hi, an anyone tell me how to do auto dial to a SIP in Asterisk using a script .any example will be more helpful ... ! Regards Upendra -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/73cb21fe/attachment.htm>
Depends on your coding preference. you need to put the files in your spool directory (IE /var/spool/asterisk/outgoing) Just search google for asterisk call files. should help you out ;) On 12-02-28 07:59 AM, upendra wrote:> hi, > > an anyone tell me how to do auto dial to a SIP in Asterisk using a > script .any example will be more helpful ... ! > > > Regards > Upendra > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Kind regards, Keith Sloan Vianet Internet Solutions Voice Operations Center OFFICE - (800) 788 0363 ext 7203 Local - (705) 222 9996 ext 7203 Email - keiths at vianet.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/e9f1bb60/attachment.htm>
On Tuesday 28 February 2012, upendra wrote:> hi, > > an anyone tell me how to do auto dial to a SIP in Asterisk using a script > .any example will be more helpful ... ! > > > Regards > UpendraYou need to inject a callfile into /var/spool/asterisk/outgoing . The file should look something like this (representing a call from extension 301 -- here the technology is important -- to extension 101 in context "internal"); #################### 8< #################### Channel: SIP/301 Context: internal Extension: 101 Priority: 1 CallerId: 301 #################### >8 #################### It's easy enough to start with a complete callfile with placeholders, in a scalar variable; then have your script do regular expression substitutions to fill in the values. You must also be careful that Asterisk will not try to parse the callfile while it is in an incomplete state. The canonical method to do this is to create it somewhere else on the same physical filesystem and `mv` it to the desired location. In practice, if the callfile is smaller than one block, then you probably will get away with just creating it in situ. -- AJS Answers come *after* questions.
thnks for the reply.. i want to know is there any way to call a SIP to SIP by command line ???? regards Upendra -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120301/37968c32/attachment.htm>
On Thursday 01 March 2012, upendra wrote:> thnks for the reply.. > > > i want to know is there any way to call a SIP to SIP by command line ????Yes. Just write a script in your favourite language (even bash will do if there is nothing better) to set up a callfile, then invoke it from the command line. As this list forbids discussion of paid-for services, you will need to contact me separately if you want me to write the script for you. -- AJS Answers come *after* questions.