I always had 1.6.2.13 costantly updated day by day to the most recent SVN. About 15/20 days ago after a regular update no outbound calls were permited. Note no change was made to any of the .conf files.
At this point the only solution I had was to downgrade to 1.6.2.13 r5114 which is the tarball downloadable from asterisk.org website.
Today I have also tryed to upgrade to 1.8 r294084 but this drawback always occurs. Are you aware of some important changes?
I am sending a “sip debug report” for a 1.6.2.13 r5114 (working) and ones for 1.8 r294084 (not working)
1.6.2.13 r5114 (working)
SIP Debugging enabled
Reliably Transmitting (no NAT) to 83.211.227.21:5060:
OPTIONS sip:voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK2d0f288b;rport
Max-Forwards: 70
From: "asterisk"
<sip:asterisk@62.62.62.62:5076>;tag=as0f800792
To: <sip:voip.eutelia.it>
Contact: <sip:asterisk@62.62.62.62:5076>
Call-ID: 14d9fa8a5e7ef2f63262bf4473e70635@62.62.62.62
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 08 Nov 2010 18:14:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK2d0f288b;rport=5076
From: "asterisk" <sip:asterisk@62.62.62.62:5076>;tag=as0f800792
To: <sip:voip.eutelia.it>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.9d7c
Call-ID: 14d9fa8a5e7ef2f63262bf4473e70635@62.62.62.62
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: SPS EUT RM GW 04
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '14d9fa8a5e7ef2f63262bf4473e70635@62.62.62.62' Method: OPTIONS
Reliably Transmitting (no NAT) to 83.211.227.21:5060:
OPTIONS sip:voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK4955f085;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@62.62.62.62:5076>;tag=as44982b18
To: <sip:voip.eutelia.it>
Contact: <sip:asterisk@62.62.62.62:5076>
Call-ID: 140d0d44116b044d48f4490d12864071@62.62.62.62
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 08 Nov 2010 18:14:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK4955f085;rport=5076
From: "asterisk" <sip:asterisk@62.62.62.62:5076>;tag=as44982b18
To: <sip:voip.eutelia.it>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.3cdb
Call-ID: 140d0d44116b044d48f4490d12864071@62.62.62.62
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: SPS EUT RM GW 04
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '140d0d44116b044d48f4490d12864071@62.62.62.62' Method: OPTIONS
<--- SIP read from UDP:192.168.0.101:5060 --->
INVITE sip:035999999@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK80c40151f1
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
Contact: <sip:601@192.168.0.101:5060>
CSeq: 1 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK
Supported: replaces,100rel
Content-Type: application/sdp
User-Agent: CM5K-PHONE (810170)
Content-Length: 326
v=0
o=CMI-SIPUA 53803 0 IN IP4 192.168.0.101
s=SIP CALL
c=IN IP4 192.168.0.101
t=0 0
m=audio 10000 RTP/AVP 0 8 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:30
a=rtcp:10001
a=sendrecv
<------------->
--- (13 headers 16 lines) ---
== Using SIP RTP CoS mark 5
Sending to 192.168.0.101 : 5060 (no NAT)
Using INVITE request as basis request - 480273b018f94cbf171d9c6819077bb3@192.168.0.101
Found peer '601' for '601' from 192.168.0.101:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.101:10000
Looking for 035999999 in DLPN_DialPlan1 (domain 192.168.0.200)
list_route: hop: <sip:601@192.168.0.101:5060>
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK80c40151f1;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:035999999@192.168.0.200>
Content-Length: 0
<------------>
-- Executing [035999999@DLPN_DialPlan1:1] Macro("SIP/601-0000000f", "trunkdial-failover-0.3,SIP/Eutelia3/035999999,SIP/Eutelia4/803
) in new stack
-- Executing [s@macro-trunkdial-failover-0.3:1] GotoIf("SIP/601-0000000f", "0?1-fmsetcid,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/601-0000000f", "1?1-setgbobname,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-setgbobname,1)
-- Executing [1-setgbobname@macro-trunkdial-failover-0.3:1] Set("SIP/601-0000000f", "CALLERID(name)=Company") in new stack
-- Executing [1-setgbobname@macro-trunkdial-failover-0.3:2] Goto("SIP/601-0000000f", "s,3") in new stack
-- Goto (macro-trunkdial-failover-0.3,s,3)
-- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/601-0000000f", "CALLERID(num)=") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:4] GotoIf("SIP/601-0000000f", "0?1-dial,1") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:5] Set("SIP/601-0000000f", "CALLERID(all)=035463968") in new stack
-- Executing [s@macro-trunkdial-failover-0.3:6] Goto("SIP/601-0000000f", "1-dial,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/601-0000000f", "SIP/Eutelia3/035999999") in new stack
== Using SIP RTP CoS mark 5
Audio is at 62.62.62.62 port 10784
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 83.211.227.21:5060:
INVITE sip:035999999@voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK224cdb4e;rport
Max-Forwards: 70
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>
Contact: <sip:03519965215@62.62.62.62:5076>
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 08 Nov 2010 18:14:05 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 240407567 240407567 IN IP4 62.62.62.62
s=Asterisk PBX 1.6.2.13
c=IN IP4 62.62.62.62
t=0 0
m=audio 10784 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
---
-- Called Eutelia3/035999999
<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK224cdb4e;rport=5076
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.9286
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="voip.eutelia.it", nonce="4cd83e29faf773736066adadc3c34f4e65a5993e", qop="auth"
Server: SPS EUT RM GW 04
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 83.211.227.21:5060:
ACK sip:035999999@voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK224cdb4e;rport
Max-Forwards: 70
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.9286
Contact: <sip:03519965215@62.62.62.62:5076>
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
---
Audio is at 62.62.62.62 port 10784
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 83.211.227.21:5060:
INVITE sip:035999999@voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK1beb2041;rport
Max-Forwards: 70
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>
Contact: <sip:03519965215@62.62.62.62:5076>
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Proxy-Authorization: Digest username="03519965215", realm="voip.eutelia.it", algorithm=MD5, uri="sip:035999999@voip.eutelia.it", non
066adadc3c34f4e65a5993e", response="789b75c9e8136c39c0151db53693953d", qop=auth, cnonce="1d73a780", nc=00000001
Date: Mon, 08 Nov 2010 18:14:05 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 240407567 240407568 IN IP4 62.62.62.62
s=Asterisk PBX 1.6.2.13
c=IN IP4 62.62.62.62
t=0 0
m=audio 10784 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
---
Retransmitting #1 (no NAT) to 83.211.227.21:5060:
INVITE sip:035999999@voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK1beb2041;rport
Max-Forwards: 70
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>
Contact: <sip:03519965215@62.62.62.62:5076>
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Proxy-Authorization: Digest username="03519965215", realm="voip.eutelia.it", algorithm=MD5, uri="sip:035999999@voip.eutelia.it", non
066adadc3c34f4e65a5993e", response="789b75c9e8136c39c0151db53693953d", qop=auth, cnonce="1d73a780", nc=00000001
Date: Mon, 08 Nov 2010 18:14:05 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 240407567 240407568 IN IP4 62.62.62.62
s=Asterisk PBX 1.6.2.13
c=IN IP4 62.62.62.62
t=0 0
m=audio 10784 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
---
<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK1beb2041;rport=5076
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 103 INVITE
Server: SPS EUT RM GW 04
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK1beb2041;rport=5076
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 103 INVITE
Server: SPS EUT RM GW 04
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.101:5060 --->
INVITE sip:035999999@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK80c40151f1
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
Contact: <sip:601@192.168.0.101:5060>
CSeq: 1 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK
Supported: replaces,100rel
Content-Type: application/sdp
User-Agent: CM5K-PHONE (810170)
Content-Length: 326
v=0
o=CMI-SIPUA 53803 0 IN IP4 192.168.0.101
s=SIP CALL
c=IN IP4 192.168.0.101
t=0 0
m=audio 10000 RTP/AVP 0 8 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:30
a=rtcp:10001
a=sendrecv
<------------->
--- (13 headers 16 lines) ---
Ignoring this INVITE request
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK80c40151f1;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:035999999@192.168.0.200>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.0.100:5060 --->
REGISTER sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bK3676e938a8
From: "602 Betty" <sip:602@192.168.0.200>;tag=5e16cd01
To: "602 Betty" <sip:602@192.168.0.200>
Call-ID: 52a4b1c43a92e53f0208a92406987b87@192.168.0.100
Contact: <sip:602@192.168.0.100:5060>
CSeq: 549 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest username="602",realm="asterisk",nonce="27f24dbc",response="e2f05b712b4ea53920e564fcd251fc24",uri="sip:192.1
D5
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.100 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.100:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3676e938a8;received=192.168.0.100;rport=5060
From: "602 Betty" <sip:602@192.168.0.200>;tag=5e16cd01
To: "602 Betty" <sip:602@192.168.0.200>;tag=as5c9ba9b4
Call-ID: 52a4b1c43a92e53f0208a92406987b87@192.168.0.100
CSeq: 549 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38567e51"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '52a4b1c43a92e53f0208a92406987b87@192.168.0.100' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.0.100:5060 --->
REGISTER sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bK838378113a
From: "602 Betty" <sip:602@192.168.0.200>;tag=5e16cd01
To: "602 Betty" <sip:602@192.168.0.200>
Call-ID: 52a4b1c43a92e53f0208a92406987b87@192.168.0.100
Contact: <sip:602@192.168.0.100:5060>
CSeq: 550 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest username="602",realm="asterisk",nonce="38567e51",response="b3afc77cef64e0d78a8eb11ff0cb3a9e",uri="sip:192.1
D5
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.100 : 5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.0.100:5060:
OPTIONS sip:602@192.168.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK781fdacb;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.200>;tag=as0ad37f00
To: <sip:602@192.168.0.100:5060>
Contact: <sip:asterisk@192.168.0.200>
Call-ID: 59f6749401610e9e77b6dabe767e319f@192.168.0.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 08 Nov 2010 18:14:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 192.168.0.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK838378113a;received=192.168.0.100;rport=5060
From: "602 Betty" <sip:602@192.168.0.200>;tag=5e16cd01
To: "602 Betty" <sip:602@192.168.0.200>;tag=as5c9ba9b4
Call-ID: 52a4b1c43a92e53f0208a92406987b87@192.168.0.100
CSeq: 550 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:602@192.168.0.100:5060>;expires=60
Date: Mon, 08 Nov 2010 18:14:11 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '52a4b1c43a92e53f0208a92406987b87@192.168.0.100' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.0.101:5060 --->
REGISTER sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bKdc9ae581c7
From: "601 Alex" <sip:601@192.168.0.200>;tag=175128ca
To: "601 Alex" <sip:601@192.168.0.200>
Call-ID: 7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101
Contact: <sip:601@192.168.0.101:5060>
CSeq: 549 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest username="601",realm="asterisk",nonce="289076ab",response="13a8cacb6eee000ce27209050b2d1214",uri="sip:192.1
D5
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.101 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKdc9ae581c7;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=175128ca
To: "601 Alex" <sip:601@192.168.0.200>;tag=as71f363db
Call-ID: 7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101
CSeq: 549 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="159d1023"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.0.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;rport=5060;received=192.168.0.200;branch=z9hG4bK781fdacb
From: "asterisk" <sip:asterisk@192.168.0.200>;tag=as0ad37f00
To: <sip:602@192.168.0.100:5060>;tag=4d4def7d
Call-ID: 59f6749401610e9e77b6dabe767e319f@192.168.0.200
CSeq: 102 OPTIONS
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '59f6749401610e9e77b6dabe767e319f@192.168.0.200' Method: OPTIONS
<--- SIP read from UDP:192.168.0.101:5060 --->
REGISTER sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK96fb8daf52
From: "601 Alex" <sip:601@192.168.0.200>;tag=175128ca
To: "601 Alex" <sip:601@192.168.0.200>
Call-ID: 7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101
Contact: <sip:601@192.168.0.101:5060>
CSeq: 550 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest username="601",realm="asterisk",nonce="159d1023",response="d45722dac5edf3f9f1afe8acfc31c61c",uri="sip:192.1
D5
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.101 : 5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.0.101:5060:
OPTIONS sip:601@192.168.0.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK4d389fcf;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.200>;tag=as789af9a7
To: <sip:601@192.168.0.101:5060>
Contact: <sip:asterisk@192.168.0.200>
Call-ID: 2e3d74b24e63f2850fb6306125f3f7b3@192.168.0.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 08 Nov 2010 18:14:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK96fb8daf52;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=175128ca
To: "601 Alex" <sip:601@192.168.0.200>;tag=as71f363db
Call-ID: 7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101
CSeq: 550 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:601@192.168.0.101:5060>;expires=60
Date: Mon, 08 Nov 2010 18:14:12 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.0.101:5060 --->
REGISTER sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK96fb8daf52
From: "601 Alex" <sip:601@192.168.0.200>;tag=175128ca
To: "601 Alex" <sip:601@192.168.0.200>
Call-ID: 7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101
Contact: <sip:601@192.168.0.101:5060>
CSeq: 550 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest username="601",realm="asterisk",nonce="159d1023",response="d45722dac5edf3f9f1afe8acfc31c61c",uri="sip:192.1
D5
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.101 : 5060 (no NAT)
[Nov 8 19:14:12] NOTICE[1016]: chan_sip.c:12841 check_auth: Correct auth, but based on stale nonce received from '"601 Alex" <si
tag=175128ca'
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK96fb8daf52;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=175128ca
To: "601 Alex" <sip:601@192.168.0.200>;tag=as71f363db
Call-ID: 7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101
CSeq: 550 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47f3e2c0", stale=true
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7e3dce090ba1f19f34fe03b010a432fe@192.168.0.101' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.0.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;rport=5060;received=192.168.0.200;branch=z9hG4bK4d389fcf
From: "asterisk" <sip:asterisk@192.168.0.200>;tag=as789af9a7
To: <sip:601@192.168.0.101:5060>;tag=0a94d168
Call-ID: 2e3d74b24e63f2850fb6306125f3f7b3@192.168.0.200
CSeq: 102 OPTIONS
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '2e3d74b24e63f2850fb6306125f3f7b3@192.168.0.200' Method: OPTIONS
<--- SIP read from UDP:192.168.0.101:5060 --->
INVITE sip:035999999@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK80c40151f1
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
Contact: <sip:601@192.168.0.101:5060>
CSeq: 1 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK
Supported: replaces,100rel
Content-Type: application/sdp
User-Agent: CM5K-PHONE (810170)
Content-Length: 326
v=0
o=CMI-SIPUA 53803 0 IN IP4 192.168.0.101
s=SIP CALL
c=IN IP4 192.168.0.101
t=0 0
m=audio 10000 RTP/AVP 0 8 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:30
a=rtcp:10001
a=sendrecv
<------------->
--- (13 headers 16 lines) ---
Ignoring this INVITE request
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK80c40151f1;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:035999999@192.168.0.200>
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.0.105:5060 --->
REGISTER sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.105:5060;rport;branch=z9hG4bK65ed27afb2
From: "611 Box" <sip:611@192.168.0.200>;tag=597e0186
To: "611 Box" <sip:611@192.168.0.200>
Call-ID: 0c667d981cbf8c8463a23cda30493f50@192.168.0.105
Contact: <sip:611@192.168.0.105:5060>
CSeq: 549 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest username="611",realm="asterisk",nonce="5f6a05d8",response="e5fe587366aac0c9408e681268a39fd6",uri="sip:192.1
D5
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.105 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bK65ed27afb2;received=192.168.0.105;rport=5060
From: "611 Box" <sip:611@192.168.0.200>;tag=597e0186
To: "611 Box" <sip:611@192.168.0.200>;tag=as59048377
Call-ID: 0c667d981cbf8c8463a23cda30493f50@192.168.0.105
CSeq: 549 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="321d4bf2"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '0c667d981cbf8c8463a23cda30493f50@192.168.0.105' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 62.62.62.62:5076;received=62.62.62.62;branch=z9hG4bK1beb2041;rport=5076
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>;tag=5CD9ADAC-20E0
Date: Mon, 08 Nov 2010 18:14:05 GMT
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:491035999999@62.94.71.28:5060>
Record-Route: <sip:83.211.227.21;lr=on;ftag=as1e334b3f>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 248
v=0
o=CiscoSystemsSIP-GW-UserAgent 2222 9013 IN IP4 62.94.71.28
s=SIP Call
c=IN IP4 83.211.227.13
t=0 0
m=audio 54696 RTP/AVP 0 101
c=IN IP4 83.211.227.13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (15 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 83.211.227.13:54696
<--- SIP read from UDP:192.168.0.105:5060 --->
REGISTER sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.105:5060;rport;branch=z9hG4bKbcade2f6ec
From: "611 Box" <sip:611@192.168.0.200>;tag=597e0186
To: "611 Box" <sip:611@192.168.0.200>
Call-ID: 0c667d981cbf8c8463a23cda30493f50@192.168.0.105
Contact: <sip:611@192.168.0.105:5060>
CSeq: 550 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest username="611",realm="asterisk",nonce="321d4bf2",response="447406781d6df9c982a54e868e08f8e1",uri="sip:192.1
D5
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.105 : 5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.0.105:5060:
OPTIONS sip:611@192.168.0.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK1c127d35;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.200>;tag=as46b489ba
To: <sip:611@192.168.0.105:5060>
Contact: <sip:asterisk@192.168.0.200>
Call-ID: 7103404c53275dce6b7859016961a46b@192.168.0.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 08 Nov 2010 18:14:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bKbcade2f6ec;received=192.168.0.105;rport=5060
From: "611 Box" <sip:611@192.168.0.200>;tag=597e0186
To: "611 Box" <sip:611@192.168.0.200>;tag=as59048377
Call-ID: 0c667d981cbf8c8463a23cda30493f50@192.168.0.105
CSeq: 550 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:611@192.168.0.105:5060>;expires=60
Date: Mon, 08 Nov 2010 18:14:15 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '0c667d981cbf8c8463a23cda30493f50@192.168.0.105' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.0.105:5060 --->
REGISTER sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.105:5060;rport;branch=z9hG4bKbcade2f6ec
From: "611 Box" <sip:611@192.168.0.200>;tag=597e0186
To: "611 Box" <sip:611@192.168.0.200>
Call-ID: 0c667d981cbf8c8463a23cda30493f50@192.168.0.105
Contact: <sip:611@192.168.0.105:5060>
CSeq: 550 REGISTER
Max-Forwards: 70
Expires: 60
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
Authorization: Digest username="611",realm="asterisk",nonce="321d4bf2",response="447406781d6df9c982a54e868e08f8e1",uri="sip:192.1
D5
User-Agent: CM5K-PHONE (810170)
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.0.105 : 5060 (no NAT)
[Nov 8 19:14:15] NOTICE[1016]: chan_sip.c:12841 check_auth: Correct auth, but based on stale nonce received from '"611 Box" <sip
ag=597e0186'
<--- Transmitting (no NAT) to 192.168.0.105:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.105:5060;branch=z9hG4bKbcade2f6ec;received=192.168.0.105;rport=5060
From: "611 Box" <sip:611@192.168.0.200>;tag=597e0186
To: "611 Box" <sip:611@192.168.0.200>;tag=as59048377
Call-ID: 0c667d981cbf8c8463a23cda30493f50@192.168.0.105
CSeq: 550 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76cd7c69", stale=true
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '0c667d981cbf8c8463a23cda30493f50@192.168.0.105' in 32000 ms (Method: REGISTER)
-- SIP/Eutelia3-00000010 is making progress passing it to SIP/601-0000000f
Audio is at 192.168.0.200 port 10512
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK80c40151f1;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>;tag=as48a20915
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:035999999@192.168.0.200>
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 294325996 294325996 IN IP4 192.168.0.200
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.0.200
t=0 0
m=audio 10512 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:192.168.0.105:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;rport=5060;received=192.168.0.200;branch=z9hG4bK1c127d35
From: "asterisk" <sip:asterisk@192.168.0.200>;tag=as46b489ba
To: <sip:611@192.168.0.105:5060>;tag=2f575813
Call-ID: 7103404c53275dce6b7859016961a46b@192.168.0.200
CSeq: 102 OPTIONS
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '7103404c53275dce6b7859016961a46b@192.168.0.200' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.0.102:5060:
OPTIONS sip:603@192.168.0.102;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK4e63390b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.200>;tag=as31220c6f
To: <sip:603@192.168.0.102;transport=UDP>
Contact: <sip:asterisk@192.168.0.200>
Call-ID: 26594f8e7371c9fc78b9c21148480ba6@192.168.0.200
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 08 Nov 2010 18:14:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:192.168.0.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK4e63390b;rport=5060;received=192.168.0.200
To: <sip:603@192.168.0.102>;tag=ek782vgttlhc6uj0uhhl
From: "asterisk" <sip:asterisk@192.168.0.200>;tag=as31220c6f
Call-ID: 26594f8e7371c9fc78b9c21148480ba6@192.168.0.200
CSeq: 102 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '26594f8e7371c9fc78b9c21148480ba6@192.168.0.200' Method: OPTIONS
<--- SIP read from UDP:192.168.0.101:5060 --->
CANCEL sip:035999999@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK80c40151f1
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
Contact: <sip:601@192.168.0.101:5060>
CSeq: 1 CANCEL
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Sending to 192.168.0.101 : 5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK80c40151f1;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>;tag=as48a20915
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK80c40151f1;received=192.168.0.101;rport=5060
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>;tag=as48a20915
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
CSeq: 1 CANCEL
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Nov 8 19:14:18] WARNING[24020]: app_dial.c:1028 wait_for_answer: Unable to write frame
[Nov 8 19:14:18] WARNING[24020]: app_dial.c:1028 wait_for_answer: Unable to write frame
[Nov 8 19:14:18] WARNING[24020]: app_dial.c:1028 wait_for_answer: Unable to write frame
[Nov 8 19:14:18] WARNING[24020]: app_dial.c:1028 wait_for_answer: Unable to write frame
[Nov 8 19:14:18] WARNING[24020]: app_dial.c:1028 wait_for_answer: Unable to write frame
[Nov 8 19:14:18] WARNING[24020]: app_dial.c:1028 wait_for_answer: Unable to write frame
Scheduling destruction of SIP dialog '774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 83.211.227.21:5060:
CANCEL sip:035999999@voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK1beb2041;rport
Max-Forwards: 70
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
---
Scheduling destruction of SIP dialog '774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.0.101:5060 --->
ACK sip:035999999@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK80c40151f1
From: "601 Alex" <sip:601@192.168.0.200>;tag=55d2785c
To: <sip:035999999@192.168.0.200>;tag=as48a20915
Call-ID: 480273b018f94cbf171d9c6819077bb3@192.168.0.101
Contact: <sip:601@192.168.0.101:5060>
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 200 canceling
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK1beb2041;rport=5076
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>;tag=4b366c6191700ae3d7c4537c33bc6b4d-e436
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 103 CANCEL
Server: SPS EUT RM GW 04
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
== Spawn extension (macro-trunkdial-failover-0.3, 1-dial, 1) exited non-zero on 'SIP/601-0000000f' in macro 'trunkdial-failover
== Spawn extension (DLPN_DialPlan1, 035999999, 1) exited non-zero on 'SIP/601-0000000f'
<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 62.62.62.62:5076;received=62.62.62.62;branch=z9hG4bK1beb2041;rport=5076
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>;tag=5CD9ADAC-20E0
Date: Mon, 08 Nov 2010 18:14:19 GMT
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 83.211.227.21:5060:
ACK sip:035999999@voip.eutelia.it SIP/2.0
Via: SIP/2.0/UDP 62.62.62.62:5076;branch=z9hG4bK1beb2041;rport
Max-Forwards: 70
From: "035463968" <sip:03519965215@voip.eutelia.it>;tag=as1e334b3f
To: <sip:035999999@voip.eutelia.it>;tag=5CD9ADAC-20E0
Contact: <sip:03519965215@62.62.62.62:5076>
Call-ID: 774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0
---
Really destroying SIP dialog '774ffb6a13e43e160923e4256826fbd5@voip.eutelia.it' Method: INVITE
Really destroying SIP dialog '480273b018f94cbf171d9c6819077bb3@192.168.0.101' Method: ACK
server1*CLI> sip set debug off1.8 r294084 (not working)
[code]server1*CLI> sip set debug on
SIP Debugging enabled
<— SIP read from UDP:192.168.0.101:5060 —>
INVITE sip:035999999@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK800960e1f8
From: “601 Alex” sip:601@192.168.0.200;tag=6b9a2eb0
To: sip:035999999@192.168.0.200
Call-ID: 5e3ddcdf2f6fe4cf5b16ea5d3d74ae63@192.168.0.101
Contact: sip:601@192.168.0.101:5060
CSeq: 1 INVITE
Max-Forwards: 70
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE,PRACK
Supported: replaces,100rel
Content-Type: application/sdp
User-Agent: CM5K-PHONE (810170)
Content-Length: 326
v=0
o=CMI-SIPUA 35406 0 IN IP4 192.168.0.101
s=SIP CALL
c=IN IP4 192.168.0.101
t=0 0
m=audio 10000 RTP/AVP 0 8 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:30
a=rtcp:10001
a=sendrecv
<------------->
— (13 headers 16 lines) —
Sending to 192.168.0.101:5060 (no NAT)
Using INVITE request as basis request - 5e3ddcdf2f6fe4cf5b16ea5d3d74ae63@192.168.0.101
Found peer ‘601’ for ‘601’ from 192.168.0.101:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combi
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.101:10000
Looking for 035999999 in DLPN_DialPlan1 (domain 192.168.0.200)
list_route: hop: sip:601@192.168.0.101:5060
<— Transmitting (no NAT) to 192.168.0.101:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK800960e1f8;received=192.168.0.101;rport=5060
From: “601 Alex” sip:601@192.168.0.200;tag=6b9a2eb0
To: sip:035999999@192.168.0.200
Call-ID: 5e3ddcdf2f6fe4cf5b16ea5d3d74ae63@192.168.0.101
CSeq: 1 INVITE
Server: Asterisk PBX SVN-branch-1.8-r294084
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:035999999@192.168.0.200:5060
Content-Length: 0
<------------>
– Executing [035999999@DLPN_DialPlan1:1] Macro(“SIP/601-0000000a”, “trunkdial-failover-0.3,SIP/Eutelia3/035999999,SIP/Eut
,Eutelia4”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:1] GotoIf(“SIP/601-0000000a”, “0?1-fmsetcid,1”) in new stack
– Executing [s@macro-trunkdial-failover-0.3:2] GotoIf(“SIP/601-0000000a”, “1?1-setgbobname,1”) in new stack
– Goto (macro-trunkdial-failover-0.3,1-setgbobname,1)
– Auto fallthrough, channel ‘SIP/601-0000000a’ status is 'UNKNOWN’
Scheduling destruction of SIP dialog ‘5e3ddcdf2f6fe4cf5b16ea5d3d74ae63@192.168.0.101’ in 6400 ms (Method: INVITE)
<— Reliably Transmitting (no NAT) to 192.168.0.101:5060 —>
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK800960e1f8;received=192.168.0.101;rport=5060
From: “601 Alex” sip:601@192.168.0.200;tag=6b9a2eb0
To: sip:035999999@192.168.0.200;tag=as56de80ef
Call-ID: 5e3ddcdf2f6fe4cf5b16ea5d3d74ae63@192.168.0.101
CSeq: 1 INVITE
Server: Asterisk PBX SVN-branch-1.8-r294084
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Retransmitting #1 (no NAT) to 192.168.0.101:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK800960e1f8;received=192.168.0.101;rport=5060
From: “601 Alex” sip:601@192.168.0.200;tag=6b9a2eb0
To: sip:035999999@192.168.0.200;tag=as56de80ef
Call-ID: 5e3ddcdf2f6fe4cf5b16ea5d3d74ae63@192.168.0.101
CSeq: 1 INVITE
Server: Asterisk PBX SVN-branch-1.8-r294084
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:192.168.0.101:5060 —>
ACK sip:035999999@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK800960e1f8
From: “601 Alex” sip:601@192.168.0.200;tag=6b9a2eb0
To: sip:035999999@192.168.0.200;tag=as56de80ef
Call-ID: 5e3ddcdf2f6fe4cf5b16ea5d3d74ae63@192.168.0.101
Contact: sip:601@192.168.0.101:5060
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:192.168.0.101:5060 —>
ACK sip:035999999@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;rport;branch=z9hG4bK800960e1f8
From: “601 Alex” sip:601@192.168.0.200;tag=6b9a2eb0
To: sip:035999999@192.168.0.200;tag=as56de80ef
Call-ID: 5e3ddcdf2f6fe4cf5b16ea5d3d74ae63@192.168.0.101
Contact: sip:601@192.168.0.101:5060
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
— (9 headers 0 lines) —
server1*CLI> sip set debug off[/code]