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Unable To Create Channel Of Type Sip Cause 3

Logged Sophie from Montréal soprom Wiki & Docs Team Offline Posts: 589 Re: Unable to create channel of type 'SIP' (cause 3...) « Reply #3 on: April 11, 2008, 04:35:32 PM I am getting the following call log. Finally, if you HAVE put forth effort, LET US KNOW!!! If it's not running at 3 or higher, set verbose to at least 3. have a peek at this web-site

Also, have you ever been able to place a call over this trunk?Did the carrier include any documentation for setting up their service? [email protected]:~# asterisk -r Asterisk, Copyright (C) 1999 - 2010 Digium, Inc. It does give some solutions for In iax2.conf set rtignoreregexpire = yes and in sip.conf set ignoreregexpire = yes Reload Asterisk #4 blanchae, Apr 16, 2010 (You must log in share|improve this answer edited Mar 30 at 20:31 Alexei 4,82081738 answered Mar 30 at 20:13 M Bagchi 1 add a comment| Your Answer draft saved draft discarded Sign up or

Il est actuellement 17h09. -- English (US) -- français Nous contacter - Asterisk-France Forum - Archives - Haut de page Édité par : vBulletin version 3.8.0 Copyright © 2000 - 2016, Forum owner bears no responsibility for accuracy of participant comments and bears no legal liability for posted discussion content. The provider only gave me username and password and sipserver and no calls from asterisk with this trunk [Nov 20 11:18:55] -- Remote UNIX connection h-87-109*CLI> sip show registry Host Username

Logged Sophie from Montréal SARK devs Home away from home Offline Posts: 2,806 Re: Unable to create channel of type 'SIP' (cause 3...) « Reply #6 on: April 11, 2008, 10:52:33 dicko 2013-03-05 16:18:30 UTC #5 Sorry I don't do windows. Did the carrier include any documentation for setting up their service? Not the answer you're looking for?

Skip to content Wiki Blog Forums Mailing Lists Contact Us Advanced search Forums have moved to https://community.asterisk.org Board index RSS RSS Change font size FAQ Information The requested topic does not Related 0Asterisk and a2billing call problems1SIP channel format. Do streams take advantage of branch-prediction? this Time: 9
[Apr 11 20:01:11] NOTICE[4574]: chan_iax2.c:8580 __iax2_poke_noanswer: Peer '5145552222peer' is now UNREACHABLE!

and others. It's not clear yet. system (system) 2014-06-04 19:51:54 UTC #7 Home Categories FAQ/Guidelines Terms of Service Privacy Policy Powered by Discourse, best viewed with JavaScript enabled Browse other questions tagged linux ubuntu sip asterisk or ask your own question.

General Help namezero111111 2013-03-05 15:59:08 UTC #1 Dear folks, I have a problem debugging a TAPI issue (we are running in device and user mode).ActivaTSP dials extension 555 in this case.Everything UNDERSTANDING at least 25% of how VoIP works is handy! This page has been accessed 7,355 times. also, i recommend not using _X!

Help, my office wants infinite branch merges as policy; what other options do we have? http://brrian.net/unable-to/unable-to-create-channel-on-youtube.html Unfortunately the 200 OK never arrives back at asterisk. It lessens the chance of you getting flamed by some guy who's been working for over 40 hours STRAIGHT and is just tired of seeing email after email after email containing Advanced Softswitch and Billing?

My question is, which two subscribers are absent??? What's the meaning of "farmer by trade"? 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 > Now,Sanjay, don't take this personally, http://brrian.net/unable-to/unable-to-create-channel-of-type-sip-freepbx.html Also, have you ever been able to place a call over this trunk?

Calls can come in but can't get out. Vicidial Installation and Repair, plus Hosting and ColocationSugarCRM integration - Customization and Add-ons - We Bring It All Together.http://www.PoundTeam.com # 352-269-0000 # +44 (203) 769-2294 # +506 4001-8914 williamconley Posts: Please help me to resolve this.

Learn More. "Unable to create channel of type 'SIP' Discussion in 'Help' started by ghurty, Apr 16, 2010.

Man, I'm starting to just get pissed...That's what, 3 questions I've seen in the last 12 or less hours where the person asking the question OBVIOUSLY doesn't want to put forth Explain it to me like I'm a physics grad: Global Warming One for All, and All for One Bought agency bond (FANNIE MAE 0% 04/08/2027), now what? Validate Random Die Tippers 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 Now it's been stable for half a day...

You might want to check this on your server. namezero111111 2013-03-05 16:17:23 UTC #4 The thing is, though, SIP/102 actually rings and the call connects properly.That's what is confusing me. linux ubuntu sip asterisk share|improve this question edited Apr 1 '14 at 8:38 asked Apr 1 '14 at 8:13 Ram 1773721 add a comment| 3 Answers 3 active oldest votes up have a peek here This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions.

Useful tip: Another thing you could do, set qualify=yes on your sip endpoints' configurations, since this is a "no route to host" issue, you'll see failure on at least one of exten => _0.,1,Dial(IAX2/1111/${EXTEN},Ttm) ... That way you will see the dialplan executions that occur just before that message. namezero111111 2013-03-05 16:20:02 UTC #6 I would like to add that once SIP/101 is logged off, none of the two messages occur, so it seems as though Asterisk logs the warning

ghurty Expand Collapse Senior Member Joined: Jan 13, 2009 Messages: 850 Likes Received: 4 I am not sure what happened, but I started getting: "Unable to create channel of type 'SIP' The Anti-Santa: Dealing with the Naughty List Where should a galactic capital be? GOOGLE is your friend! Logged soprom Wiki & Docs Team Offline Posts: 589 Re: Unable to create channel of type 'SIP' (cause 3...) « Reply #2 on: April 11, 2008, 06:37:52 AM » Thanks for

Important changes Recent changes Random page Search Toolbox What links here Related changes Special pages Printable version Permanent link This page was last modified on 14 October 2008, at 14:38. Please note the last two lines with subscriber absent: -- Executing [[email protected]:1] Set("[email protected];2", "__RINGTIMER=15") in new stack -- Executing [[email protected]:2] Macro("[email protected];2", "exten-vm,555,555,0,0,0") in new stack -- Executing [[email protected]:1] Macro("[email protected];2", "user-callerid,") in If you watch the TCP packets (with wireshark) you will see the sip register being sent out from asterisk and then nothing coming back. inbound does but not outgoing.

It seems as though this solves my "problem", but it interesting nonetheless... Alerts Alert Preferences Show All... MOR 0.7+ and does not allow this, for lower versions change login value for provider or device so usernames do not match. So you need register your softphone/phone on asterisk.

Time: 2010
[Apr 11 20:02:03] NOTICE[4574]: chan_iax2.c:7671 socket_process: Peer '4185553333peer' is now REACHABLE!