On Thu, Sep 08, 2005 at 02:56:28PM +0200, Marek Zachara wrote: > i have a box running debian sarge with asterisk installed from distribution > packages: > > CLI> show version > Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64 > running Linux > > I have managed to configure a simple dialplan (the PBX task is quite simple > as > this is a small office with just a few phones) > I have one Zap (PSTN) line connected to it and one SIP to a local provider. > > After some googling most things seem to work well, but i'm having problems > with Hangup. This affects both the Zap interface and the SIP connection to > the provider. > > No matter who tries to hang up an established call, its not properly > finished. > In such case, with the other node just silence is heard (and not > congestion/busy signal). This includes calls initiated both from outside and > from inside with a voip phone connected to the asterisk. > > on asterisk console i'm getting these messages: > > == Spawn extension (incoming, s, 2) exited non-zero on 'SIP/1012082-8408' > == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/1012082-8408' > > here is the relevant part of the dialplan: > > [ Context 'incoming' created by 'pbx_config' ] > 'h' => 1. Hangup() [pbx_config] > 's' => 1. Ringing() [pbx_config] > 2. Dial(SIP/11|5) [pbx_config] > 3. Dial(SIP/11&SIP/21|20) [pbx_config]
You dial. After a hangup you attempt to dial again. Right? What do you try to do there? Ring the one that is free? > 5. Hangup() [pbx_config] > 't' => 1. Hangup() [pbx_config] -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
