Here is the output I get when I attempt to call. > > > console dial 2103 > [Jan 8 14:47:37] WARNING: chan_oss.c:489 setformat: Unable to > re-open DSP device /dev/dsp: The OSS sound system is very old now and modern systems normally use ALSA sound instead. It connects via port 5060 (sip) This service is provided by echunga.lemis.com. Help, my office wants infinite branch merges as policy; what other options do we have? http://webjak.net/unable-to/xmlreader-open-unable-to-open-source-data.html
My problem comes afterwards: after attempting to call from the console to one of the IP phones as a test, I got the following error WARNING: chan_oss.c:485 setformat: Unable to re-open Please turn \ off on client if possible. Son's music tastes What's the difference between ls and la? Suchen Sie sich einfach das Forum aus, das Sie am meisten interessiert.
The book does not use the default installed file, but starts again from scratch (on page 73), making it difficult to correlate the entries. Dec 29 11:44:02 WARNING: Unable to lookup 'static' Dec 29 11:44:02 WARNING: Unable to open IAX timing interface: No such file or directory Dec 29 11:44:02 WARNING: Unable to get our Sign up for the SourceForge newsletter: I agree to receive quotes, newsletters and other information from sourceforge.net and its partners regarding IT services and products. Wie kann ich evtl.
No, thanks Registrieren Hilfe Angemeldet bleiben? Ich kämpfe seit zwei Tagen mit folgender Meldung bei meinem ersten "Hello World"-Anruf: Code: *CLI> WARNING: chan_oss.c:486 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Was hat difficult to keep up. Yes, its simply a warning, but its also giving : == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'Console/dsp' status is 'CHANUNAVAIL' For dialing a console should
Thanks. SRV records are not VoIP specific: they're a general way of describing network accessible services. und vBulletin Solutions, Inc. Here's what I have so far: asterisk.conf describes the directories which asterisk uses.
I connected 2 IP phones to the server over the network and they registered fine. Ich höre "Hello World.": Code: *CLI> console dial 1001 *CLI> << Console call has been answered >> [Jul 18 21:34:12] ERROR: chan_alsa.c:457 alsa_read: Read error: Resource temporarily unavailable [Jul 18 21:34:12] RE: Unable to re-open DSP device /dev/dsp - Added by Urmi L over 3 years ago Thank you for your reply. I don't know if I did something wrong during compilation or what...
For some reason it gets read twice. Do you have calling in from > the phone/BTS working? > > > On Tue, Sep 10, 2013 at 2:55 PM, Impossible Is Nothing
Briefly describe the problem (required): Upload screenshot of ad (required): Select a file, or drag & drop file here. âœ” âœ˜ Please provide the ad click URL, if possible: Home Browse this contact form Asterisk is indeed a very hairy beast. anyone got an idea? Here is the output I get when I attempt to call. > console dial 2103 [Jan 8 14:47:37] WARNING: chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or
Client IP: 188.8.131.52
== 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, Powered by vBulletin Version 4.2.3 (Deutsch)Copyright ©2016 Adduco Digital e.K. I suppose it's not surprising that it should have its oven views of what the console output should look like: Fortunately you can turn these colours off with the -n option. have a peek here What is a "frozen ATPL"?
They were able to call each other successfully. 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 The file isn't the default file installed by Asterisk, which includes lots of comments; instead it's hand written, and the text doesn't even describe all the entries. 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 > > Re: [Openbts-discuss] Asterisk 'console dial' From: Impossible Is Nothing
Ergebnis 1 bis 3 von 3 Thema: chan_oss.c:486 setformat: Unable to re-open DSP device /dev/dsp: No such file or dir Themen-Optionen Druckbare Version zeigen Anzeige Linear-Darstellung Zur Hybrid-Darstellung wechseln Zur Baum-Darstellung I understand that I can withdraw my consent at any time. For example, at the end of Chapter 1, the rest of the book is described with chapter numbers off by one. Check This Out I'd assume that a PBX corresponds more to a typical Internet server (for example, a mail transport agent or a web server).
Please turn off on client if > possible. No idea. The book also doesn't describe the entries; the default config file does. kolucoms6 Posts: 432Joined: Tue Aug 14, 2007 5:55 pmLocation: India YIM Top Reply with quote by williamconley » Sun Feb 08, 2009 10:44 pm Asterisk dial from the command line.
Please don't fill out this field. 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 And how? Everything seemed to run smoothly and no errors were apparent during the install process.
Please don't fill out this field. This is the only mention of SRV records in the book, at least as far as I can tell from the index, and it gives no further information. 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: chan_oss.c:687 setformat: 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:
If there is any guide or any link is available then also pls do let me know. It describes parts of a sip.conf file, the configuration file for SIP. Alle Anrufe funktionieren. The configuration file distributed with Asterisk, which this book doesn't use, is more informative: context=default ; Default context for incoming calls So what kind of context do you use for outgoing
But Chapter 4 starts with a description of how to configure Zaptel (not part of Asterisk) for a specific VoIP end user card (Digium Dev-Lite, not easily available in Australia) so