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Unable To Re Open Dsp Device Asterisk

asked 5 years ago viewed 2221 times active 5 years ago Blog Stack Overflow Gives Back 2016 Visit Chat Related 3How to know what is the default audio device? /dev/audio or But the current state of affairs is that I'm continuing with the book, and the next chapter looks (marginally) better. I originally made the mistake of installing using the default Ubuntu apt-get repos', which are almost always outdated by 1-2 versions. The next section tells you how to set up a soft phone, though the Asterisk configuration isn't complete yet. http://webjak.net/unable-to/unable-to-re-open-dsp-device-dev-dsp-asterisk.html

RE: Unable to re-open DSP device /dev/dsp - Added by striker 247 over 3 years ago you need to mention the context console dial [email protected] make sure the 1005 exist in If you're configuring only SIP (whatever that may be; this is the first mention I've seen in the book, another problem arising from the missing overview chapter), you continue on page Powered by vBulletin Version 4.2.3 (Deutsch)Copyright ©2016 Adduco Digital e.K. The Baum-Sweet Sequence What is the speed of the Force?

All Rights Reserved. Details: Code: /etc/asterisk/extensions.conf ------------------------------------ [default] exten => 1001,1,Answer() exten => 1001,2,Playback(hello-world) exten => 1001,3,Hangup() ------------------------------------ *CLI> console dial 1001 *CLI> [Jul 18 21:02:07] WARNING[29544]: chan_oss.c:486 setformat: Unable to re-open DSP device Ich habe jetzt zum Test drei unterschidliche VoIP-Softphones im Netzwerk. Once I found the log files (in FreeBSD they're in in /var/log/asterisk), I found the messages: Dec 29 11:44:02 WARNING[63086]: Unable to open pseudo channel for timing...

Asterisk Forums Please hold while I try that extension. And how? rtp.conf is undocumented beyond stating that it's an RTP configuration file, whatever that might mean. Aktivitäten Erweiterte Suche Forum VoIP-(Software) TK-Anlagen Asterisk Asterisk Allgemein chan_oss.c:486 setformat: Unable to re-open DSP device /dev/dsp: No such file or dir Wenn dies Ihr erster Besuch hier ist, lesen Sie

Alle Rechte vorbehalten. Do you have calling in from > the phone/BTS working? > > > On Tue, Sep 10, 2013 at 2:55 PM, Impossible Is Nothing > wrote: > >> Hi, Powered by Redmine © 2006-2011 Jean-Philippe Lang [asterisk-users] chan_oss.c:585 setformat: Unable to re-open DSP device David @ULC ucoms2001 at gmail.com Sun Feb 8 22:29:53 CST 2009 Previous message: [asterisk-users] meetme application Hätte nicht gedacht das davon was in der CLI steht wenn ich das eigentlich am Switch/Router/Firewall festlege.Tim hat geschrieben:/dev/dsp ist das "Audio"-Inferface im Linux, also die Soundkarte, damit der Asterisk auf

Default-Wert ist /dev/dsp.GrußTim Dipl.-Ing. Client IP: 216.168.169.103 > == Spawn extension (local, 919545090201, 2) exited non-zero on 'OSS/dsp' > -- Executing DeadAGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new > stack > -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 Everything seemed to run smoothly and no errors were apparent during the install process. Client IP: 216.168.169.103 == Spawn extension (local, 919545090201, 2) exited non-zero on 'OSS/dsp' -- Executing DeadAGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing DeadAGI("OSS/dsp", "agi://

oddly enough, no other program are running that utilize sound, so I haven't the foggiest idea why it is happening! Conclusion I can't really conclude yet, of course; I'm not done. The next line is srvlookup=yes The text continues to explain that this means that Asterisk will use DNS SRV records, “a way of setting up a logical, resolvable address where you Hot Network Questions What's this round token depicting a knight?

At the time of writing, I still don't have a functional configuration. this contact form It's read in on startup and (according to the comments) on login. modules.conf describes the how to handle loadable modules. Maybe, by the time I've fought my way through to the end of the book, I'll be happier with it.

Okt 2011, 15:58 Asterisk Anfänger Problem Zitat Beitragvon Hitman4022 » Mi 5. The book has three authors, something that I've always found a problem. This is the kind of difference which makes me wonder why I bothered to buy the book. have a peek here Lightning Component - maintaining a counter in Server Controller using Static Variable Are zipped EXE files harmless for Linux servers?

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Tim GrünbergdCAP zertifiziert seit 2007Debian Wheezy x86_64Asterisk 13.5.0 mit JNET-DAHDI 1.0.14Junghanns QuardBRI, TDM400PSNOM 370, SNOM M3 & Grandstream GXP2000AMD Athlon II X4 605e -- 10 GB DDR3 RAM -- 2 TB

Configuration files I'm still trying to understand the configuration files. It expires in 3600 seconds, and it's an Internet domain SRV record (this is the generic syntax for DNS records) It has a priority of 0 and order 0. from the below link :http://goautodial.org/projects/goautodialce/wiki/64bit I want to dial a number from console as below and I m getting issues : *CLI> console dial 1005 [Jun 18 11:44:08] WARNING[20447]: chan_oss.c:687 setformat: sip.conf, extensions.conf und voicemail.conf sind angepasst.

Alle Anrufe funktionieren. Public keys are named /var/lib/asterisk/keys/.pub while private keys are named /var/lib/asterisk/keys/.key. It differs from the FXO by generating a dial tone. Check This Out After making the following change, I was able to start Asterisk and leave it running. --- modules.conf-dist Thu Dec 29 10:32:13 2005 +++ modules.conf Thu Dec 29 13:03:07 2005 @@ -30,7

Greg's diary Greg's photos Greg's links Greg's home page $Id: asterisk.html,v 1.2 2005/12/30 02:05:35 grog Exp $ Registrieren Hilfe Angemeldet bleiben? Mac OSX: Remove Inactive iMessage Account In Finder use the Go menu and hold the ALT key down to see the Library Option. Dec 29 11:44:02 WARNING[63086]: Unable to lookup 'static' Dec 29 11:44:02 WARNING[63086]: Unable to open IAX timing interface: No such file or directory Dec 29 11:44:02 WARNING[63086]: Unable to get our Do you have calling in from the phone/BTS working?

Implement zero-touch automation to replace manual, redundant tasks >> >> http://pubads.g.doubleclick.net/gampad/clk?id=51271111&iu=/4140/ostg.clktrk >> _______________________________________________ >> Openbts-discuss mailing list >> [email protected] >> https://lists.sourceforge.net/lists/listinfo/openbts-discuss >> >> > SourceForge About Site Status @sfnet_ops Powered by 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 Suchen Sie sich einfach das Forum aus, das Sie am meisten interessiert. Output the first position in your program for each input character Help, my office wants infinite branch merges as policy; what other options do we have?

Chapter 2 describes how to prepare a system, and chapter 3 explains how to install Asterisk, something that the FreeBSD ports collection does for me, so I started at chapter 4, On Sep 10, 2013 6:44 PM, "Kurtis Heimerl" wrote: > Asterisk is a hairy beast, it's hard to say. It was less help than I had hoped; it seems to have been sloppily and hurriedly produced. Nov 2008, 00:02 Wohnort: Freilassing / Oberbayern / Deutschland Kontaktdaten: Kontaktdaten von Tim ICQ Website Re: Asterisk Anfänger Problem Zitat Beitragvon Tim » Di 18.

Tim GrünbergdCAP zertifiziert seit 2007Debian Wheezy x86_64Asterisk 13.5.0 mit JNET-DAHDI 1.0.14Junghanns QuardBRI, TDM400PSNOM 370, SNOM M3 & Grandstream GXP2000AMD Athlon II X4 605e -- 10 GB DDR3 RAM -- 2 TB Vinz Zitieren « Vorheriges Thema | Nächstes Thema » Ähnliche Themen Unable to re-open DSP device /dev/dsp: Permission denied Von EiserneGarde im Forum Asterisk Allgemein Antworten: 7 Letzter Beitrag: 04.02.2010, 12:31 To resolve, enter: console hangup Posted by Tim Salmon at 05:47 Email ThisBlogThis!Share to TwitterShare to FacebookShare to Pinterest Labels: Asterisk No comments: Post a Comment Newer Post Older Post Home vorzunehmen.Wenn du auf der Asterisk-Konsole den "dial" Befehl verwendest, bietet Asterisk die Möglichkeit, direkt den Server als "Telefon" zu verwenden, dafür wird eine Sound-Karte benötigt - eh klar - es ist

The documentation I started by buying the book Asterisk: The Future of Telephony , from O'Reilly. If so then kindly let me know. Here's what I have so far: asterisk.conf describes the directories which asterisk uses. It seems that default is, in fact, a name for this particular entry, and there can be more.

com [Download message RAW] [Attachment #2 (multipart/alternative)] Any Help ? Deshalb habe ich diese genommen: http://downloads.asterisk.org/pub/te....6.2.19.tar.gz Zitieren 19.07.2011,17:53 #2 Vin2 Profil Beiträge anzeigen Private Nachricht IPPF-Einsteiger Registriert seit 19.07.2011 Beiträge 4 chan_alsa VS chan_oss Hallo nochmal, 1, ist es möglich, dass