Home > Failed To > Chan_sip C Handle_response_invite Failed To Authenticate On Invite To

Chan_sip C Handle_response_invite Failed To Authenticate On Invite To

Contents

URL: Previous message: [asterisk-users] Failed to authenticate on INVITE to Anonymous Next message: [asterisk-users] Failed to authenticate on INVITE to Anonymous Messages sorted by: [ date ] [ thread ] while if i >>>>> registered this trunk in softphone like Xlite, there is no problem with >>>>> outbound calls. Code: [2013-07-31 21:46:09] NOTICE[1660] chan_sip.c: Failed to authenticate on INVITE to '"0208802xxxx" ;tag=as1d319b8c' "0208802xxxx" is my number and I am calling an other "020...." number which equates to "134yyyy" - the After your pointing to the PEER settings, I reviewed them. Check This Out

i use sip phones & sip trunk sip.conf & extensions.conf is attached asterisk output is also attached for dial prefix in my campaign i use X i have country code added while if i > registered this trunk in softphone like Xlite, there is no problem > with outbound calls. up vote 1 down vote Your DialPlan is not correct clearly from your configuration files. Аs a first step change your register string like: register => username:[email protected]\Myprovider and then add the Learn More.

Chan_sip C Handle_response_invite Failed To Authenticate On Invite To

So that number is not coming from my system - it is being returned by Sipgate. [*] Talking to Sipgate is a misnomer - email support only! #1 LesD, Jul There they specify SIP-ID rather than the phone number for end of the register string but that also makes no difference.Click to expand... What is the Allure with VDSL ? [TekSavvy] by EdT385.

SOLVED Failed to authenticate on INVITE Discussion in 'Help' started by LesD, Jul 31, 2013. asterisk cli> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status user1/user1 68.198.. Is there any term for this when a movie doesn't end as its plot suggests? My thought is, it is time to buy a SSD drive, grab the latest PBX in a Flash, and have a fun weekend...

Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -------------- next part -------------- An Freepbx Failed To Authenticate On Invite To I am thinking I may miss some items which are required by FPL... · actions · 2012-Aug-27 8:24 am · N9MDToo busy to chatPremium Memberjoin:2005-10-08Boca Raton, FL·VOIPO·voip.ms·Callcentric

N9MD to Trimline Premium Forum content is licensed under a Creative Commons Attribution-ShareAlike 4.0 International License. [asterisk-users] Failed to authenticate on INVITE to Anonymous Jayesh Labade jayesh.labade at gmail.com Wed Jan 4 05:14:20 CST 2012 When I call her number it is routed through one of my Sipgate trunks and now it fails with the message below.

Asterisk®, Digium® and Asterisk logo are registered trademarks of Digium, Inc. Similiar experience? · actions · 2012-Aug-28 8:06 am · Dan_voipjoin:2007-01-03Saint-Hubert, QC Dan_voip Member 2012-Aug-28 6:02 pm When you check your Asterisk, FPL is it registered?asterisk -rx 'sip show registry' voip.freephoneline.ca:5060 N I'm surprised I know more than you about the certainty that you had messed with them. 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

Freepbx Failed To Authenticate On Invite To

The lines now seem OK but I have an other issue now. (I am not sure if this issue arose when the lines failed or only some time after they were At least you had told us, which is +1. Chan_sip C Handle_response_invite Failed To Authenticate On Invite To By using, accessing, or advertising on this site, you agree to waive all legal claims against the following entities and members: PBX in a Flash Development Team, Incredible PBX Development Team, Handle_request_invite Failed To Authenticate Device Do you use asterisk as well, and works well now?

Help me. > > please find sip.conf file in http://pastebin.com/zBGVmdcY > > I have pasted sip debug with verbosity of failed call > http://pastebin.com/jL2ki0s8 > > > Best Regards, > *Jayesh his comment is here So there is nothing we can do? · actions · 2012-Aug-25 3:54 pm · TrimlinePremium Memberjoin:2004-10-24Windermere, FL·voip.ms Trimline Premium Member 2012-Aug-25 4:32 pm I have 5 outbound trunks, it only happened Does every data type just boil down to nodes with pointers? What is the meaning of the message? Chan_sip.c: Failed To Authenticate Device

Outils de la discussion Modes d'affichage #1 01/10/2009, 13h38 ip04mate Junior Member Date d'inscription: octobre 2009 Messages: 12 [ippi] Failed to authenticate on INVITE to ... I removed that but still no improvement. Il est actuellement 04h28. -- English (US) -- franais Nous contacter - Asterisk-France Forum - Archives - Haut de page dit par : vBulletin version 3.8.0 Copyright 2000 - 2017, this contact form Merci.

Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.27 currently running on vicidialnow (pid = 2619) vicidialnow*CLI> Verbosity is at least 21 vicidialnow*CLI> -- Remote UNIX connection vicidialnow*CLI> == Parsing Alerts Alert Preferences Show All... New $200 activation fee for 300MBps Internet?

I then tried using my Voiptalk trunk and that seems to work reliably.

Yes, my password is: Forgot your password? PBX in a Flash would appear to fit the definition of a "project".I am, however, curious as to why the bottom of the home page »www.pbxinaflash.net/ reads:Copyright © 2004-2011, Ward MundyAll E-Mail Just Now From Xfinity..100Mbps [ComcastXFINITY] by hayc59282. No, create an account now.

ForumsJoin Search similar:CircleNet [ONLY BRIEFLY] Downcouple voip questionsiNum down?[Asterisk] SIP URI calls between 2 Asterisk servers[Equipment] Generic OEM Firware For Innomedia MTA6328-2Re?[Anveo] Anveo Direct with OBI Forums → VOIP etc → gmane ! My Thoughts: I feel like I am missing a part of the process, like how User1 is set up to handle outgoing calls... navigate here in the mean time, just check your logs... · actions · 2012-Aug-25 4:32 pm · tbrummell2join:2002-02-09Ottawa, ON tbrummell2 to akoei Member 2012-Aug-27 6:31 am to akoeiWhat is your useragent set

New Home HVAC Setup [HomeImprovement] by daparker219. Asterisk Forums Please hold while I try that extension. thank you. –M. It's why I know you made two changes while having issues: have tried adding [outboundproxy=proxy.live.sipgate.co.uk] per http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asterisk but no change.