[Asterisk-Users] Call Placing timeouts

[Anand S. Katti]

Hello All,


	I have a problem that is alien to me and obvious for some of you
:). I have asterisk setup with few sip clients.

In a proper context, I have mentioned extensions 107 as
simputer at bogus.com

Asterisk Server-------------------------simputer(sip ua)

I can make calls from sipua to asterisk but not reverse way.

I get the following display on asterisk terminal
---------------------
*CLI>
    -- Executing Dial("SIP/clienta-30c9", "SIP/simputer|20|tr") in new
stack
    -- Called simputer
Dec 28 12:00:05 WARNING[-1116775504]: chan_sip.c:665 retrans_pkt: Maximum
retries exceeded on call 05accd28324a5f5a60ccb4d807be5d9d at 144.16.94.105
for seqno 102 (Critical Request)
  == No one is available to answer at this time
Dec 28 12:00:11 WARNING[-1116775504]: chan_sip.c:665 retrans_pkt: Maximum
retries exceeded on call 05accd28324a5f5a60ccb4d807be5d9d at 144.16.94.105
for seqno 102 (Non-critical Request)
Dec 28 12:00:15 WARNING[36199344]: pbx.c:1996 ast_pbx_run: Timeout, but no
rule 't' in context 'sip'

-------

I do tcpdump and see 12 INVITES going from asterisk to sipua,and then 12
CANCEL in a span of 5 secs and the session terminates with being call
established.

Please tell me what could be the problem here ?

----*-------
for you anaysis: i have put my extensions.conf here..

[globals]
;AbsoluteTimeout(3)

[incoming]
;incoming context is tied with channel 4 FXO device
exten => s,1,Answer
exten => s,2,Background(beep)
exten => s,3,Dial(Zap/1,40,tr)
exten => s,4,Playback(vm-isunavail)
;exten => s,5,Dial(SIP/clienta,10,tr)
exten => s,5,Background(vm-enter-num-to-call)
exten => s,6,NoOp,${CALLERID}
;exten => s,8,Dial(Zap/1,20,tr)
;exten =>s,9,Hangup
include =>sip

;This context is used to record voicemenu and its recording properly.
; used to record prompts

[playback]
exten =>20,1,Playback(vm-sorry)
exten =>20,2,Hangup

[record]
 exten => 205,1,Wait(2)
 exten => 205,2,Record(/tmp/asterisk-recording1:gsm)
 exten => 205,3,Wait(2)
 exten => 205,4,Playback(/tmp/asterisk-recording1)
 exten => 205,5,Wait(2)

 exten => 205,6,Hangup
 include =>playback

;sip users
[sipextensions]
exten => 100,1,Dial(SIP/clienta,20,tr)
exten => 101,1,Dial(SIP/salisa,20,tr)
exten => 102,1,Dial(SIP/salisd,20,tr)
exten => 103,1,Dial(SIP/sourabha,20,tr)
exten => 104,1,Dial(SIP/laptop,20,tr)
exten => 105,1,Dial(SIP/anurag,20,tr)
exten => 106,1,Dial(SIP/askatti,20,tr)
exten => 107,1,Dial(SIP/simputer,20,tr)
exten => 108,1,Dial(SIP/geetha,20,tr)
;exten => clienta,1,Dial(SIP/clienta,20,tr)
include=> record

;working
[extensions]
exten => _3X,1,Dial,Zap/4/${EXTEN}
exten => _4X,1,Dial,Zap/4/${EXTEN}
include =>sipextensions

;working
[centrix]
ignorepat => 9
exten => 9,1,Dial,Zap/4/${EXTEN}
exten => _93XXX,2,Dial,Zap/4/${EXTEN:1}
include =>extensions

;working
[local]
ignorepat =>0
exten => 0,1,Dial,Zap/4/${EXTEN}
exten => _0NXXXXXXXXX,2,Dial(Zap/4/${EXTEN:1})
exten => _0NXXXXXXX,2,Dial(Zap/4/${EXTEN:1})
include =>centrix

;working
[longdistance]
ignorepat => 0
exten => 25,1,Dial,Zap/4/${EXTEN}
exten => _250NXNXXXXXXX,2,Dial,Zap/4/${EXTEN:2}
include =>local

;FOR SIP
[sip]
;exten => s,1,Wait(1)
exten => 1000,1,Dial(Zap/1,20,t)
exten => 1000,2,Hangup
include =>sipextensions
include=>local
--------------------
--sip.conf----------------
;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = X.X.X.X        ; Address to bind to
context = sip   ; Default for incoming calls
;context = longdistance
disallow=all
allow=ulaw
alloq=gsm
allow=alaw
allow=iLbc
maxexpirey=180
canreinvite=yes
nat=no
defaultexpirey=160

[askatti]
type=friend
secret=oneday
host=dynamic
username=askatti

[simputer]
type=friend
username=simputer
host=dynamic

------------------------

Warm Regards,
Anand