[Asterisk-Users] No Sound (2nd post)
Hello anyone who can help I have two Asterisk boxes with identical hardware (Dev Production). I recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head. The hardware is an Intel CA810e, onboard everything with a PIII processor. The config is pure VOIP using IAX2 ilBC with Virbiage Firefly soft clients. I also use Ztdummy which seems to be working ok - no error messages. My problem Is that none of the sounds work, there is no sound for any of the following features 1. Voicemail prompts 2. the menu macro in Dial 3. Music on hold 4. conversation Here's everything I have tried so far. 1. update fedora (I have compiled asterisk off the disk release and also after Redhat updates) 2. update Asterisk ( I have recompiled several times over the past month with different HEAD versions) 3. recompile mpg-123 using both 'r' and 'q' versions I am getting a console message from time to time which say Application asterisk uses obsolete OSS audio interface But Dial, Voicemail and Music on hold report no errors. The production system works fine on the older CVS head from Jan 26 2005. With an out of date Fedora install off the CDs. Thanks Craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Sound at all
Hello anyone who can help I have two Asterisk boxes with identical hardware (Dev Production). I recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head. The hardware is an Intel CA810e, onboard everything with a PIII processor. The config is pure VOIP using IAX2 ilBC with Virbiage Firefly soft clients. I also use Ztdummy which seems to be working ok - no error messages. My problem Is that none of the sounds work, there is no sound for any of the following features 1. Voicemail prompts 2. the menu macro in Dial 3. Music on hold 4. conversation Here's everything I have tried so far. 1. update fedora (I have compiled asterisk off the disk release and also after Redhat updates) 2. update Asterisk ( I have recompiled several times over the past month with different HEAD versions) 3. recompile mpg-123 using both 'r' and 'q' versions I am getting a console message from time to time which say Application asterisk uses obsolete OSS audio interface But Dial, Voicemail and Music on hold report no errors. The production system works fine on the older CVS head from Jan 26 2005. With an out of date Fedora install off the CDs. Thanks Craig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ztdummy is Loaded but Asterisk is not using it
Hello, I have a problem with * on Fedora Core 3 Kernel 2.6. I set up the ztdummy module by following the instructions here http://www.voip-info.org/wiki-Asterisk+timer+ztdummy. The compile worked ok and I edited the files mention in the wiki. Here is a screen grab of what I see when I run lsmod [EMAIL PROTECTED] /]# lsmod Module Size Used by ztdummy 3924 0 zaptel207364 1 ztdummy crc_ccitt 2113 1 zaptel And this is what I see when I do a reload at the astersisk console Apr 1 16:17:18 WARNING[2400]: chan_iax2.c:7311 build_user: Unable to support trunking on user '2277' without zaptel timing I seems like Asterisk is not aware of the presence of ztdummy. Anyone got any suggestions ? Rockwater ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Screen Macro Not Exiting when call rejected
Thanks everyone for your help. The code in the dialplan was ok. I had to switch to CVS head and everything worked straight away. Any clues on when this will be working in the stable release ? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Screen Macro Not Exiting when call rejected
This is a followup to the posting earlier about Hunt Groups with Call
Screening.
I have implemented the following macro and for some reason the Macro does
not exit and continue the context it was called from when the called party
rejects the call - It always drops through to the NoOp at the end and
connects the call.
Below are two examples of the dial commands I am using to call the macro.
The macro itself is below and following is the output from the console. In
the example on the console the 'User Entered '2' ' and conrol is passed to
priority 5 which is correct. At this point the macro should exit and go
back to the next priority in the context that called it.
The system is running Asterisk v 1.0.3
[example_context]
exten =
2,2,DIAL(IAX2/${USER}:[EMAIL PROTECTED]/${MOBILE_CRAIG},15,mM(screen))
..
exten = 3,1,DIAL(IAX2/${TEST},15,mgM(screen))
[macro-screen]
; Prompt operator to accept,reject or transfer the incoming call
; before the call is connected to them.
exten = s,1,Wait(0.2)
exten = s,2,Playback(og-welcome)
exten = s,3,Read(ACCEPT||1)
exten = s,4,GotoIf($[${ACCEPT} = 1]?6:5) ;1 = connect else return
exten = s,5,SetVar(MACRO_RESULT=CONTINUE)
exten = s,6,NOOP()
Output from Console --
-- Executing Wait(IAX2/1001/14, 0.2) in new stack
-- Executing Playback(IAX2/1001/14, og-accept_reject) in new stack
-- Playing 'og-accept_reject' (language 'en')
-- Executing Read(IAX2/1001/14, ACCEPT||1) in new stack
-- Accepting a maximum of 1 digits.
-- User entered '2'
-- Executing GotoIf(IAX2/1001/14, 0?6:5) in new stack
-- Goto (macro-screen,s,5)
-- Executing SetVar(IAX2/1001/14, MACRO_RESULT=CONTINUE) in new
stack
-- Executing NoOp(IAX2/1001/14, ) in new stack
-- Stopped music on hold on IAX2/[EMAIL PROTECTED]/13
-- Attempting native bridge of IAX2/[EMAIL PROTECTED]/13 and IAX2/1001/14
-- Channel 'IAX2/1001/14' ready to transfer
-- Channel 'IAX2/[EMAIL PROTECTED]/13' ready to transfer
-- Releasing IAX2/[EMAIL PROTECTED]/13 and IAX2/1001/14
-- Hungup 'IAX2/1001/14'
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[Asterisk-Users] Call Screen Macro Not Exiting when call rejected
Thanks for the prompt responses. I am aware of the patch but i can't figure out how to install it ? I have emailed mantis who told me the patch is included in the latest versions * which I beleive is 1.0.3.. or do I have to run the latest unstable version ? I would prefer to just apply the patch by itself to my system. Thanks for your help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hunt group with Accept/Reject Option
Hi, Could someone please give me some advice on how to get * to perform the following :- Answer incoming call (from IAX Trunk) Play a prompt to the caller like Thanks for calling please hold while you call is transferred to first available operator The caller hears MusicOnHold Asterisk begins 'hunting' for someone to take the call by dialing through a list of 'available' extensions When an extension is found * dials it and plays a prompt like You have a call, Press 1 to Accept 2 to Reject - * will play the prompt several times and wait for the operator to respond until timeout. If the operator accepts the call by pressing '1' the call is transferred to them. if the operator rejects the call by pressing '2' * disconnects from this operator and 'hunts' for the next available extension if the operator does not answer or does not respond before the timeout * disconnects from this operator hunts for the next available extension If no operators are available the caller is transferred to voicemail Thanks Brendan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
