search for: americani

..._s-.,1,Congestion [macro-gw-nufone] exten => s,1,SetCallerID(${CALLERIDNAME}) exten => s,2,Dial,IAX2/rpingar@NuFone/${ARG1} exten => s,3,Goto(s-${DIALSTATUS},1) exten => s-BUSY,1,Busy exten => s-CHANUNAVAIL,1,Noop exten => _s-.,1,Congestion [ser] ; combinazione 81 - per provider americani - destinaione usa e canada exten => _81.,1,Macro(gw-nufone,${EXTEN:1}) ; NuFone exten => _81.,2,Macro(gw-voipjet,${EXTEN:1}) ; VoipJet.com exten => _81.,3,Congestion ; combinazione 8011 - per provider americani - destinaione rotteinternazionali exten => _8011.,1,Macro(gw-voipjet,${EXT...

Have Asterisk us at running fine, but have run into a small snag. It's not emailing the voicemails to the users. I have attach=yes set, sendmail is configured and works from from the commandline (sent an email to myself). Unless I'm wrong, or missing something, asterisk is configured by default to send an email when a users receives a voicemail, correct? Thanx A

...blocking the emails because I had WAV files in the block list. One of those things that doesn't occur to you until you've had a little bit of sleep. Thanx for the help! A *-------------------------------- Date: Fri, 25 Mar 2005 06:03:42 -0800 From: "Richard J. Sears" <rsears@americanIS.net> Subject: Re: [Asterisk-Users] Re: Emailed voicemail To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <20050325060017.6D80.RSEARS@americanIS.net> Content-Type: text/plain; charset="US-ASCII" Yes Andy - that...

Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Hey Everyone, I am using NuFone for 866 inbound service and I am trying to figure out the callerid part of it. Any call into my * system just shows "Toll Free Call" and will not give me the calling party's caller ID info. Is this just something I have to live with using NuFOne, or did I miss some type of config in * that will grab the callerID other than the inbound 866 number...?

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Hi Everyone - I just got my hands on a Cisco 7970 and was told that I should do a factory reset before trying to configure it to work with Asterisk. After the factory reset, it will not boot at all, instead sits with the line button lights flashing one at a time in sequence. I have had no luck trying to figure it out - anyone run into the same problem that can lend a hand..? Thanks

Hey Everyone - I was going to create a visio diagram outlining how my extensions will flow out. I was just wondering if anyone on the list may have an example they have already done up so I can get some ideas. Thanks ****************************************** Richard J. Sears Vice President American Internet Services

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1" 200 9438

Hey Everyone, This is not an operational issue, and to my knowledge only effects the look of the command, but when I issue a "sip reload" then a "sip show peers" I see all of the actual usernames I have assigned in my sip.conf. However, five minutes later I reissue the sip show peers and all of the usernames have disappeared and are replaced by the sip ID. The only way to get

Hey Everyone, I am playing around with my * box, and I have a few different phones hanging off it it right now. I have a Cisco 7960 capable of g729, ulaw and alaw, I have a Cisco ATA186 with a Panasonic cordless phone attached to it, I have a Digum IAXy with a dumb analog phone attached to it, and I have a Linksys PAP2-NA with an AT&T 959 analog phone attached to it. I also have several

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit