SkykingOH 2013-04-05 04:29:03 UTC #11 Just the IVR module. Why not just add the IVR from FreePBX web interface? An Array of Challenges #2: Separate a Nested Array How can I keep the computers on my spaceship from dying after a hull breach? Check your format and make sure that you're saving it in the format Asterisk expects. this contact form
Extensions should be in from-internal and trunks in from-pstn starjef 2013-04-05 12:04:28 UTC #15 Hello, I have fixed the problem. cc_card_expired Card expired. cc_end_ivr_1 Press one to dial new number. GBP_one ...pound... pop over to these guys
asked 4 years ago viewed 2171 times active 4 years ago Blog Stack Overflow Gives Back 2016 Developers, Webmasters, and Ninjas: What’s in a Job Title? Act Now! share|improve this answer answered Dec 30 '11 at 23:01 Peter Grace♦ 2,49711735 This worked Thanks! –Jacob Dec 31 '11 at 0:23 add a comment| Your Answer draft saved
VOIP Event Calendar PBX Internet Speed Test About Voip-info.org Business VOIP Business Voip Providers IP PBX Asterisk Based PBX Hosted PBX Virtual PBX VOIP Billing PBX Phone System SBCs / Softswitch Ast_openstream_full File Does Not Exist In Any Format If your channel is encoded as GSM, Asterisk will not look for a .wav of the same name if a .gsm is available. Can the product of two nonsymmetric matrices be symmetric? Asterisk needs to translate the audio between the stored format and the format it sends out over the VOIP connection.
If your channel is encoded as GSM, Asterisk will not look for a .wav of the same name if a .gsm is available. (format (ulaw)): No Such File Or Directory By console dial I can make calls. confirmed file formats. cc_please_enter_number Please enter the number you wish to call followed by the hash (#) key.
By console dial I can make calls. minus minus... Asterisk File Does Not Exist In Any Format seconds ...seconds. Ast_streamfile: Unable To Open starjef 2013-04-05 08:10:52 UTC #13 Hello, I tried this but no luck it stays stereo SkykingOH 2013-04-05 08:53:35 UTC #14 Did you just makeup the context from-localtelecom?
VOIP Event Calendar PBX Internet Speed Test About Voip-info.org Business VOIP Business Voip Providers IP PBX Asterisk Based PBX Hosted PBX Virtual PBX VOIP Billing PBX Phone System SBCs / Softswitch http://popupjammer.com/unable-to/octave-unable-to-determine-file-format.html Business VoIP Residential VoIP Last modif pagesBicom Systemsvoip-info.orgThirdlane Business PBXVoIP Providers IndiaSIP Trunk Providers IndiaSIP Trunk Providers UKPBXwwVoIP OriginationVoIP TerminationVoIP InnovationsShow More… VoIP Speed Test Get HelpAsk a questionSIP Trunk Provider Once the sampling rate was correctly applied at 8000 Hz, MOH began working. Press five to increase your Calling Card balance by using another card. Asterisk Convert Wav To Gsm
When do "dial s" \ from console it wil play demo files that I can here from headphone connected to \ asterisk running system(Android OS).If I play gsm file noise is Press two to redial same number. In which case you might run into an artifact of the way Asterisk starts up its codecs.Asterisk computes the above table by running transcoding tests for each entry in the table navigate here Where I can see the channels is encodes as GSM,and how to change to wav.? *CLI> dial s Recommended Misc Links AstFAQs Store About Me Recent Questions Pjsip Realtime - Endpoints
That'll show the formats available and the file extensions they should have. Sox Convert Wav To Ulaw Asterisk will still work, but will give the error on playing any audio files that aren't ogg.The situation can be much improved by disabling some codecs. enter_voucher_number Please enter voucher number.
Is format_wav.so loaded? 2) Your WAV file is not encoded correctly. Make sure it is in Asterisk compatible format. Nikhil says: March 2, 2011 at 1:50 am Hi I am using Asterisks as client. Asterisk Sounds To see the active translations, do: devonian*CLI> show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
cc_end_ivr_9 Press one to dial new number. Where I can see the channels is encodes as GSM,and how to change to wav.? *CLI> dial s -- Executing [s at default:1] Wait("ALSA/hw:0,0", "1") in new stack The 'dial' command What are those "sticks" on Jyn Erso's back? his comment is here enter_pin Please enter your PIN number.
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 20 June 2016, at 15:12. From a shell, as the user your instance of Asterisk runs as, can you access the file? Without any console log, these are just guesses: 1) Don't specify the file type in your dialplan. Press two to clear saved PIN number.
That's not a FreePBX context. Press two to redial same number. This page has been accessed 28,861 times.