RE: [Asterisk-Users] Hunting S(n)IPs
[EMAIL PROTECTED] wrote: Another observation of something which doesn't work: exten = 3200,1,Dial(SIP/3200,20,tTr) exten = 3200,2,Playback(tt-weasels) exten = 3200,3,Hangup exten = 3200,102,Dial(SIP/3201,20,tTr) exten = 3200,103,Playback(tt-weasels) exten = 3200,104,Hangup exten = 3200,203,Dial(SIP/3202,20,tTr) exten = 3200,204,Playback(tt-weasels) exten = 3200,205,Hangup The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer the first call has been answered. Therefore, Call#2 happily dials 3200 again, although 3200 is currently talking. I also tried to limit the number of calls going to the phone with outgoinglimit=1 in the sip.conf, but that makes no difference either. According to the wiki that functionality is broken. Two things: 1) Have you looked at call queue's? 2) I think you should have been looking at incominglimit, not outgoinglimit, or possibly both of them together in some combination. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hunting S(n)IPs
On Tue, 13 Apr 2004, Andrew Thompson wrote: Two things: 1) Have you looked at call queue's? 2) I think you should have been looking at incominglimit, not outgoinglimit, or possibly both of them together in some combination. In response to [EMAIL PROTECTED], who wrote: The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer the first call has been answered. Therefore, Call#2 happily dials 3200 again, although 3200 is currently talking. I also tried to limit the number of calls going to the phone with outgoinglimit=1 in the sip.conf, but that makes no difference either. According to the wiki that functionality is broken. Another thing to try is to disable call waiting on the [EMAIL PROTECTED] phone (if call waiting is enabled, it's doing what you've asked it to)... Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hunting S(n)IPs
Andrew Thompson wrote: [EMAIL PROTECTED] wrote: Another observation of something which doesn't work: exten = 3200,1,Dial(SIP/3200,20,tTr) exten = 3200,2,Playback(tt-weasels) exten = 3200,3,Hangup exten = 3200,102,Dial(SIP/3201,20,tTr) exten = 3200,103,Playback(tt-weasels) exten = 3200,104,Hangup exten = 3200,203,Dial(SIP/3202,20,tTr) exten = 3200,204,Playback(tt-weasels) exten = 3200,205,Hangup The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer the first call has been answered. Therefore, Call#2 happily dials 3200 again, although 3200 is currently talking. I also tried to limit the number of calls going to the phone with outgoinglimit=1 in the sip.conf, but that makes no difference either. According to the wiki that functionality is broken. Two things: 1) Have you looked at call queue's? 2) I think you should have been looking at incominglimit, not outgoinglimit, or possibly both of them together in some combination. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I may be missing something here, but I'll make this suggestion just in case you haven't already considered it. Have your phone register multiple call appearances with the same DN. For instance, my 7960 has three appearances of 2205. Calls are automatically offered to the first available appearance, kind of like what you'd expect. I think this is the behavior you're looking for, but you may be trying to do it he hard way. Cheers, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hunting S(n)IPs
snip 2) I think you should have been looking at incominglimit, not outgoinglimit, or possibly both of them together in some combination. /snip Another perspective issue. Apparantly 'incoming' means into the [*] box, and outgoing is leaving the [*]. In any case, I tried both, but 'outgoing' is confirmed broken. WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hunting S(n)IPs
Another thing to try is to disable call waiting on the [EMAIL PROTECTED] phone (if call waiting is enabled, it's doing what you've asked it to)... Yep, except on the Polycom, we have found no way to disable call-waiting. WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hunting S(n)IPs
Hi Akk, If this has been discussed/done then apologies be-4-hand. I did not find it in the Wiki or the Archives. Here's the question. We have incoming PRI lines, all on the same main number. An attendant is supposed to handle all incoming calls. Now, let's say I have a multi-line SIP phone. For argument's sake (and to keep it simple) say I only have two lines. We'll call them SIP/att-0 and SIP/att-1. Here's the desired behavior: Call comes in. Gets to Dial(SIP/att-0) Other call comes in b4 first one is answered. Gets to Dial(SIP/att-1) Or, if Line-0 is busy (however) I still want to ring line-1. Kinda-like a hunt group. The problem I am having is that I cannot find out (real-time - in the dial plan) whether a particular channel is already in use. Otherwise a GotIf() might do the trick. I tried to set a parameter in the DB to indicate that a chan is in use, e.g. exten = s,1,DBPut(inuse/chan${ARG1}=1) exten = s,2,Dial(SIP/att-${ARG1}) however, I do not seem to be able to catch the event wich releases the channel in order to reset the DB variable. exten = h,1,DCPut(inuse/chan${ARG1}=0) ; this never gets executed Any ideas? Cheers, WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hunting S(n)IPs
Another observation of something which doesn't work: exten = 3200,1,Dial(SIP/3200,20,tTr) exten = 3200,2,Playback(tt-weasels) exten = 3200,3,Hangup exten = 3200,102,Dial(SIP/3201,20,tTr) exten = 3200,103,Playback(tt-weasels) exten = 3200,104,Hangup exten = 3200,203,Dial(SIP/3202,20,tTr) exten = 3200,204,Playback(tt-weasels) exten = 3200,205,Hangup The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer the first call has been answered. Therefore, Call#2 happily dials 3200 again, although 3200 is currently talking. I also tried to limit the number of calls going to the phone with outgoinglimit=1 in the sip.conf, but that makes no difference either. According to the wiki that functionality is broken. - Original Message Follows - Hi Akk, If this has been discussed/done then apologies be-4-hand. I did not find it in the Wiki or the Archives. Here's the question. We have incoming PRI lines, all on the same main number. An attendant is supposed to handle all incoming calls. Now, let's say I have a multi-line SIP phone. For argument's sake (and to keep it simple) say I only have two lines. We'll call them SIP/att-0 and SIP/att-1. Here's the desired behavior: Call comes in. Gets to Dial(SIP/att-0) Other call comes in b4 first one is answered. Gets to Dial(SIP/att-1) Or, if Line-0 is busy (however) I still want to ring line-1. Kinda-like a hunt group. The problem I am having is that I cannot find out (real-time - in the dial plan) whether a particular channel is already in use. Otherwise a GotIf() might do the trick. I tried to set a parameter in the DB to indicate that a chan is in use, e.g. exten = s,1,DBPut(inuse/chan${ARG1}=1) exten = s,2,Dial(SIP/att-${ARG1}) however, I do not seem to be able to catch the event wich releases the channel in order to reset the DB variable. exten = h,1,DCPut(inuse/chan${ARG1}=0) ; this never gets executed Any ideas? Cheers, WW Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users