Doug Millsaps
2005-May-05 13:19 UTC
[Asterisk-Users] 7777 (simulate incoming call) not working
I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP. I can make out going calls. The problem I'm having now is the digital receptionist greeting (aa_1). If I set it to automatically forward to an extension it works. But, if I have it play a message (press 200 for Joe, etc), you can't here the message at all. I can dial the extension number and * will accept and forward me to that extension. I can see on the CLI that it is suppose to be playing the message. If I dial 7777 (simulate incoming call), I get the same thing, can't hear voice but can dial extensions. I've adjusted the txgain and rxgain in zapata. This only increased echo. I have googled this list and SF, I can't find anything else to try yet, or I'm using the wrong search terms. Probably unrelated, but when I "stop gracefully" and then restart *, I get the following error: [app_zapbarge.so] => (Barge in on Zap channel application) == Registered application 'ZapBarge' [app_zapscan.so] => (Scan Zap channels application) == Registered application 'ZapScan' [app_rxfax.so][root@asterisk1 root]# Ouch ... error while writing audio data: : Broken pipe The only thing I can do at this point is reboot the machine. I don't see any failures on the boot up. My search for this error appears to be related to mpg123. But, I never found where somebody had a solution for it. I have tried to install fax capability (install-pdf), but that doesn't work either. I get this error: [root@asterisk1 root]# install-pdf ------------------------------------------- installing Fax PostScript support ------------------------------------------- Gathering header information file(s) from server(s) Server: CentOS-3 - Addons retrygrab() failed for: http://mirror.centos.org/centos/3/addons/i386/headers/header.info Executing failover method failover: out of servers to try Error getting file http://mirror.centos.org/centos/3/addons/i386/headers/header.info [Errno 4] IOError: <urlopen error > [root@asterisk1 root]# There is a pretty long delay after the "Server: CentOS-3 - Addons" line Thanks, Doug
asterisk@txpe.net
2005-May-06 07:43 UTC
[Asterisk-Users] 7777 (simulate incoming call) not working
I don't know if this is related, but the last two mornings I've come in, the newer AAH 1.0 computer has been locked-up. The Caps Lock and Scroll Lock lights on the keyboard are flashing (apparently in a specific pattern). The computer is a HP 7960 w/ ASUS mobo, P4, 1.3Ghz, 256MB RDram. Not being a Linux person, I don't know if this is a Linux issue and/or a hardware issue. Is there a specific log I can look at that might tell me what happened? Thanks, Doug At 03:19 PM 5/5/2005, you wrote:>I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the >new box, I've installed a generic ebay X100P. I don't have my livevoip or >voicepulse accounts set up yet on the new box (can both boxes be >registered at the same time?). I've set up one IP phone (SPA841) with the >new box. I have my SBC POTS line plugged into the fxo card. I set up >everything in AMP. I can make out going calls. The problem I'm having >now is the digital receptionist greeting (aa_1). If I set it to >automatically forward to an extension it works. But, if I have it play a >message (press 200 for Joe, etc), you can't here the message at all. I >can dial the extension number and * will accept and forward me to that >extension. I can see on the CLI that it is suppose to be playing the >message. If I dial 7777 (simulate incoming call), I get the same thing, >can't hear voice but can dial extensions. > >I've adjusted the txgain and rxgain in zapata. This only increased >echo. I have googled this list and SF, I can't find anything else to try >yet, or I'm using the wrong search terms. > >Probably unrelated, but when I "stop gracefully" and then restart *, I get >the following error: >[app_zapbarge.so] => (Barge in on Zap channel application) > == Registered application 'ZapBarge' > [app_zapscan.so] => (Scan Zap channels application) > == Registered application 'ZapScan' > [app_rxfax.so][root@asterisk1 root]# Ouch ... error while writing audio > data: : Broken pipe > >The only thing I can do at this point is reboot the machine. I don't see >any failures on the boot up. My search for this error appears to be >related to mpg123. But, I never found where somebody had a solution for it. > >I have tried to install fax capability (install-pdf), but that doesn't >work either. I get this error: > >There is a pretty long delay after the "Server: CentOS-3 - Addons" line > >Thanks, >Doug
Gregory Wiktor - ADCom Corp.
2005-May-06 09:42 UTC
[Asterisk-Users] 7777 (simulate incoming call) not working
Either its suspending or APM, or watchdog... Check your hardware... Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk@txpe.net Sent: Friday, May 06, 2005 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7777 (simulate incoming call) not working I don't know if this is related, but the last two mornings I've come in, the newer AAH 1.0 computer has been locked-up. The Caps Lock and Scroll Lock lights on the keyboard are flashing (apparently in a specific pattern). The computer is a HP 7960 w/ ASUS mobo, P4, 1.3Ghz, 256MB RDram. Not being a Linux person, I don't know if this is a Linux issue and/or a hardware issue. Is there a specific log I can look at that might tell me what happened? Thanks, Doug At 03:19 PM 5/5/2005, you wrote:>I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on >the new box, I've installed a generic ebay X100P. I don't have my >livevoip or voicepulse accounts set up yet on the new box (can both >boxes be registered at the same time?). I've set up one IP phone >(SPA841) with the new box. I have my SBC POTS line plugged into the >fxo card. I set up everything in AMP. I can make out going calls. >The problem I'm having now is the digital receptionist greeting (aa_1).>If I set it to automatically forward to an extension it works. But, if>I have it play a message (press 200 for Joe, etc), you can't here the >message at all. I can dial the extension number and * will accept and >forward me to that extension. I can see on the CLI that it is suppose >to be playing the message. If I dial 7777 (simulate incoming call), I >get the same thing, can't hear voice but can dial extensions. > >I've adjusted the txgain and rxgain in zapata. This only increased >echo. I have googled this list and SF, I can't find anything else to >try yet, or I'm using the wrong search terms. > >Probably unrelated, but when I "stop gracefully" and then restart *, I >get the following error: >[app_zapbarge.so] => (Barge in on Zap channel application) > == Registered application 'ZapBarge' > [app_zapscan.so] => (Scan Zap channels application) > == Registered application 'ZapScan' > [app_rxfax.so][root@asterisk1 root]# Ouch ... error while writing >audio > data: : Broken pipe > >The only thing I can do at this point is reboot the machine. I don't >see any failures on the boot up. My search for this error appears to >be related to mpg123. But, I never found where somebody had a solutionfor it.> >I have tried to install fax capability (install-pdf), but that doesn't >work either. I get this error: > >There is a pretty long delay after the "Server: CentOS-3 - Addons" line > >Thanks, >Doug_______________________________________________ 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