Extension rings but does not answer call

system · February 8, 2010, 3:09am

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

david55 · February 8, 2010, 12:01pm

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

Do you have any codec restrictions on the phone?

system · February 8, 2010, 4:09pm

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