[asterisk-bugs] [Asterisk 0016436]: asterisk is not able to register with SIP server
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Dec 13 11:14:39 CST 2009
The following issue has been SUBMITTED.
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https://issues.asterisk.org/view.php?id=16436
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Reported By: sam chan
Assigned To:
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Project: Asterisk
Issue ID: 16436
Category: Channels/chan_sip/Registration
Reproducibility: have not tried
Severity: major
Priority: normal
Status: new
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:
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Date Submitted: 2009-12-13 11:14 CST
Last Modified: 2009-12-13 11:14 CST
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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
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Issue History
Date Modified Username Field Change
======================================================================
2009-12-13 11:14 sam chan New Issue
2009-12-13 11:14 sam chan Asterisk Version => 1.2.X
2009-12-13 11:14 sam chan Regression => No
2009-12-13 11:14 sam chan SVN Branch (only for SVN checkouts, not tarball
releases) => N/A
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