similar to: MP3player problem

hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm

Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback("file_which_dont_exist") just before the Moh or MP3Player I can hear the music. Actually I observed the

Hi all, Here another guy working with ztdummy and having problems with music. MP3Player does not work for me both from a telephone entering through a passive ISDN-adapter as well as a SIP-client in the LAN. Ztdummy works with conferences. Here are my messages. I have seen several discussion threads in the mail-archive regarding this subject, but could not find an answer. Is this something I

I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I was trying to test cmd Playback, MusicOnHold, MP3Player but when I call to extension 100 I don't hear the sound ( mp3 or gsm that I put) , I only hear noise If I leave a message in a mailbox the same, all the record is noise --------- extensionns.conf

Hi, I'm running the latest Asterisk (built last Saturday) and can't get mp3's to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I don't hear any sound, and the following displays on the console:

OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM

[I'm reposting this to the asterisk-users list, since it seems to be a bit more active.] Hello, I started messing with Asterisk few days ago, so my overall knoledge about it is still fairy superficial. I think I found an issue with MP3Player; it can be reproducted with this extension: exten => 6001,1,Answer exten => 6001,2,Background(blahblah) exten => 6001,3,Ringing exten =>

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

Okay, I've looked around for a FAQ on this, and it's kinda driving me up the wall. I recently installed a new Asterisk system, and built mpg123 v. 0.59s from sources, and the audio comes out waaaay overmodulated. Any thoughts?

Hi, I would like to prepare postlogin a script that allow imap connection to roundcube for all but restrict imap access for selected users. My question is that: Is possible in condition IF use IP addresses as range or with mask (because I've more than one web servers) ? My script: #!/bin/sh if [ "$IP" = "172.11.0.28" ] ; then printf "* [ALERT] Access allowed from

I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten => s,1,Hangup (I'm just using the line for outbound dialing) -----Original Message----- From: asterisk-users-admin@lists.digium.com

What is this? chan_iax2.c line 4695 (build_peer): Unable to support trunking on peer 'lamas-tigris' without zaptel timing codec_g729b.c Line 413 (load_module): Unable to initialize va stuff: -1 This is why I can't start asterisk in the background -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy

Thx, prips works as I expected, gr8 tool, not available in Gentoo repository but after compilation Dovecot doing what I wanted. Regards, Jack 2017-11-09 21:19 GMT+01:00 Gedalya <gedalya at gedalya.net>: > A bit clunky but perhaps you could find another command. > > https://packages.debian.org/stretch/netmask > > $ IP=172.11.0.28 > $ if [ "$(netmask -n $IP/24)"

Hi all, I could use MP3Player to play local sound (e.g: /usr/sound/abc.mp3) but I could not use it to run a remote stream, if I use mpg123 in command line, I can hear the audio ( /usr/bin/mpg123 http://...), but the same remote mp3 file could not be replay with asterisk. I would appreciate with any suggestion. Phuong Here are the log message: -- Starting simple switch on 'Zap/3-1'

Hi All, I'm having problem with MP3Player app. I want the caller to hear mp3 when he is waiting until I answer my phone. -- from extentions.conf -- exten => 200,1,Answer() exten => 200,2,MP3Player(/home/user200/mp3/hanna-hais.mp3) exten => 200,3,Dial(SIP/200|20|tTrR) exten => 200,4,Hangup() -- end -- here is debug from CLI: -- Executing

My experience told me that mp3player (command) was rather unreliable with shoutcast. I heard nothing. If I use madplay in musiconhold.conf, everything is ok. But, using mp3player in extensions.conf is better than using MoH. Pls kindly advise if we could use madplay in extensions.conf. Could we put it into system command? Best Rgds, ????????????????????????????????????????

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",