Re: [asterisk-users] Play blank sound while VM recording?
On Tuesday 03 April 2007 20:09, [EMAIL PROTECTED] wrote: Charles Ulrich wrote: I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This is fine and probably should be the expected behavior, except that after 30 seconds to a minute of not seeing any RTP traffic coming from the PBX, the trunk appears to make the faulty assumption that the PBX is gone and hangs up the call. Maybe this is what you need?: ;rtpkeepalive=secs; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) (in sip.conf, [general] section) Regards, Philipp That was exactly what I needed, thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play blank sound while VM recording?
Greetings, (Apologies if this is an FAQ, but I've Googled for hours and haven't come up with anything yet.) I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This is fine and probably should be the expected behavior, except that after 30 seconds to a minute of not seeing any RTP traffic coming from the PBX, the trunk appears to make the faulty assumption that the PBX is gone and hangs up the call. I've called the trunk provider and they said two things. 1) This is indeed what their trunk was programmed to do. 2) No, they won't change it. We're working on switching the customer to a trunk provider with a bit more clue, but in the meantime, how can I have Asterisk play an empty sound file while the caller is leaving a voicemail message just to keep the RTP traffic flowing? This installation of Asterisk was designed by someone else and I have limited personal experience with Asterisk configuration files, so an example would be appreciated if possible. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play blank sound while VM recording?
Charles Ulrich wrote: I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This is fine and probably should be the expected behavior, except that after 30 seconds to a minute of not seeing any RTP traffic coming from the PBX, the trunk appears to make the faulty assumption that the PBX is gone and hangs up the call. Maybe this is what you need?: ;rtpkeepalive=secs; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) (in sip.conf, [general] section) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
