[Asterisk-Users] Re: [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation

2004-02-22 Thread Fran Boon
On Sat, 2004-02-21 at 09:06, Costa Tsaousis wrote: > >> callgroup= ; UP > >> pickupgroup= ; UP > >> Q4: Since a user cannot accept calls, why to setup call pickup for > >> him/her? > > Sorry, haven't used or checked call groups. Anyone else? > No answer on this yet... I use pickup groups just fin

Re: [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation

2004-02-21 Thread Olle E. Johansson
Costa Tsaousis wrote: Sorry, I was on the wrong topic, canreinvite has yes|no|update as keywords. with UPDATE a SIP method UPDATE is initiatied to change the media path. with YES, a new INVITE is issued within the current call. (a "re-invite") with NO, the call stays within asterisk. Sorry for

Re: [Asterisk-Users] SIP config documentation

2004-02-21 Thread Fran Boon
On Sat, 2004-02-21 at 19:30, Costa Tsaousis wrote: > I believe there are three possible paths for asterisk: > 1. Stick to the switched world (as the common denominator for telephony). > This means that * can have any number of gateways on it, but always, it > will be a "switching-like" PBX with som

[Asterisk-Users] Re: [Asterisk-Users]  SIP config documentation

2004-02-21 Thread Costa Tsaousis
Hi again, > Feel we need to document various solutions here. Yes we do. I still have a filling that I don't exactly get it... > Sorry, I was on the wrong topic, canreinvite has > yes|no|update as keywords. > with UPDATE a SIP method UPDATE is initiatied to change the media path. > with YES, a n

[Asterisk-Users] Re: [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation

2004-02-21 Thread Fran Boon
On Sat, 2004-02-21 at 09:06, Costa Tsaousis wrote: > >> incominglimit= ; U- concurrent call limitations ( >= 0 ) > >> outgoinglimit= ; U- concurrent call limitations ( >= 0 ) > >> Q6: How is it possible for a type=user phone to have BOTH incoming and > >> outgoing limits? > > Interesting question.

Re: [Asterisk-Users] SIP config documentation

2004-02-21 Thread Fran Boon
On Sat, 2004-02-21 at 11:48, Olle E. Johansson wrote: > canreinvite has yes|no|update as keywords. > with UPDATE a SIP method UPDATE is initiated to change the media path. > with YES, a new INVITE is issued within the current call. (a "re-invite") > with NO, the call stays within asterisk. Any ide

Re: [Asterisk-Users] SIP config documentation

2004-02-21 Thread Olle E. Johansson
Costa Tsaousis wrote: context= ; UP, the context name for placing calls Q1: Why is there a context for peers? We use peers in some other situations as well. This is strange and rather undocumented, but an incoming call is first matched by username with the defined users (including 'friends'). Af

[Asterisk-Users] Re: [Asterisk-Users] SIP config documentation

2004-02-21 Thread Costa Tsaousis
Hi all, (oej, I have lost your e-mail somehow, so I have replied to some other reply to you... sorry!) >> context= ; UP, the context name for placing calls >> >> Q1: Why is there a context for peers? > > We use peers in some other situations as well. This is strange and > rather undocumented, bu

Re: [Asterisk-Users] SIP config documentation

2004-02-18 Thread Fran Boon
On Wed, 2004-02-18 at 16:06, Arretni VoIP Tech wrote: > Can musiconhold= be included in sip.conf? I want to play music on > hold for calling users on the VoIP side. Currently, I can only play moh > when the call came from the PSTN (zapata). Use Olle's chan_sip2: http://bugs.digium.com/bug_view_pa

Re: [Asterisk-Users] SIP config documentation

2004-02-18 Thread Arretni VoIP Tech
IL PROTECTED]> Sent: Wednesday, February 18, 2004 2:58 AM Subject: Re: [Asterisk-Users] SIP config documentation > Costa Tsaousis wrote: > > > > I was trying to figure out all the valid options for a sip.conf and I > > believe I found a few weird things (or just a few things that

Re: [Asterisk-Users] SIP config documentation

2004-02-18 Thread Olle E. Johansson
Costa Tsaousis wrote: I was trying to figure out all the valid options for a sip.conf and I believe I found a few weird things (or just a few things that are weird to me :) Anyway, I decided to post this here together with my questions and notes in case other people need this info too or have simil

[Asterisk-Users] SIP config documentation

2004-02-17 Thread Costa Tsaousis
Hi all, I was trying to figure out all the valid options for a sip.conf and I believe I found a few weird things (or just a few things that are weird to me :) Anyway, I decided to post this here together with my questions and notes in case other people need this info too or have similar questions.