[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 (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' [EMAIL PROTECTED] 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: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] root]# 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
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
I've only programmed my IP500 through the web interface. On the web interface, I set the GMT Offset to -6 (for Central). It works for me. BTW, I'm using pool.ntg.org as my sntp server. Now what doesn't work, is that when I change something in the web interface and the phone reboots, the time goes back to last time I booted. The only way to reset it is to power cycle the phone. The SNTP Resync Period doesn't seem to work. Doug At 01:11 PM 4/28/2005, you wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= Are these the correct ones? Thanks, Wiley ___ 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
Re: [Asterisk-Users] Sipura SPA-841 Phone Review
I use a headset w/out any problems, except for if my cell phone is close by and rings. Otherwise, volume is ok and no humming. Could it be your headset? At 01:56 PM 4/10/2005, you wrote: Just make sure you don't have a cordless or cell phone near by or the headset jack will receive a considerable amount of interference into your conversation (when NOT using a headset). Also don't even try using a headset... volume is low and there is a loud humming noise. ___ 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
Re: [Asterisk-Users] AAH 0.06 - IAX Connection Over NAT Firewall
Which provider, I have my AAH 0.6 box set up with VoicePulse using IAX2. At 12:59 AM 3/11/2005, you wrote: Hello all, I am having trouble getting my IAX based Voip provider setup. Any pointers are welcome. So here is the deal. I am registered up and I can make outgoing calls but incoming calls fail. Configs all look good I thought. My PBX is behind our firewall with a direct NAT of one to one for an external IP. IAX port is forwarded UDP and TCP to the internal IP. * shows good registration and Ips and ports show solid. Within my AAH I have the registration like the provier said to do. I get absolutely nothing on the incoming. IAX2 debug shows nothing on incoming. Just a fast busy. Outgoing works perfectly however. I have a defined DID in the AMP interface and verified it is written to confs and have reloaded. Can anyone tell me another way to verify that something is coming in? Or did I just miss something on the whole IAX over NAT? Thanks all, Wiley ___ 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
Re: [Asterisk-Users] DID in the U.S.
At 04:15 PM 3/8/2005, you wrote: Hello! Have a look at the following page: http://www.tex-an-2000.com/plxr.html Block of 10.000 DID numbers: No charge Is there something comparable in the LA area? Andreas I believe it's only free if you pay for the other services listed on that page. ___ 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
