similar to: Console\DSP

Hello, I'm trying to build a page system using a Dell Desktop PC optiplex 170L, My sound card is working fine under /dev/snd/ exten => s,1,Dial(Console/snd/,20,A(trek)) exten => s,2,Hangup But won't work! I get the following error [Apr 27 11:44:46] WARNING[2950]: chan_oss.c:377 find_desc: could not find <snd> [Apr 27 11:44:46] WARNING[2950]: chan_oss.c:850 oss_request:

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040915/65a2736a/attachment.htm -------------- next part -------------- Hi friends, I tried to dial 111 from CLI without any hard/soft phones. I used the following config when i called 111 from CLI by CLI> dial 111 I got these errors -- Executing Dial("OSS/dsp",

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for

== 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

Hi, I hope someone could help me, or at least give me a hint about this error. ERROR => Aug 18 10:48:51 WARNING[17120]: chan_oss.c:838 oss_indicate: Don't know how to display condition 16 on OSS/dsp -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Aug 18 10:48:53 WARNING[17120]: chan_oss.c:838 oss_indicate: Don't know how to

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 have 4 systems and each one of them has a partition I'd like to be remotely accessible on the other 3 systems. In other words System1 has Partition1. Systems 2,3,4 should be able to remotely mount Partition1 from System1. Also System2 has Partition2. Then systems 1,3,4 should be able to remotely mount Partition2 from System2 and so on. I tried NFS and it works but only in the ideal

Hi, I installed openssh3.1p1 on two IRIX test systems which have both ipv6/v4 stack. When tried to execute ssh, it says permission denied and it fails. I did add --with-pam when i did configure, installed pam libraries also on the test systems. user1 at system1~ 59 % /usr/freeware/bin/ssh user1 at system1 /usr/bin/true user1 at system1's password: Permission denied, please try again. user1

I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In

Yes this scenario works on my 2 systems which are at LAN. I made one system as server (192.168.0.20) and registered from other system... it is fine but now there is a different scene. actually there is a registered user named abc at system1 (192.168.0.20) having context [payasyougo] which is used to do outbound calls. we want to use this user's context and account so that when we register

Greetings! I am working with Lustre-2.1.2 on RHEL 6.2. First I configured it using the standard defaults over TCP/IP. Everything worked very nicely usnig a real, static --mgsnode=a.b.c.x value which was the actual IP of the MGS/MDS system1 node. I am now trying to integrate it with Pacemaker-1.1.7. I believe I have most of the set-up completed with a particular exception. The "lctl

I'm rsyncing files on system1 to its external HD. system2 is remote and pulls those files from the external HD. system2 does not have root privileges on system1 so I chown the files to pull. Can I somehow use --fake-super or something similar to save the original ownership info to ACLs? - Grant

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"?

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

https://bugzilla.mindrot.org/show_bug.cgi?id=1945 Bug #: 1945 Summary: Only 1 of the 2 krb cache files is removed on closing the ssh connection with UsePrivilegeSeparation=yes Classification: Unclassified Product: Portable OpenSSH Version: 5.8p1 Platform: All OS/Version: HP-UX Status: NEW

I'm trying to set up the console as an extension (so I can set up overhead paging), but I can't seem to get it to work. When I call my paging extension, I get an error that it can't open the device: -- Executing Ringing("Zap/9-1", "") in new stack -- Executing Dial("Zap/9-1", "Console/dsp0|18|A(new/whistle)") in new stack << Call

hi all I have a samba pdc with ldap backed, samba version being 3.0.21c, and openldap 2.3.19. All clients get ip through dhcp and dynamically updates dns also The problem i am facing is from a windows 2000 client if i go to run and browse another system it connects to some other system for example from START-> RUN->\\system1 , it will open some other system say system2 when i ping to

Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] ??? exten => 99,1,ANSWER() ??? exten => 99,2,PLAYBACK(tt-monkeys) ??? exten => 99,3,HANGUP() The phone

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"