Two voip system with the same extension

thanhh83 · January 17, 2013, 5:52am

Hello All,

I got problem with two voip system with the same extention.

Detail in link bellow:

community.spiceworks.com/topic/2 … -extension

Please help me.

Thanks
Thanh

david55 · January 17, 2013, 8:02am

It looks like you have a users.conf and Asterisk only uses the existence of a value for trunkname, not the actual value. I’d therefore suspect that you are using a GUI. That may constrain what you can do.

You also haven’t provided details of the “extension” entries in sip.conf.

However, my guess would be this is a combination of one or more of:

not using type=peer for “extensions”;
using the Asterisk extension number for the device name for the “extension”, rather than unique (and preferably unguessable) value.

thanhh83 · January 18, 2013, 1:02am

This did help me solve my problem.

Thanks you
Thanh

thanhh83 · January 18, 2013, 2:37am

but now the user the same number with extension can not reachable from other call.

Detail of config:
sip.conf:
[1151]
host=192.168.1.9
context=DID_QuangNam
insecure=very
disallow=all
allow=ulaw,alaw,gsm,g726
type=peer
nat=yes
dtmfmode=rfc2833

extensions.conf
[DID_QuangNam]
type=peer
include=DID_QuangNam_default

[DID_QuangNam_default]
type=peer
exten=_83.,1,Goto(default|${EXTEN:2.}|1)

[DLPN_DialPlan1]
include=default

[default]
exten=o,1,Goto(default,0,1)

users.conf:
[1151]
fullname=IT
registersip=no
host=dynamic
callgroup=1
mailbox=1151
call-limit=100
type=user
username=1151
transfer=yes
callcounter=yes
context=DLPN_DialPlan1
cid_number=1151
hasvoicemail=no
vmsecret=
email=
threewaycalling=no
hasdirectory=yes
callwaiting=no
hasmanager=no
hasagent=no
hassip=yes
hasiax=no
secret=1151
nat=no
canreinvite=no
dtmfmode=rfc2833
insecure=very
pickupgroup=1
macaddress=000b8234ebdd
autoprov=yes
label=1151
linenumber=1
LINEKEYS=1
trustrpid=yes
requirecalltoken=no
disallow=all
allow=ulaw,gsm,alaw,g723,g726

Please help

Thanks
Thanh

david55 · January 18, 2013, 11:29am

type=peer is a sip.conf option.

include= should be include =>

Verbose level 3 or higher trace output is likely to be needed to see what is happening.

thanhh83 · January 21, 2013, 12:37am

sip.conf option:
[general]
context=default
allowguest=yes
allowoverlap=yes
allowtransfer=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
subscribecontext=default
pedantic=yes
maxexpiry=600
minexpiry=300
relaxdtmf=no
sendrpid=yes
promiscredir=yes
compactheaders=no
maxcallbitrate=384
callevents=yes
alwaysauthreject=no
g726nonstandard=yes
recordhistory=yes
dumphistory=yes
allowsubscribe=yes
notifyringing=yes
rtcachefriends=no
rtsavesysname=no
rtupdate=no
ignoreregexpire=no
autodomain=no
allowexternaldomains=yes
allowexternalinvites=yes
jbenable=yes
jbforce=yes
jblog=no
checkmwi=10
defaultexpiry=120
externrefresh=10
mohinterpret=default
progressinband=never
registerattempts=0
registertimeout=20
sipdebug=yes
t1min=100
t38pt_udptl=no
tos_audio=none
tos_sip=none
tos_video=none
trustrpid=no
useragent=Rieker Asterisk PBX
usereqphone=no
videosupport=no
insecure=very
nat=yes
realm=asterisk
disallow=all
allow=undefined,ulaw,alaw,gsm,g726,g729,g723
type=peer

Change verbose level 3 and try to call:

[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 0: INVITE sip:831151@172.20.151.10:5060 SIP/2.0 (44)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK44AB1DF2 (56)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 2: Remote-Party-ID: “Mr. Thanh IT” sip:192.168.1.9;party=calling;screen=no;privacy=off (85)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 3: From: “unknown” sip:192.168.1.9;tag=E7760110-109A (51)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 4: To: sip:831151@172.20.151.10 (30)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 5: Date: Mon, 21 Jan 2013 00:20:42 GMT (35)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 6: Call-ID: 39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9 (56)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 7: Supported: 100rel,timer,resource-priority,replaces (50)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 8: Min-SE: 1800 (12)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 9: Cisco-Guid: 951072208-1654067682-3125124931-1206966072 (54)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 10: User-Agent: Cisco-SIPGateway/IOS-12.x (37)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 11: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER (97)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 12: CSeq: 101 INVITE (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 13: Max-Forwards: 70 (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 14: Timestamp: 1358727642 (21)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 15: Contact: sip:192.168.1.9:5060 (31)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 16: Expires: 180 (12)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 17: Allow-Events: telephone-event (29)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 18: Content-Type: multipart/mixed;boundary=uniqueBoundary (53)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 19: Mime-Version: 1.0 (17)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 20: Content-Length: 761 (19)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 21: (0)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: Content-Type: application/sdp (29)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: Content-Disposition: session;handling=required (46)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: v=0 (3)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: o=CiscoSystemsSIP-GW-UserAgent 8067 1510 IN IP4 192.168.1.9 (59)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: s=SIP Call (10)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: c=IN IP4 192.168.1.9 (20)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: t=0 0 (5)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: m=audio 16548 RTP/AVP 18 8 0 98 99 102 4 101 (44)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: c=IN IP4 192.168.1.9 (20)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=fmtp:18 annexb=no (19)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=rtpmap:98 G726-16/8000 (24)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=rtpmap:99 G726-24/8000 (24)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=rtpmap:102 G726-32/8000 (25)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=fmtp:4 bitrate=6.3;annexa=yes (31)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=fmtp:101 0-16 (15)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: a=direction:passive (19)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: Content-Type: application/gtd (29)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: Content-Disposition: signal;handling=optional (45)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: IAM, (4)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: GCI,38b035d0629711e2ba459f4347f0d738 (36)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: GEN,u,n,0,Mr. Thanh IT (22)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary-- (18)
[Jan 21 07:22:33] VERBOSE[5016] logger.c: — (21 headers 34 lines) —
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Setting NAT on RTP to On
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Allocating new SIP dialog for 39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9 - INVITE (With RTP)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Begin: parsing SIP “Supported: 100rel,timer,resource-priority,replaces”
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Found SIP option: -100rel-
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Matched SIP option: 100rel
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Found SIP option: -timer-
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Matched SIP option: timer
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Found SIP option: -resource-priority-
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Matched SIP option: resource-priority
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Found SIP option: -replaces-
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Matched SIP option: replaces
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Sending to 192.168.1.9 : 57949 (NAT)
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Using INVITE request as basis request - 39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found peer ‘1151’
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Setting NAT on RTP to On
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing session-level SDP o=CiscoSystemsSIP-GW-UserAgent 8067 1510 IN IP4 192.168.1.9… UNSUPPORTED.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing session-level SDP s=SIP Call… UNSUPPORTED.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.9… OK.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing session-level SDP t=0 0… UNSUPPORTED.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found RTP audio format 18
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found RTP audio format 8
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found RTP audio format 0
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found RTP audio format 98
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found RTP audio format 99
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found RTP audio format 102
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found RTP audio format 4
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found RTP audio format 101
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 192.168.1.9… OK.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found audio description format G729 for ID 18
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000… OK.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no… UNSUPPORTED.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found audio description format PCMA for ID 8
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000… OK.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found audio description format PCMU for ID 0
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000… OK.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found unknown media description format G726-16 for ID 98
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000… UNSUPPORTED.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found unknown media description format G726-24 for ID 99
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:99 G726-24/8000… UNSUPPORTED.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found audio description format G726-32 for ID 102
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 G726-32/8000… OK.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found audio description format G723 for ID 4
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000… OK.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=fmtp:4 bitrate=6.3;annexa=yes… UNSUPPORTED.
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Found audio description format telephone-event for ID 101
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16… UNSUPPORTED.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Processing media-level (audio) SDP a=direction:passive… UNSUPPORTED.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: T38 state changed to 0 on channel
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x11d (g723|ulaw|alaw|g729|g726aal2)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Peer audio RTP is at port 192.168.1.9:16548
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: We’re settling with these formats: 0xc (ulaw|alaw)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Checking SIP call limits for device 1151
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Updating call counter for incoming call
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Call from user ‘1151’ is 1 out of 0
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Looking for 831151 in DID_QuangNam (domain 172.20.151.10)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: *** Our native formats are 0x4 (ulaw)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: *** Our capabilities are 0x80e (gsm|ulaw|alaw|g726)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: This channel will not be able to handle video.
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: build_route: Contact hop: sip:192.168.1.9:5060
[Jan 21 07:22:33] VERBOSE[5016] logger.c: list_route: hop: sip:192.168.1.9:5060
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: SIP/1151-00001b75: New call is still down… Trying…
[Jan 21 07:22:33] VERBOSE[5016] logger.c:
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 0: SIP/2.0 100 Trying (18)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK44AB1DF2;received=192.168.1.9 (77)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 2: From: “unknown” sip:192.168.1.9;tag=E7760110-109A (51)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 3: To: sip:831151@172.20.151.10 (30)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 4: Call-ID: 39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9 (56)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 5: CSeq: 101 INVITE (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 6: User-Agent: Rieker Asterisk PBX (31)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 8: Supported: replaces (19)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 9: Contact: sip:831151@172.20.151.10 (35)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 10: Content-Length: 0 (17)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 11: (0)
[Jan 21 07:22:33] DEBUG[5016] devicestate.c: Notification of state change to be queued on device/channel SIP/1151
[Jan 21 07:22:33] DEBUG[6110] pbx.c: Launching ‘Goto’
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] VERBOSE[6110] logger.c: – Executing [831151@DID_QuangNam:1] Goto(“SIP/1151-00001b75”, “default|1151|1”) in new stack
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] VERBOSE[6110] logger.c: – Goto (default,1151,1)
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: Changing state for SIP/1151 - state 1 (Not in use)
[Jan 21 07:22:33] DEBUG[6110] pbx.c: Launching ‘Dial’
[Jan 21 07:22:33] DEBUG[5011] app_queue.c: Device ‘SIP/1151’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] VERBOSE[6110] logger.c: – Executing [1151@default:1] Dial(“SIP/1151-00001b75”, “SIP/1151”) in new stack
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Setting NAT on RTP to On
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: *** Our native formats are 0x4 (ulaw)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: *** Joint capabilities are 0x0 (nothing)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: *** Our capabilities are 0x80e (gsm|ulaw|alaw|g726)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: This channel will not be able to handle video.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable DIALEDTIME.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable ANSWEREDTIME.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable DIALEDPEERNAME.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable DIALEDPEERNUMBER.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable DIALSTATUS.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable SIPCALLID.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable SIPUSERAGENT.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable SIPDOMAIN.
[Jan 21 07:22:33] DEBUG[6110] channel.c: Not copying variable SIPURI.
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Outgoing Call for 1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Updating call counter for outgoing call
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Call to peer ‘1151’ is 1 out of 100
[Jan 21 07:22:33] DEBUG[6110] devicestate.c: Notification of state change to be queued on device/channel SIP/1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Our T38 capability (0), joint T38 capability (0)
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: ** Our capability: 0x80e (gsm|ulaw|alaw|g726) Video flag: False
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: Changing state for SIP/1151 - state 6 (Ringing)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
[Jan 21 07:22:33] DEBUG[5011] app_queue.c: Device ‘SIP/1151’ changed to state ‘6’ (Ringing) but we don’t care because they’re not a member of any queue.
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] VERBOSE[6110] logger.c: Audio is at 172.20.151.10 port 12650
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] VERBOSE[6110] logger.c: Adding codec 0x4 (ulaw) to SDP
[Jan 21 07:22:33] VERBOSE[6110] logger.c: Adding codec 0x8 (alaw) to SDP
[Jan 21 07:22:33] VERBOSE[6110] logger.c: Adding codec 0x2 (gsm) to SDP
[Jan 21 07:22:33] VERBOSE[6110] logger.c: Adding codec 0x800 (g726) to SDP
[Jan 21 07:22:33] VERBOSE[6110] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: – Done with adding codecs to SDP
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Done building SDP. Settling with this capability: 0x80e (gsm|ulaw|alaw|g726)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 0: INVITE sip:1151@172.20.151.100:5060 SIP/2.0 (43)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.20.151.10:5060;branch=z9hG4bK776743ad;rport (64)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 2: From: “Mr. Phuoc IT QN” sip:1151@172.20.151.10;tag=as15336dbf (63)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 3: To: sip:1151@172.20.151.100:5060 (34)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 4: Contact: sip:1151@172.20.151.10 (33)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 5: Call-ID: 57bdf7e80092242a7908b7d75381c6ca@172.20.151.10 (55)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 6: CSeq: 102 INVITE (16)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 7: User-Agent: Rieker Asterisk PBX (31)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 8: Max-Forwards: 70 (16)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 9: Remote-Party-ID: “Mr. Phuoc IT QN” sip:1151@172.20.151.10;privacy=off;screen=no (81)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 10: Date: Mon, 21 Jan 2013 00:22:33 GMT (35)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 12: Supported: replaces (19)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 13: Content-Type: application/sdp (29)
[Jan 21 07:22:33] VERBOSE[6110] logger.c: Reliably Transmitting (NAT) to 192.168.1.9:5060:
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 0: INVITE sip:1151@172.20.151.100:5060 SIP/2.0 (43)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.20.151.10:5060;branch=z9hG4bK776743ad;rport (64)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 2: From: “Mr. Phuoc IT QN” sip:1151@172.20.151.10;tag=as15336dbf (63)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 3: To: sip:1151@172.20.151.100:5060 (34)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 4: Contact: sip:1151@172.20.151.10 (33)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 5: Call-ID: 57bdf7e80092242a7908b7d75381c6ca@172.20.151.10 (55)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 6: CSeq: 102 INVITE (16)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 7: User-Agent: Rieker Asterisk PBX (31)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 8: Max-Forwards: 70 (16)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 9: Remote-Party-ID: “Mr. Phuoc IT QN” sip:1151@172.20.151.10;privacy=off;screen=no (81)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 10: Date: Mon, 21 Jan 2013 00:22:33 GMT (35)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 12: Supported: replaces (19)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 13: Content-Type: application/sdp (29)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 14: Content-Length: 291 (19)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 15: (0)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: v=0 (3)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: o=root 4768 4768 IN IP4 172.20.151.10 (37)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: s=session (9)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: c=IN IP4 172.20.151.10 (22)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: t=0 0 (5)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: m=audio 12650 RTP/AVP 0 8 3 111 101 (35)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: a=rtpmap:111 G726-32/8000 (25)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: a=fmtp:101 0-16 (15)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: a=ptime:20 (10)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Line: a=sendrecv (10)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
[Jan 21 07:22:33] VERBOSE[6110] logger.c: – Called 1151
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Reliably Transmitting (NAT) to 192.168.3.50:63320:
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 0: NOTIFY sip:1152@192.168.3.50:63320 SIP/2.0 (42)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.20.151.10:5060;branch=z9hG4bK597dab89;rport (64)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 2: Route: sip:1152@192.168.3.50:63320 (36)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 3: From: “asterisk” sip:asterisk@172.20.151.10;tag=as7a697336 (60)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 4: To: sip:1152@192.168.3.50:63320;tag=fd6a2a59 (46)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 5: Contact: sip:asterisk@172.20.151.10 (37)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 6: Call-ID: 375b0043cc60c017NzVkNTY4NTg0MDJjYTE3OTQwYWYxZmQ4YzI0ZTU2ZWM. (69)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 7: CSeq: 103 NOTIFY (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 8: User-Agent: Rieker Asterisk PBX (31)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 9: Max-Forwards: 70 (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 10: Event: message-summary (22)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 11: Content-Type: application/simple-message-summary (48)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 12: Subscription-State: active (26)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 13: Content-Length: 93 (18)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 14: (0)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: Messages-Waiting: no (20)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: Message-Account: sip:asterisk@172.20.151.10 (43)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Line: Voice-Message: 0/0 (0/0) (24)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
[Jan 21 07:22:33] VERBOSE[5016] logger.c:
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.20.151.10:5060;branch=z9hG4bK597dab89;rport=5060 (69)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 2: Contact: sip:1152@192.168.3.50:63320 (38)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 3: To: sip:1152@192.168.3.50:63320;tag=fd6a2a59 (46)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 4: From: "asterisk"sip:asterisk@172.20.151.10;tag=as7a697336 (59)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 5: Call-ID: 375b0043cc60c017NzVkNTY4NTg0MDJjYTE3OTQwYWYxZmQ4YzI0ZTU2ZWM. (69)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 6: CSeq: 102 NOTIFY (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 7: User-Agent: X-Lite release 1002tx stamp 29712 (45)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 8: Content-Length: 0 (17)
[Jan 21 07:22:33] VERBOSE[5016] logger.c: — (9 headers 0 lines) —
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Stopping retransmission on ‘375b0043cc60c017NzVkNTY4NTg0MDJjYTE3OTQwYWYxZmQ4YzI0ZTU2ZWM.’ of Request 102: Match Not Found
[Jan 21 07:22:33] VERBOSE[5016] logger.c:
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 0: SIP/2.0 400 Bad Request - ‘Invalid Host’ (40)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.20.151.10:5060;branch=z9hG4bK776743ad;rport (64)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 2: From: “Mr. Phuoc IT QN” sip:1151@172.20.151.10;tag=as15336dbf (63)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 3: To: sip:1151@172.20.151.100:5060;tag=E7760138-E47 (51)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 4: Date: Mon, 21 Jan 2013 00:20:42 GMT (35)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 5: Call-ID: 57bdf7e80092242a7908b7d75381c6ca@172.20.151.10 (55)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 6: Server: Cisco-SIPGateway/IOS-12.x (33)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 7: CSeq: 102 INVITE (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 8: Allow-Events: telephone-event (29)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 9: Content-Length: 0 (17)
[Jan 21 07:22:33] VERBOSE[5016] logger.c: — (10 headers 0 lines) —
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Acked pending invite 102
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #632733
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Stopping retransmission on ‘57bdf7e80092242a7908b7d75381c6ca@172.20.151.10’ of Request 102: Match Found
[Jan 21 07:22:33] VERBOSE[5016] logger.c: – Got SIP response 400 “Bad Request - ‘Invalid Host’” back from 192.168.1.9
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Transmitting (NAT) to 192.168.1.9:5060:
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 0: ACK sip:1151@172.20.151.100:5060 SIP/2.0 (40)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 172.20.151.10:5060;branch=z9hG4bK776743ad;rport (64)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 2: From: “Mr. Phuoc IT QN” sip:1151@172.20.151.10;tag=as15336dbf (63)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 3: To: sip:1151@172.20.151.100:5060;tag=E7760138-E47 (51)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 4: Contact: sip:1151@172.20.151.10 (33)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 5: Call-ID: 57bdf7e80092242a7908b7d75381c6ca@172.20.151.10 (55)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 6: CSeq: 102 ACK (13)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 7: User-Agent: Rieker Asterisk PBX (31)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 8: Max-Forwards: 70 (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 9: Remote-Party-ID: “Mr. Phuoc IT QN” sip:1151@172.20.151.10;privacy=off;screen=no (81)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 10: Content-Length: 0 (17)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 11: (0)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Setting SIP_ALREADYGONE on dialog 57bdf7e80092242a7908b7d75381c6ca@172.20.151.10
[Jan 21 07:22:33] VERBOSE[6110] logger.c: – SIP/1151-00001b76 is circuit-busy
[Jan 21 07:22:33] DEBUG[6110] channel.c: Hanging up channel ‘SIP/1151-00001b76’
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Hangup call SIP/1151-00001b76, SIP callid 57bdf7e80092242a7908b7d75381c6ca@172.20.151.10)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: update_call_counter(1151) - decrement call limit counter on hangup
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Updating call counter for outgoing call
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Call to peer ‘1151’ removed from call limit 100
[Jan 21 07:22:33] DEBUG[6110] devicestate.c: Notification of state change to be queued on device/channel SIP/1151
[Jan 21 07:22:33] DEBUG[6110] devicestate.c: Notification of state change to be queued on device/channel SIP/1151
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] VERBOSE[6110] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
[Jan 21 07:22:33] DEBUG[6110] rtp.c: Channel ‘’ has no RTP, not doing anything
[Jan 21 07:22:33] DEBUG[6110] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] VERBOSE[6110] logger.c: == Auto fallthrough, channel ‘SIP/1151-00001b75’ status is ‘CONGESTION’
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: Changing state for SIP/1151 - state 1 (Not in use)
[Jan 21 07:22:33] VERBOSE[6110] logger.c:
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] DEBUG[5011] app_queue.c: Device ‘SIP/1151’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 0: SIP/2.0 503 Service Unavailable (31)
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK44AB1DF2;received=192.168.1.9 (77)
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 2: From: “unknown” sip:192.168.1.9;tag=E7760110-109A (51)
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: Changing state for SIP/1151 - state 1 (Not in use)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 3: To: sip:831151@172.20.151.10;tag=as3d5b46c4 (45)
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] DEBUG[5011] app_queue.c: Device ‘SIP/1151’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 4: Call-ID: 39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9 (56)
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 5: CSeq: 101 INVITE (16)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 6: User-Agent: Rieker Asterisk PBX (31)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 8: Supported: replaces (19)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 9: X-Asterisk-HangupCause: Normal Clearing (39)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 10: X-Asterisk-HangupCauseCode: 16 (30)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 11: Content-Length: 0 (17)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Header 12: (0)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Setting SIP_ALREADYGONE on dialog 39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9
[Jan 21 07:22:33] DEBUG[6110] channel.c: Soft-Hanging up channel ‘SIP/1151-00001b75’
[Jan 21 07:22:33] DEBUG[6110] devicestate.c: Notification of state change to be queued on device/channel SIP/1151
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] DEBUG[6110] channel.c: Soft-Hanging up channel ‘SIP/1151-00001b75’
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] DEBUG[6110] channel.c: Hanging up channel ‘SIP/1151-00001b75’
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: Changing state for SIP/1151 - state 1 (Not in use)
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Hangup call SIP/1151-00001b75, SIP callid 39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9)
[Jan 21 07:22:33] DEBUG[5011] app_queue.c: Device ‘SIP/1151’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: update_call_counter(1151) - decrement call limit counter on hangup
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Updating call counter for incoming call
[Jan 21 07:22:33] DEBUG[6110] chan_sip.c: Call from user ‘1151’ removed from call limit 0
[Jan 21 07:22:33] DEBUG[6110] devicestate.c: Notification of state change to be queued on device/channel SIP/1151
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: Changing state for SIP/1151 - state 1 (Not in use)
[Jan 21 07:22:33] DEBUG[5011] app_queue.c: Device ‘SIP/1151’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Jan 21 07:22:33] DEBUG[4787] devicestate.c: No provider found, checking channel drivers for SIP - 1151
[Jan 21 07:22:33] DEBUG[4787] chan_sip.c: Checking device state for peer 1151
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Really destroying SIP dialog ‘57bdf7e80092242a7908b7d75381c6ca@172.20.151.10’ Method: INVITE
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c:
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: * SIP Call
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: 001. NewChan Channel SIP/1151-00001b76 - from 57bdf7e80092242a7908b7d75381c6
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: 002. TxReqRel INVITE / 102 INVITE - -UNKNOWN-
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: 003. Rx SIP/2.0 / 102 INVITE / 400 Bad Request - ‘Invalid Host’
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: 004. TxReq ACK / 102 ACK - -UNKNOWN-
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: 005. Hangup Cause Normal Clearing
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c:
[Jan 21 07:22:33] VERBOSE[5016] logger.c:
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 0: ACK sip:831151@172.20.151.10:5060 SIP/2.0 (41)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK44AB1DF2 (56)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 2: From: “unknown” sip:192.168.1.9;tag=E7760110-109A (51)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 3: To: sip:831151@172.20.151.10;tag=as3d5b46c4 (45)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 4: Date: Mon, 21 Jan 2013 00:20:42 GMT (35)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 5: Call-ID: 39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9 (56)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 6: Max-Forwards: 70 (16)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 7: CSeq: 101 ACK (13)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 8: Allow-Events: telephone-event (29)
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Header 9: Content-Length: 0 (17)
[Jan 21 07:22:33] VERBOSE[5016] logger.c: — (10 headers 0 lines) —
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #632736
[Jan 21 07:22:33] DEBUG[5016] chan_sip.c: Stopping retransmission on ‘39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9’ of Response 101: Match Found
[Jan 21 07:22:33] VERBOSE[5016] logger.c: Really destroying SIP dialog ‘39B6A9ED-629711E2-BA4A9F43-47F0D738@192.168.1.9’ Method: ACK

Thanks
Thanh

david55 · January 21, 2013, 8:22am

[Jan 21 07:22:33] VERBOSE[5016] logger.c: – Got SIP response 400 “Bad Request - ‘Invalid Host’” back from 192.168.1.9

Possibly the other side expects a domain name. Possibly it is not happy with the default port number being supplied explicitly.

david55 · January 21, 2013, 8:26am

Or it may be simply that the IP address to which you sent the request is not that in the request, so maybe there should not be a proxy, or maybe the proxy is misconfigured.

The relevant configuratoin is missing, or only on the other forum.

thanhh83 · January 22, 2013, 6:17am

Hello,

I think problem can not call because asterisk always check “from” user who call to other user.

Do you know any way, command or option to tell asterisk don’t check that?

Thanks
Thanh

david55 · January 22, 2013, 8:07am

That’s not why you got the Bad Request response.

I’ve already told you how to avoid Asterisk trying to match the From: user; you use type=peer.

thanhh83 · January 23, 2013, 1:11am

Hello David,

Thanks you.

I already to put type=peer in users, extensions and in option of sip.conf but the same problem. Asterisk till try to match From: user. I use Asterisk 1.4.44

Thanks
Thanh

david55 · January 23, 2013, 8:28am

You haven’t provided error messages showing a from matching issue. You have provided error messages showing the remote system objecting to a domain name. Please provide a trace that shows the first without the second.

thanhh83 · January 24, 2013, 1:01am

Hello David,

With current config:
sip.conf:
[general]
type=peer
context=default
allowguest=yes
allowoverlap=yes
allowtransfer=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
subscribecontext=default
pedantic=yes
maxexpiry=600
minexpiry=300
relaxdtmf=no
sendrpid=yes
promiscredir=yes
compactheaders=no
maxcallbitrate=384
callevents=yes
alwaysauthreject=no
g726nonstandard=yes
recordhistory=yes
dumphistory=yes
allowsubscribe=yes
notifyringing=yes
rtcachefriends=no
rtsavesysname=no
rtupdate=no
ignoreregexpire=no
autodomain=no
allowexternaldomains=yes
allowexternalinvites=yes
jbenable=yes
jbforce=yes
jblog=no
checkmwi=10
defaultexpiry=120
externrefresh=10
mohinterpret=default
progressinband=never
registerattempts=0
registertimeout=20
sipdebug=yes
t1min=100
t38pt_udptl=no
tos_audio=none
tos_sip=none
tos_video=none
trustrpid=no
useragent=Rieker Asterisk PBX
usereqphone=no
videosupport=no
insecure=very
nat=yes
realm=asterisk
disallow=all
allow=undefined,ulaw,alaw,gsm,g726,g729,g723

[quang]
host=192.168.1.9
context=DID_QuangNam
insecure=very
disallow=all
allow=ulaw,alaw,gsm,g726
type=peer
nat=no
dtmfmode=rfc2833

extension.conf:
[DID_QuangNam]
type=peer
include=DID_QuangNam_default

[DID_QuangNam_default]
type=peer
exten=_83.,1,Goto(default|${EXTEN:2.}|1)

[default]
type=peer
exten=o,1,Goto(default,0,1)

[DLPN_DialPlan1]
include=default
type=peer

users.conf:
[1151]
fullname=Mr. Phuoc IT QN
registersip=no
host=dynamic
callgroup=1
mailbox=1151
call-limit=100
type=peer
username=1151
transfer=yes
callcounter=yes
context=DLPN_DialPlan1
cid_number=1151
hasvoicemail=no
vmsecret=
email=
threewaycalling=no
hasdirectory=yes
callwaiting=no
hasmanager=no
hasagent=no
hassip=yes
hasiax=no
secret=****
nat=no
canreinvite=no
dtmfmode=rfc2833
insecure=very
pickupgroup=1
macaddress=000b8234ebdd
autoprov=yes
label=1151
linenumber=1
LINEKEYS=1
trustrpid=yes
requirecalltoken=no
disallow=all
allow=ulaw,gsm,alaw,g723,g726

[1152]
fullname=Mr. Thanh IT QN
registersip=no
host=dynamic
callgroup=1
mailbox=1152
call-limit=100
type=peer
username=1152
transfer=yes
callcounter=yes
context=DLPN_DialPlan1
cid_number=1152
hasvoicemail=no
vmsecret=
email=
threewaycalling=no
hasdirectory=no
callwaiting=no
hasmanager=no
hasagent=no
hassip=yes
hasiax=no
secret=****
nat=no
canreinvite=no
dtmfmode=rfc2833
insecure=very
pickupgroup=1
macaddress=
autoprov=yes
label=1152
linenumber=1
LINEKEYS=1
trustrpid=yes
requirecalltoken=no
disallow=all
allow=ulaw,gsm

Error when call 831152 from 192.168.1.9 with extension: 1151
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 0: INVITE sip:831152@172.20.151.10:5060 SIP/2.0 (44)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK52CF16AB (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 2: Remote-Party-ID: “Mr. Thanh IT” sip:1151@192.168.1.9;party=calling;screen=no;privacy=off (90)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 3: From: “Mr. Thanh IT” sip:1151@192.168.1.9;tag=F6EDD580-273 (60)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 4: To: sip:831152@172.20.151.10 (30)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 5: Date: Thu, 24 Jan 2013 00:25:55 GMT (35)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 6: Call-ID: 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9 (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 7: Supported: 100rel,timer,resource-priority,replaces (50)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 8: Min-SE: 1800 (12)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 9: Cisco-Guid: 1929903383-1693651426-2701958979-1206966072 (55)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 10: User-Agent: Cisco-SIPGateway/IOS-12.x (37)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 11: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER (97)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 12: CSeq: 101 INVITE (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 13: Max-Forwards: 70 (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 14: Timestamp: 1358987155 (21)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 15: Contact: sip:1151@192.168.1.9:5060 (36)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 16: Expires: 180 (12)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 17: Allow-Events: telephone-event (29)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 18: Content-Type: multipart/mixed;boundary=uniqueBoundary (53)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 19: Mime-Version: 1.0 (17)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 20: Content-Length: 737 (19)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 21: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: Content-Type: application/sdp (29)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: Content-Disposition: session;handling=required (46)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: v=0 (3)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: o=CiscoSystemsSIP-GW-UserAgent 8627 3971 IN IP4 192.168.1.9 (59)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: s=SIP Call (10)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: c=IN IP4 192.168.1.9 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: t=0 0 (5)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: m=audio 19360 RTP/AVP 18 8 0 98 99 102 4 101 (44)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: c=IN IP4 192.168.1.9 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=fmtp:18 annexb=no (19)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:98 G726-16/8000 (24)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:99 G726-24/8000 (24)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:102 G726-32/8000 (25)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=fmtp:4 bitrate=6.3;annexa=yes (31)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=fmtp:101 0-16 (15)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=direction:passive (19)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: Content-Type: application/gtd (29)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: Content-Disposition: signal;handling=optional (45)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: IAM, (4)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: GCI,7307fd1764f311e2a10c9f4347f0d738 (36)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary-- (18)
[Jan 24 07:27:45] VERBOSE[5016] logger.c: — (21 headers 33 lines) —
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Setting NAT on RTP to On
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Allocating new SIP dialog for 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9 - INVITE (With RTP)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Begin: parsing SIP “Supported: 100rel,timer,resource-priority,replaces”
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Found SIP option: -100rel-
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Matched SIP option: 100rel
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Found SIP option: -timer-
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Matched SIP option: timer
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Found SIP option: -resource-priority-
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Matched SIP option: resource-priority
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Found SIP option: -replaces-
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Matched SIP option: replaces
[Jan 24 07:27:45] VERBOSE[5016] logger.c: Sending to 192.168.1.9 : 56256 (NAT)
[Jan 24 07:27:45] VERBOSE[5016] logger.c: Using INVITE request as basis request - 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Setting NAT on RTP to Off
[Jan 24 07:27:45] VERBOSE[5016] logger.c:
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 0: SIP/2.0 407 Proxy Authentication Required (41)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK52CF16AB;received=192.168.1.9 (77)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 2: From: “Mr. Thanh IT” sip:1151@192.168.1.9;tag=F6EDD580-273 (60)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 3: To: sip:831152@172.20.151.10;tag=as1da3e209 (45)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 4: Call-ID: 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9 (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 5: CSeq: 101 INVITE (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 6: User-Agent: Rieker Asterisk PBX (31)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 8: Supported: replaces (19)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 9: Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7d45d83d” (76)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 10: Content-Length: 0 (17)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 11: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
[Jan 24 07:27:45] VERBOSE[5016] logger.c: Scheduling destruction of SIP dialog ‘73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9’ in 32000 ms (Method: INVITE)
[Jan 24 07:27:45] VERBOSE[5016] logger.c: Found user ‘1151’
[Jan 24 07:27:45] VERBOSE[5016] logger.c:
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 0: ACK sip:831152@172.20.151.10:5060 SIP/2.0 (41)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK52CF16AB (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 2: From: “Mr. Thanh IT” sip:1151@192.168.1.9;tag=F6EDD580-273 (60)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 3: To: sip:831152@172.20.151.10;tag=as1da3e209 (45)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 4: Date: Thu, 24 Jan 2013 00:25:55 GMT (35)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 5: Call-ID: 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9 (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 6: Max-Forwards: 70 (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 7: CSeq: 101 ACK (13)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 8: Allow-Events: telephone-event (29)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 9: Content-Length: 0 (17)
[Jan 24 07:27:45] VERBOSE[5016] logger.c: — (10 headers 0 lines) —
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #694243
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Stopping retransmission on ‘73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9’ of Response 101: Match Found
[Jan 24 07:27:45] VERBOSE[5016] logger.c:
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 0: INVITE sip:831152@172.20.151.10:5060 SIP/2.0 (44)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK52D01594 (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 2: Remote-Party-ID: “Mr. Thanh IT” sip:1151@192.168.1.9;party=calling;screen=no;privacy=off (90)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 3: From: “Mr. Thanh IT” sip:1151@192.168.1.9;tag=F6EDD580-273 (60)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 4: To: sip:831152@172.20.151.10 (30)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 5: Date: Thu, 24 Jan 2013 00:25:55 GMT (35)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 6: Call-ID: 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9 (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 7: Supported: 100rel,timer,resource-priority,replaces (50)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 8: Min-SE: 1800 (12)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 9: Cisco-Guid: 1929903383-1693651426-2701958979-1206966072 (55)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 10: User-Agent: Cisco-SIPGateway/IOS-12.x (37)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 11: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER (97)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 12: CSeq: 102 INVITE (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 13: Max-Forwards: 70 (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 14: Timestamp: 1358987155 (21)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 15: Contact: sip:1151@192.168.1.9:5060 (36)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 16: Expires: 180 (12)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 17: Allow-Events: telephone-event (29)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 18: Proxy-Authorization: Digest username=“quang”,realm=“asterisk”,uri=“sip:831152@172.20.151.10:5060”,response=“c12335e71e9389500f3d895333a856ef”,nonce=“7d45d83d”,algorithm=MD5 (172)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 19: Content-Type: multipart/mixed;boundary=uniqueBoundary (53)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 20: Mime-Version: 1.0 (17)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 21: Content-Length: 737 (19)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 22: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: Content-Type: application/sdp (29)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: Content-Disposition: session;handling=required (46)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: v=0 (3)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: o=CiscoSystemsSIP-GW-UserAgent 8627 3971 IN IP4 192.168.1.9 (59)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: s=SIP Call (10)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: c=IN IP4 192.168.1.9 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: t=0 0 (5)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: m=audio 19360 RTP/AVP 18 8 0 98 99 102 4 101 (44)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: c=IN IP4 192.168.1.9 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=fmtp:18 annexb=no (19)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:98 G726-16/8000 (24)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:99 G726-24/8000 (24)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:102 G726-32/8000 (25)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=fmtp:4 bitrate=6.3;annexa=yes (31)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=fmtp:101 0-16 (15)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: a=direction:passive (19)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: Content-Type: application/gtd (29)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: Content-Disposition: signal;handling=optional (45)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: IAM, (4)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: GCI,7307fd1764f311e2a10c9f4347f0d738 (36)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Line: --uniqueBoundary-- (18)
[Jan 24 07:27:45] VERBOSE[5016] logger.c: — (22 headers 33 lines) —
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jan 24 07:27:45] VERBOSE[5016] logger.c: Sending to 192.168.1.9 : 5060 (no NAT)
[Jan 24 07:27:45] VERBOSE[5016] logger.c: Using INVITE request as basis request - 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Setting NAT on RTP to Off
[Jan 24 07:27:45] WARNING[5016] chan_sip.c: username mismatch, have <1151>, digest has
[Jan 24 07:27:45] VERBOSE[5016] logger.c: Found user ‘1151’
[Jan 24 07:27:45] NOTICE[5016] chan_sip.c: Failed to authenticate user “Mr. Thanh IT” sip:1151@192.168.1.9;tag=F6EDD580-273
[Jan 24 07:27:45] VERBOSE[5016] logger.c:
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 0: SIP/2.0 403 Forbidden (21)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK52D01594;received=192.168.1.9 (77)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 2: From: “Mr. Thanh IT” sip:1151@192.168.1.9;tag=F6EDD580-273 (60)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 3: To: sip:831152@172.20.151.10;tag=as1da3e209 (45)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 4: Call-ID: 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9 (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 5: CSeq: 102 INVITE (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 6: User-Agent: Rieker Asterisk PBX (31)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 8: Supported: replaces (19)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 9: Content-Length: 0 (17)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 10: (0)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1
[Jan 24 07:27:45] VERBOSE[5016] logger.c: Scheduling destruction of SIP dialog ‘73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9’ in 32000 ms (Method: INVITE)
[Jan 24 07:27:45] VERBOSE[5016] logger.c:
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 0: ACK sip:831152@172.20.151.10:5060 SIP/2.0 (41)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK52D01594 (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 2: From: “Mr. Thanh IT” sip:1151@192.168.1.9;tag=F6EDD580-273 (60)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 3: To: sip:831152@172.20.151.10;tag=as1da3e209 (45)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 4: Date: Thu, 24 Jan 2013 00:25:55 GMT (35)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 5: Call-ID: 73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9 (56)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 6: Max-Forwards: 70 (16)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 7: CSeq: 102 ACK (13)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 8: Allow-Events: telephone-event (29)
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Header 9: Content-Length: 0 (17)
[Jan 24 07:27:45] VERBOSE[5016] logger.c: — (10 headers 0 lines) —
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #694245
[Jan 24 07:27:45] DEBUG[5016] chan_sip.c: Stopping retransmission on ‘73E45608-64F311E2-A1119F43-47F0D738@192.168.1.9’ of Response 102: Match Found

Thanks and Best regards
Thanh