] VERBOSE[31457][C-0000005a] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
[2013-07-22 20:24:02] VERBOSE[1922] chan_sip.c: Retransmitting #6 (NAT) to 204.11.192.163:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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
Require: timer
Content-Length: 259

v=0
o=root 2027894801 2027894801 IN IP4 71.82.125.24
s=Asterisk PBX 11.4.0
c=IN IP4 71.82.125.24
t=0 0
m=audio 18142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2013-07-22 20:24:03] VERBOSE[1922] chan_sip.c: Really destroying SIP dialog ‘[email protected][::1]’ Method: REGISTER
[2013-07-22 20:24:06] VERBOSE[1922] chan_sip.c: Retransmitting #7 (NAT) to 204.11.192.163:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
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
Require: timer
Content-Length: 259

v=0
o=root 2027894801 2027894801 IN IP4 71.82.125.24
s=Asterisk PBX 11.4.0
c=IN IP4 71.82.125.24
t=0 0
m=audio 18142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2013-07-22 20:24:06] VERBOSE[1922] chan_sip.c:
<— SIP read from UDP:204.11.192.163:5060 —>
BYE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-4a8de0c51f9f9182fae3e4d65c700cca
f: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
t: sip:[email protected];tag=as06294d18
i: [email protected]
CSeq: 2 BYE
Max-Forwards: 10
l: 0

<------------->
[2013-07-22 20:24:06] VERBOSE[1922] chan_sip.c: — (8 headers 0 lines) —
[2013-07-22 20:24:06] VERBOSE[1922][C-0000005a] chan_sip.c:
<— Reliably Transmitting (NAT) to 204.11.192.163:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-25cc571eadb3b02e209f1c7f084a5d40;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[2013-07-22 20:24:06] VERBOSE[1922][C-0000005a] chan_sip.c: Sending to 204.11.192.163:5060 (NAT)
[2013-07-22 20:24:06] VERBOSE[1922][C-0000005a] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: BYE)
[2013-07-22 20:24:06] VERBOSE[1922][C-0000005a] chan_sip.c:
<— Transmitting (NAT) to 204.11.192.163:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.163:5060;branch=z9hG4bK-4a8de0c51f9f9182fae3e4d65c700cca;received=204.11.192.163;rport=5060
From: “Dobbs Keith” sip:[email protected];tag=3583531426-109964
To: sip:[email protected];tag=as06294d18
Call-ID: [email protected]
CSeq: 2 BYE
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

.
.
[2013-07-22 20:23:46] VERBOSE[1922][C-0000005a] chan_sip.c: No matching peer for ‘19524356989’ from ‘204.11.192.163:5060’
.
.
you will need a “matching peer”