similar to: help with waning on OSS/dsp, condition 16 and 17

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 >>

Hi, I have to develop a phone application using asterisk's chan_oss. When the phone is idle, i.e. the last command was a hangup, one hears a "toot, toot, toot, ..." But unforuntaly its use is in Germany, where one expects a continous "toooooooooooooooooooooooooooooooooo ..." before dialing. Is there anything to define the tone indicating "ready to dial"?

== 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

I am trying to get a console/dsp application going with 1.4.18 and not hearing any audio. In the CLI I see the call coming in, I see the Dial(Console/dsp) I see <auto answered> I see ALSA default but I hear no audio. What can I do to tell what is happening here. I have in modules.conf: noload chan_oss.so load chan_alsa.so For kicks I tried it the other way to noload chan_alsa.so and load

I am using a extension to dial the console which has autoanswer enabled. I am getting a strange warning, has anyone seen this before? Nothing on Google, or Voip-Info [Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request: oss_request ty <console> data 0x0xb7851e00 <dsp> << Call to device 'dsp' dnid '(null)' rdnis '(null)' on console from

Hi list! Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch. Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian 9). Well, I copied the configuration and changed what needed, so basically, it works, at least with my tests. But when Asterisk will be started, in the message log I get this error: [Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel

I've been trying to get a console channel working without success. The sound card, which is built into the motherboard, is a VIA Technologies, Inc. VT82C686 AC97 Audio Controller. Using the oss drivers (vi82cxxx_audio) in kernel 2.4.23 and chan_oss, I just get beeps and screeches. Using alsa drivers (snd-via82cxx) and chan_oss (using the alsa oss emulation), playing sound works, but

Hello, I am getting this kind of Warning in the Asterisk console, but i don't know why. WARNING[8200]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call 46925e274af3b022219246016414b107@192.168.1.101 for seqno 102 (Non-critical Request) Could you give some clue to solve this problem? Thanks in advice. Ismael.

hi, it'd be very useful to include the quota warning patch in official release. without it the quota support is not really useful since mail simple dropped when quota is over. and most enduser never know what happend, they just recognize mails are not coming:-( thanks. -- Levente "Si vis pacem para bellum!"

Hello. I am new to Asterisk but I have been reading like for a month all de documentation and tutorials that i have found. Most of them from de voip-info.org. Recently I set up an Asterisk server based on Debian 'Sarge' 3.1. Everthing was looking fine. I even locate an old modem that would work like a X100P Clone (an Ambient MD3200) and after compiling and installing zaptel it is

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andrew > Kohlsmith > Sent: Thursday, September 29, 2005 2:23 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Best echo canceller? > > > On Thursday 29 September 2005 17:04, Claudio Canseco wrote: >

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

Hi, Can someone help me understand why I can't make calls to cellphones? Last week I edited my extensions.conf and created a context for outbound calls [outgoing] exten => _92XXXXXX,1,NoOp("Call for "${EXTEN:1}) exten => _92XXXXXX,2,Dial(Zap/1/${EXTEN:1}) exten => _04466.,1,NoOp("Call for "(${EXTEN}) ;Llamadas a Celular exten =>

Hi, I recently installed a web interface for webmail, from asterisk CVS-HEAD from date 2005-08-10. Looks like it is script (on pearl) that builds all the web interface. The problem is that I can't play the wav files, and doesnt display the default pictures (animlogo.gif, play.gif ). The error: > Hrm, can't seem to open /var/spool/asterisk/voicemail/x-lite/6678/Work/msg0000.wav >

Hi I had have trouble to get sound working under freebsd 8.1 amd64 arch. So i decide to dig in wine code and I create a simple "proof of concept" patch to get work my sound card.? ********* SND STAT ********FreeBSD Audio Driver (newpcm: 64bit 2009061500/amd64) Installed devices: pcm0: <HDA Realtek ALC272 PCM #0 Analog> (play/rec) default <- This is my primary snd pcm1: <HDA

Hi! When I want to monitor the OSS/dsp channel through the Asterisk management interface, I get a "permission denied" error: Action: Monitor Monitor: OSS/dsp File: 1124096949 Mix: 1 Response: Error Message: Permission denied My permissions for /var/spool/asterisk look like this: pound:~# ls -la /var/spool/asterisk/ total 40 drwxr-xr-x 10 asterisk asterisk 4096 Aug 9 10:19 .

I recently upgraded a server to Asterisk 1.4.22 with OpenR2. Previously I was using 1.4.18. It seems that 1.4.22 has a big bug using chan_alsa.so for overhead paging. After rebooting the server it would work once or twice and then I just got an error on the CLI: [Jan 7 10:35:14] ERROR[26164]: chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable I had to switch to chan_oss

The subject says it all. After digging through latency and other issues with all kinds of linux softphones, I've found that only * works alright for me as a VoIP client. Problem now is that, unlike other apps, chan_oss resp. chan_alsa grab the card once and won't release it until shutdown, while other clients are friendly enough to grab the card only on calls. So, before getting lost in

Hello I have some older programs that require OSS and /dev/dsp I have tried the pulseaudio trick with padsp but some work and some don't. I have also edited the /etc/modprobe/dist-oss-conf and uncommented the line that says install snd-pcm /sbin/modprobe --ignore-install snd-pcm && /sbin/modprobe snd-pcm-oss && /sbin/modprobe snd-seq-device && /sbin/modprobe

Hello everyone, I'm streaming audio on CentOS 5.8 with no problem, even on a cheap sound card using DarkIce as the input tool. For the input under CentOS5, I use: device = /dev/dsp # OSS DSP soundcard device for the audio input But under CentOS 6.2, there is no such device. I see /dev/snd, and it has: controlC0 hwC0D2 midiC0D1 pcmC0D0p pcmC0D2p pcmC1D0p seq controlC1 hwC1D0