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Chan_sip C Handle_response_invite Failed To Authenticate On Invite To

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Why call it a "major" revision if the suggested changes are seemingly minor? i created a sip trunk for them to connect..here it is [general] context=users realm=training.com bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=gsm language=en trustrpid=yes sendrpid=yes [examconfig](!) type=friend host=dynamic secret=1qaz1qaz qualify=yes callgroup=1 pickupgroup=1 context=users Asterisk®, Digium® and Asterisk logo are registered trademarks of Digium, Inc. I'd suspect Distributel as well. · actions · 2012-Aug-28 6:38 am · akoeijoin:2005-11-03Brampton, ON

akoei Member 2012-Aug-28 8:06 am Any reason you suspect ISP as well? have a peek here

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Chan_sip C Handle_response_invite Failed To Authenticate On Invite To

I am starting to suspect maybe my ISP cause the issue: since distributel bought CIA (owned 3web), not only FPL, my vbuzzer line works not good as well... · actions · 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 ] Cable Modem + Switch = Multiple DHCP IPs? [ComcastXFINITY] by Samir256. I got below output ast18*CLI> originate sip/test02 application dial == Using SIP RTP CoS mark 5 [Jan 4 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to '"Anonymous"

C'était exactement ca ! Why study finite-dimensional vector spaces in the abstract if they are all isomorphic to R^n? Browse other questions tagged asterisk freepbx or ask your own question. I got below output

ast18*CLI> \ originate sip/test02 application dial
  == Using SIP RTP CoS mark 5
[Jan  4 \ 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to \ authenticate on INVITE to '"Anonymous" \

Support forum for the ViciBox ISO Server Install and ISO LiveCD Demo Moderators: enjay, williamconley, Staydog, mflorell, MJCoate, mcargile, Kumba Post a reply 3 posts • Page 1 of 1 Reply Try re-saving your trunk and outbound rule. I want to use a softphone to make outgoing calls, when I make outgoing calls on the softphone it needs to route through my asterisk server and then out to the my company The wait is over! [AT&TU-verse] by dslwanter643.

How normal is it to have published as an undergraduate? Il est actuellement 02h49. -- English (US) -- français Nous contacter - Asterisk-France Forum - Archives - Haut de page Édité par : vBulletin version 3.8.0 Copyright © 2000 - 2016, I see things, not in my soup but in your posts... #4 Hyksos, Aug 2, 2013 LesD Expand Collapse Member Joined: Nov 8, 2009 Messages: 430 Likes Received: 18 Don't more hot questions question feed about us tour help blog chat data legal privacy policy work here advertising info mobile contact us feedback Technology Life / Arts Culture / Recreation Science

Freepbx Failed To Authenticate On Invite To

My daughter also has a Sipgate number. http://pbxinaflash.com/community/threads/failed-to-authenticate-on-invite.13147/ Tous droits réservés. Chan_sip C Handle_response_invite Failed To Authenticate On Invite To Good to see, I'll probably remove that from my conf file now. · actions · 2012-Aug-28 7:28 pm ·

Forums → VOIP etc → VOIP → VOIP Tech Chat« Voip.ms SOLVED Failed to authenticate on INVITE Discussion in 'Help' started by LesD, Jul 31, 2013.

Whose murder is it? navigate here Thanks Vicidial forum. How can I monitor the progress of a slow upgrade? How should I position two shelf supports for the best distribution of load?

When I call her number it is routed through one of my Sipgate trunks and now it fails with the message below. After your pointing to the PEER settings, I reviewed them. by ralph06143 » Fri Jan 04, 2013 3:56 am hi need help,i am from the philippines and i need to call singapore using goautodial.dial plan:exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@goautodial,,tTo)exten => _91XXXXXXXXXX,3,Hangup Check This Out Links given below.

While Dialing \ call fro Xlite send following Sip header F=sip:[email protected].

Has nothing been added since 1969?!?! · actions · 2012-Aug-27 8:49 am · torojoin:2006-01-27Scarborough, ON

toro to akoei Member 2012-Aug-27 9:14 am to akoeiHere's my FPL configuration:register => 1416477xxxx:[email protected]/416477xxxx [fpl_peer] type=friend unique stamp per SSH login Clone yourself! 8-year-old received tablet as gift, but he does not have the self-control or maturity to own a tablet more hot questions question feed about Voici le resultat des differents output : Citation: ip04*CLI> sip show registry Host Username Refresh State Reg.Time ippi.fr:5060 usersip 105 Registered Wed, 07 Oct 2009 19:06:09 ip04*CLI> sip show peers Name/username

This will force asterisk to use the configured trunk [myprovider] instead of directly forward the call to gw.sip.us directly.

Alerts Alert Preferences Show All... like: e.g: you use 6XXX series to dial to the provider: [outgoing] exten => _6XXX,1,Dial(SIP/Myprovider/${EXTEN:0}) exten => _6XXX,2,Hangup and for incoming calls [incoming] include = users ; this will go into share|improve this answer edited Sep 9 '15 at 7:10 answered Sep 8 '15 at 9:31 Sergey S. 7617 add a comment| Your Answer draft saved draft discarded Sign up or Learn More.

After a bit more fiddling, I disabled the line and then re-enabled it and it now works. 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 I've been with FPL for almost 3 years now, with an Asterisk server from day 1, and it didn't work in the beginning. this contact form while if i registered this trunk in softphone like Xlite, \ there is no problem with outbound calls.