Keep in touch with Experts ExchangeTech news and trends delivered to your inbox every month Membership How it Works Gigs Live Careers Plans and Pricing For Business Become an Expert Resource To do so you need to not check the static port box.So could someone please describe a bit more detailed where this setting has to be done?The only other option I For active calls, this should not affect you as you have already bonded to the server. I'm just curious to see the registration state. 0 Message Author Comment by:timbo007 ID: 242688232009-04-30 hi rdbnz.. http://smartnewsolutions.com/timed-out/registration-for-timed-out-trying-again-attempt.html
fromdomain=tim host=sip.kiwilink.co.nz ;usereqphone=yes ; This provider requires ";user=phone" on URI call-limit=2 ; permit only 5 simultaneous outgoing calls to this peer the peer canreinvite=no qualify=yes context=from-sipout ;context=maincontext ;disallow=all Scheduling for restart.May 3 01:27:42 init: starting pid 1756, tty '/dev/tty1': '/etc/rc.initial'May 3 01:27:48 asterisk: NOTICE: chan_sip.c:13673 in sip_reg_timeout: -- Registration for 'XXXXXX' timed out, trying again (Attempt #5)May 3 01:28:08 Is there any way to take stable Long exposure photos without using Tripod? as the error seems to suggest that Asterisk cant get hold of the server.
Give an indeterminate limit of a function that is always indeterminate with iterated attempts at l'Hopital's Rule. Thanks ! any ideas? This option is NOT turned on by default!!!A SRV lookup is only performed when the FQDN hostname is specified in the Dial() command; if instead in Dial() you specify a peername
This site is not affiliated with Linus Torvalds or The Open Group in any way. I suppose, that when the problem occurs your ADSL connection is being reset for some reason, at least most providers here in Europe drop the line every 24 hours and then This option is NOT turned on by default!!!A SRV lookup is only performed when the FQDN hostname is specified in the Dial() command; if instead in Dial() you specify a peername Chan_sip.c Sip_reg_timeout I will try rebooting the router every few days and see if that solves it, if it does then that makes it a very annoying problem as I don't want to
Shop Now LVL 4 Overall: Level 4 IP Telephony 2 Networking 1 Message Expert Comment by:rbdnz ID: 242758852009-04-30 i've had a similar problem with one of my sip trunk providers. Sip Registration Timed Out more stack exchange communities company blog Stack Exchange Inbox Reputation and Badges sign up log in tour help Tour Start here for a quick overview of the site Help Center Detailed traceroute command is your friend. · actions · 2014-Sep-3 5:06 pm · TrimlinePremium Memberjoin:2004-10-24Windermere, FL·voip.ms Trimline to Livadia Premium Member 2014-Sep-3 5:27 pm to LivadiaI have seen this on my Asterisk A single word for "the space in between" Why isn't the religion of R'hllor, The Lord of Light, dominant?
Please login or register. 1 Hour 1 Day 1 Week 1 Month Forever Login with username, password and session length Home Help Search Login Register Askozia Forums>AskoziaPBX>Bug Registration For Sip Flowroute Com Timed Out Trying Again Either NAT, DNS, or SIP proxy would be my guess, but more information is needed to find out. Some wrong setting in my config files or the 'yyyy' server down for a minute or so? and I really should fix this problem so it doesn't happen again.
Maybe this is the problem ... 0 LVL 9 Overall: Level 9 IP Telephony 1 Message Expert Comment by:tkalchev ID: 242594392009-04-29 Maybe a simple corn job which restarts asterisk every What to do about a player who takes risks and dies (without consequence)? Freepbx Registration For Timed Out Trying Again asked 1 year ago viewed 477 times active 1 year ago Related 3VoIP dialer for SIP Hot Network Questions How can "USB stick" online identification possibly work? Chan_sip C Registration Timed Out Below is the "sip show registry" during when it was down, reconnecting and then online... Connected to Asterisk 184.108.40.206 currently running on tim-desktop (pid = 22628) tim-desktop*CLI> sip show registry
Logged mircsicz Jr. http://smartnewsolutions.com/timed-out/what-does-timed-out-mean-on-internet.html O > Asterisk está alarmando sempre, se deslogando e logando novamente. > Estou usando codec g729 e existe uma banda garantida de 512kbps. Logged BaFu Newbie Karma: 0 Posts: 23 Re: asterisk: NOTICE: chan_sip.c:13673 in sip_reg_timeout « Reply #1 on: July 29, 2013, 09:28:18 AM » I'm observing same issue using Askozia 2.2.4.I have ForumsJoin Search similar:Cisco 877 losing NTP servers after "reload" IOS 12.4Weekly packet loss issue - Detroit, MI[General] My Incoming / Outgoing Cost: What are you paying?[TekTalk] Failure of TEKTALK Forums → Freepbx Trunk Registration Timeout
SvenV 2012-09-05 13:52:08 UTC #5 Hello, The asterisk server can indeed ping the provider XXXX.XXXX.weepee.org .Even when i ping with the port 5060Result: icmp_seq=1 ttl=55 time=9.84 msI already set Qualify to Abaixo configuração da conta e log do Asterisk. > Obs.: Existe mais de uma conta configurada no meu asterisk. > >  > type=friend > username=2121580613 > secret=XXXXXXXXXX > domain=220.127.116.11 > Board index The team • Delete all board cookies • All times are UTC - 6 hours Powered by phpBB © 2000, 2002, 2005, 2007 phpBB Group Log In Sip registration Source SvenV 2012-09-05 08:53:00 UTC #3 Hello, Thanks for your fast response.Please find here my settings: Outgoing peer details:Trunk Name: s username=32XXXXXXXXXtype=friendtrustrpid=yessendrpid=yessecret=XXXXXXXXXXXXqualify=yesinsecure=veryhost=sswX.XXXXXXX.weepee.orgfromuser=32XXXXXXXXXfromdomain=sswX.XXXXXXX.weepee.orgdtmfmode=inbanddisallow=allcontext=from-trunkallow=alaw No sttings set yet for incoming Register String32XXXXXXXXXXX:[email protected] SIP Log:
All rights reserved. Asterisk Sip Registration Timeout Thanks ! I have 1 external sip account to my sip provider sip.kiwilink.co.nz this works fine for a certain period then I have to restart my whole system before it will work again.
you could be having NAT issues also. Join our community for more solutions or to ask questions. That seems logical so I will monitor my IP and see if it changes. Logged tom76dc Full Member Karma: 2 Posts: 80 Re: asterisk: NOTICE: chan_sip.c:13673 in sip_reg_timeout « Reply #5 on: July 29, 2013, 02:58:57 PM » Hi againI'm also using pfsense.
I think for sure the problem lies somewhere in the linksys and I think you have helped prove it. all these can call each other no problem via asterisk using sip accounts. What effect does it have (when it happens)a) on receiving calls,b) on placing calls,c) on an active call?My apologies, if this issue has been discussed in the past. · actions · have a peek here Asterisk Forums Please hold while I try that extension.
my configuration is as follows: I have a ubuntu linux box (8.04) which runs my asterisk. that's what i have in my current production config at the moment and am chugging along. 0 Message Author Comment by:timbo007 ID: 242763692009-04-30 Thanks rdbnz, (are you in NZ? Cheers, Tim here is part of my sip.conf file: [general] context=default ; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) New $200 activation fee for 300MBps Internet?
Last qualify: 2710:23:32: Peer 'yyyy' is now REACHABLE!where 'yyyy' is the VoSP. (I did not want to name a particular one, as this happens at different times to all of those