Extension rings but does not answer call

I can make outgoing calls fine but incoming calls ring the extension, but when I pick it up nothing happens and the caller still hears ringing!

This is driving me nuts for 12+ hours. I am thinking about replacing my SPA9000 system with Asterisk and snom 870 phones. I setup a PC with AsteriskNow and updated it to AN 1.6 and Asterisk 1.6. I have a peer for CallCentric setup. I can use the snom phone with the SPA9000 fine, also by itself configured to CallCentric. Both in and out fine. But with Asterisk… I could not find anyone one the web who had this issue. Anyone here have an idea? I sure would appreciate any help.

type=peer
trustrpid=no
session-timers=refuse
session-refresher=uas
session-minse=90
session-expires=180
sendrpid=yes
secret=xxxxxxxxxx
qualify=yes
nat=yes
insecure=port,invite
host=callcentric.com
fromuser=1777########
fromdomain=callcentric.com
dtmfmode=rfc2833
disallow=all
defaultuser=1777########
context=from-callcentric
allow=ulaw&alaw

Please supply sip set debug on output, with a suitable verbosity and debug level.

Do you have any codec restrictions on the phone?

I am at the end of anything else I can find, any help to point me in the right directions would be greatly appreciated. Thanks!

I think is this what you want, seems there is no way to send attachments to you so I had to put the last 621 lines here. The trace is long and I also noticed some warning/error messages I put first.

17 Lines of possible issues?:
[Feb 8 09:26:02] ERROR[2823] chan_unistim.c: Unable to load config unistim.conf
[Feb 8 09:26:03] NOTICE[2823] chan_sip.c: The ‘username’ field for sip peers has been deprecated in favor of the term ‘defaultuser’
[Feb 8 09:26:03] ERROR[2823] pbx_dundi.c: Unable to load config dundi.conf
[Feb 8 09:26:03] WARNING[2823] app_festival.c: No such configuration file festival.conf
[Feb 8 09:26:03] NOTICE[2823] app_queue.c: No queuerules.conf file found, queues will not follow penalty rules
[Feb 8 09:26:03] NOTICE[2823] chan_agent.c: No agent configuration found – agent support disabled
[Feb 8 09:26:03] NOTICE[2823] chan_skinny.c: Unable to load config skinny.conf, Skinny disabled
[Feb 8 09:26:03] NOTICE[2823] chan_mgcp.c: Unable to load config mgcp.conf, MGCP disabled
[Feb 8 09:26:03] NOTICE[2823] pbx_ael.c: Starting AEL load process.
[Feb 8 09:26:03] NOTICE[2823] pbx_ael.c: File /etc/asterisk/extensions.ael not found;
[Feb 8 09:26:03] WARNING[2823] pbx_config.c: The use of ‘_.’ for an extension is strongly discouraged and can have unexpected behavior. Please use ‘_X.’ instead at line 726
[Feb 8 09:26:03] WARNING[2823] pbx.c: Unable to register extension ‘s’, priority 1 in ‘macro-outisbusy’, already in use
[Feb 8 09:26:03] WARNING[2823] pbx.c: Unable to register extension ‘s’, priority 2 in ‘macro-outisbusy’, already in use
[Feb 8 09:26:03] WARNING[2823] pbx.c: Unable to register extension ‘s’, priority 3 in ‘macro-outisbusy’, already in use
[Feb 8 09:26:03] ERROR[2823] app_amd.c: Configuration file amd.conf missing.
[Feb 8 09:26:03] NOTICE[2834] chan_sip.c: Peer ‘CallCentric’ is now Reachable. (93ms / 2000ms)
[Feb 8 09:26:03] NOTICE[2834] chan_sip.c: Peer ‘600’ is now Reachable. (35ms / 2000ms)[Feb 8 09:26:03] WARNING[2823] cdr_csv.c: unable to load config: cdr.conf

The last 621 lines are here, shows calling in to hang up:
<------------->
[Feb 8 09:42:50] VERBOSE[2834] logger.c: — (8 headers 0 lines) —
[Feb 8 09:42:50] VERBOSE[2834] logger.c: Scheduling destruction of SIP dialog ‘63aa520e4efc75da6cd5dd6e232225c7@10.246.1.24’ in 32000 ms (Method: REGISTER)
[Feb 8 09:42:50] NOTICE[2834] chan_sip.c: Outbound Registration: Expiry for callcentric.com is 71 sec (Scheduling reregistration in 56 s)
[Feb 8 09:42:53] VERBOSE[2834] logger.c:
<— SIP read from UDP://204.11.192.37:5060 —>
INVITE sip:17777654321@10.246.1.24 SIP/2.0
v: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b
f: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
t: sip:14151234567@ss.callcentric.com
i: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
Max-Forwards: 13
m: sip:742b7490b026844cf99f83f36c3166fd@204.11.192.37:5060;transport=udp
Supported: timer
c: application/sdp
l: 311

v=0
o=NexTone-MSW 313478145 1265645542 IN IP4 204.11.192.37
s=sip call
c=IN IP4 204.11.192.37
t=0 0
m=audio 50252 RTP/AVP 18 0 101
a=ptime:20
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:0 PCMU/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=silenceSupp:off - - - -
a=setup:actpass

<------------->
[Feb 8 09:42:53] VERBOSE[2834] logger.c: — (11 headers 14 lines) —
[Feb 8 09:42:53] VERBOSE[2834] logger.c: == Using SIP RTP TOS bits 184
[Feb 8 09:42:53] VERBOSE[2834] logger.c: == Using SIP RTP CoS mark 5
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Sending to 204.11.192.37 : 5060 (NAT)
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Using INVITE request as basis request - 48061422-3474634843-426617@msw1.telengy.net
[Feb 8 09:42:53] VERBOSE[2834] logger.c: No user ‘14092551234’ in SIP users list
[Feb 8 09:42:53] VERBOSE[2834] logger.c: No matching peer for ‘14092551234’ from ‘204.11.192.37:5060’
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Found RTP audio format 18
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Found RTP audio format 0
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Found RTP audio format 101
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Found audio description format telephone-event for ID 101
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Found audio description format PCMU for ID 0
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Found audio description format G729 for ID 18
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Peer audio RTP is at port 204.11.192.37:50252
[Feb 8 09:42:53] VERBOSE[2834] logger.c: Looking for 17777654321 in from-sip-external (domain 10.246.1.24)
[Feb 8 09:42:53] VERBOSE[2834] logger.c: list_route: hop: sip:742b7490b026844cf99f83f36c3166fd@204.11.192.37:5060;transport=udp
[Feb 8 09:42:53] VERBOSE[2834] logger.c:
<— Transmitting (NAT) to 204.11.192.37:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Length: 0

<------------>
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-sip-external:1] NoOp(“SIP/66.193.176.35-00000002”, “Received incoming SIP connection from unknown peer to 17777654321”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-sip-external:2] Set(“SIP/66.193.176.35-00000002”, “DID=17777654321”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-sip-external:3] Goto(“SIP/66.193.176.35-00000002”, “s,1”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Goto (from-sip-external,s,1)
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@from-sip-external:1] GotoIf(“SIP/66.193.176.35-00000002”, “1?checklang:noanonymous”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Goto (from-sip-external,s,2)
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@from-sip-external:2] GotoIf(“SIP/66.193.176.35-00000002”, “0?setlanguage:from-trunk,17777654321,1”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Goto (from-trunk,17777654321,1)
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-trunk:1] Set(“SIP/66.193.176.35-00000002”, “__FROM_DID=17777654321”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-trunk:2] Gosub(“SIP/66.193.176.35-00000002”, “app-blacklist-check,s,1”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@app-blacklist-check:1] GotoIf(“SIP/66.193.176.35-00000002”, “0?blacklisted”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@app-blacklist-check:2] Set(“SIP/66.193.176.35-00000002”, “CALLED_BLACKLIST=1”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@app-blacklist-check:3] Return(“SIP/66.193.176.35-00000002”, “”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-trunk:3] ExecIf(“SIP/66.193.176.35-00000002”, “0 ?Set(CALLERID(name)=14092551234)”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-trunk:4] Set(“SIP/66.193.176.35-00000002”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-trunk:5] Set(“SIP/66.193.176.35-00000002”, “CALLERPRES()=allowed_not_screened”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [17777654321@from-trunk:6] Goto(“SIP/66.193.176.35-00000002”, “from-did-direct,600,1”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Goto (from-did-direct,600,1)
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [600@from-did-direct:1] Macro(“SIP/66.193.176.35-00000002”, “exten-vm,600,600”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:1] Macro(“SIP/66.193.176.35-00000002”, “user-callerid”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:1] Set(“SIP/66.193.176.35-00000002”, “AMPUSER=14092551234”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:2] GotoIf(“SIP/66.193.176.35-00000002”, “0?report”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:3] ExecIf(“SIP/66.193.176.35-00000002”, “1?Set(REALCALLERIDNUM=14092551234)”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:4] Set(“SIP/66.193.176.35-00000002”, “AMPUSER=”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:5] Set(“SIP/66.193.176.35-00000002”, “AMPUSERCIDNAME=”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:6] GotoIf(“SIP/66.193.176.35-00000002”, “1?report”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Goto (macro-user-callerid,s,10)
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:10] GotoIf(“SIP/66.193.176.35-00000002”, “0?continue”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:11] Set(“SIP/66.193.176.35-00000002”, “__TTL=64”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:12] GotoIf(“SIP/66.193.176.35-00000002”, “1?continue”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Goto (macro-user-callerid,s,19)
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-user-callerid:19] NoOp(“SIP/66.193.176.35-00000002”, “Using CallerID “M G TECHNOLOGIE” <14092551234>”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:2] Set(“SIP/66.193.176.35-00000002”, “RingGroupMethod=none”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:3] Set(“SIP/66.193.176.35-00000002”, “VMBOX=600”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:4] Set(“SIP/66.193.176.35-00000002”, “EXTTOCALL=600”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:5] Set(“SIP/66.193.176.35-00000002”, “CFUEXT=”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:6] Set(“SIP/66.193.176.35-00000002”, “CFBEXT=”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:7] Set(“SIP/66.193.176.35-00000002”, “RT=15”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:8] Macro(“SIP/66.193.176.35-00000002”, “record-enable,600,IN”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-record-enable:1] GotoIf(“SIP/66.193.176.35-00000002”, “1?check”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Goto (macro-record-enable,s,4)
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Executing [s@macro-record-enable:4] AGI(“SIP/66.193.176.35-00000002”, “recordingcheck,20100208-094253,1265650973.2”) in new stack
[Feb 8 09:42:53] VERBOSE[3515] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Feb 8 09:42:55] VERBOSE[3515] logger.c: recordingcheck,20100208-094253,1265650973.2: Inbound recording not enabled
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – <SIP/66.193.176.35-00000002>AGI Script recordingcheck completed, returning 0
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – Executing [s@macro-record-enable:5] MacroExit(“SIP/66.193.176.35-00000002”, “”) in new stack
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – Executing [s@macro-exten-vm:9] Macro(“SIP/66.193.176.35-00000002”, “dial,15,tr,600”) in new stack
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – Executing [s@macro-dial:1] GotoIf(“SIP/66.193.176.35-00000002”, “1?dial”) in new stack
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – Goto (macro-dial,s,3)
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – Executing [s@macro-dial:3] AGI(“SIP/66.193.176.35-00000002”, “dialparties.agi”) in new stack
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[Feb 8 09:42:55] VERBOSE[3515] logger.c: dialparties.agi: Starting New Dialparties.agi
[Feb 8 09:42:55] VERBOSE[3515] logger.c: dialparties.agi: Caller ID name is ‘M G TECHNOLOGIE’ number is ‘14092551234’
[Feb 8 09:42:55] VERBOSE[3515] logger.c: dialparties.agi: Methodology of ring is ‘none’
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – dialparties.agi: Added extension 600 to extension map
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – dialparties.agi: Extension 600 cf is disabled
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – dialparties.agi: Extension 600 do not disturb is disabled
[Feb 8 09:42:55] VERBOSE[3515] logger.c: dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE)
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – dialparties.agi: dbset CALLTRACE/600 to 14092551234
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – dialparties.agi: Filtered ARG3: 600
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – <SIP/66.193.176.35-00000002>AGI Script dialparties.agi completed, returning 0
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – Executing [s@macro-dial:7] Dial(“SIP/66.193.176.35-00000002”, “SIP/600,15,tr”) in new stack
[Feb 8 09:42:55] VERBOSE[3515] logger.c: == Using SIP RTP TOS bits 184
[Feb 8 09:42:55] VERBOSE[3515] logger.c: == Using SIP RTP CoS mark 5
[Feb 8 09:42:55] VERBOSE[3515] logger.c: Audio is at 10.246.1.24 port 18192
[Feb 8 09:42:55] VERBOSE[3515] logger.c: Adding codec 0x4 (ulaw) to SDP
[Feb 8 09:42:55] VERBOSE[3515] logger.c: Adding codec 0x8 (alaw) to SDP
[Feb 8 09:42:55] VERBOSE[3515] logger.c: Adding codec 0x2 (gsm) to SDP
[Feb 8 09:42:55] VERBOSE[3515] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 8 09:42:55] VERBOSE[3515] logger.c: Reliably Transmitting (NAT) to 10.246.1.13:3072:
INVITE sip:600@10.246.1.13:3072 SIP/2.0
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK1bce35c9;rport
Max-Forwards: 70
From: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
To: sip:600@10.246.1.13:3072
Contact: sip:14092551234@10.246.1.24
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Date: Mon, 08 Feb 2010 17:42:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 1228995745 1228995745 IN IP4 10.246.1.24
s=Asterisk PBX 1.6.0.21
c=IN IP4 10.246.1.24
t=0 0
m=audio 18192 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


[Feb 8 09:42:55] VERBOSE[3515] logger.c: – Called 600
[Feb 8 09:42:55] VERBOSE[3515] logger.c:
<— Transmitting (NAT) to 204.11.192.37:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com;tag=as61cfc947
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Length: 0

<------------>
[Feb 8 09:42:55] VERBOSE[2834] logger.c:
<— SIP read from UDP://10.246.1.13:3072 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK1bce35c9;rport=5060
From: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
To: sip:600@10.246.1.13:3072;tag=jb93f9erih
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 102 INVITE
Contact: sip:600@10.246.1.13:3072;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

<------------->
[Feb 8 09:42:55] VERBOSE[2834] logger.c: — (10 headers 0 lines) —
[Feb 8 09:42:55] VERBOSE[3515] logger.c: – SIP/600-00000003 is ringing
[Feb 8 09:42:55] VERBOSE[3515] logger.c:
<— Transmitting (NAT) to 204.11.192.37:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com;tag=as61cfc947
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Length: 0

<------------>
[Feb 8 09:42:56] VERBOSE[2834] logger.c:
<— SIP read from UDP://10.246.1.13:3072 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK1bce35c9;rport=5060
From: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
To: sip:600@10.246.1.13:3072;tag=jb93f9erih
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 102 INVITE
Contact: sip:600@10.246.1.13:3072;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

<------------->
[Feb 8 09:42:56] VERBOSE[2834] logger.c: — (10 headers 0 lines) —
[Feb 8 09:42:56] VERBOSE[3515] logger.c: – SIP/600-00000003 is ringing
[Feb 8 09:42:57] VERBOSE[2834] logger.c:
<— SIP read from UDP://10.246.1.13:3072 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK1bce35c9;rport=5060
From: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
To: sip:600@10.246.1.13:3072;tag=jb93f9erih
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 102 INVITE
Contact: sip:600@10.246.1.13:3072;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

<------------->
[Feb 8 09:42:57] VERBOSE[2834] logger.c: — (10 headers 0 lines) —
[Feb 8 09:42:57] VERBOSE[3515] logger.c: – SIP/600-00000003 is ringing
[Feb 8 09:42:59] VERBOSE[2834] logger.c:
<— SIP read from UDP://10.246.1.13:3072 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK1bce35c9;rport=5060
From: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
To: sip:600@10.246.1.13:3072;tag=jb93f9erih
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 102 INVITE
Contact: sip:600@10.246.1.13:3072;reg-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Content-Length: 0

<------------->
[Feb 8 09:42:59] VERBOSE[2834] logger.c: — (10 headers 0 lines) —
[Feb 8 09:42:59] VERBOSE[3515] logger.c: – SIP/600-00000003 is ringing
[Feb 8 09:42:59] VERBOSE[2834] logger.c:
<— SIP read from UDP://10.246.1.13:3072 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK1bce35c9;rport=5060
From: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
To: sip:600@10.246.1.13:3072;tag=jb93f9erih
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 102 INVITE
Contact: sip:600@10.246.1.13:3072;reg-id=1
User-Agent: snom870/8.3.6
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 977943656 977943657 IN IP4 10.246.1.13
s=call
c=IN IP4 10.246.1.13
t=0 0
m=audio 51014 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
[Feb 8 09:42:59] VERBOSE[2834] logger.c: — (13 headers 13 lines) —
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Found RTP audio format 0
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Found RTP audio format 8
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Found RTP audio format 3
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Found RTP audio format 101
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Found audio description format pcmu for ID 0
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Found audio description format pcma for ID 8
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Found audio description format gsm for ID 3
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Found audio description format telephone-event for ID 101
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Peer audio RTP is at port 10.246.1.13:51014
[Feb 8 09:42:59] VERBOSE[2834] logger.c: list_route: hop: sip:600@10.246.1.13:3072
[Feb 8 09:42:59] VERBOSE[2834] logger.c: set_destination: Parsing sip:600@10.246.1.13:3072 for address/port to send to
[Feb 8 09:42:59] VERBOSE[2834] logger.c: set_destination: set destination to 10.246.1.13, port 3072
[Feb 8 09:42:59] VERBOSE[2834] logger.c: Transmitting (NAT) to 10.246.1.13:3072:
ACK sip:600@10.246.1.13:3072 SIP/2.0
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK30708441;rport
Max-Forwards: 70
From: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
To: sip:600@10.246.1.13:3072;tag=jb93f9erih
Contact: sip:14092551234@10.246.1.24
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.21
Content-Length: 0


[Feb 8 09:42:59] VERBOSE[3515] logger.c: – SIP/600-00000003 answered SIP/66.193.176.35-00000002
[Feb 8 09:42:59] VERBOSE[3515] logger.c: Audio is at 10.246.1.24 port 18984
[Feb 8 09:42:59] VERBOSE[3515] logger.c: Adding codec 0x4 (ulaw) to SDP
[Feb 8 09:42:59] VERBOSE[3515] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 8 09:42:59] VERBOSE[3515] logger.c:
<— Reliably Transmitting (NAT) to 204.11.192.37:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com;tag=as61cfc947
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1248691410 1248691410 IN IP4 10.246.1.24
s=Asterisk PBX 1.6.0.21
c=IN IP4 10.246.1.24
t=0 0
m=audio 18984 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Feb 8 09:43:00] VERBOSE[2834] logger.c: Retransmitting #1 (NAT) to 204.11.192.37:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com;tag=as61cfc947
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1248691410 1248691410 IN IP4 10.246.1.24
s=Asterisk PBX 1.6.0.21
c=IN IP4 10.246.1.24
t=0 0
m=audio 18984 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


[Feb 8 09:43:01] VERBOSE[2834] logger.c: Retransmitting #2 (NAT) to 204.11.192.37:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com;tag=as61cfc947
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1248691410 1248691410 IN IP4 10.246.1.24
s=Asterisk PBX 1.6.0.21
c=IN IP4 10.246.1.24
t=0 0
m=audio 18984 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


[Feb 8 09:43:03] VERBOSE[2834] logger.c: Retransmitting #3 (NAT) to 204.11.192.37:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com;tag=as61cfc947
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1248691410 1248691410 IN IP4 10.246.1.24
s=Asterisk PBX 1.6.0.21
c=IN IP4 10.246.1.24
t=0 0
m=audio 18984 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


[Feb 8 09:43:04] VERBOSE[2834] logger.c:
<— SIP read from UDP://10.246.1.13:3072 —>
BYE sip:14092551234@10.246.1.24 SIP/2.0
Via: SIP/2.0/UDP 10.246.1.13:3072;branch=z9hG4bK-lw1kviceoib4;rport
From: sip:600@10.246.1.13:3072;tag=jb93f9erih
To: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 1 BYE
Max-Forwards: 70
Contact: sip:600@10.246.1.13:3072;reg-id=1
User-Agent: snom870/8.3.6
RTP-RxStat: Total_Rx_Pkts=0,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=0,Tx_Pkts=251,Remote_Tx_Pkts=0
Content-Length: 0

<------------->
[Feb 8 09:43:04] VERBOSE[2834] logger.c: — (12 headers 0 lines) —
[Feb 8 09:43:04] VERBOSE[2834] logger.c: Sending to 10.246.1.13 : 3072 (NAT)
[Feb 8 09:43:04] VERBOSE[2834] logger.c:
<— Transmitting (NAT) to 10.246.1.13:3072 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.246.1.13:3072;branch=z9hG4bK-lw1kviceoib4;received=10.246.1.13;rport=3072
From: sip:600@10.246.1.13:3072;tag=jb93f9erih
To: “M G TECHNOLOGIE” sip:14092551234@10.246.1.24;tag=as4ee6573b
Call-ID: 551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24
CSeq: 1 BYE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Feb 8 09:43:04] VERBOSE[3515] logger.c: – Executing [h@macro-dial:1] Macro(“SIP/66.193.176.35-00000002”, “hangupcall”) in new stack
[Feb 8 09:43:04] VERBOSE[3515] logger.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/66.193.176.35-00000002”, “1?skiprg”) in new stack
[Feb 8 09:43:04] VERBOSE[3515] logger.c: – Goto (macro-hangupcall,s,4)
[Feb 8 09:43:04] VERBOSE[3515] logger.c: – Executing [s@macro-hangupcall:4] GotoIf(“SIP/66.193.176.35-00000002”, “1?skipblkvm”) in new stack
[Feb 8 09:43:04] VERBOSE[3515] logger.c: – Goto (macro-hangupcall,s,7)
[Feb 8 09:43:04] VERBOSE[3515] logger.c: – Executing [s@macro-hangupcall:7] GotoIf(“SIP/66.193.176.35-00000002”, “1?theend”) in new stack
[Feb 8 09:43:04] VERBOSE[3515] logger.c: – Goto (macro-hangupcall,s,9)
[Feb 8 09:43:04] VERBOSE[3515] logger.c: – Executing [s@macro-hangupcall:9] Hangup(“SIP/66.193.176.35-00000002”, “”) in new stack
[Feb 8 09:43:04] VERBOSE[3515] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/66.193.176.35-00000002’ in macro ‘hangupcall’
[Feb 8 09:43:04] VERBOSE[3515] logger.c: == Spawn extension (macro-dial, h, 1) exited non-zero on ‘SIP/66.193.176.35-00000002’
[Feb 8 09:43:04] VERBOSE[3515] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/66.193.176.35-00000002’ in macro ‘dial’
[Feb 8 09:43:04] VERBOSE[3515] logger.c: == Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/66.193.176.35-00000002’ in macro ‘exten-vm’
[Feb 8 09:43:04] VERBOSE[3515] logger.c: == Spawn extension (from-did-direct, 600, 1) exited non-zero on ‘SIP/66.193.176.35-00000002’
[Feb 8 09:43:04] VERBOSE[3515] logger.c: Scheduling destruction of SIP dialog '48061422-3474634843-426617@msw1.telengy.net’ in 32000 ms (Method: INVITE)
[Feb 8 09:43:04] VERBOSE[2834] logger.c: Reliably Transmitting (NAT) to 204.11.192.22:5060:
OPTIONS sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK67fefbea;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.246.1.24;tag=as0b7523e0
To: sip:callcentric.com
Contact: sip:Unknown@10.246.1.24
Call-ID: 47fff3ce6c7677372ba1113239584c91@10.246.1.24
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.21
Date: Mon, 08 Feb 2010 17:43:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


[Feb 8 09:43:04] VERBOSE[2834] logger.c: Really destroying SIP dialog ‘551721263e49b3dc3e1ea61f27f6cc76@10.246.1.24’ Method: BYE
[Feb 8 09:43:04] VERBOSE[2834] logger.c:
<— SIP read from UDP://204.11.192.22:5060 —>
SIP/2.0 200 Ok
v: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK67fefbea;rport
f: “Unknown” sip:Unknown@10.246.1.24;tag=as0b7523e0
t: sip:callcentric.com
i: 47fff3ce6c7677372ba1113239584c91@10.246.1.24
CSeq: 102 OPTIONS
l: 0

<------------->
[Feb 8 09:43:04] VERBOSE[2834] logger.c: — (7 headers 0 lines) —
[Feb 8 09:43:04] VERBOSE[2834] logger.c: Really destroying SIP dialog ‘47fff3ce6c7677372ba1113239584c91@10.246.1.24’ Method: OPTIONS
[Feb 8 09:43:07] VERBOSE[2834] logger.c: Retransmitting #4 (NAT) to 204.11.192.37:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com;tag=as61cfc947
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1248691410 1248691410 IN IP4 10.246.1.24
s=Asterisk PBX 1.6.0.21
c=IN IP4 10.246.1.24
t=0 0
m=audio 18984 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


[Feb 8 09:43:09] VERBOSE[2834] logger.c: Reliably Transmitting (NAT) to 10.246.1.13:3072:
OPTIONS sip:600@10.246.1.13:3072 SIP/2.0
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK1f6af7cd;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.246.1.24;tag=as6d4d23d2
To: sip:600@10.246.1.13:3072
Contact: sip:Unknown@10.246.1.24
Call-ID: 0066c6c268b6eaed27af4a375341828a@10.246.1.24
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.21
Date: Mon, 08 Feb 2010 17:43:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


[Feb 8 09:43:09] VERBOSE[2834] logger.c:
<— SIP read from UDP://10.246.1.13:3072 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.246.1.24:5060;branch=z9hG4bK1f6af7cd;rport=5060
From: “Unknown” sip:Unknown@10.246.1.24;tag=as6d4d23d2
To: sip:600@10.246.1.13:3072
Call-ID: 0066c6c268b6eaed27af4a375341828a@10.246.1.24
CSeq: 102 OPTIONS
Contact: sip:600@10.246.1.13:3072;reg-id=1
User-Agent: snom870/8.3.6
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0

<------------->
[Feb 8 09:43:09] VERBOSE[2834] logger.c: — (14 headers 0 lines) —
[Feb 8 09:43:09] VERBOSE[2834] logger.c: Really destroying SIP dialog ‘0066c6c268b6eaed27af4a375341828a@10.246.1.24’ Method: OPTIONS
[Feb 8 09:43:11] VERBOSE[2834] logger.c: Retransmitting #5 (NAT) to 204.11.192.37:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.11.192.37:5060;branch=z9hG4bK-d97f0ea0ee0625b3f3ce9afbf8fc069b;received=204.11.192.37
From: “M G TECHNOLOGIE” sip:14092551234@66.193.176.35;tag=3474634843-426645
To: sip:14151234567@ss.callcentric.com;tag=as61cfc947
Call-ID: 48061422-3474634843-426617@msw1.telengy.net
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: sip:17777654321@10.246.1.24
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1248691410 1248691410 IN IP4 10.246.1.24
s=Asterisk PBX 1.6.0.21
c=IN IP4 10.246.1.24
t=0 0
m=audio 18984 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

You can sometimes use wireshark. However, if this is a codec problem, the sip set debug output will be more informative.

I posted a link to the SIP bug reporting guidelines earlier today.

Does the post above have what you need? BTW, Asterisk is 10.246.1.24 and the phone is .13.

The device at from 204.11.192.37:5060 is ignoring everything that Asterisk is replying to it. I’d suspect you have some sort of routing or firewall problem on that side. Are you sure that the …37 is actually generating ringback tone, as it would seem strange that it should accept 180 Ringing, but ignore 6 attempts to send 200 OK. Maybe it fakes ringing immediately.

The 204 address is the ITSP and I have no control over that, keep in mind all this works fine with the SPA9000 or phone directly to the ITSP, it’s only with Asterisk. I have a Linksys RV042 with many ports open that I found documented for Asterisk.

Asterisk is doing the right thing, according to that trace.

Are you sure that you have a route to your router configured into the OS that is hosting Asterisk?

Also, something is modifying the incoming SIP from the ITSP, to replace the public IP address with the private one. Are you sure that that something is doing the reverse, going the other way. If not, you would be better not having it do it at all, and letting Asterisk handle NAT issues.

Specifically, Asterisk sending a local address in the Contact: header.

The router (Linksys RV042) has NAT/Firewall turned on as it needs it. I am using 2 DSL links balanced. I did test with one turned off and that made no difference. I passed your info to the ITSP and they did a trace. Their info is below but it also made no difference. Thanks so much for your valuable input. I really appreciate what you’ve done so far and anything else.

I have FreePBX set to NAT ON and IP Config to Public. That machine is temporary and uses DHCP, I can ping back and forth from anywhere. If I get this to work the final machine will have a static address. They address has not changed through the last several days.

From the ITSP:
We are showing that your most recent incoming calls are dropping due to a “Media Timeout” which basically means that you are not sending/ receiving audio in either direction. Please add the following lines within your sip_general_custom.conf file:

videosupport=no
canreinvite=no

Your trace showed you were not getting far enough for canreinvite to make a difference. My guess is that the ITSP is trying to send the ACK to the private address in the Contact header. It is probably also sending the RTP stream there and may be expecting to receive it from there.

You are also not accepting any video codecs.

Your router is doing more than NATting, it is also modifying the first line of the incoming SIP requests. On the other hand, Asterisk doesn’t seem to know its own public address. That means that the router also needs to re-write the private address back to the public address, going the other way.

With that configuration, you probably need to tell Asterisk its public address, or enable it to use STUN to discover it for itself.

Thanks, I an trying to unravel what you mean and what I need to change. BTW, I put a rule in to open everything on the FW and that made no difference. I wonder what Asterisk does differently from the SPA9000 and snom 870 that allow them to work but not Asterisk? Oh yes… I also changed Asterisk to use static IP and the made no difference either.

I spent the last 2 hours trying to setup a different ITSP on Asterisk to see if it’s any different. I have been using both ITSPs for about a year. Works with the SPA9000 or snom 870 fine. With Asterisk and ITSP2 (Voipyourlife), I can call out but not in, but it seems a different problem since I don’t even get a ring. I have the inbound route setup the same as I had with the first ITSP( CallCentric) but no luck so far, it wants to go to a phone number instead of extension 600. I thought if I can get further I can see if the two ITSPs are having the same inbound problem, or not. Many thanks again. Oh, I also turned video on in FreePbx but not any codecs.

[Feb 8 14:59:39] VERBOSE[2835] logger.c: Looking for 4081234567 in from-voipyourphone (domain 99.31.165.5)
[Feb 8 14:59:39] VERBOSE[2835] logger.c: <— Reliably Transmitting (NAT) to 216.143.130.36:5060 —>
SIP/2.0 404 Not Found

host=sip.voipyourphone.com
defaultip=sip.voipyourphone.com
fromdomain=sip.voipyourphone.com
defaultuser=myid
secret=xxxxxx
type=peer
authname=myid
fromuser=myid
allow=ulaw
canrenvite=no
context=from-voipyourphone
insecure=very
nat=yes
qualify=yes

[Feb 8 15:11:10] VERBOSE[2835] logger.c: — (14 headers 0 lines) —
[Feb 8 15:11:10] VERBOSE[2835] logger.c: Really destroying SIP dialog ‘6f1f69886573ed393a82af571fd12649@10.246.1.24’ Method: OPTIONS
[Feb 8 15:11:21] VERBOSE[2835] logger.c:
<— SIP read from UDP://216.143.130.36:5060 —>
INVITE sip:4081234567@99.31.165.5:5060 SIP/2.0
Record-Route: sip:216.143.130.36;lr=on;ftag=as3ceb0510
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKb33f.575f79b1.0
Via: SIP/2.0/UDP 216.143.130.112:5060;received=216.143.130.112;branch=z9hG4bK7dc78602;rport=5060
From: “San Jose CA” sip:4087654321@216.143.130.112;tag=as3ceb0510
To: sip:myid@216.143.130.36
Contact: sip:4087654321@216.143.130.112
Call-ID: 28087ea12edb900b74effcba58742e31@216.143.130.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Mon, 08 Feb 2010 21:49:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 26673 26673 IN IP4 216.143.130.112
s=session
c=IN IP4 216.143.130.112
b=CT:384
t=0 0
m=audio 11794 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 52646 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=sendrecv

<------------->
[Feb 8 15:11:21] VERBOSE[2835] logger.c: — (16 headers 16 lines) —
[Feb 8 15:11:21] VERBOSE[2835] logger.c: == Using SIP RTP TOS bits 184
[Feb 8 15:11:21] VERBOSE[2835] logger.c: == Using SIP RTP CoS mark 5
[Feb 8 15:11:21] VERBOSE[2835] logger.c: == Using SIP VRTP TOS bits 136
[Feb 8 15:11:21] VERBOSE[2835] logger.c: == Using SIP VRTP CoS mark 6
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Sending to 216.143.130.36 : 5060 (NAT)
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Using INVITE request as basis request - 28087ea12edb900b74effcba58742e31@216.143.130.112
[Feb 8 15:11:21] VERBOSE[2835] logger.c: No user ‘4087654321’ in SIP users list
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found peer ‘myid’ for ‘4087654321’ from 216.143.130.36:5060
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found RTP audio format 0
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found RTP audio format 8
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found audio description format PCMU for ID 0
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found audio description format PCMA for ID 8
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found RTP video format 31
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found RTP video format 34
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found video description format H261 for ID 31
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Found video description format H263 for ID 34
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0xc0000 (h261|h263)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Peer audio RTP is at port 216.143.130.112:11794
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Looking for 4081234567 in from-voipyourphone (domain 99.31.165.5)
[Feb 8 15:11:21] VERBOSE[2835] logger.c:
<— Reliably Transmitting (NAT) to 216.143.130.36:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKb33f.575f79b1.0;received=216.143.130.36
Via: SIP/2.0/UDP 216.143.130.112:5060;received=216.143.130.112;branch=z9hG4bK7dc78602;rport=5060
From: “San Jose CA” sip:4087654321@216.143.130.112;tag=as3ceb0510
To: sip:myid@216.143.130.36;tag=as622c0618
Call-ID: 28087ea12edb900b74effcba58742e31@216.143.130.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.21
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Feb 8 15:11:21] NOTICE[2835] chan_sip.c: Call from ‘myid’ to extension ‘4081234567’ rejected because extension not found.
[Feb 8 15:11:21] VERBOSE[2835] logger.c: Scheduling destruction of SIP dialog ‘28087ea12edb900b74effcba58742e31@216.143.130.112’ in 6400 ms (Method: INVITE)
[Feb 8 15:11:21] VERBOSE[2835] logger.c:
<— SIP read from UDP://216.143.130.36:5060 —>
ACK sip:4081234567@99.31.165.5:5060 SIP/2.0
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKb33f.575f79b1.0
From: “San Jose CA” sip:4087654321@216.143.130.112;tag=as3ceb0510
Call-ID: 28087ea12edb900b74effcba58742e31@216.143.130.112
To: sip:myid@216.143.130.36;tag=as622c0618
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.6.1-notls (i386/freebsd))
Content-Length: 0

This was killing me until I opened port 5060 on the router, for sip traffic.

That port has always been open since I had the SPA9000. Wish it was that easy. Thanks anyway!