Guys. I would like to hear tips and tricks on extention config best practices, for example, naming, etc. and most of all, how to deal with extention that have a full time hardphone configured and assigned and then a softphone connecting to the same extention, for example, one employee has its hardphone on the office but sometimes when he travel, he uses his softphone to work with, what happens when two phones have the same user id and connect to the same asterisk? How are calls routed or how to handle this kind of scenarios. Thx for any comments Guys.
> I would like to hear tips and tricks on extention config best practices, for > example, naming, etc. and most of all, how to deal with extention that have > a full time hardphone configured and assigned and then a softphone > connecting to the same extention, for example, one employee has its > hardphone on the office but sometimes when he travel, he uses his softphone > to work with, what happens when two phones have the same user id and connect > to the same asterisk? How are calls routed or how to handle this kind of > scenarios.In general terms and without being able to see how the extension is defined in sip.conf, the last phone to register with * will get the call. Assuming both the hard and soft phones register every hour, it is entirely possible the hard phone will get the call for the first 30 minutes and the soft phone for the next 30 minutes.
Rich is indeed correct, Asterisk does not yet support multiple registrations for a single peer entry. Thus when you register the previous registration is discarded and the new one is used. Thus like he said, the last one that registered gets the call. - Joshua Colp. On 6/21/05 9:39 AM, "Rich Adamson" <radamson@routers.com> wrote:>> I would like to hear tips and tricks on extention config best practices, for >> example, naming, etc. and most of all, how to deal with extention that have >> a full time hardphone configured and assigned and then a softphone >> connecting to the same extention, for example, one employee has its >> hardphone on the office but sometimes when he travel, he uses his softphone >> to work with, what happens when two phones have the same user id and connect >> to the same asterisk? How are calls routed or how to handle this kind of >> scenarios. > > In general terms and without being able to see how the extension is > defined in sip.conf, the last phone to register with * will get the > call. > > Assuming both the hard and soft phones register every hour, it is > entirely possible the hard phone will get the call for the first > 30 minutes and the soft phone for the next 30 minutes. > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
In environments where users have their hard and soft phones... How do you glue everything together? |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Rich Adamson |Sent: Martes, 21 de Junio de 2005 07:39 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice | |> I would like to hear tips and tricks on extention config best |> practices, for example, naming, etc. and most of all, how to |deal with |> extention that have a full time hardphone configured and |assigned and |> then a softphone connecting to the same extention, for example, one |> employee has its hardphone on the office but sometimes when |he travel, |> he uses his softphone to work with, what happens when two |phones have |> the same user id and connect to the same asterisk? How are calls |> routed or how to handle this kind of scenarios. | |In general terms and without being able to see how the |extension is defined in sip.conf, the last phone to register |with * will get the call. | |Assuming both the hard and soft phones register every hour, it |is entirely possible the hard phone will get the call for the |first 30 minutes and the soft phone for the next 30 minutes. | | |_______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |
Ok, so how are you guys coping with scenarios like this? Managers working in the office during the day or mid day and then in the afternoon, working remotely using their laptops? |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Joshua Colp |Sent: Martes, 21 de Junio de 2005 08:20 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice | |Rich is indeed correct, Asterisk does not yet support multiple |registrations for a single peer entry. Thus when you register |the previous registration is discarded and the new one is |used. Thus like he said, the last one that registered gets the call. | |- Joshua Colp. | | |On 6/21/05 9:39 AM, "Rich Adamson" <radamson@routers.com> wrote: | |>> I would like to hear tips and tricks on extention config best |>> practices, for example, naming, etc. and most of all, how to deal |>> with extention that have a full time hardphone configured and |>> assigned and then a softphone connecting to the same extention, for |>> example, one employee has its hardphone on the office but sometimes |>> when he travel, he uses his softphone to work with, what |happens when |>> two phones have the same user id and connect to the same asterisk? |>> How are calls routed or how to handle this kind of scenarios. |> |> In general terms and without being able to see how the extension is |> defined in sip.conf, the last phone to register with * will get the |> call. |> |> Assuming both the hard and soft phones register every hour, it is |> entirely possible the hard phone will get the call for the first 30 |> minutes and the soft phone for the next 30 minutes. |> |> |> _______________________________________________ |> Asterisk-Users mailing list |> Asterisk-Users@lists.digium.com |> http://lists.digium.com/mailman/listinfo/asterisk-users |> To UNSUBSCRIBE or update options visit: |> http://lists.digium.com/mailman/listinfo/asterisk-users | | |_______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |
Goolsby, Daniel S (Daniel)
2005-Jun-21 07:38 UTC
[Asterisk-Users] Extension Configuration Best Practice
Could you just configure the extention to be a ring group instead of an actual extention, or ring queue.. then have his phone/laptop log in whenever he's at the office/coffee shop? I know AMP has the functionality, but I haven't gone behind the scenes and looked at the sip.conf or extensions.conf to see what the script or macro is doing in a ring group/queue. Daniel -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Anton Krall Sent: Tuesday, June 21, 2005 9:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Extension Configuration Best Practice Ok, so how are you guys coping with scenarios like this? Managers working in the office during the day or mid day and then in the afternoon, working remotely using their laptops? |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Joshua Colp |Sent: Martes, 21 de Junio de 2005 08:20 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice | |Rich is indeed correct, Asterisk does not yet support multiple |registrations for a single peer entry. Thus when you register |the previous registration is discarded and the new one is |used. Thus like he said, the last one that registered gets the call. | |- Joshua Colp. | | |On 6/21/05 9:39 AM, "Rich Adamson" <radamson@routers.com> wrote: | |>> I would like to hear tips and tricks on extention config best |>> practices, for example, naming, etc. and most of all, how to deal |>> with extention that have a full time hardphone configured and |>> assigned and then a softphone connecting to the same extention, for |>> example, one employee has its hardphone on the office but sometimes |>> when he travel, he uses his softphone to work with, what |happens when |>> two phones have the same user id and connect to the same asterisk? |>> How are calls routed or how to handle this kind of scenarios. |> |> In general terms and without being able to see how the extension is |> defined in sip.conf, the last phone to register with * will get the |> call. |> |> Assuming both the hard and soft phones register every hour, it is |> entirely possible the hard phone will get the call for the first 30 |> minutes and the soft phone for the next 30 minutes. |> |> |> _______________________________________________ |> Asterisk-Users mailing list |> Asterisk-Users@lists.digium.com |> http://lists.digium.com/mailman/listinfo/asterisk-users |> To UNSUBSCRIBE or update options visit: |> http://lists.digium.com/mailman/listinfo/asterisk-users | | |_______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
In addition to the call forwarding approach noted in an earlier response, you can also have both the hard and soft phones register as different exetensions, then use something like: exten => 1234,2,Dial(SIP/1234&SIP/1235,15) to ring both phones, and the first one to answer gets the call. ------------------------> Ok, so how are you guys coping with scenarios like this? Managers working in > the office during the day or mid day and then in the afternoon, working > remotely using their laptops? > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com > |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > |Joshua Colp > |Sent: Martes, 21 de Junio de 2005 08:20 a.m. > |To: Asterisk Users Mailing List - Non-Commercial Discussion > |Subject: Re: [Asterisk-Users] Extension Configuration Best Practice > | > |Rich is indeed correct, Asterisk does not yet support multiple > |registrations for a single peer entry. Thus when you register > |the previous registration is discarded and the new one is > |used. Thus like he said, the last one that registered gets the call. > | > |- Joshua Colp. > | > | > |On 6/21/05 9:39 AM, "Rich Adamson" <radamson@routers.com> wrote: > | > |>> I would like to hear tips and tricks on extention config best > |>> practices, for example, naming, etc. and most of all, how to deal > |>> with extention that have a full time hardphone configured and > |>> assigned and then a softphone connecting to the same extention, for > |>> example, one employee has its hardphone on the office but sometimes > |>> when he travel, he uses his softphone to work with, what > |happens when > |>> two phones have the same user id and connect to the same asterisk? > |>> How are calls routed or how to handle this kind of scenarios. > |> > |> In general terms and without being able to see how the extension is > |> defined in sip.conf, the last phone to register with * will get the > |> call. > |> > |> Assuming both the hard and soft phones register every hour, it is > |> entirely possible the hard phone will get the call for the first 30 > |> minutes and the soft phone for the next 30 minutes. > |> > |> > |> _______________________________________________ > |> Asterisk-Users mailing list > |> Asterisk-Users@lists.digium.com > |> http://lists.digium.com/mailman/listinfo/asterisk-users > |> To UNSUBSCRIBE or update options visit: > |> http://lists.digium.com/mailman/listinfo/asterisk-users > | > | > |_______________________________________________ > |Asterisk-Users mailing list > |Asterisk-Users@lists.digium.com > |http://lists.digium.com/mailman/listinfo/asterisk-users > |To UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > | > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >---------------End of Original Message-----------------