Portal Forum Neue Beiträge Hilfe Kalender Community Benutzerliste Aktionen Alle Foren als gelesen markieren Nützliche Links Heutige Beiträge Forum-Mitarbeiter anzeigen Was ist neu? Registration and outbound calls do work as expected, but after 3-4 minutes from registration process or an outgoing call, when testing incoming calls I do get this message on the server: Why is this error showing up? Is it possible for the server to open a new socket to the client that s behind a nat? navigate here
Can three +1/+1 counters be considered one +3/+3 counter? Alles funktioniert prima, inbound & outbound Anrufe, bis ich die FreePBX VM neu starte. What is 'sparrow bath' and how do you do it in airport bathroom? Up loading shortly. http://community.freepbx.org/t/solved-after-reboot-tcptls-c-unable-to-connect-sip-socket-to-ip-of-external-extension-52224-no-route-to-host/28050
The other warnings have disappeared. We recommend upgrading to the latest Safari, Google Chrome, or Firefox. Correct behavour of you app - send SIP OPTIONS message, if timeout - do registration again. why does it take 20-40 minutes for the phone to become available again when the qualify time is set to 60?
Can a creature with multiattack make more than one attack as part of a readied attack? As Rob said, you'll need a static port forward across your internet router to address. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is See how .13 and .14 both trigger errors but in my peer conf file I just have .14 defined. [Oct 13 10:09:53] ERROR  : tcptls.c:350 ast_tcptls_client_start: Unable to connect SIP
Last qualify: 0 [Oct 13 10:10:00] ERROR: tcptls.c:350 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.0.13:5060: Connection refused [Oct 13 10:10:07] ERROR: tcptls.c:350 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.0.14:5060: No ETA can be given. Related 2Using Asterisk as SIP relay server5SIP to PSTN gateway connection from asterisk?0Asterisk SIP server not working for wifi client or client out side LAN3CANCEL request from Android SIP demo ignored An idiom or phrase for when you're about to be ill An item in IEnumerable does not equal an item in List A Page of Puzzling Explain it to me like
the intruision detection is only fail2ban which doesn't block outgoing traffic and reregistering the phone wouldn't help. http://lists.digium.com/pipermail/asterisk-bugs/2010-August/084210.html Asterisk RTP ports sind 10000-20000, Telefon und FreePBX sind beide jeweils hinter einem NAT. Show Paul Belanger added a comment - 03/Aug/10 9:01 AM I can confirm. 'sip show peers' is empty, but tcptls.c still generates errors. Board index All times are UTC All About Voipfone Customer Testimonials Case Studies Press And Awards Our Network Disaster Recovery What Is VoIP?
Sounds like Intrusion Detection been triggered due to TCP traffic. http://popupjammer.com/asterisk-unable/chan-dongle-asterisk-13.html Sign in to comment Contact GitHub API Training Shop Blog About © 2016 GitHub, Inc. also no logs in the firewall of the host system. danach gehen nur noch outbound aber keine inbound Anrufe mehr.
r294734 | jpeeler | 2010-11-11 15:58:25 -0600 (Thu, 11 Nov 2010) | 32 lines Merged revisions 294733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ Powered by vBulletin Version 4.2.3 (Deutsch)Copyright ©2016 Adduco Digital e.K. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. his comment is here The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is
The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. if i reregister the extension on the phone site everything works fine again. r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
I thougth of a cryptolifetime problem, but setting it to yes/no on the pbx and phone side didn't change anything. tcom (Paul Thompson) 2015-03-22 09:45:59 UTC #3 Sam, Is your firewall running? the first "tcptls.c: Unable to connect SIP socket" appeared at 10:23 (immediately after the reboot) and the last one followed by the "Registered SIP" message appeared at 11:12. Jobs At Voipfone Voipfone User Forum Voipfone Blog Quality Mark Guarantee Phone Numbers For Business Voipfone For Business Voipfone For Home Voipfone For Charities FSB Members Free Trial UKBF Members Free
Is it possible to configure a stun sending keep alive messages to the client in order to mantain the socket open and when having incoming messages the server will be able everything works fine, inbound and outbound calls unless I reboot the free-pbx Virtual machine. The patch works! http://popupjammer.com/asterisk-unable/asterisk-unable-to-get-your-ip-address.html Geändert von sam-j (03.04.2015 um 12:59 Uhr) Zitieren 22.03.2015,13:42 #2 sam-j Profil Beiträge anzeigen Private Nachricht IPPF-Einsteiger Registriert seit 21.03.2015 Beiträge 3 next try, diesmal nur 21 Minuten, bis das Telefon
And yes, you need send keepalives(recomended method - OPTIONS message) or setup keepalive on asterisk side and setup in your side correct answer. But between these 10 minutes there is no keep alive mechanism neither on the client neither on the server. the log says: [2015-03-21 09:24:06] ERROR tcptls.c: Unable to connect SIP socket to[IP of external extension]:52224: No route to host all external non tls & srtp encrypted phones are still working. issues.asterisk.org runs on a server provided by Digium, Inc.
Aktivitäten Erweiterte Suche Forum VoIP-(Software) TK-Anlagen Asterisk FreePBX, TrixBox ([email protected]) [Gelöst] fpbx distro reboot: tcptls.c: Unable to connect SIP socket to [...] No route to host Wenn dies Ihr erster Besuch If the Châ€™in dynasty was so short-lived, why was China named for it? This scenario has the potential to progress to the point of saturating a link just from options packets. 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
The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. Join them; it only takes a minute: Sign up asterisk Unable to connect SIP socket to ip:port Connection timed out up vote 0 down vote favorite I am working on a The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is Show Simon M added a comment - 13/Oct/10 9:48 AM Ok, so before applying the patch I could reproduce the issue.
Is there a way to bypass this or is this a bug? [Aug 2 14:53:56] ERROR: tcptls.c:350 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.0.228:5060: Connection refused [Aug 2 14:54:10] ERROR: Registered SIP '9881' at [IP of external extension]:52475 if I expose the external extension in a dmz after reboot I get ERROR tcptls.c: Unable to connect SIP socket to [IP of Atlassian [asterisk-bugs] [Asterisk 0017779]: tcptls.c:350 Unable to connect SIP socket Connection refused Asterisk Bug Tracker noreply at bugs.digium.com Thu Aug 5 15:15:56 CDT 2010 Previous message: [asterisk-bugs] [Asterisk 0017779]: tcptls.c:350 Unable Not the answer you're looking for?