Re: [asterisk-users] Avoided deadlock Error(solved)
Stefan, I had the same problem. And I upgrade to 1.4.37 to resolv. Leonardo Silva 2010/11/25 bayardo.sanc...@gmail.com The proble is dialplan I configure fine -- Sent from my BlackBerry® VoIP, Windows/Linux Administration and Network Management US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -Original Message- From: Stefan Schmidt s...@sil.at Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 24 Nov 2010 22:59:56 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Avoided deadlock Error Am 24.11.2010 13:48, schrieb Bayardo Sanchez: My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem is : Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x861f6d8', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85a6420', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85bc510', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85f9e68', 9 retries! Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided deadlock for '0x85e1db0', 9 retries! this error comes only when I call spain saturated my CLI with the message error hello, as tilghman noticed 1.2 is EOL, but i still use it too and i see a bunch of this messages on different servers and they dont cause any problem at all. if you have some problems with this (except the warning message) you should upgrade. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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 -- Leonardo Silva fone: 16 8143-1146 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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
Re: [asterisk-users] help with channelbank audiocodes MP-124
Carlos, What's the help do you need? Leonardo Silva 2007/9/26, Carlos Hernandez [EMAIL PROTECTED]: Hi: We're offering some sort of reward to that one who can help us For this site we are using trixbox and Asterisk 1.2 More info off list. Many thanks, Carlos ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Leonardo Silva fone: 16 8143-1146 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source VoIP client (on a webpage)
Dear Jason, Here in my company we use an applet it java IAX, and it functions very well! If to want to visit the URL is http://www.virgos.com.br, calls the service as 0800Web. Leonardo Silva 2007/4/5, Jason Wolfe [EMAIL PROTECTED]: I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Leonardo Silva fone: 16 8143-1146 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration problem
2006/10/31, Jon Farmer [EMAIL PROTECTED]: Sergio R. D'Ippolito wrote: Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: */SIP/2.0 401 Unauthorized/* /Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x/ /From: SPA922 sip:[EMAIL PROTECTED] ;tag=685bbad1fae3325do0/ /To: SPA922 sip:[EMAIL PROTECTED];tag=as4da6f6ce/ /Call-ID: [EMAIL PROTECTED] / /CSeq: 5503 REGISTER/ /User-Agent: incore-PBX/ /Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY/ /WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=372b2479/ Asterisk is asking the phone to resend the registration withWWW-Authenticate using MD5 hash. Make sure the phone supports this andretry. Or you could turn this option off in the sip.conf.Regards Jon--Jon FarmerTelford, Shropshire, UK___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersMaybe a Firewall ? -- Leonardo Silvafone: 16 8143-1146 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] alive check for HA constellation
Hi Sebastian, This url http://underlinux.com.br/content/view/6330/70/ have some thinks that you need.Leonardo Silva 2006/10/10, Sebastian Reitenbach [EMAIL PROTECTED]: Hi,I have setup two asterisks with ucarp, to build a HA cluster. Everything worksfine, if one of the machines is going to die completely. But if the asterisksoftware is running, but behaving not correctly, this cannot be detected by the ucarp software.I think I need a script that periodically checks the master, and if the answeris not the expected one, the slave shall try to take over the master.I can imagine this will work when I try to check whether I can successfully authenticate via SIP to the asterisk. Just pipe it through netcat, and waitfor the answer. but I have the feeling that I am not the first one with thatproblem, so I want to ask for more easily/robust tests to make sure the master is running or not.any suggestions are appreciated.kind regardsSebastian___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Leonardo Silvafone: 16 8143-1146 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hangs up on incoming PSTN line to analog extension
Robert, I have the same problem, and I discover that when you use de flash or hang up you need a time toasterisk detect that you not do a flash function. A suggest is put de little time umflash. []' Leonardo Silva 2006/4/23, Robert La Ferla [EMAIL PROTECTED]: I have encountered the following problem with the latest Asterisk source(as of 4/23/2006):Someone calls me on my PSTN line, it then dials my analog extension (I have both SIP and analog phones where all analog phones are a sharedextension.)After a while, I get a busy signal.How can I furtherdiagnose this?What could be the problem?___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Leonardo Silvafone: 16 8146-1143 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
