[Asterisk-Dev] Unable to create channel
Chintan Thakker
cthakker at ipnetfusion.com
Mon Jul 7 16:32:08 MST 2003
Hi,
We have a test environment which is capable of emulating SIP
endpoints/proxy. I am trying to make a SIP-> asterisx -> SIP call to run
asterisx against some test scenarios.
The registration of both the endpoints is successful.
Next when I try to establish a call, I get the above message. Also
attached is the trace with this message.
Once an endpoint registers, asterix keeps sending OPTIONS message to
these endpoints. Since the endpoints do not respond to these OPTION
messages, asterisx sees them as unreachable. Can this be the reason why
it is unable to create a channel ? Am I missing something in the config
files ?
Thank you,
ps: I removed some OPTIONS messages sent by asterisx to reduce the size
of this message.
-- start sip.conf --
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
;context = default ; Default for incoming calls
context = sip
;tos=lowdelay
;tos=184
maxexpirey=3600 ; Max length of incoming registration we
allow
defaultexpirey=3600 ; Default length of incoming/outoing
registration
[9727610001]
type=friend
username=9727610001
fromuser=9727610001
;secret=9727610001
host=dynamic
qualify=2000 ; Qualify peer is no more than 2000ms away
defaultip=192.1.2.223
[9727619271]
type=friend
username=9727619271
fromuser=9727619271
;secret=9727619271
host=dynamic
qualify=2000 ; Qualify peer is no more than 2000ms away
defaultip=192.1.2.88
-- end sip.conf --
-- start extentions.conf --
;contexts relevant to EAST
[sip]
include => sip_term
include => sip_orig
[sip_term]
exten => 9727610001,1,Dial,SIP/9727610001
include => demo
[sip_orig]
exten => 9727619271,1,Dial,SIP/9727619271
include => demo
-- end extentions.conf --
-- Start asterisx trace --
[root at linux17 asterisk-0.4.0]# ./asterisk -vvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc.
Written by Mark Spencer <markster at linux-support.net>
=========================================================================
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
== Manager registered action Ping
== Manager registered action Logoff
== Manager registered action Hangup
== Manager registered action Status
== Manager registered action Redirect
== Manager registered action Originate
== Manager registered action MailboxStatus
== Manager registered action Command
== Manager registered action ExtensionState
== Parsing '/etc/asterisk/manager.conf': Found
Asterisk PBX Core Initializing
Registering builtin applications:
[Answer]
== Registered application 'Answer'
[Goto]
== Registered application 'Goto'
[Hangup]
== Registered application 'Hangup'
[DigitTimeout]
== Registered application 'DigitTimeout'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[BackGround]
== Registered application 'BackGround'
[Wait]
== Registered application 'Wait'
[StripMSD]
== Registered application 'StripMSD'
[Prefix]
== Registered application 'Prefix'
[SetLanguage]
== Registered application 'SetLanguage'
[Ringing]
== Registered application 'Ringing'
[Congestion]
== Registered application 'Congestion'
[Busy]
== Registered application 'Busy'
[SetVar]
== Registered application 'SetVar'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[NoOp]
== Registered application 'NoOp'
[GotoIf]
== Registered application 'GotoIf'
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[chan_modem.so] => (Generic Voice Modem Driver)
== Parsing '/etc/asterisk/modem.conf': Found
== Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell
Chipset) ITU-2 VoiceModem Driver)
== Registered channel type 'Modem' (Generic Voice Modem Channel Driver)
[res_musiconhold.so] => (Music On Hold Resource)
== Parsing '/etc/asterisk/musiconhold.conf': Found
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
[res_adsi.so] => (Call Parking Resource)
== Parsing '/etc/asterisk/adsi.conf': Found
[res_parking.so] => (Call Parking Resource)
== Parsing '/etc/asterisk/parking.conf': Found
-- Registered extension context 'parkedcalls'
-- Added extension '701' priority 1 to parkedcalls
-- Added extension '702' priority 1 to parkedcalls
-- Added extension '703' priority 1 to parkedcalls
-- Added extension '704' priority 1 to parkedcalls
-- Added extension '705' priority 1 to parkedcalls
-- Added extension '706' priority 1 to parkedcalls
-- Added extension '707' priority 1 to parkedcalls
-- Added extension '708' priority 1 to parkedcalls
-- Added extension '709' priority 1 to parkedcalls
-- Added extension '710' priority 1 to parkedcalls
-- Added extension '711' priority 1 to parkedcalls
-- Added extension '712' priority 1 to parkedcalls
-- Added extension '713' priority 1 to parkedcalls
-- Added extension '714' priority 1 to parkedcalls
-- Added extension '715' priority 1 to parkedcalls
-- Added extension '716' priority 1 to parkedcalls
-- Added extension '717' priority 1 to parkedcalls
-- Added extension '718' priority 1 to parkedcalls
-- Added extension '719' priority 1 to parkedcalls
-- Added extension '720' priority 1 to parkedcalls
== Registered application 'ParkedCall'
[res_crypto.so] => (Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'iaxtel'
[res_indications.so] => (Indications Configuration)
== Parsing '/etc/asterisk/indications.conf': Found
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
-- Registered indication country 'de'
-- Registered indication country 'nl'
-- Registered indication country 'uk'
-- Setting default indication country to 'us'
== Registered application 'Playtones'
== Registered application 'StopPlaytones'
[res_monitor.so] => (Call Monitoring Resource)
== Registered application 'Monitor'
== Registered application 'StopMonitor'
== Registered application 'ChangeMonitor'
== Manager registered action Monitor
== Manager registered action StopMonitor
== Manager registered action ChangeMonitor
[chan_iax.so] => (Inter Asterisk eXchange)
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
== Registered channel type 'IAX' (Inter Asterisk eXchange Drver)
== Using TOS bits 16
== IAX Ready and Listening on 0.0.0.0 port 5036
[chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
== SIP Listening on 0.0.0.0:5060
== Using TOS bits 0
== Registered channel type 'sip' (Session Initiation Protocol (SIP))
[chan_oss.so] => (OSS Console Channel Driver)
== Parsing '/etc/asterisk/oss.conf': Found
WARNING[8192]: File chan_oss.c, Line 342 (setformat): Requested 8000 Hz,
got 8125 Hz -- sound may be choppy
WARNING[8192]: File chan_oss.c, Line 970 (load_module): XXX I don't work
right with non-full duplex sound cards XXX
== Registered channel type 'Console' (OSS Console Channel Driver)
[chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem
Driver)
[chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver)
[chan_agent.so] => (Agent Proxy Channel)
== Registered channel type 'Agent' (Call Agent Proxy Channel)
== Registered application 'AgentLogin'
== Parsing '/etc/asterisk/agents.conf': Found
[chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
== Parsing '/etc/asterisk/mgcp.conf': Found
== MGCP Listening on 0.0.0.0:2427
== Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
WARNING[49159]: File chan_oss.c, Line 228 (sound_thread): Read error on
sound device: Resource temporarily unavailable
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
WARNING[8192]: File chan_iax2.c, Line 4952 (set_config): Ignoring port
for now
== Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
== Using TOS bits 16
== IAX Ready and Listening on 0.0.0.0 port 4569
[chan_local.so] => (Local Proxy Channel)
== Registered channel type 'Local' (Local Proxy Channel Driver)
[chan_phone.so] => (Linux Telephony API Support)
== Parsing '/etc/asterisk/phone.conf': Found
== Registered channel type 'Phone' (Standard Linux Telephony API Driver)
[pbx_config.so] => (Text Extension Configuration)
== Parsing '/etc/asterisk/extensions.conf': Found
-- Setting global variable 'CONSOLE' to 'Console/dsp'
-- Setting global variable 'IAXINFO' to 'guest'
-- Setting global variable 'TRUNK' to 'Zap/g2'
-- Setting global variable 'TERMSERV' to '192.1.2.223'
-- Setting global variable 'ORIGSERV' to '192.1.2.88'
-- Setting global variable 'SERV' to '192.1.2.17'
-- Setting global variable 'TERMID' to '9727610001'
-- Setting global variable 'ORIGID' to '9727619271'
-- Registered extension context 'macro-dialtermorig'
-- Added extension 's' priority 1 to macro-dialtermorig
-- Added extension 's' priority 2 to macro-dialtermorig
-- Registered extension context 'iaxtel700'
-- Added extension '_91700NXXXXXX' priority 1 to iaxtel700
-- Registered extension context 'iaxprovider'
-- Registered extension context 'trunkint'
-- Added extension '9011.' priority 1 to trunkint
-- Added extension '9011.' priority 2 to trunkint
-- Registered extension context 'trunkld'
-- Added extension '_91NXXNXXXXXX' priority 1 to trunkld
-- Added extension '_91NXXNXXXXXX' priority 2 to trunkld
-- Registered extension context 'trunklocal'
-- Added extension '_9NXXXXXX' priority 1 to trunklocal
-- Added extension '_9NXXXXXX' priority 2 to trunklocal
-- Registered extension context 'trunktollfree'
-- Added extension '_91800NXXXXXX' priority 1 to trunktollfree
-- Added extension '_91800NXXXXXX' priority 2 to trunktollfree
-- Added extension '_91888NXXXXXX' priority 1 to trunktollfree
-- Added extension '_91888NXXXXXX' priority 2 to trunktollfree
-- Added extension '_91877NXXXXXX' priority 1 to trunktollfree
-- Added extension '_91877NXXXXXX' priority 2 to trunktollfree
-- Added extension '_91866NXXXXXX' priority 1 to trunktollfree
-- Added extension '_91866NXXXXXX' priority 2 to trunktollfree
-- Registered extension context 'international'
-- Including context 'longdistance' in context 'international'
-- Including context 'trunkint' in context 'international'
-- Registered extension context 'longdistance'
-- Including context 'local' in context 'longdistance'
-- Including context 'trunkld' in context 'longdistance'
-- Registered extension context 'local'
-- Including context 'default' in context 'local'
-- Including context 'parkedcalls' in context 'local'
-- Including context 'trunklocal' in context 'local'
-- Including context 'iaxtel700' in context 'local'
-- Including context 'trunktollfree' in context 'local'
-- Including context 'iaxprovider' in context 'local'
-- Registered extension context 'macro-stdexten'
-- Added extension 's' priority 1 to macro-stdexten
-- Added extension 's' priority 2 to macro-stdexten
-- Added extension 's' priority 3 to macro-stdexten
-- Added extension 's' priority 102 to macro-stdexten
-- Added extension 's' priority 103 to macro-stdexten
-- Registered extension context 'demo'
-- Added extension 's' priority 1 to demo
-- Added extension 's' priority 2 to demo
-- Added extension 's' priority 3 to demo
-- Added extension 's' priority 4 to demo
-- Added extension 's' priority 5 to demo
-- Added extension 's' priority 6 to demo
-- Added extension '2' priority 1 to demo
-- Added extension '2' priority 2 to demo
-- Added extension '3' priority 1 to demo
-- Added extension '3' priority 2 to demo
-- Added extension '1000' priority 1 to demo
-- Added extension '1234' priority 1 to demo
-- Added extension '1234' priority 2 to demo
-- Added extension '1235' priority 1 to demo
-- Added extension '1236' priority 1 to demo
-- Added extension '1236' priority 2 to demo
-- Added extension '#' priority 1 to demo
-- Added extension '#' priority 2 to demo
-- Added extension 't' priority 1 to demo
-- Added extension 'i' priority 1 to demo
-- Added extension '500' priority 1 to demo
-- Added extension '500' priority 2 to demo
-- Added extension '500' priority 3 to demo
-- Added extension '500' priority 4 to demo
-- Added extension '600' priority 1 to demo
-- Added extension '600' priority 2 to demo
-- Added extension '600' priority 3 to demo
-- Added extension '600' priority 4 to demo
-- Added extension '8500' priority 1 to demo
-- Added extension '8500' priority 2 to demo
-- Registered extension context 'default'
-- Including context 'demo' in context 'default'
-- Registered extension context 'sip'
-- Including context 'sip_term' in context 'sip'
-- Including context 'sip_orig' in context 'sip'
-- Registered extension context 'sip_term'
-- Added extension '9727610001' priority 1 to sip_term
-- Including context 'demo' in context 'sip_term'
-- Registered extension context 'sip_orig'
-- Added extension '9727619271' priority 1 to sip_orig
-- Including context 'demo' in context 'sip_orig'
[pbx_wilcalu.so] => (Wil Cal U (Auto Dialer))
[pbx_spool.so] => (Outgoing Spool Support)
/var/spool/asterisk/outgoing
[app_dial.so] => (Dialing Application)
== Registered application 'Dial'
[app_playback.so] => (Trivial Playback Application)
== Registered application 'Playback'
[app_voicemail.so] => (Comedian Mail (Voicemail System))
== Registered application 'VoiceMail'
== Registered application 'VoiceMailMain'
[app_directory.so] => (Extension Directory)
== Registered application 'Directory'
[skipping app_intercom.so]
[app_mp3.so] => (Silly MP3 Application)
== Registered application 'MP3Player'
[app_system.so] => (Generic System() application)
== Registered application 'System'
[app_echo.so] => (Simple Echo Application)
== Registered application 'Echo'
[app_record.so] => (Trivial Record Application)
== Registered application 'Record'
[app_image.so] => (Image Transmission Application)
== Registered application 'SendImage'
[app_url.so] => (Send URL Applications)
== Registered application 'SendURL'
[app_disa.so] => (DISA (Direct Inward System Access) Application)
== Registered application 'DISA'
[app_agi.so] => (Asterisk Gateway Interface (AGI))
== Registered application 'EAGI'
== Registered application 'AGI'
[app_qcall.so] => (Call from Queue)
[app_adsiprog.so] => (Asterisk ADSI Programming Application)
== Registered application 'ADSIProg'
[app_getcpeid.so] => (Get ADSI CPE ID)
== Registered application 'GetCPEID'
[app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
== Registered application 'Milliwatt'
[app_zapateller.so] => (Block Telemarketers with Special Information Tone)
== Registered application 'Zapateller'
[app_datetime.so] => (Date and Time)
== Registered application 'DateTime'
[app_setcallerid.so] => (Set CallerID Application)
== Registered application 'SetCallerID'
[app_festival.so] => (Simple Festival Interface)
== Registered application 'Festival'
[app_queue.so] => (True Call Queueing)
== Registered application 'Queue'
== Manager registered action Queues
== Parsing '/etc/asterisk/queues.conf': Found
[app_senddtmf.so] => (Send DTMF digits Application)
== Registered application 'SendDTMF'
[app_parkandannounce.so] => (Call Parking and Announce Application)
== Registered application 'ParkAndAnnounce'
[app_striplsd.so] => (Strip trailing digits)
== Registered application 'StripLSD'
[app_setcidname.so] => (Set CallerID Name)
== Registered application 'SetCIDName'
[app_lookupcidname.so] => (Look up CallerID Name from local database)
== Registered application 'LookupCIDName'
[app_substring.so] => (Save substring digits in a given variable)
== Registered application 'SubString'
[app_macro.so] => (Extension Macros)
== Registered application 'Macro'
[app_authenticate.so] => (Authentication Application)
== Registered application 'Authenticate'
[app_softhangup.so] => (Hangs up the requested channel)
== Registered application 'SoftHangup'
[app_lookupblacklist.so] => (Look up Caller*ID name/number from
blacklist database)
== Registered application 'LookupBlacklist'
[app_waitforring.so] => (Waits until first ring after time)
== Registered application 'WaitForRing'
[app_privacy.so] => (Require phone number to be entered, if no CallerID
sent)
== Registered application 'PrivacyManager'
[app_db.so] => (Database access functions for Asterisk extension logic)
== Registered application 'DBget'
== Registered application 'DBput'
== Registered application 'DBdel'
== Registered application 'DBdeltree'
[app_chanisavail.so] => (Check if channel is available)
== Registered application 'ChanIsAvail'
[codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
== Registered translator 'gsmtolin' from format 1 to 6, cost 1
== Registered translator 'lintogsm' from format 6 to 1, cost 4
[codec_mp3_d.so] => (MP3/PCM16 (signed linear) Translator (Decoder only))
== Registered translator 'mp3tolin' from format 4 to 6, cost 7
[codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
== Registered translator 'lpc10tolin' from format 7 to 6, cost 3
== Registered translator 'lintolpc10' from format 6 to 7, cost 9
[codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
== Registered translator 'adpcmtolin' from format 5 to 6, cost 1
== Registered translator 'lintoadpcm' from format 6 to 5, cost 1
[codec_ulaw.so] => (Mu-law Coder/Decoder)
== Registered translator 'ulawtolin' from format 2 to 6, cost 1
== Registered translator 'lintoulaw' from format 6 to 2, cost 1
[codec_alaw.so] => (A-law Coder/Decoder)
== Registered translator 'alawtolin' from format 3 to 6, cost 1
== Registered translator 'lintoalaw' from format 6 to 3, cost 1
[codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
== Registered translator 'alawtoulaw' from format 3 to 2, cost 1
== Registered translator 'ulawtoalaw' from format 2 to 3, cost 1
[format_g723.so] => (G.723.1 Simple Timestamp File Format)
== Registered file format g723sf, extension(s) g723
[format_wav.so] => (Microsoft WAV format (8000hz Signed Linear))
== Registered file format wav, extension(s) wav
[format_mp3.so] => (MPEG-1,2 Layer 3 File Format Support)
== Registered file format mp3, extension(s) mp3|mpeg3
[format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
== Registered file format wav49, extension(s) WAV
[format_gsm.so] => (Raw GSM data)
== Registered file format gsm, extension(s) gsm
[format_vox.so] => (Dialogic VOX (ADPCM) File Format)
== Registered file format vox, extension(s) vox
[format_pcm.so] => (Raw uLaw 8khz Audio support (PCM))
== Registered file format pcm, extension(s) pcm|ulaw|ul|mu
[format_g729.so] => (Raw G729 data)
== Registered file format g729, extension(s) g729
[format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
== Registered file format alaw, extension(s) alaw|al
[format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format)
== Registered format 'jpg' (JPEG (Joint Picture Experts Group))
[cdr_csv.so] => (Comma Separated Values CDR Backend)
Asterisk Ready.
*CLI> sip debug
SIP Debugging Enabled
*CLI> Sip read:
REGISTER sip:192.1.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK1388
Max-Forwards: 70
From: 9727619271 <sip:9727619271 at 192.1.2.17>;tag=1388
To: 9727619271 <sip:9727619271 at 192.1.2.17>
Call-ID: 1388 at 192.1.2.88
CSeq: 1 REGISTER
Contact: <sip:9727619271 at 192.1.2.88>
Content-Length: 0
Expires: 3600
10 headers, 0 lines
Interface is eth0
IP Address is 192.1.2.17
Using latest request as basis request
Sending to 192.1.2.88 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK1388
From: 9727619271 <sip:9727619271 at 192.1.2.17>;tag=1388
To: 9727619271 <sip:9727619271 at 192.1.2.17>;tag=as782ddb96
Call-ID: 1388 at 192.1.2.88
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Contact: <sip:9727619271 at 192.1.2.17>
Content-Length: 0
to 192.1.2.88:5060
Interface is eth0
IP Address is 192.1.2.17
9 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK6c80f180
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as417a16df
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 4c5f01f278abc2416ce6f50a37e40d19 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.1.2.88:5060
-- Registered SIP '9727619271' at 192.1.2.88 port 5060 expires 3600
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK1388
From: 9727619271 <sip:9727619271 at 192.1.2.17>;tag=1388
To: 9727619271 <sip:9727619271 at 192.1.2.17>;tag=as782ddb96
Call-ID: 1388 at 192.1.2.88
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Expires: 3600
Contact: <sip:9727619271 at 192.1.2.17>;expires=3600
Date: Mon, 07 Jul 2003 23:16:52 GMT
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #1 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK6c80f180
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as417a16df
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 4c5f01f278abc2416ce6f50a37e40d19 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
ÿ±
to 192.1.2.88:5060
NOTICE[40966]: File chan_sip.c, Line 4636 (sip_poke_noanswer): Peer
'9727619271' is now UNREACHABLE!
Sip read:
REGISTER sip:192.1.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.223:5060;branch=z9hG4bK1656
Max-Forwards: 70
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=1656
To: 9727610001 <sip:9727610001 at 192.1.2.17>
Call-ID: 1656 at 192.1.2.223
CSeq: 1 REGISTER
Contact: <sip:9727610001 at 192.1.2.223>
Content-Length: 0
Expires: 3600
10 headers, 0 lines
Interface is eth0
IP Address is 192.1.2.17
Using latest request as basis request
Sending to 192.1.2.223 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.2.223:5060;branch=z9hG4bK1656
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=1656
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as7d35f746
Call-ID: 1656 at 192.1.2.223
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Contact: <sip:9727610001 at 192.1.2.17>
Content-Length: 0
(no NAT) to 192.1.2.223:5060
-- Registered SIP '9727610001' at 192.1.2.223 port 5060 expires 3600
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.2.223:5060;branch=z9hG4bK1656
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=1656
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as7d35f746
Call-ID: 1656 at 192.1.2.223
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Expires: 3600
Contact: <sip:9727610001 at 192.1.2.17>;expires=3600
Date: Mon, 07 Jul 2003 23:17:00 GMT
Content-Length: 0
±
to 192.1.2.223:5060
NOTICE[40966]: File chan_sip.c, Line 4636 (sip_poke_noanswer): Peer
'9727610001' is now UNREACHABLE!
Interface is eth0
IP Address is 192.1.2.17
9 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK510f49b9
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as6ed8172a
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 6cc6420c2cc20d5e4afa46f924aa4f25 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.1.2.88:5060
Retransmitting #1 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK510f49b9
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as6ed8172a
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 6cc6420c2cc20d5e4afa46f924aa4f25 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #2 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK510f49b9
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as6ed8172a
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 6cc6420c2cc20d5e4afa46f924aa4f25 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #3 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK510f49b9
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as6ed8172a
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 6cc6420c2cc20d5e4afa46f924aa4f25 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Sip read:
INVITE sip:9727610001 at 192.1.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK057250637
Max-Forwards: 70
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
To: 9727610001 <sip:9727610001 at 192.1.2.17>
Call-ID: 057250637 at 192.1.2.88
CSeq: 1 INVITE
Contact: <sip:9727619271 at 192.1.2.88>
Content-Type: application/sdp
Content-Length: 138
v=0
o=username 057250637 057250637 IN IP4 192.1.2.88
s=Session SDP
c=IN IP4 192.1.2.88
t=0 0
m=audio 54454 RTP/AVP 0
a=rtpmap:0 PCMU/8000
10 headers, 7 lines
Interface is eth0
IP Address is 192.1.2.17
Using latest request as basis request
Sending to 192.1.2.88 : 5060 (non-NAT)
Capabilities: us - 14, them - 4, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 9727610001 in sip
list_route: hop: <sip:9727619271 at 192.1.2.88>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK057250637
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
Call-ID: 057250637 at 192.1.2.88
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:9727610001 at 192.1.2.17>
Content-Length: 0
to 192.1.2.88:5060
-- Executing Dial("SIP/9727619271-f956", "SIP/9727610001") in new stack
NOTICE[106510]: File app_dial.c, Line 476 (dial_exec): Unable to create
channel of type 'SIP'
== Everyone is busy at this time
Interface is eth0
IP Address is 192.1.2.17
9 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK7c1302b3
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as2826cca6
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 1f0164fa53ced0ac6b7ba9a94bc37258 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.1.2.223:5060
Retransmitting #1 (no NAT):
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK7c1302b3
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as2826cca6
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 1f0164fa53ced0ac6b7ba9a94bc37258 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
©
to 192.1.2.223:5060
Retransmitting #2 (no NAT):
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK7c1302b3
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as2826cca6
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 1f0164fa53ced0ac6b7ba9a94bc37258 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
©
to 192.1.2.223:5060
Retransmitting #3 (no NAT):
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK7c1302b3
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as2826cca6
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 1f0164fa53ced0ac6b7ba9a94bc37258 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
©
to 192.1.2.223:5060
Interface is eth0
IP Address is 192.1.2.17
9 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK3e1cedb2
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as55995ec2
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 3e24dde234780b48561d6b4d78787c4a at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.1.2.88:5060
Retransmitting #1 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK3e1cedb2
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as55995ec2
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 3e24dde234780b48561d6b4d78787c4a at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #2 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK3e1cedb2
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as55995ec2
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 3e24dde234780b48561d6b4d78787c4a at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #3 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK3e1cedb2
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as55995ec2
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 3e24dde234780b48561d6b4d78787c4a at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
-- Timeout on SIP/9727619271-f956
-- Executing Goto("SIP/9727619271-f956", "#|1") in new stack
-- Goto (sip,#,1)
-- Executing Playback("SIP/9727619271-f956", "demo-thanks") in new stack
We're at 192.1.2.17 port 14992
Answering with capability 4
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK057250637
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
Call-ID: 057250637 at 192.1.2.88
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:9727610001 at 192.1.2.17>
Content-Type: application/sdp
Content-Length: 127
v=0
o=root 9657 9657 IN IP4 192.1.2.17
s=session
c=IN IP4 192.1.2.17
t=0 0
m=audio 14992 RTP/AVP 0
a=rtpmap:0 PCMU/8000
to 192.1.2.88:5060
-- Playing 'demo-thanks'
Sip read:
ACK sip:9727610001 at 192.1.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK057250637
Max-Forwards: 70
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
Call-ID: 057250637 at 192.1.2.88
CSeq: 1 ACK
Content-Length: 0
8 headers, 0 lines
Interface is eth0
IP Address is 192.1.2.17
9 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK34fa6a70
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as7abe0d0c
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 4acb187a664b125a21cd5de640869736 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.1.2.223:5060
-- Executing Hangup("SIP/9727619271-f956", "") in new stack
== Spawn extension (sip, #, 2) exited non-zero on 'SIP/9727619271-f956'
set_destination: Parsing <sip:9727619271 at 192.1.2.88> for address/port to
send to
set_destination: set destination to 192.1.2.88, port 5060
Reliably Transmitting:
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK2ac72fe9
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 057250637 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.1.2.88:5060
Retransmitting #1 (no NAT):
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK34fa6a70
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as7abe0d0c
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 4acb187a664b125a21cd5de640869736 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
I
to 192.1.2.223:5060
Retransmitting #1 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK2ac72fe9
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 057250637 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #2 (no NAT):
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK34fa6a70
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as7abe0d0c
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 4acb187a664b125a21cd5de640869736 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
I
to 192.1.2.223:5060
Retransmitting #2 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK2ac72fe9
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 057250637 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #3 (no NAT):
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK34fa6a70
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as7abe0d0c
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 4acb187a664b125a21cd5de640869736 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
I
to 192.1.2.223:5060
Retransmitting #3 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK2ac72fe9
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 057250637 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #4 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK2ac72fe9
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 057250637 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #5 (no NAT):
BYE sip:9727619271 at 192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK2ac72fe9
From: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
To: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
Contact: <sip:9727610001 at 192.1.2.17>
Call-ID: 057250637 at 192.1.2.88
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Interface is eth0
IP Address is 192.1.2.17
9 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK507c23a0
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as0c923538
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 363b7b3108f9a5ae4be228406191425d at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.1.2.88:5060
WARNING[40966]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries
exceeded on call 057250637 at 192.1.2.88 for seqno 102 (Request)
Retransmitting #1 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK507c23a0
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as0c923538
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 363b7b3108f9a5ae4be228406191425d at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #2 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK507c23a0
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as0c923538
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 363b7b3108f9a5ae4be228406191425d at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Retransmitting #3 (no NAT):
OPTIONS sip:192.1.2.88 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK0218a4a0
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as069f51b9
To: <sip:192.1.2.88>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 4f799cb84e1339230efaa64044dbf237 at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
Sip read:
BYE sip:9727610001 at 192.1.2.17 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK057250637
Max-Forwards: 70
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
Call-ID: 057250637 at 192.1.2.88
CSeq: 1 BYE
Contact: <sip:9727619271 at 192.1.2.88>
Content-Length: 0
9 headers, 0 lines
Interface is eth0
IP Address is 192.1.2.17
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.1.2.88:5060;branch=z9hG4bK057250637
From: 9727619271 <sip:9727619271 at 192.1.2.88>;tag=057250637
To: 9727610001 <sip:9727610001 at 192.1.2.17>;tag=as55bc686a
Call-ID: 057250637 at 192.1.2.88
CSeq: 1 BYE
User-Agent: Asterisk PBX
Contact:
Content-Length: 0
to 192.1.2.88:5060
Interface is eth0
IP Address is 192.1.2.17
9 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:192.1.2.223 SIP/2.0
Via: SIP/2.0/UDP 192.1.2.17:5060;branch=z9hG4bK7348e199
From: "asterisk" <sip:asterisk at 192.1.2.17>;tag=as17b7572f
To: <sip:192.1.2.223>
Contact: <sip:asterisk at 192.1.2.17>
Call-ID: 0db2b29c2fba1b3c3476957620438c7b at 192.1.2.17
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Content-Length: 0
to 192.1.2.88:5060
y
Waiting for inactivity to perform halt...
Asterisk cleanly ending (0).
[root at linux17 asterisk-0.4.0]#
-- End asterisk trace --
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