similar to: Some warnings when starting Asterisk

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("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 Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

Greetings I was getting bad static crackle on a phone, so I reload from the latest CVS and did a make clean ; make install on zaptel, libpri and asterisk Now I get this error [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled Urgent handler [chan_oss.so] => (OSS

Hello, I need to send a tiff via fax with my asterisk 1.6.1.0. I tried in the dialplan [default] exten => _X.,1,SendFax(/root/test.tiff) but I have: salledeconf1*CLI> console dial 111 at default [Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory -- Executing [111 at default:1]

Hi, I get the following message when I try to run Asterisk. This happened after I changed some of the configuration files: extensions.conf, zaptel.conf and zapata.conf. I have both TDM40B and X100P cards and am trying to use them together. Please let me know if there is something else I should do. Thnx, Gladys [chan_local.so] => (Local Proxy Channel) == Registered channel type

hi list, when I run asterisk, appears the following: .... WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested 8000 Hz, got 8178 Hz -- sound may be choppy WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I don't work right with non-full duplex sound cards XXX WARNING[1133735216]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

All, I'm trying to hear the asterisk's auto attendant in its default configuration. According to VoIP Hacks in Chapter 4, I found the following excerpt after successfully configuring my SIP IP Phone (Cisco 7960G): In its default configuration, Asterisk has an auto-attendant that can route calls. To try it out, take the IP phone off the hook and dial 2. Then dial the BudgeTone's Send

Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. [chan_oss.so] => (OSS

Hi, Anybody seen this error? Getting an odd error on the console when I place a call from console to a SIP station. As soon as the station answers, here is the error. SIP/172.16.10.24-527b answered OSS/dsp WARNING[1298960704]: File chan_oss.c, Line 679 (oss_indicate): Don't know how to display condition -1 on OSS/dsp << Console call has been answered >>

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

I'm trying to install on a 233Mz PII running RH7.1 kernel 2.4.2 The "astinstall" script pulled files from CVS ok and compiled with no warnings. When I try to run "modprobe wcfxo" and "modprobe wcusb" I get the following identical set of error messages: 15. Creating sample environment...SKIPPED ================= Installation Completed! ====================

Hello, i have a big problem... i want to connect my asterisk server to a lancom 1722 device (ISDN/SIP) Gateway. sip.conf: [general] context=default allowguest=yes realm=10.1.1.209 bindport=5060 bindaddr=0.0.0.0 tos_sip=cs3 ; f?r SIP-Pakete (Kommunikationsaufbau) tos_audio=ef ; f?r RTP-Audio-Pakete tos_video=af41 ; f?r RTP-Video-Pakete allow=all dtmfmode=rfc2833 canreinvite=yes

Hello, This is debug output I get: Jul 28 15:05:49 WARNING[8249]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable [chan_zap.so] => (Zapata Telephony w/PRI) Jul 28 15:05:49 WARNING[8249]: chan_zap.c:924 zt_open: Unable to specify channel 1: No such device or address Jul 28 15:05:49 ERROR[8249]: chan_zap.c:6460 mkintf: Unable to open

Dear Asterisk developers, I run the same asterisk version on the home machine and on the work. On the home machine I have Slackware 10.0 (kernel 2.4.24) while on the work machine I have Mandrake 10.1 (kernel 2.6.8.1 <http://2.6.8.1>). When I run asterisk on the work machine, these warnings and error appear (there are no warnings or error at home): [ Booting......Oct 17 18:19:04

All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing

Does anybody know what would be causing the errors below? I get these errors continuously until asterisk finally quits. This happens when I make 20 simultaneous SIP calls with the Dial Command. chan_oss.c:291 sound_thread: Failed to write sound chan_oss.c:200 send_sound: Unable to read output space __________________________________ Do you Yahoo!? Yahoo! Mail - Find what you need with new

Hello. Although I can hear the demo etc. now, I notice during asterisk launch I get :- [chan_oss.so] => (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource

Does anyone know whether the linux machine running * have to have a sound card on it in order for musiconhold to work for sip phones? I've tried about everything (including tons of google searching) to get it to work, and nothing. When a call is placed on hold between two C7960's, the CLI indicates: -- Executing Dial("SIP/3002-c418", "SIP/3000|20") in new stack

Hi My motto is to connect two computers on the same network with Voip without using any special hardware,i have downloaded Asterisk, I was suggested to use LinPhone as a soft phone as it is very easy to install I have installed Asterisk on my computer and iam using it as a server. And whe i DAIL 1234 at CLI i get the following errors repeatedly Apr 21 17:29:13 WARNING[1167272128]:

Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great!!!! Here is the output asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding

Looking at the error I tried moving chan_modem* out of the modules folder and asterisk started and its working again... Not sure what changed in the chan_modem_i4l.so but removing it from the folder fixed my problem. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Hall Sent: Sunday, January 23, 2005