[asterisk-bugs] [Asterisk 0016436]: asterisk is not able to register with SIP server

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Dec 14 09:27:54 CST 2009


The following issue has been UPDATED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16436 
====================================================================== 
Reported By:                sam chan
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16436
Category:                   Channels/chan_sip/Registration
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.2.X 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 no change required
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-12-13 11:14 CST
Last Modified:              2009-12-14 09:27 CST
====================================================================== 
Summary:                    asterisk is not able to register with SIP server
Description: 
HI,
I have installed ViCiDialNow 2.0.5 with Asterisk version of 1.2.30.2 for
OUTBOUND dialing. This server is not able to register with SIP server as
SIP SHOW REGISTRY shows unregistered. Also, I continuously recieves
following messages on the "vici*CLI>" CLI 

...

Dec 13 12:50:16 NOTICE[2586]: chan_sip.c:5529 sip_reg_timeout:    --
Registration for 'USER ID at 69.1.224.14' timed out, trying again (Attempt
https://issues.asterisk.org/view.php?id=59)
Dec 13 12:50:36 NOTICE[2586]: chan_sip.c:5529 sip_reg_timeout:    --
Registration for 'USER ID at 69.1.224.14' timed out, trying again (Attempt
https://issues.asterisk.org/view.php?id=60)
Dec 13 12:50:57 NOTICE[2586]: chan_sip.c:5529 sip_reg_timeout:    --
Registration for 'USER ID at 69.1.224.14' timed out, trying again (Attempt
https://issues.asterisk.org/view.php?id=61)
Dec 13 12:51:17 NOTICE[2586]: chan_sip.c:5529 sip_reg_timeout:    --
Registration for 'USER ID at 69.1.224.14' timed out, trying again (Attempt
https://issues.asterisk.org/view.php?id=62)
  == Refreshing DNS lookups.
...

My Asterisk Server Configurations are as follows:

Network Configuration:
IPTables are configured and IP Forwarding is all set as the LAN PC's are
able to browse INTERNET through this server.

/etc/asterisk/sip.conf:
[general]
context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is
yes, this can also be set to 'osp'
;realm=mydomain.tld             ; Realm for digest authentication
bindport=5744               ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound
calls
;domain=mydomain.tld            ; Set default domain for this host
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4                 ; Add IP address as local domain
;allowexternalinvites=no        ; Disable INVITE and REFER to non-local
domains
;autodomain=yes                 ; Turn this on to have Asterisk add local
host
;pedantic=yes                   ; Enable slow, pedantic checking for
Pingtel
;tos=184                        ; Set IP QoS to either a keyword or
numeric val
tos=lowdelay                    ;
lowdelay,throughput,reliability,mincost,none
maxexpiry=3600                  ; Max length of incoming registration we
allow
defaultexpiry=300               ; Default length of incoming/outgoing
registration
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI
NOTIFY
;checkmwi=10                    ; Default time between mailbox checks for
peers
;vmexten=voicemail      ; dialplan extension to reach mailbox sets the
;videosupport=yes               ; Turn on support for SIP video
;recordhistory=yes              ; Record SIP history by default
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=gsm                       ;
musicclass=default              ; Sets the default music on hold class for
all SIP calls
language=en                     ; Default language setting for all
users/peers
relaxdtmf=yes                   ; Relax dtmf handling
rtptimeout=60                   ; Terminate call if 60 seconds of no RTP
activity
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP
activity
trustrpid = no                  ; If Remote-Party-ID should be trusted
sendrpid = yes                  ; If Remote-Party-ID should be sent
progressinband=no               ; If we should generate in-band ringing
always
;useragent=Asterisk PBX         ; Allows you to change the user agent
string
promiscredir = no       ; If yes, allows 302 or REDIR to non-local SIP
address
;usereqphone = no               ; If yes, ";user=phone" is added to uri
that contains
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF.
Default: rfc2833
;compactheaders = yes           ; send compact sip headers.
;sipdebug = yes                 ; Turn on SIP debugging by default, from
;subscribecontext = default     ; Set a specific context for SUBSCRIBE
requests
;notifyringing = yes            ; Notify subscriptions on RINGING state
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is
to be rejected,
;regcontext=sipregistrations
;registertimeout=20             ; retry registration calls every 20
seconds (default)
;registerattempts=10            ; Number of registration attempts before
we give up
callevents=no                   ; generate manager events when sip ua
performs events (e.g. hold)
externip = MY PUBLIC STATIC IP     ; Address that we're going to put in
outbound SIP messages
;externhost=foo.dyndns.net      ; Alternatively you can specify an
;externrefresh=10               ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local
networks
localnet=10.0.0.0/255.0.0.0     ; Also RFC1918
localnet=172.16.0.0/12          ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes                         ; Global NAT settings  (Affects all peers
and users)
canreinvite=no
;rtcachefriends=yes             ; Cache realtime friends by adding them to
the internal list
;rtupdate=yes                   ; Send registry updates to database using
realtime? (yes|no)
;rtautoclear=yes                ; Auto-Expire friends created on the fly
on the same schedule
;ignoreregexpire=yes            ; Enabling this setting has two
functions:
; domain=myasterisk.dom
; domain=customer.com,customer-context
; autodomain=yes
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test at 10.10.10.16:5060

register => VOIP SWITCH ID:VOIP SWITCH PASSWORD at 69.1.224.14:5744
; setup account for SIP trunking:
 [SIPtrunk]
 disallow=all
 allow=ulaw
 allow=alaw
 type=friend
 username= VOIP SWITCH ID
 secret= VOIP SWITCH PASSWORD
host=69.1.224.14
 dtmfmode=inband
 qualify=1000









Following is the dial plan in /etc/asterisk/extensions.conf:

...

; dial a long distance outbound number through a SIP provider

 exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
 exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
 exten => _91NXXNXXXXXX,3,Hangup

...


I would appreciate if I recieve help over this issue of registration time
out.

Thank you.

Sam
====================================================================== 

---------------------------------------------------------------------- 
 (0115194) lmadsen (administrator) - 2009-12-14 09:27
 https://issues.asterisk.org/view.php?id=16436#c115194 
---------------------------------------------------------------------- 
Please either utilize the ViciDial support forums, or the asterisk-users
mailing lists for support issues. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-14 09:27 lmadsen        Note Added: 0115194                          
2009-12-14 09:27 lmadsen        Status                   new => closed       
2009-12-14 09:27 lmadsen        Resolution               open => no change
required
======================================================================




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