similar to: cmd page crashes Asterisk SVN-branch-1.4-r57207

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

I have created the following stuff: BRANDS: Brand Name Language Published Number DID Inc Markup (in 1/100 of 1%) Gamma en 5000 6 0 TRUNKS: Trunk Name Technology Peer/Trunk Gamma SIP 200.68.XXX.XXX ROUTES: Pattern Comment Trunks Connect Fee Inc. Seconds Cost per additional minute 63 Parana Gamma 0 60 500 CARDS: I have created some custom numbered cards. So, every call started with

I'm getting Number but when I look at the CDR database. I do see the name -----Original Message----- From: Lacy Moore - Aspendora [mailto:aspendora@gmail.com] Sent: Mon May 01 17:10:26 2006 To: Asterisk Users Mailing 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

Using this link http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt This is a Dell PowerEdge 1950 running Whitbox 4 and Asterisk 1.4.0-beta3 I get the following errors on make install Any help would be GREAT! Thanks [CC] app_cepstral.c -> app_cepstral.o In file included from /usr/src/asterisk/include/asterisk/linkedlists.h:23, from

There seems to be an error in the summary() function when applied to "ts" class objects. The results of a call to summary( ), on the R "ts" data set USAccDeaths , reports the wrong value for Max. The value reported by the summary function is 11320. The max( ) function returns the correct value 11317, the July 1993 value. Coercing the data to a data.frame and calling summary

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

wondering if anybody knows this......does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. Nauman -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, July 08, 2004 5:48 PM To: asterisk-users@lists.digium.com Subject:

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.

Hello, i have a server which supposed to control three Eaton UPS connected by USB cables. Each UPS is Eaton NV 2000H. (eaton_1, eaton_2, eaton_3) The connection is fine with each UPS, if it is configured alone. If i configure two or all three, the connections broke up. After nut started, tried to start upsdrv manually (eaton_1. The log: Apr 4 00:43:46 mail blazer_usb[10799]: Startup

Simplest version that I can think of: x <- scan("~/Downloads/digits.txt") x <- x[-seq(1,220000,11)] length(x) # 200000 hist(x) Now, because it's Friday: How does one work out the theoretical distribution of the following table? > table(table(factor(x,levels=0:99999))) 0 1 2 3 4 5 6 7 8 9 10 11 13497 27113 27010 18116 9122

The problem is actually very simple to describe. I manage a Linux fileserver. Most of the clients use Win98 or NT4 and that seems to work just fine - but w2k is an entirely different story. What happens is that no w2k can see the shares in the linux machine an when trying to access it w2k comes with something like "Service not avaible - connection closed". Now the really fun part starts.

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?

Hi, Every day I try to write some small R programs to improve my R-skills. Yesterday I wrote a small program to read the digits from "A Million Random Digits" from RAND. My code works but it is very slow and I guess the code is not optimal. The digits.txt file downloaded from https://www.rand.org/pubs/monograph_reports/MR1418.html contains 20000 lines which looks like this: 00000

====================================================== "Thousands of candles can be lighted from a single candle, and the life of the candle will not be shortened. Happiness never decreases by being shared." Buddha ======================================================

====================================================== "Thousands of candles can be lighted from a single candle, and the life of the candle will not be shortened. Happiness never decreases by being shared." Buddha ======================================================

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.

I've been running with Dovecot 2.2.15 on my mail server and Thunderbird on workstations with Kerberos/GSSAPI authentication. This has been working for over a year for 10 users. The other day, I replaced a user's workstation and set up this user with a Thunderbird client. Unfortunately, I got the error: "The Kerberos/GSSAPI ticket was not accepted by the IMAP server ... please

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

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: