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News: Need fast expert assistance?https://www.pfsense.org/support Home Help Search Login Register pfSense Forum» pfSense English Support» Firewalling» Asterisk can't connect to SIP-Provider after DSL reconnect « previous next » Print Pages: [1] Sven SvenV 2012-09-06 05:57:22 UTC #6 Hello SkykingOH, I connected the server at another place and it works !So, I think it's something with the firewall or ??? The defaultvalue for FXS1 is 5004, FXS2 is 5008.Xlite default to port 8000 and Samsung WiFiPhone use 12000.But, as you said, RTP port is assigned dinamically ???I'm confused (more) ! If you can ping enable SIP debug to the IP address of voip.eutelia.it' (sip set debug ip xx.xx.xx.xx) and see if the data in the log (/var/log/asterisk/full) offers any clues. weblink

Will it work for us? 3 35 2016-12-16 You are currently not connected to any networks windows 10 5 58 3d Why Voice-Over-IP is Unreliable for Faxing and how the Cloud my configuration is as follows: I have a ubuntu linux box (8.04) which runs my asterisk. Thanks ! What I discovered is that in these days my provider is working on the net, with some interruption between 2:00 and 4:00 AM.May be the original cause.So I suppose tomorrow morning http://forums.asterisk.org/viewtopic.php?f=1&t=84744

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Thanks !!Sven SkykingOH 2012-09-04 16:11:25 UTC #2 can you pint the domain from the Asterisk server? Thanks thanks thanks.Charles SkykingOH 2011-09-04 21:52:17 UTC #9 I have never seen a fixed RTP port configuration. Logged BaFu Newbie Karma: 0 Posts: 23 Re: asterisk[1559]: NOTICE[1599]: 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 It is the base RTP port for channel 0.

Cheers! 0 LVL 4 Overall: Level 4 IP Telephony 2 Networking 1 Message Expert Comment by:rbdnz ID: 242634992009-04-29 curiously, what does the output of asterisk -rx 'sip show registry' look system (system) 2014-06-04 19:41:09 UTC #8 Home Categories FAQ/Guidelines Terms of Service Privacy Policy Powered by Discourse, best viewed with JavaScript enabled Log In Eutelia Registration TimeOut General Help cguasco 2011-09-03 Unfortunatly I'm affected. Registration For Sip Flowroute Com Timed Out Trying Again 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

try the above &see how it goes... Sip Registration Timed Out SkykingOH 2012-09-05 13:35:57 UTC #4 I meant to say "ping" can the Asterisk server resolve and ping the host defined in the trunk and registration string? This is a starting point to debugThere is an open issue in pfsense, not killing firewall states correctly when wan ip address changes or link goes down. have a peek at this web-site at 16:44:42 ?

Would more Full Nodes help scaling and transaction speed? Freepbx Registration Expiry Is there a reason why similar or the same musical instruments would develop? This morning the problem has repeated.Follow a summary of "full" log file. below is it but its fine now as you are meaning when the problem happens, so I will have to wait for it to break again :( I think I remember

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Join & Ask a Question Need Help in Real-Time? Covered by US Patent. Freepbx Registration For Timed Out Trying Again it's caused us nothing but problems. Freepbx Trunk Registration Timeout Unix & Linux Stack Exchange works best with JavaScript enabled Welcome, Guest.

I will try doing that though and see what the response is. (also I don't think NZ has any SIP servers that could possibly be overloaded due to our monopolistic Telcos have a peek at these guys VOIP Event Calendar PBX Internet Speed Test About Voip-info.org Business VOIP Business Voip Providers IP PBX Asterisk Based PBX Hosted PBX Virtual PBX VOIP Billing PBX Phone System SBCs / Softswitch As I have upgraded the version of Asterisk to 1.6 and the problems keep happening I figured it must be to do with the .conf files as there is so many 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? Chan_sip C Registration Timed Out

we use them extensively ourselves, but make sure if you have the "sip application layer gateway" function in your router, you disable it. 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 There's an article about VoIP-Config:https://doc.pfsense.org/index.php/VoIP_ConfigurationWhich contains a link to https://doc.pfsense.org/index.php/Static_Port but I don't get it right:QuoteIn many cases you must enable advanced outbound NAT and not rewrite the source port on check over here for RTP port, this is from Grandstream HT386 manualLocal RTP port:This parameter defines the local RTP-RTCP port pair the HT-386 will listenand transmit.

Many tks in advanceCharles SkykingOH 2011-09-03 17:50:50 UTC #2 First stop yelling (with the capitals and explanation points), it doesn't make anyone want to help you. Asterisk Sip Registration Timeout Do they wish to personify BBC Worldwide? Please use '_X.' instead at line 844 of extensions.confJul 29 09:02:15 asterisk[1594]: NOTICE[1618]: chan_sip.c:21734 in handle_response_peerpoke: Peer 'SIP-PROVIDER-19090294034e9de54b147e6' is now Reachable. (42ms / 2000ms)And a week or some time later I

I can reset my ADSL router and then restart asterisk without rebooting PC and it will start working again!

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Nov 7 04:01:28 asterisk[1819]: NOTICE[1847]: chan_sip.c:13706 in sip_reg_timeout: -- Registration for '[email protected]' timed out, trying again (Attempt #2)
Nov 7 04:01:29 asterisk[1819]: NOTICE[1847]: chan_sip.c:13706 in sip_reg_timeout: -- Registration So its looking like my ADSL router may have something to do with it.. Now that both accounts are configured on my Askozia Box they don't reconnect after my DSL reconnect at 4AM...First thing I tried was rebooting the Askozia Box, guess what no success...Then Sip_reg_timeout Get a free login here: Register Thanks! - Find us on Google+ Page Changes | Comments Featured - Business VoIP Residential VoIP Last modif pagesVoIP Providers CanadaHow to start a VOIP

Asterisk SIP option srvlookup (sip.conf)Synopsis:srvlookup = yes | noDefaultsrvlookup=yes (As of version 1.4.14*)srvlookup=no (Prior to version 1.4.14)* https://issues.asterisk.org/bug_view_page.php?bug_id=10954If srvlookup is turned on, Asterisk supports DNS SRV lookups partially. A single word for "the space in between" Sunlight and Vampires Move directories despite of errors Are people of Nordic Nations "happier, healthier" with "a higher standard of living overall than Join our community for more solutions or to ask questions. this content Can the integral of a function be larger than function itself?

up untill a few weeks ago I was running this on a debian server and would just reboot the system each time this problem happens and it would fix itself, but cguasco 2011-09-13 06:28:59 UTC #10 Sorry for the delay, but I was traveling. here's a couple of things you can do: when it happens just issue a "sip reload" at your asterisk cli prompt. it's caused us nothing but problems.

obelisk 2011-09-03 18:43:42 UTC #3 during a night ? 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 Example: FWD (Free World Dialup) ; We match on IP address of the proxy for incoming calls ; since we can not match on username (caller id) type=peer context=from-kiwi host=202.180.76.167 [sipout] we'll see if we can find anything... 0 Message Author Comment by:timbo007 ID: 243606892009-05-11 so it broke again....

obviously, xx.yy.zz.ww is my public IP. ...................omissis................[2011-09-04 04:02:21] VERBOSE[15047] logger.c: Asterisk Queue Logger restarted[2011-09-04 04:02:21] VERBOSE[15047] asterisk.c: -- Remote UNIX connection disconnected[2011-09-04 04:02:25] NOTICE[2702] chan_sip.c: -- Registration for '[email protected]' timed out, 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 Providers offering unlimited calling plans may have restrictions. we use them extensively ourselves, but make sure if you have the "sip application layer gateway" function in your router, you disable it.

With drives up to 8TB, WD Red offers a wide array of solutions for customers looking to build the biggest, best-performing NAS storage solution. So i choosed a safe 120secondsI will check this out, maybe I'm observing the same issue using PfSense.Thank you very much, for your help! any ideas? RTP ports are assigned in the SDP as part of the invite.

FreePBX provides "hooks" if you must modify in the form of the files with the "custom" filename. Browse other questions tagged ubuntu asterisk or ask your own question. Impossible to troubleshoot a network from forum messages. you could be having NAT issues also.