Summary: | ASTERISK-15308: asterisk is not able to register with SIP server | ||
Reporter: | Salman Khan (sam chan) | Labels: | |
Date Opened: | 2009-12-13 11:14:38.000-0600 | Date Closed: | 2011-06-07 14:01:01 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Registration |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
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@69.1.224.14' timed out, trying again (Attempt ASTERISK-55) Dec 13 12:50:36 NOTICE[2586]: chan_sip.c:5529 sip_reg_timeout: -- Registration for 'USER ID@69.1.224.14' timed out, trying again (Attempt ASTERISK-56) Dec 13 12:50:57 NOTICE[2586]: chan_sip.c:5529 sip_reg_timeout: -- Registration for 'USER ID@69.1.224.14' timed out, trying again (Attempt ASTERISK-57) Dec 13 12:51:17 NOTICE[2586]: chan_sip.c:5529 sip_reg_timeout: -- Registration for 'USER ID@69.1.224.14' timed out, trying again (Attempt ASTERISK-58) == 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@10.10.10.16:5060 register => VOIP SWITCH ID:VOIP SWITCH PASSWORD@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 | ||
Comments: | By: Leif Madsen (lmadsen) 2009-12-14 09:27:54.000-0600 Please either utilize the ViciDial support forums, or the asterisk-users mailing lists for support issues. |