Alberto Pastore
2007-Apr-12 07:17 UTC
[asterisk-users] Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
Hi.
I'm stuck into an odd situation.
Here's what happens:
4 Thomson ST2030S
2 Cisco 7912
3 Cisco 7940
2 AAstra 480i
Asterisk 1.2.17
Diva 4BRI + chan_capi
I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17.
Until yesterday, everything was just fine with 1.2.13.
Immediately after the upgrade, *all* the 7940 are no more able to
make calls, just receive them, while 7912 models as well as any
other phone work fine.
Firmware on 7940 is 8.6 (the latest one).
The configuration for asterisk is really simple. After many hours
guessing and reloading configuration changes, I've traced the full
debug output from both asterisk logger and one 7940.
Here's what happens
1) I dial the number on the 7940 (which, by the way
is regularly registered as a peer and REACHABLE by asterisk)
2) the 7940 sends an INVITE to asterisk
3) Asterisk sends back a "407 Authorization required"
4) The 7940 sends back an ACK
5) The 7940 sends a new INVITE which includes the MD5 challenge response
6) nothing happens in asterisk (nothing logged, even with full debug
enabled)
7) the 7940 retries sending the INVITE many times, until it times out
8) I hang up the handset
What on Earth is happening????
Why is not Asterisk logging the subsequent INVITEs from the phone?
(BTW, these sip packets are logged by iptables, I just wanted to make
sure they were received on the asterisk ethernet interface)
######################################################################
Here's an extract from asterisk log:
######################################################################
smtp-ms*CLI>
<-- SIP read from 10.0.10.136:50393:
INVITE sip:212@10.0.10.5;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5
From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d
To: <sip:212@10.0.10.5;user=phone>
Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
Max-Forwards: 70
Date: Thu, 12 Apr 2007 13:39:56 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:215@10.0.10.136:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "Cisco 7940"
<sip:215@10.0.10.5>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 19361 0 IN IP4 10.0.10.136
s=SIP Call
t=0 0
m=audio 16946 RTP/AVP 8 0 18 101
c=IN IP4 10.0.10.136
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
0: INVITE sip:212@10.0.10.5;user=phone SIP/2.0 (43)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
1: Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5 (56)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
2: From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d (76)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
3: To: <sip:212@10.0.10.5;user=phone> (34)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
4: Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136 (56)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
5: Max-Forwards: 70 (16)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
6: Date: Thu, 12 Apr 2007 13:39:56 GMT (35)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
7: CSeq: 101 INVITE (16)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
8: User-Agent: Cisco-CP7940G/8.0 (29)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
9: Contact: <sip:215@10.0.10.136:5060;transport=udp> (49)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
10: Expires: 180 (12)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
11: Accept: application/sdp (23)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
12: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE (65)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
13: Remote-Party-ID: "Cisco 7940"
<sip:215@10.0.10.5>;party=calling;id-type=subscriber;privacy=off;screen=yes
(105)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
14: Supported: replaces,join,norefersub (35)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
15: Content-Length: 274 (19)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
16: Content-Type: application/sdp (29)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
17: Content-Disposition: session;handling=optional (46)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
18: (0)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
v=0 (3)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
o=Cisco-SIPUA 19361 0 IN IP4 10.0.10.136 (40)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
s=SIP Call (10)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
t=0 0 (5)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
m=audio 16946 RTP/AVP 8 0 18 101 (32)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
c=IN IP4 10.0.10.136 (20)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
a=rtpmap:8 PCMA/8000 (20)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
a=rtpmap:0 PCMU/8000 (20)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
a=rtpmap:18 G729/8000 (21)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
a=fmtp:18 annexb=no (19)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
a=rtpmap:101 telephone-event/8000 (33)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
a=fmtp:101 0-15 (15)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3474 parse_request: Line:
a=sendrecv (10)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3191 sip_alloc: Allocating
new SIP dialog for 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136 -
INVITE (With RTP)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:11320 handle_request: ****
Received INVITE (5) - Command in SIP INVITE
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1015 parse_sip_options:
Begin: parsing SIP "Supported: replaces,join,norefersub"
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1027 parse_sip_options:
Found SIP option: -replaces-
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1033 parse_sip_options:
Matched SIP option: replaces
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1027 parse_sip_options:
Found SIP option: -join-
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1033 parse_sip_options:
Matched SIP option: join
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1027 parse_sip_options:
Found SIP option: -norefersub-
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1038 parse_sip_options:
Found no match for SIP option: norefersub (Please file bug report!)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1044 parse_sip_options: *
SIP extension value: 17 for call
0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
Using INVITE request as basis request -
0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
Sending to 10.0.10.136 : 5060 (non-NAT)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:7291 check_user_full:
Setting NAT on RTP to 0
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:7295 check_user_full:
Setting NAT on VRTP to 0
Reliably Transmitting (no NAT) to 10.0.10.136:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.0.10.136:5060;branch=z9hG4bK0cc1ada5;received=10.0.10.136
From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d
To: <sip:212@10.0.10.5;user=phone>;tag=as1ae4df20
Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="msoft",
nonce="733b51d0"
Content-Length: 0
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1307 __sip_reliable_xmit:
*** SIP TIMER: Initalizing retransmit timer on packet: Id #6103
Scheduling destruction of call
'0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136' in 15000 ms
Found user '215'
<-- SIP read from 10.0.10.136:50394:
ACK sip:212@10.0.10.5;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5
From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d
To: <sip:212@10.0.10.5;user=phone>;tag=as1ae4df20
Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
Date: Thu, 12 Apr 2007 13:39:57 GMT
CSeq: 101 ACK
Content-Length: 0
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
0: ACK sip:212@10.0.10.5;user=phone SIP/2.0 (40)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
1: Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5 (56)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
2: From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d (76)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
3: To: <sip:212@10.0.10.5;user=phone>;tag=as1ae4df20 (49)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
4: Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136 (56)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
5: Date: Thu, 12 Apr 2007 13:39:57 GMT (35)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
6: CSeq: 101 ACK (13)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
7: Content-Length: 0 (17)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header
8: (0)
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3239 find_call: = Found
Their Call ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136 Their Tag
0013c3677fdf00ae6752cb07-7fbc304d Our tag: as1ae4df20
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:11320 handle_request: ****
Received ACK (6) - Command in SIP ACK
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1403 __sip_ack: ** SIP
TIMER: Cancelling retransmit of packet (reply received) Retransid #6103
2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1415 __sip_ack: Stopping
retransmission on '0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136' of
Response 101: Match Found
######################################################################
Here's an extract from the 7940 log:
######################################################################
SIPTaskProcessListEvent: cmd = 0x161700
sip_cc_event LINE 0/1: --0x0004ea7d-- :
SIP_STATE_IDLE <- E_CC_SETUP
idle_ev_cc_setup: All digits collected. Placing the call
SIPSM 0/1/18: idle_ev_cc_setup : Setup
SIPSPISendInvite: Sending INVITE...
get_next_request_trx_index: Getting next TRX index, sent = 1
get_next_request_trx_index: Got TRX(0) for sent req
get_last_request_trx_index: Getting last TRX index, sent = 1
get_last_request_trx_index: Got TRX(0) for sent req
get_last_request_trx_index: Getting last TRX index, sent = 1
get_last_request_trx_index: Got TRX(0) for sent req
get_last_request_trx_index: Getting last TRX index, sent = 1
get_last_request_trx_index: Got TRX(0) for sent req
sipTransportSendMessage: ccb <0>: config <10.0.10.5>:<5060> -
remote
<10.0.10.5>:<5060>
sipTransportSendMessage: Got handle 2
sipTransportSendMessage: Opened a one-time UDP send channel to server
<10.0.10.5>:<5060>, handle = 8 local port= 0
sipTransportSendMessage:Sent SIP message to <10.0.10.5>:<5060>,
handle=<8>, length=<1056>, message
INVITE sip:212@10.0.10.5;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5
From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d
To: <sip:212@10.0.10.5;user=phone>
Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
Max-Forwards: 70
Date: Thu, 12 Apr 2007 13:39:56 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:215@10.0.10.136:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "Cisco 7940"
<sip:215@10.0.10.5>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 19361 0 IN IP4 10.0.10.136
s=SIP Call
t=0 0
m=audio 16946 RTP/AVP 8 0 18 101
c=IN IP4 10.0.10.136
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
LINE 0/1: sipTransportSendMessage : Stopping reTx timer
LINE 0/1: sipTransportSendMessage : Starting reTx timer (500
msec)
CHANGE STATE: LINE 0/1: : State
change: SIP_STATE_IDLE -> SIP_STATE_SENT_INVITE
SIPTaskProcessListEvent: cmd = 0x160200
SIPProcessUDPMessage: recv UDP message from <10.0.10.5>:<50195>,
length=<517>, message
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.0.10.136:5060;branch=z9hG4bK0cc1ada5;received=10.0.10.136
From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d
To: <sip:212@10.0.10.5;user=phone>;tag=as1ae4df20
Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="msoft",
nonce="733b51d0"
Content-Length: 0
SIPTaskProcessSIPMessage: Line filter: Determining destination line...
get_method_request_trx_index: Getting TRX for method(INVITE), sent = 1
get_method_request_trx_index: Got TRX(0) for sent method(INVITE)
sip_sm_determine_ccb: Matched branch_id & CSeq
SIPTaskProcessSIPMessage: Line filter: Call ID match: Destination line
= <0/1>.
SIPTaskProcessSIPMessage: Received SIP response.
get_method_request_trx_index: Getting TRX for method(INVITE), sent = 1
get_method_request_trx_index: Got TRX(0) for sent method(INVITE)
sipSPICheckResponse: Response match:
callid=0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136, cseq=101,
cseq_method=INVITE
SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers...
LINE 0/1: sip_sm_check_retx_timers : Stopping reTx timer.
(callid=0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136, cseq=101,
cseq_method=INVITE)
SIPTaskProcessSIPMessage: Recv 4xx/5xx/6xx message.
sip_sm_process_event LINE 0/1: --0x00050839-- :
SIP_STATE_SENT_INVITE <- E_SIP_FAILURE_RESPONSE
LINE 0/1: SIP 407 Proxy Authentication required
get_method_request_trx_index: Getting TRX for method(INVITE), sent = 1
get_method_request_trx_index: Got TRX(0) for sent method(INVITE)
clean_method_request_trx: Removing TRX for method(INVITE), sent = 1
clean_method_request_trx: Removed TRX(0) for method(INVITE)
SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
sipRelDevCoupledMessageStore: Storing for reTx (cseq=101, method=INVITE,
to_tag=<as1ae4df20>)
sipTransportSendMessage: Opened a one-time UDP send channel to server
<10.0.10.5>:<5060>, handle = 8 local port= 0
sipTransportSendMessage:Sent SIP message to <10.0.10.5>:<5060>,
handle=<8>, length=<360>, message
ACK sip:212@10.0.10.5;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK0cc1ada5
From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d
To: <sip:212@10.0.10.5;user=phone>;tag=as1ae4df20
Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
Date: Thu, 12 Apr 2007 13:39:57 GMT
CSeq: 101 ACK
Content-Length: 0
sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
Proxy-Authenticate= Digest algorithm=MD5, realm="msoft",
nonce="733b51d0"
sipSPISendInviteMidCall: Sending INVITE...
sipSPIGenRequestURI: Forming Req-URI (Caller): using original Req-URI
get_next_request_trx_index: Getting next TRX index, sent = 1
get_next_request_trx_index: Got TRX(0) for sent req
get_last_request_trx_index: Getting last TRX index, sent = 1
get_last_request_trx_index: Got TRX(0) for sent req
get_last_request_trx_index: Getting last TRX index, sent = 1
get_last_request_trx_index: Got TRX(0) for sent req
SIPSPIAddRouteHeaders: Route info not available; will not add Route header.
get_last_request_trx_index: Getting last TRX index, sent = 1
get_last_request_trx_index: Got TRX(0) for sent req
sipTransportSendMessage: ccb <0>: config <10.0.10.5>:<5060> -
remote
<10.0.10.5>:<5060>
sipTransportSendMessage: Got handle 2
sipTransportSendMessage: Opened a one-time UDP send channel to server
<10.0.10.5>:<5060>, handle = 8 local port= 0
sipTransportSendMessage:Sent SIP message to <10.0.10.5>:<5060>,
handle=<8>, length=<1224>, message
INVITE sip:212@10.0.10.5;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.10.136:5060;branch=z9hG4bK2e21f6c7
From: "Cisco 7940"
<sip:215@10.0.10.5>;tag=0013c3677fdf00ae6752cb07-7fbc304d
To: <sip:212@10.0.10.5;user=phone>
Call-ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136
Max-Forwards: 70
Date: Thu, 12 Apr 2007 13:39:57 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:215@10.0.10.136:5060;transport=udp>
Proxy-Authorization: Digest
username="215",realm="msoft",uri="sip:212@10.0.10.5;user=phone",response="25d8a11faab3a8e3ff6c7fa74f142475",nonce="733b51d0",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "Cisco 7940"
<sip:215@10.0.10.5>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 19361 0 IN IP4 10.0.10.136
s=SIP Call
t=0 0
m=audio 16946 RTP/AVP 8 0 18 101
c=IN IP4 10.0.10.136
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
LINE 0/1: sipTransportSendMessage : Stopping reTx timer
LINE 0/1: sipTransportSendMessage : Starting reTx timer (500
msec)
SIPTaskProcessListEvent: cmd = 0x0
sip_sm_process_event LINE 0/1: --0x000557d9-- :
SIP_STATE_SENT_INVITE <- E_SIP_TIMER
sipTransportSendMessage: ccb <0>: config <10.0.10.5>:<5060> -
remote
<10.0.10.5>:<5060>
sipTransportSendMessage: Got handle 2
sipTransportSendMessage: Opened a one-time UDP send channel to server
<10.0.10.5>:<5060>, handle = 8 local port= 0
sipTransportSendMessage:Sent SIP message to <10.0.10.5>:<5060>,
handle=<8>, length=<1224>, message
......many other retransmissions follow................
--
--
Alberto Pastore
B-Press Srl - Gruppo MSoft
P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
http://www.msoft.it
Doug Lytle
2007-Apr-12 07:59 UTC
[asterisk-users] Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
Alberto Pastore wrote:> > Firmware on 7940 is 8.6 (the latest one). >I had the same issue. I ended up moving back to firmware P0S3-07-4-00 on the phone. I did a telnet into the phone, did a show register and shaw some very weird info. Normally, I would see: LINE REGISTRATION TABLE Proxy Registration: ENABLED, state: REGISTERED line APR state timer expires proxy:port ---- --- ------------- ---------- ---------- ---------------------------- 1 111 REGISTERED 115 106 drdos.info:5060 2 111 REGISTERED 115 38 drdos.info:5060 3 ... NONE 0 0 undefined:0 4 ... NONE 0 0 undefined:0 5 ... NONE 0 0 undefined:0 6 ... NONE 0 0 undefined:0 1-BU .1x NONE 0 0 undefined:0 Note: APR is Authenticated, Provisioned, Registered But, under 8x firmware the timers would be some huge number and the state would be registering. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
Alberto Pastore
2007-Apr-12 10:05 UTC
[asterisk-users] Asterisk 1.2.17 and Cisco 7940/SIP: bug or what?
Doug Lytle ha scritto:> Alberto Pastore wrote: >> >> Firmware on 7940 is 8.6 (the latest one). >> > I had the same issue. I ended up moving back to firmware P0S3-07-4-00 > on the phone. I did a telnet into the phone, did a show register and > shaw some very weird info. Normally, I would see:>... But why does 8.6 seem to work with previous asterisk 1.2.13??
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