Re: [asterisk-users] CWI, call-limit and incominglimit
On 2/24/07, Pavel Jezek [EMAIL PROTECTED] wrote: Brian Capouch wrote: But the included comments say, The user part of a type=friend call will still be affected by the call limit Those seem to be in conflict, but perhaps it's just my parser :-) Could someone clueful explain? I interpret this that asterisk _internally_ still counting calls for both user and peer, but actually limits calls only for peers... :-\ Thanks for all of the pointers on this - I think merging the limitonpeers change from trunk into 1.2.15 is my favourite option right now. Or should I just take chan_sip.c from trunk? Would that be fairly safe? I think that the problem I was having is actually related to sip reload not clearing down an old call-limit setting to its default if the option is removed from a sip user/peer (as opposed to setting call-limit: 0 to disable it.) I have not confirmed this to the point where I can open a bug, but someone might want to check that out? :) Also, from reading the code... Is it worth updating sip show inuse to reflect the setting of limitonpeers ? Regards, Steve ___ --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] CWI, call-limit and incominglimit
On 2/27/07, Steve Davies [EMAIL PROTECTED] wrote: Thanks for all of the pointers on this - I think merging the limitonpeers change from trunk into 1.2.15 is my favourite option right now. Or should I just take chan_sip.c from trunk? Would that be fairly safe? Err... What I meant was shall I take chan_sip.c from the head of the 1.2 branch, but now I see that the limitonpeers option is a 1.4 feature. Steve ___ --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] CWI, call-limit and incominglimit
23 feb 2007 kl. 12.42 skrev Steve Davies: Hi, In older versions of asterisk I used to be able to use incominglimit=1 to effectively disable call waiting on a specific SIP channel (Where broken phones do not allow this on the handset itself) In 1.2.x this became call-limit=1, but this prevents the phone from opening a 2nd line in order to transfer a call using attended transfer. The WiKi suggests using SetGroup() etc, but this does not cater for the case where you are Dialling several different phones simultaneously. You can still set one call-limit for the user and another for the peer. The peer call-limit would be used to prevent call waiting and the user limit could be set to a reasonable level so the phone can do transfers. I _could_ dial a whole bunch of Local channels, each of which checked for an extension usage count, but the additional load and complexity in the dialplan seems a bit over-the-top to me, especially when there used to be a one-line solution to this. I also considered separate user and peer sections in sip.conf, but the hosts are dynamic, and there is no way to link the IP address of the peer to the user. Why is that an issue? The user authenticates on the incoming call, no IP address is needed since the auth is done on the From: header. /O ___ --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] CWI, call-limit and incominglimit
Olle E Johansson wrote: 23 feb 2007 kl. 12.42 skrev Steve Davies: Hi, In older versions of asterisk I used to be able to use incominglimit=1 to effectively disable call waiting on a specific SIP channel (Where broken phones do not allow this on the handset itself) In 1.2.x this became call-limit=1, but this prevents the phone from opening a 2nd line in order to transfer a call using attended transfer. The WiKi suggests using SetGroup() etc, but this does not cater for the case where you are Dialling several different phones simultaneously. You can still set one call-limit for the user and another for the peer. The peer call-limit would be used to prevent call waiting and the user limit could be set to a reasonable level so the phone can do transfers. it can be also usefull to use 'limitonpeer' option in sip.conf with this, it would not be needed to define separate type=user and type=peer for each phone, instead define one type=friend and apply limitonpeers=yes ;limitonpeers=no; Apply all call limits (limit=) only to peers, never ; to users. This improves handling of call limits ; and device states in certain situations. The user part ; of a type=friend will still be affected by the call ; limit, but Asterisk will only use one object for ; counting the simultaneous calls. ___ --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] CWI, call-limit and incominglimit
24 feb 2007 kl. 11.07 skrev Pavel Jezek: Olle E Johansson wrote: 23 feb 2007 kl. 12.42 skrev Steve Davies: Hi, In older versions of asterisk I used to be able to use incominglimit=1 to effectively disable call waiting on a specific SIP channel (Where broken phones do not allow this on the handset itself) In 1.2.x this became call-limit=1, but this prevents the phone from opening a 2nd line in order to transfer a call using attended transfer. The WiKi suggests using SetGroup() etc, but this does not cater for the case where you are Dialling several different phones simultaneously. You can still set one call-limit for the user and another for the peer. The peer call-limit would be used to prevent call waiting and the user limit could be set to a reasonable level so the phone can do transfers. it can be also usefull to use 'limitonpeer' option in sip.conf with this, it would not be needed to define separate type=user and type=peer for each phone, instead define one type=friend and apply limitonpeers=yes ;limitonpeers=no; Apply all call limits (limit=) only to peers, never ; to users. This improves handling of call limits ; and device states in certain situations. The user part ; of a type=friend will still be affected by the call ; limit, but Asterisk will only use one object for ; counting the simultaneous calls. Well, yes. That option does not exist in 1.2, it's someting I have implemented in svn trunk. And in this particular case, different call limits on the user and the peer seemed useful. /O ___ --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] CWI, call-limit and incominglimit
Pavel Jezek wrote: it can be also usefull to use 'limitonpeer' option in sip.conf with this, it would not be needed to define separate type=user and type=peer for each phone, instead define one type=friend and apply limitonpeers=yes ;limitonpeers=no; Apply all call limits (limit=) only to peers, never ; to users. This improves handling of call limits ; and device states in certain situations. The user part ; of a type=friend will still be affected by the call ; limit, but Asterisk will only use one object for ; counting the simultaneous calls. I'm a little confused about the comments shown above, which I assume are from sip.conf. limitonpeers=yes would seem to imply that the limit= value would only apply to the peer portion of the sip user. But the included comments say, The user part of a type=friend call will still be affected by the call limit Those seem to be in conflict, but perhaps it's just my parser :-) Could someone clueful explain? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] CWI, call-limit and incominglimit
Brian Capouch wrote: But the included comments say, The user part of a type=friend call will still be affected by the call limit Those seem to be in conflict, but perhaps it's just my parser :-) Could someone clueful explain? I interpret this that asterisk _internally_ still counting calls for both user and peer, but actually limits calls only for peers... :-\ ___ --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] CWI, call-limit and incominglimit
Hi, In older versions of asterisk I used to be able to use incominglimit=1 to effectively disable call waiting on a specific SIP channel (Where broken phones do not allow this on the handset itself) In 1.2.x this became call-limit=1, but this prevents the phone from opening a 2nd line in order to transfer a call using attended transfer. The WiKi suggests using SetGroup() etc, but this does not cater for the case where you are Dialling several different phones simultaneously. I _could_ dial a whole bunch of Local channels, each of which checked for an extension usage count, but the additional load and complexity in the dialplan seems a bit over-the-top to me, especially when there used to be a one-line solution to this. I also considered separate user and peer sections in sip.conf, but the hosts are dynamic, and there is no way to link the IP address of the peer to the user. My best thought so far is a Macro to check each SIP entry that has CWI disabled, using SetGroup(), and removing it from the dial string if it is in use... Any better suggestions out there? Thanks, Steve ___ --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
