similar to: Connecting 2 asterisk servers

Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R

Hi guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

hi all, i just finished developing some incoming call features in a macro. that macro gets executed everytime an incoming call is received and a new mysql connection is made using the MYSQL cmd in dialplan. i want to use a single mysql connection for every incoming call. my idea of doing it is like this, i want to get a mysql connection in a global variable, just to share the connection with

Is it possible to select a portion of string? For example, in many websites, they give the first paragraph in the main page and link for the article on the other page. I remember this is available in PHP. Any help is appreciated. Thanks Rizwan -- Posted via http://www.ruby-forum.com/.

Hi all, I need a fully developed web based remote provisioning system. I cant find anything reliable on the internet. Have already checked ataconfig.com and voxilla-ays.com. have tried to contact them but got no response. So if anybody knows a good provisioning system then plz tell me about it. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough.

Hi all, We are planning to shift our sip users from one platform to another. (basically from one asterisk server to another). the problem we are facing is that both asterisk servers are using different ports to listen for sip. and both have live customers on them. provisioning their ata's is not a good option for us coz of our settup. we cant just ask the customers to change their ports for

Hi all, Is there any way of originating calls in future without using call files? We have 2 servers (1 active at a time). If we use call files with modification date in future, on the 1st server and it dies and, we activate the second server but we lose the call files. I could have a cronjob on both servers and create callfiles reading execution time from database, but this involves some other

Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this

Hi all, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling

Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have several controllers for my webapp. I''d like to make a view that could take processed data through several controllers. Is this possible? For example, I have two controllers named ''upload'' with an action list_uploads and a controller named notes with list_notes action. I want to bring both of these action under a single view which could mean a table with

Hi, I am new to Opus, I want to decode a RTP packet which has opus payload. Following can describe my intention: I have RTP-packet----------> rtp_header+ opus_payload(20ms/960/48000/1channel) I have now opus_palyload after removing the rtp_header, what I want, a simple code using the opus_decoder which can give me the 20 msec opus paylaod consists of 960 samples. Can anybody put a simple

Hi all, i need an XML file format which is used in remote provisioning of different spa devices. Please somebody tell me the format or tell me where can i find it on the internet. I also need a list of parameters which are configured using auto-provisioning. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was

Hi, I have a dataframe "gexp_1" with 115 rows of samples and 27000 columns of gene expression measurements with each column corresponding to one gene. I now have a smaller vector of genes "tp" for which I need to pull out the data form the data frame. # first gene from the list with 16 genes x <- tp[1,] x [1] geneA 16 Levels: geneA, geneB... #unsuccessful gexp_1$x NULL

Hi all, I dont have any problem, my asterisk is working fine. but on the cli, asterisk keeps saying "Got SIP response 603 "Declined (no dialog)" back from 192.168.0.100". trixbox running on another machine is registered to our server from address 192.168.0.100. whats the reason of this msg? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An

Hi all, I am planning to do clustering for my company's asterisk servers. I dont know much about it, just read some articles on the internet and learned how to use DUNDi and some basic information about clustering. What I need to know is: 1. can i register end user with multiple asterisk servers at a time? 2. If not, Can I re-route registeration requests to different servers using 1 asterisk

Hi All, Please refresh my memory. I am trying to install asterisk after 2 years. I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 1.8.0-rc2 on centos 5.5 but getting the following errors. app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory app_mysql.c: In function ?mysql_ds_destroy?: app_mysql.c:135: warning: implicit declaration of function ?mysql_close?