Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM
Basically you need to disable call waiting on your SIP device (if it supports call waiting to begin with). When the second call comes into the SIP device with call waiting disabled, it should send a 486 SIP message (mine says "486 Busy Here") back to the Asterisk. You can see this in "sip debug" mode on the console. Then setup your extensions.conf to take the appropriate action on Busy like any other extension. Sean _______________________________________________ Sean Robertson NETXUSA p. 800-289-6389 f. 864-233-4344 "Ask me about Voice over IP." http://www.netxusa.com/ ----- Original Message ----- From: "Paulo Mannheimer" <paulohm@instant.com.br> To: <asterisk-users@lists.digium.com> Sent: Thursday, September 11, 2003 4:32 PM Subject: [Asterisk-Users] SIP busy> Hi, > > I would like * to treat a SIP extension as a normal extension, when it > comes to the busy functionality. In other words, if someone tries to > call the SIP phone and there is already an ongoing conversation, the new > caller should get a busy message/tone > > Is there any parameter that I can set? Is this something that should be > configured at my softphone? > > Best, > > PHM > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
[message re-ordered]>----- Original Message ----- >From: "Paulo Mannheimer" <paulohm@instant.com.br> >To: <asterisk-users@lists.digium.com> >Sent: Thursday, September 11, 2003 4:32 PM >Subject: [Asterisk-Users] SIP busy > > >> Hi, >> >> I would like * to treat a SIP extension as a normal extension, when it >> comes to the busy functionality. In other words, if someone tries to >> call the SIP phone and there is already an ongoing conversation, the new >> caller should get a busy message/tone >> >> Is there any parameter that I can set? Is this something that should be >> configured at my softphone? >> >> Best, >> > > PHM>Basically you need to disable call waiting on your SIP device (if it >supports call waiting to begin with). When the second call comes into the >SIP device with call waiting disabled, it should send a 486 SIP message >(mine says "486 Busy Here") back to the Asterisk. You can see this in "sip >debug" mode on the console. > >Then setup your extensions.conf to take the appropriate action on Busy like >any other extension. > >Sean >_______________________________________________ > >Sean Robertson > >NETXUSA >p. 800-289-6389 >f. 864-233-4344 "Ask me about Voice over IP." >http://www.netxusa.com/Another method would simply be to keep a call counter for existing calls, and increment it/decrement it when calls are made and then hung up. Put a short GotoIf before your Dial statement to check if the line is "occupied" and then reject the call if that is the case. [test] exten => 1234,1,DBGet(STATUS=${EXTEN}/OFFHOOK) exten => 1234,2,SetVar(CALLEDNUMBER=${EXTEN}) exten => 1234,3,GotoIf($[${STATUS} = 1}]?106:3) exten => 1234,4,DBPut(${EXTEN}/OFFHOOK=1) exten => 1234,5,Dial(SIP/1234,20) exten => 1234,6,DBPut(${EXTEN}/OFFHOOK=0) exten => 1234,7,Voicemail2(u1234) exten => 1234,106,DBPut(${EXTEN}/OFFHOOK=0) exten => 1234,107,Voicemail2(b1234) exten => h,1,DBPut(${CALLEDNUNMBER}/OFFHOOK=0) exten => h,2,Hangup JT
Thanks John and all, Unfortunatelly this will not work for me, because the SIP phones are agents and I'm managing incomming calls through a queue. Anyone knows a SIP softphone that supports disabling call waiting? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of John Todd Sent: September 11, 2003 8:20 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP busy [message re-ordered]>----- Original Message ----- >From: "Paulo Mannheimer" <paulohm@instant.com.br> >To: <asterisk-users@lists.digium.com> >Sent: Thursday, September 11, 2003 4:32 PM >Subject: [Asterisk-Users] SIP busy > > >> Hi, >> >> I would like * to treat a SIP extension as a normal extension, when >> it comes to the busy functionality. In other words, if someone tries>> to call the SIP phone and there is already an ongoing conversation, >> the new caller should get a busy message/tone >> >> Is there any parameter that I can set? Is this something that should >> be configured at my softphone? >> >> Best, >> > > PHM>Basically you need to disable call waiting on your SIP device (if it >supports call waiting to begin with). When the second call comes into >the SIP device with call waiting disabled, it should send a 486 SIP >message (mine says "486 Busy Here") back to the Asterisk. You can see >this in "sip debug" mode on the console. > >Then setup your extensions.conf to take the appropriate action on Busy >like any other extension. > >Sean >_______________________________________________ > >Sean Robertson > >NETXUSA >p. 800-289-6389 >f. 864-233-4344 "Ask me about Voice over IP." >http://www.netxusa.com/Another method would simply be to keep a call counter for existing calls, and increment it/decrement it when calls are made and then hung up. Put a short GotoIf before your Dial statement to check if the line is "occupied" and then reject the call if that is the case. [test] exten => 1234,1,DBGet(STATUS=${EXTEN}/OFFHOOK) exten => 1234,2,SetVar(CALLEDNUMBER=${EXTEN}) exten => 1234,3,GotoIf($[${STATUS} = 1}]?106:3) exten => 1234,4,DBPut(${EXTEN}/OFFHOOK=1) exten => 1234,5,Dial(SIP/1234,20) exten => 1234,6,DBPut(${EXTEN}/OFFHOOK=0) exten => 1234,7,Voicemail2(u1234) exten => 1234,106,DBPut(${EXTEN}/OFFHOOK=0) exten => 1234,107,Voicemail2(b1234) exten => h,1,DBPut(${CALLEDNUNMBER}/OFFHOOK=0) exten => h,2,Hangup JT _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users