Re: [asterisk-users] RE: [asterisk-dev] Phone status
Your are right, I dont have to invent the wheel again, and I'm getting cleverer by looking at other peoples code. But this does not solve my problems, I have worked in the PABX business as a software developer for about 8 years, and coming to * is not all that easy. For instance, * does not give you very good information of the state of extensions (like we are used to in the old-fashioned PABX business), or maybe I'm not good at finding the information. I'm trying to port an existing Windows application to *, its a dialer, used to dial and se information about received calls. I know how to dial new calls, by using ORIGINATE on the AMI. I can receive some status information via the AMI, but consider this example: I receive a call, which I accept. I get an event from theAMI,that the call is now in the UP state. I receive another call, I get en event from the AMI, that the new call is in the RINGING state. So far, so good. I now answer the other call (for instance by the line button on my phone). Both calls are now in the UP state, who am I talking to? This, and many other questions, are currently making me even more thin haired than normal :-) Michael 2006/8/25, C F [EMAIL PROTECTED]: So how about inventing a car? The auto industry is much more profitable.The point; there is no point in reinventing the wheel, why are you writing this from scratch?On 8/24/06, Mir [EMAIL PROTECTED] wrote: What do you mean? I'm not looking for someone elses work, I'm developing an application from scratch. Michael 2006/8/24, Andrew Kirch [EMAIL PROTECTED]: Umm… Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.comSubject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQL table. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: Status Privilege: Call Channel: SIP/310-08697fb8 CallerID: 310 CallerIDName: unknown Account: State: Up Link: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: Status Privilege: Call Channel: SIP/311-0868fd98 CallerID: 311 CallerIDName: Snom Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 13 Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: Status Privilege: Call Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: unknown Account: State: Up Link: IAX2/MR-1 Uniqueid: 1156442974.76 Event: Status Privilege: Call Channel: IAX2/MR-1 CallerID: 35436121 CallerIDName: unknown Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --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 ___ --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 ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] RE: [asterisk-dev] Phone status
IIRC, you'll want to look at 'hint' extensions, and possibly subscriptions to get status updates From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MirSent: Monday, August 28, 2006 9:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] RE: [asterisk-dev] Phone status Your are right, I dont have to invent the wheel again, and I'm getting cleverer by looking at other peoples code. But this does not solve my problems, I have worked in the PABX business as a software developer for about 8 years, and coming to * is not all that easy. For instance, * does not give you very good information of the state of extensions (like we are used to in the "old-fashioned" PABX business), or maybe I'm not good at finding the information. I'm trying to port an existing Windows application to *, its a dialer, used to dial and se information about received calls. I know how to dial new calls, by using ORIGINATE on the AMI. I can receive some status information via the AMI, but consider this example: I receive a call, which I accept. I get an event from theAMI,that the call is now in the UP state. I receive another call, I get en event from the AMI, that the new call is in the RINGING state. So far, so good. I now answer the other call (for instance by the line button on my phone). Both calls are now in the UP state, who am I talking to? This, and many other questions, are currently making me even more thin haired than normal :-) Michael 2006/8/25, C F [EMAIL PROTECTED]: So how about inventing a car? The auto industry is much more profitable.The point; there is no point in reinventing the wheel, why are you writing this from scratch?On 8/24/06, Mir [EMAIL PROTECTED] wrote: What do you mean? I'm not looking for someone elses work, I'm developing an application from scratch. Michael 2006/8/24, Andrew Kirch [EMAIL PROTECTED]: Umm Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.comSubject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a "Status" to the manager-interface, and processing the return data and then put the result into a MySQL table. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: Status Privilege: Call Channel: SIP/310-08697fb8 CallerID: 310 CallerIDName: unknown Account: State: Up Link: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: Status Privilege: Call Channel: SIP/311-0868fd98 CallerID: 311 CallerIDName: Snom Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 13 Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: Status Privilege: Call Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: unknown Account: State: Up Link: IAX2/MR-1 Uniqueid: 1156442974.76 Event: Status Privilege: Call Channel: IAX2/MR-1 CallerID: 35436121 CallerIDName: unknown Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] RE: [asterisk-dev] Phone status
What do you mean? I'm not looking for someone elses work, I'm developing an application from scratch. Michael 2006/8/24, Andrew Kirch [EMAIL PROTECTED]: Umm… Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of MirSent: Thursday, August 24, 2006 2:18 PMTo: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQLtable. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: StatusPrivilege: CallChannel: SIP/310-08697fb8CallerID: 310CallerIDName: unknownAccount: State: UpLink: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: StatusPrivilege: CallChannel: SIP/311-0868fd98CallerID: 311CallerIDName: SnomAccount: State: UpContext: macro-vm Extension: sPriority: 5Seconds: 13Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: StatusPrivilege: CallChannel: SIP/311-08695698CallerID: 35254390CallerIDName: unknownAccount: State: UpLink: IAX2/MR-1 Uniqueid: 1156442974.76 Event: StatusPrivilege: CallChannel: IAX2/MR-1CallerID: 35436121CallerIDName: unknownAccount: State: UpContext: macro-vm Extension: sPriority: 5Seconds: 9Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] RE: [asterisk-dev] Phone status
I guess what Andrew was saying is what are you trying to do specifically that Flash Operator Panel doesnt already give you Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Thursday, 24 August 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: [asterisk-dev] Phone status What do you mean? I'm not looking for someone elses work, I'm developing an application from scratch. Michael 2006/8/24, Andrew Kirch [EMAIL PROTECTED]: Umm Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQLtable. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: Status Privilege: Call Channel: SIP/310-08697fb8 CallerID: 310 CallerIDName: unknown Account: State: Up Link: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: Status Privilege: Call Channel: SIP/311-0868fd98 CallerID: 311 CallerIDName: Snom Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 13 Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: Status Privilege: Call Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: unknown Account: State: Up Link: IAX2/MR-1 Uniqueid: 1156442974.76 Event: Status Privilege: Call Channel: IAX2/MR-1 CallerID: 35436121 CallerIDName: unknown Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --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 ___ --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] RE: [asterisk-dev] Phone status
So how about inventing a car? The auto industry is much more profitable. The point; there is no point in reinventing the wheel, why are you writing this from scratch? On 8/24/06, Mir [EMAIL PROTECTED] wrote: What do you mean? I'm not looking for someone elses work, I'm developing an application from scratch. Michael 2006/8/24, Andrew Kirch [EMAIL PROTECTED]: Umm… Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQL table. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: Status Privilege: Call Channel: SIP/310-08697fb8 CallerID: 310 CallerIDName: unknown Account: State: Up Link: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: Status Privilege: Call Channel: SIP/311-0868fd98 CallerID: 311 CallerIDName: Snom Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 13 Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: Status Privilege: Call Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: unknown Account: State: Up Link: IAX2/MR-1 Uniqueid: 1156442974.76 Event: Status Privilege: Call Channel: IAX2/MR-1 CallerID: 35436121 CallerIDName: unknown Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --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 ___ --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 ___ --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] RE: [asterisk-dev] Phone status
Umm Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.com Subject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a Status to the manager-interface, and processing the return data and then put the result into a MySQLtable. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: Status Privilege: Call Channel: SIP/310-08697fb8 CallerID: 310 CallerIDName: unknown Account: State: Up Link: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: Status Privilege: Call Channel: SIP/311-0868fd98 CallerID: 311 CallerIDName: Snom Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 13 Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: Status Privilege: Call Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: unknown Account: State: Up Link: IAX2/MR-1 Uniqueid: 1156442974.76 Event: Status Privilege: Call Channel: IAX2/MR-1 CallerID: 35436121 CallerIDName: unknown Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --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
