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Hi all, In my Asterisk setup, incoming calls through Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be terminated properly after hangup. However, when calls were forwarded to voicemail, after recording & hangup the PSTN calls and cisco FXO port remained connected unless cisco port was manually shut/no shut. # key used to hang up the call did NOT help. Did anyone

...behave identically, shouldn't it? If by design, is there a workaround? /Stig ------------------------------------------------------------------------------------- N Y H E T E R! - IP-telefoni, spara tusenlappar om ?ret! - Rikst?ckande ADSL 0,25-24Mbit - Internetaccess (Modem/ISDN64+128 via Ymex - utan abonnemangskostnad! - Eposttj?nster, ?ven UUCP, Uppringd SMTP, MX fallback, Dom?nPOP - Surf24 - en billig bredbandstj?nst fr?n Ymex f?r kunder i H?rn?sand/?landsbro. ------------------------------------------------------------------------------------- Get your emailed Web-forms into a databa...

...ine beginning with a semicolon? Or should a comment now be terminated too? /Stig ------------------------------------------------------------------------------------- N Y H E T E R! - IP-telefoni, spara tusenlappar om ?ret! - Rikst?ckande ADSL 0,25-24Mbit - Internetaccess (Modem/ISDN64+128 via Ymex - utan abonnemangskostnad! - Eposttj?nster, ?ven UUCP, Uppringd SMTP, MX fallback, Dom?nPOP - Surf24 - en billig bredbandstj?nst fr?n Ymex f?r kunder i H?rn?sand/?landsbro. ------------------------------------------------------------------------------------- Get your emailed Web-forms into a databa...

...; ; Typical config for 12SP+ > [flex] > device=SEPXXXXXXXXXXXX > version=P002F202 > context=demo > line => 133 > > > > --__--__-- > > Message: 8 > Date: Wed, 31 Mar 2004 09:08:53 +0200 > To: asterisk-users@lists.digium.com > From: Stig Andersson <stig@ymex.se> > Subject: Re: [Asterisk-Users] Caller entered digits ignored during > wait.... > Reply-To: asterisk-users@lists.digium.com > > Asterisk doesn't accept keys during wait, use Background > and play 1 sec silence instead. > > /Stig > > At 23:46 2004-03-30 -06...

...; Typical config for 12SP+ > [flex] > device=SEPXXXXXXXXXXXX > version=P002F202 > context=demo > line => 133 > > > > -- __--__-- > > Message: 8 > Date: Wed, 31 Mar 2004 09:08:53 +0200 > To: asterisk-users@lists.digium.com > From: Stig Andersson <stig@ymex.se> > Subject: Re: [Asterisk-Users] Caller entered digits ignored during > wait.... > Reply-To: asterisk-users@lists.digium.com > > Asterisk doesn't accept keys during wait, use Background > and play 1 sec silence instead. > > /Stig > > At 23:46 2004-03-30 -06...

Hi, Using latest cvs. I (as many otheres it seems) can't get Attended transfer to work with Cisco ATA186 (using SIP) Has anyone else had any luck? Same with 3-part calling, if one drops off, all are disconnected... /Stig