search for: amaxx

Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid = 8002) Verbosity is at least 3 -- Attempting call on ZAP/G1/6144994925 for 36652@amaxx:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEEC...

...exten => _**2,3,Hangup CLI output ******************** Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid = 11317) -- Remote UNIX connection Verbosity is at least 3 Extension Changed 36652 new state InUse for Notify User 36653 -- Executing [**2@amaxx:1] SIPAddHeader("SIP/36652-b7d0c1f0", "Call-Info: answer-after=0") in new stack -- Executing [**2@amaxx:2] Page("SIP/36652-b7d0c1f0", "SIP/36651") in new stack -- Called 36651 -- <SIP/36652-b7d0c1f0> Playing 'beep' (language '...

I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give.

I search Google to find how to get the message light to flash on my Cisco 7960 running (Application Load ID POS3-06-3-00) (Boot Load ID PC03M030) (DSP Load ID PS03AT38) All I see is about the sip.conf file witch mine has the mailbox=XXXX but still no light. Also the messages button does not work. Any ideas?

...`usecount': app_cepstral.c:270: warning: implicit declaration of function `STANDARD_USECOUNT' app_cepstral.c: At top level: app_cepstral.c:305: warning: function declaration isn't a prototype make[1]: *** [app_cepstral.o] Error 1 make: *** [apps] Error 2 Eric Hall Vice-president Amaxx, Inc. "Customized IT Solutions" 5925B Wilcox Place Dublin OH 43016 614.923.6652 - Direct 614.486.3481 - Office 614.923.6652 - eFax Try our off site backup service free for 30 days. <blocked::http://www.nationalbackup.com/> http://www.nationalbackup.com <blocked::http://www.natio...

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

...CARDS: I have created some custom numbered cards. So, every call started with the pattern "63" goes to trunk or route Gamma with a price of 0.60$ and a connect fee of 5$ I hope this helps you. Regards... Nahuel Ramos. On Mon, 29 Nov 2004 23:30:35 -0500, Eric Hall <ehall@amaxx.com> wrote: > > Hello Sir > > Could you send me a copy of your route Pattern? I'm having trouble > > > > Thank You

Group I'm using realtime for voicemail the it works great.. The only problem I have is I'm not able to use directory or vmail.cgi Does anyone have a solution for this problem? Asterisk CVS-HEAD-01/24/05-07:36:37 RedHat 9.0 Any help would be great!!!! Thanks

...g List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] CallerID Name problem Do you get caller ID number? If so, WAITing is not going to help, since you already get the info. If you get caller ID number, then your telco is not sending the name. On 5/1/06, Hall, Eric M. <ehall@amaxx.com> wrote: > > Do you wait before or after the answer? Do you even need the answer? > > > > > -----Original Message----- > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com<Alex.Lopez@OpSys.com> > ] > Sent: Mon May 01 14:26:49 2006 > To: Asteris...

I'm looking to set up asterisk to call customer 3 days before the app and remind them we will be out to see them. I'm looking for any ideas on good ways to do this. Also I think it would be best to do some type of text to speech however I do not like the sound of the free one . Any ideas? Thanks!!! Eric Hall -------------- next part -------------- An HTML attachment was

D Not sure why this works exten => _3665[0-9],1,goto(test|${EXTEN}|1) but this does not. exten => _366[50-59],1,goto(test|${EXTEN}|1) I would like to route 36650 - 36700 to a Context 'test' however I'm only able to get 10 to work at a time. Any ideas? Any help would be great! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello group I just installed ASTCC and it was VERY easy to get running. I have a question about Pattern Via the web page I click the Routes link and everything makes sense to me but the pattern part. I tried _NXXNXXXXXX with the idea that everything would match this. Well it doesn't work... Does anyone have a good how-to? Thanks for all your help!!

Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI> -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1

Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G

...+---------------------------+ http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend| mailto:tzafrir@technion.ac.il +---------------------------+ ------------------------------ Message: 11 Date: Tue, 11 Jan 2005 08:35:37 -0500 From: "Eric Hall" <ehall@amaxx.net> Subject: RE: [Asterisk-Users] asterisk one number service To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <3987E97097F13A4780088C49F42F8A00037FB2@corpsrv.amaxx.net> Content-Type: text/plain; charset="us-...

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

...eople would like to see in the LCD/button interface and what kind of statistics and options would be beneficial. Kind regards, Matt Riddell --__--__-- Message: 4 Subject: RE: [Asterisk-Users] Cisco 7960 NAT question Date: Thu, 8 Jul 2004 07:54:49 -0400 From: "Hall, Eric M." <ehall@amaxx.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com I had the same problem. What I found is I needed to set register with proxy to yes in the sip config.=20 Hope this helps -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:...

I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic

Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter & Reed Wade do you still

Group Everything is working great with my * server. That's to everyone for all your help!!! I have a problem that I can't seem to find a fix for. When I'm on a call and someone calls in the system never picks up. Also I'm unable to place calls out if someone is on the phone. Here is what I have in my system. Please let me know if you need any other information! Let me start