Can anyone give me some pointers on the following, in our setup (ast 1.6.3) we use international carriers to terminate calls for a callingcard system, we have an issue where there can be a very long delay after dialing but before the far end begins to ring.
I would like to play a tone every second during this period (before ringing) but then cut off the tone once the far end either sends ringing or progress in band (183). Iv tried using Progress() and Playtones() before the dial but Playtones cuts off as soon as i hit Dial. Iv tried using the Sip/xxx&Sip/xxx trick to get Playtones to continue (which kinda works) unfortunately this also over writes the far end ringing, so the user only hears the tone every second until the far end answers. I have tried a lot of things but can't seem to get the exact behaviour we want. Ask about our great deals on mobile phone contracts from the leading networks. We also provide high speed unlimited usage Business Broadband from £15.99 per month. Contact your account manager for further details. VoIPtalk network on Twitter. Get the latest news, service information, rates and much more - http://www.twitter.com/voiptalktweets. Get latest status messages on http://www.twitter.com/voiptalkstatus This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. -- _____________________________________________________________________ -- 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
