similar to: Need Help in changing Voice message

Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq --

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable.

Hello Everyone, I am receiving following error message will making Zaptel on Cent OS 5.2. make[1]: Entering directory `/usr/src/zaptel-1.4.12.1' echo "You do not appear to have the sources for the 2.6.18-92.el5 kernel installed." You do not appear to have the sources for the 2.6.18-92.el5 kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Hello Everyone, I have write a script following script for nagios -- typeset -i CRITICAL; #Positional parameter CRITICAL=`echo $2`; ME=`basename $0`; #echo $CRITICAL if [[ "$2" == "" ]] then echo NO INPUT!!! Usage ./$ME -c N else typeset -i ASCALLS; ASCALLS=`asterisk -rx "core show channels" | grep active | grep call | awk '{print $1}'` #echo $ASCALLS;

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

dovecot 2.2.13 With configuration like plugin { expire_dict = blblbla } expiere plugins segfaults during deinit. It might also segfault if the dict service was unavaiable. attached bellow is a diff against 2.2.13 Jul 7 14:51:11 dovecot1 dovecot: lmtp(10548, lazy): Error: expire plugin: dict_init(blblbla) failed: Dictionary URI is missing ':': blblbla Jul 7 14:51:11 dovecot1

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hello, Not sure if this is implied in the documentation or something given: does the RoR framework require "every table must have an id" ? If this is the case, then what happened to the concept of normalization ? Futher, if the tables have already been build for an existing appication ( without ID column in everytable ) , will the script / generate work ? Thanks -- Posted via

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem ------------------------ I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg

Hi all, I am have some difficulty with the legend function. I need to add a legend to describe the different line types in a plot. The legend box is small. It did not include sufficient length of each line type to help distinguish the differnt line types. Is there a way to fix this. Thank you Hannah [[alternative HTML version deleted]]

Is there a place that shows how to create two plots that are stacked on top of each other where they share a common x-axis scale, but have differnt y-axis scale? Say have the following data: airquality Stack plot(airquality$Day, airquality$Wind) on top of plot(airquality$Day, airquality$Temp). I am interested in stacking the two on top of each other with no seam, or plotting the two lines with

Hi, I'm trying to implement a Voice Drop service within Asterisk dial-plan. The service is supposed to work as following: 1. A initiates a call to B 2. The call is answered by B's answering machine 3. A hears the answering machine's greeting and the recording beep 4. A speaks a few words into the recording to personalize the message 5. A presses some DTMF keys (say, '##') to

On 3/9/2016 9:02 PM, Timo Sirainen <tss at iki.fi> wrote: > On 08 Mar 2016, at 01:50, Pavel Stano <stanojr at websupport.sk> wrote: >> >> sis attachment deduplication is broken in 2.2.16 upwards. >> It is caused by this commit. >> https://github.com/dovecot/core/commit/664bf3e236c214aee86294483c379e4fa66c2e63 >> >> in src/lib-fs/fs-sis.c function

I have been frustrated with J2EE technology just like any other developer who has worked in that technology for a long time. This made me learn about Rails and I was excited about RoR. Today I came across JBoss Seam framework and I am not able to contain my excitement. My passion for JEE development is back. It seems like it is better than RoR. I would like to hear differnt opinions about this

Hi: This should be easy. I'm running 1.2.15. When a caller calls someone's voice mail, it goes straight to a beep, even though there is an unavail.wav file in that user's voice mail directory. Here is the relevant part of extensions.conf: [internal] exten => 2211,1,Dial(SIP/211,10) exten => 2211,2,VoiceMail(u211@default) exten => 2211,3,Hangup Here is the relevant part of

I can browse a site I developed/deployed to Apache 2.2.3/Mongrel from PC/s with Firefox/Explorer. I can''t browse from Mac Safari? I checked these setting and it still doesn''t work... http://docs.info.apple.com/article.html?artnum=151263 I don''t know Mac''s, what esle could this be? Jim

I setup some IPSEC between 2 networks. From 1 network I can ping the other networks local connection but not anything beyond that. Network A - 10.0.1.1 (loc) 23.23.23.23 (net) Network B - 10.0.2.1 (loc) 44.44.44.44 (net) I''m on local machine 10.0.1.10 on network A, I can ping 10.0.2.1 but I cannot ping a machine on that network ex. 10.0.2.200. I was thinking it probally has to do