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Chan_sip.c Failed To Authenticate On Register To


one is gui-less asterisk while the other one is freepbx.. I'm surprised I know more than you about the certainty that you had messed with them. How to select and output text in a string more hot questions question feed about us tour help blog chat data legal privacy policy work here advertising info mobile contact us asked 2 years ago viewed 6123 times active 2 months ago Get the weekly newsletter! http://computerhelpdev.com/failed-to/failed-to-authenticate-on-register-to.php

Does every data type just boil down to nodes with pointers? dicko 2014-06-20 23:29:04 UTC #5 no specific way, perhaps a combination of sip set debug ip th.em.ip.dudes and tcpdump port 5060 and host th.em.ip.dudes to presumably fix your router misconfiguration Home Is this definition of convergence of a sequence equivalent to the actual definition? Zenoph (Bregovich) 2014-06-20 14:31:46 UTC #2 UPDATE: This is what happens in the log files when I create a trunk: `-- Registration for '[email protected](VoIP Provider IP)' timed out, trying again (Attempt

Chan_sip.c: Failed To Authenticate Device

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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 Is it bad practice to use GET method as login username/password for administrators? Why are Zygote and Whatsapp asking for root? Failed To Authenticate Device For Subscribe These are the configuration setups I came up with: Trunk #1 disallow=all
t38pt_udptl=yes Trunk #2 username=
allow=ulaw&alaw&gsm Also the log is quite full of these kind of reports : NOTICE[2859][C-0000004f] chan_sip.c: Failed to

So no NAT. –dylan7 Jun 16 '15 at 16:20 add a comment| up vote 0 down vote I had same issue. Handle_request_invite: Failed To Authenticate Device Can the integral of a function be larger than function itself? Or it's merely an ordinary mistake? https://forum.asterisk.ru/viewtopic.php?p=12833 I did following changes: nat=yes and made sure the Zoiper used udp and STUN share|improve this answer answered Nov 12 '15 at 20:38 Jens 1 add a comment| Your Answer

how do i troubleshoot this one asterisk freepbx share|improve this question asked Sep 8 '15 at 6:51 Efren Al Añora 11 add a comment| 1 Answer 1 active oldest votes up Failed To Authenticate On Invite To Asterisk Sears Sells Craftsman Brand to Stanley [HomeImprovement] by robbin385. Come on, Spectrum [CharterSpectrum] by josephwit343. 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 chan_sip.c: Failed

Handle_request_invite: Failed To Authenticate Device

Finch May 4 '14 at 17:53 add a comment| up vote 1 down vote Your DialPlan is not correct clearly from your configuration files. Аs a first step change your register Newer Than: Search this thread only Search this forum only Display results as threads More... Chan_sip.c: Failed To Authenticate Device 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. Chan_sip.c: Failed To Authenticate Device Elastix Recreate the ASCII-table as an ASCII-table Node modules have 755 permissions, what permissions should I set so that npm don't require sudo?

A reload fixes it temporarily.Once I have some time I'll try to look in the debug for the info, but I was just wondering if anyone had a registration issue?I'll post navigate here Sent" occurring on my box, almost everyday. Hulking hurler is too powerful for campaign, player complains when countered Is it a security vulnerability if the addresses of university students are exposed? Output N in base -10 How to deal with an intern's lack of basic skills? Failed To Authenticate Device Freepbx

When I call her number it is routed through one of my Sipgate trunks and now it fails with the message below. My daughter also has a Sipgate number. 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 Check This Out I have configured two non-identical trunks with different parameters for testing purposes on asterisk (11.7.0) with Elastix (2.4.0 x64) on a Linux based machine (x86_64; 2.6.18) .

I tried a different sipgate trunk and that was OK. Check_auth: Username Mismatch, Have Give an indeterminate limit of a function that is always indeterminate with iterated attempts at l'Hopital's Rule. Is the binomial theorem actually more efficient than just distributing When jumping a car battery, why is it better to connect the red/positive cable first?

Output N in base -10 Is it bad practice to use GET method as login username/password for administrators?

johny87 Сообщений: 39Зарегистрирован: 22 ноя 2011, 05:28 Вернуться наверх Re: Не проходит регистрация (Failed to authenticate on REGIS johny87 » 22 ноя 2011, 08:24 Проблема решена.Переделал Код: выделить всеregister => Or it's merely an ordinary mistake? Zenoph (Bregovich) 2014-06-20 17:32:59 UTC #4 I have contacted both the SIP provider company and the server hosting company and they both reported that everything works just fine on their end. Freepbx Username Mismatch Have Digest Has Not the answer you're looking for?

format block of text Print all ASCII alphanumeric characters without using them Which was the last major war in which horse mounted cavalry actually participated in active fighting? Finch 4016 Note the ';', do you have that IP on your configuration? (Note that it isn't a valid IP address) –pah May 4 '14 at 17:26 Useful Searches Recent Posts PIAF - Your own Linux-based PBX Forums Forum Topics Help This site uses cookies. this contact form Join them; it only takes a minute: Sign up Asterisk: Connecting an Asterisk System To SIP Provider up vote 3 down vote favorite 1 Setup: Centos 6 OS: Linux CentOS 64-Bit

D Auto (No) No 55461 Unmonitored user2/user2 68.198.. E-Mail Just Now From Xfinity..100Mbps [ComcastXFINITY] by hayc59219. It was working for quite a while with no problem. 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

What is the name of these creatures in Harry Potter and the Deathly Hallows? i created a sip trunk for them to connect..here it is [general] context=users realm=training.com bindport=5060 bindaddr= 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 A single word for "the space in between" How can I take a photo through trees but focus on an object behind the trees? Currently when I make an outbound call it produces a "Failed to authenticate" and status is 'CONGESTION' notices.

At least you had told us, which is +1. Hope this helps. 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, Asterisk Forums Please hold while I try that extension.

SIP clients must use type=friend to let them be able to authenticate, receive and and send calls. FreePBX® is a registered trademark of Sangoma Technologies, Inc. ForumsJoin Search similar:[Internet] Asterisk fails to register with new SagemCom 2864Loss of Internet Access on XP Server Since MondayCircleNet issues this morning[CallCentric] CallCentric Down?[Voip.ms] MontrealCirclenet.us Forums → The Site → Old Now I'm getting Registration from failed for Peer is not suppose to register This happens when I try to create a Zoiper account for the peer.

johny87 Сообщений: 39Зарегистрирован: 22 ноя 2011, 05:28 Вернуться наверх Показать сообщения за: Все сообщения1 день7 дней2 недели1 месяц3 месяца6 месяцев1 год Сортировать по: АвторВремя размещенияЗаголовок по возрастаниюпо убыванию Ответить Сообщений: See an overview of how I set these two files up currently: notes: - all username and passwords have been removed for this post. - sip.us is the sip provider sip.conf: DNS SRV lookup is supposedly disabled in Asterisk by default but you will find numerous reports of people saying that they needed to add this entry into their sip.conf before it Check out the FAQ! Пожалуйста, войдите здесь. Часто задаваемые вопросы О нас вопросы тэги пользователи награды старый форум задать вопрос ВСЕ НЕОТВЕЧЕННЫЕ Задайте Ваш вопрос asterisk 1.6 и localphone.com: Failed to

It is likely the srvlookup=no that fixed your issue. Browse other questions tagged voip sip asterisk or ask your own question.