Hi everyone, I have some doubt about use or not to use SER. I need a solution using a single linux box that manages, aproximatly 500-1000 registred SIP users, but not more than 50 simultaneouly calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in diferents cities of my country (Argentine) connected through Internet (with public IP). I was searching for SER solutions (and works perfectly) but it does not support Prepaid Billing. So I post a message (on SerUsers maillist) and everybody said me to use Asterisk to use a Prepaid Billing App., so I install Asterisk. I "googled", read this maillist (and post some message) and I receive some helpful answers recomending me to install ASTCC, so I install it too and work perfectly too. My questions (if someone could help me) are : 1) What platform (hardware) do I need to support my call flow (500-1000 registers and 50 simultaneouly calls)? 2) Do I need to install SER? 3) If YES, do I need to register my SIP clients on SER and forward all the calls to Asterisk? 3) If NO, do I need to register my SIP clients on Asterisk and forward all the calls to SER? 4) Can I use only Asterisk to REGISTER, AUTH, ACC, ROUTE CALLS, etc, SIP clients? 5) Could I use extension.conf file to route my calls to my diferents Cisco PSTN GW? 6) And how can I use MySQL instead of file? (I have created the DB and tables but I do not know how to make Asterisk use it instead the extension.conf file) 7) I found easy to use only Asterisk, but I have read that it uses to much CPU and memory, is that true? 8) Could anyone some me information about how to configure Asterisk to receive calls through Cisco PSTN GW? 9) THANK YOU VERY VERY MUCH!!! Nahuel Ramos.
We use SER + Asterisk. One heck of a powerful combination. On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote:> Hi everyone, > I have some doubt about use or not to use SER. > I need a solution using a single linux box that manages, aproximatly > 500-1000 registred SIP users, but not more than 50 simultaneouly > calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in > diferents cities of my country (Argentine) connected through Internet > (with public IP). > I was searching for SER solutions (and works perfectly) but it does > not support Prepaid Billing. So I post a message (on SerUsers > maillist) and everybody said me to use Asterisk to use a Prepaid > Billing App., so I install Asterisk. > I "googled", read this maillist (and post some message) and I > receive some helpful answers recomending me to install ASTCC, so I > install it too and work perfectly too. > My questions (if someone could help me) are : > 1) What platform (hardware) do I need to support my call flow > (500-1000 registers and 50 simultaneouly calls)? > 2) Do I need to install SER? > 3) If YES, do I need to register my SIP clients on SER and forward > all the calls to Asterisk? > 3) If NO, do I need to register my SIP clients on Asterisk and > forward all the calls to SER? > 4) Can I use only Asterisk to REGISTER, AUTH, ACC, ROUTE CALLS, > etc, SIP clients? > 5) Could I use extension.conf file to route my calls to my > diferents Cisco PSTN GW? > 6) And how can I use MySQL instead of file? (I have created the DB > and tables but I do not know how to make Asterisk use it instead the > extension.conf file) > 7) I found easy to use only Asterisk, but I have read that it uses > to much CPU and memory, is that true? > 8) Could anyone some me information about how to configure Asterisk > to receive calls through Cisco PSTN GW? > 9) THANK YOU VERY VERY MUCH!!! > > Nahuel Ramos. > _______________________________________________ > 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 >
Hi i am stuck with the same dilemma, as the original poster I have setup ser now (with the helpful pointer from Girish..tks mate) and can do Ip <---> Ip calls, and IP --->pstn (via cisco box), all via ser, however I also have asterisk installed, and now am wondering where I use asterisk, it was/is suggested I use it for all pbx functions such as voicemail etc, however I cant seem to see how on a call not answered howto get ser to send to asterisk. I also am looking at the prepaid billing option, and hence the main reason for asterisk, but unless all calls flow via asterisk instead of ser I cant see the point of astcc, and if they do all flow via asterisk, then why put ser infront... tks iqbal On 10/21/2004, "Darren Sessions" <dsessions@ionosphere.net> wrote:>We use SER + Asterisk. One heck of a powerful combination. > > >On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote: > >> Hi everyone, >> I have some doubt about use or not to use SER. >> I need a solution using a single linux box that manages, aproximatly >> 500-1000 registred SIP users, but not more than 50 simultaneouly >> calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in >> diferents cities of my country (Argentine) connected through Internet >> (with public IP). >> I was searching for SER solutions (and works perfectly) but it does >> not support Prepaid Billing. So I post a message (on SerUsers >> maillist) and everybody said me to use Asterisk to use a Prepaid >> Billing App., so I install Asterisk. >> I "googled", read this maillist (and post some message) and I >> receive some helpful answers recomending me to install ASTCC, so I >> install it too and work perfectly too. >> My questions (if someone could help me) are : >> 1) What platform (hardware) do I need to support my call flow >> (500-1000 registers and 50 simultaneouly calls)? >> 2) Do I need to install SER? >> 3) If YES, do I need to register my SIP clients on SER and forward >> all the calls to Asterisk? >> 3) If NO, do I need to register my SIP clients on Asterisk and >> forward all the calls to SER? >> 4) Can I use only Asterisk to REGISTER, AUTH, ACC, ROUTE CALLS, >> etc, SIP clients? >> 5) Could I use extension.conf file to route my calls to my >> diferents Cisco PSTN GW? >> 6) And how can I use MySQL instead of file? (I have created the DB >> and tables but I do not know how to make Asterisk use it instead the >> extension.conf file) >> 7) I found easy to use only Asterisk, but I have read that it uses >> to much CPU and memory, is that true? >> 8) Could anyone some me information about how to configure Asterisk >> to receive calls through Cisco PSTN GW? >> 9) THANK YOU VERY VERY MUCH!!! >> >> Nahuel Ramos. >> _______________________________________________ >> 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 > >
I use a failure route in ser for the call to be sent to the voicemail system.. I use ser as mainly a primary router for sip messages that sits in the center of a ring of asterisk servers that feed the clients.. Iqbal wrote:>Hi > >i am stuck with the same dilemma, as the original poster > >I have setup ser now (with the helpful pointer from Girish..tks mate) and >can do Ip <---> Ip calls, and IP --->pstn (via cisco box), all via ser, >however I also have asterisk installed, and now am wondering where I use >asterisk, it was/is suggested I use it for all pbx functions such as >voicemail etc, however I cant seem to see how on a call not answered >howto get ser to send to asterisk. > >I also am looking at the prepaid billing option, and hence the main >reason for asterisk, but unless all calls flow via asterisk instead of >ser I cant see the point of astcc, and if they do all flow via asterisk, >then why put ser infront... > >tks > >iqbal > >On 10/21/2004, "Darren Sessions" <dsessions@ionosphere.net> wrote: > > > >>We use SER + Asterisk. One heck of a powerful combination. >> >> >>On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote: >> >> >> >>>Hi everyone, >>> I have some doubt about use or not to use SER. >>> I need a solution using a single linux box that manages, aproximatly >>>500-1000 registred SIP users, but not more than 50 simultaneouly >>>calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in >>>diferents cities of my country (Argentine) connected through Internet >>>(with public IP). >>> I was searching for SER solutions (and works perfectly) but it does >>>not support Prepaid Billing. So I post a message (on SerUsers >>>maillist) and everybody said me to use Asterisk to use a Prepaid >>>Billing App., so I install Asterisk. >>> I "googled", read this maillist (and post some message) and I >>>receive some helpful answers recomending me to install ASTCC, so I >>>install it too and work perfectly too. >>> My questions (if someone could help me) are : >>> 1) What platform (hardware) do I need to support my call flow >>>(500-1000 registers and 50 simultaneouly calls)? >>> 2) Do I need to install SER? >>> 3) If YES, do I need to register my SIP clients on SER and forward >>>all the calls to Asterisk? >>> 3) If NO, do I need to register my SIP clients on Asterisk and >>>forward all the calls to SER? >>> 4) Can I use only Asterisk to REGISTER, AUTH, ACC, ROUTE CALLS, >>>etc, SIP clients? >>> 5) Could I use extension.conf file to route my calls to my >>>diferents Cisco PSTN GW? >>> 6) And how can I use MySQL instead of file? (I have created the DB >>>and tables but I do not know how to make Asterisk use it instead the >>>extension.conf file) >>> 7) I found easy to use only Asterisk, but I have read that it uses >>>to much CPU and memory, is that true? >>> 8) Could anyone some me information about how to configure Asterisk >>>to receive calls through Cisco PSTN GW? >>> 9) THANK YOU VERY VERY MUCH!!! >>> >>> Nahuel Ramos. >>>_______________________________________________ >>>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 > > > >-------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041021/f8e47286/mhess.vcf
Hi Thanks for the reply, I have tried failure_route, however I think I am putting the intial detection in the wrong place, since nuthing gets sent to asterisk, on no answer --- if (uri=~"^sip:\+?[0-9]+@domain.com" ) { log(1, "Forwarding to PSTN\n"); t_relay_to_udp("213.166.5.135","5060"); t_on_failure("6"); break; }; this condition me thinks will never be met, i also tried doing the t_on_failure part early in my main route, where it looks up to see if user is part of voicemail group, but again that would not work simply becauise the pstn number that is dialed will never be part of my group. What I need to work out is how after the invite and forward to the cisco box happens (as the code above does), can I detect that no answer happened, and then drop it into the failure section tks iqbal On 10/21/2004, "Matt Hess" <mhess@livewirenet.com> wrote:>I use a failure route in ser for the call to be sent to the voicemail >system.. >I use ser as mainly a primary router for sip messages that sits in the >center of a ring of asterisk servers that feed the clients.. > > >Iqbal wrote: > >>Hi >> >>i am stuck with the same dilemma, as the original poster >> >>I have setup ser now (with the helpful pointer from Girish..tks mate) and >>can do Ip <---> Ip calls, and IP --->pstn (via cisco box), all via ser, >>however I also have asterisk installed, and now am wondering where I use >>asterisk, it was/is suggested I use it for all pbx functions such as >>voicemail etc, however I cant seem to see how on a call not answered >>howto get ser to send to asterisk. >> >>I also am looking at the prepaid billing option, and hence the main >>reason for asterisk, but unless all calls flow via asterisk instead of >>ser I cant see the point of astcc, and if they do all flow via asterisk, >>then why put ser infront... >> >>tks >> >>iqbal >> >>On 10/21/2004, "Darren Sessions" <dsessions@ionosphere.net> wrote: >> >> >> >>>We use SER + Asterisk. One heck of a powerful combination. >>> >>> >>>On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote: >>> >>> >>> >>>>Hi everyone, >>>> I have some doubt about use or not to use SER. >>>> I need a solution using a single linux box that manages, aproximatly >>>>500-1000 registred SIP users, but not more than 50 simultaneouly >>>>calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in >>>>diferents cities of my country (Argentine) connected through Internet >>>>(with public IP). >>>> I was searching for SER solutions (and works perfectly) but it does >>>>not support Prepaid Billing. So I post a message (on SerUsers >>>>maillist) and everybody said me to use Asterisk to use a Prepaid >>>>Billing App., so I install Asterisk. >>>> I "googled", read this maillist (and post some message) and I >>>>receive some helpful answers recomending me to install ASTCC, so I >>>>install it too and work perfectly too. >>>> My questions (if someone could help me) are : >>>> 1) What platform (hardware) do I need to support my call flow >>>>(500-1000 registers and 50 simultaneouly calls)? >>>> 2) Do I need to install SER? >>>> 3) If YES, do I need to register my SIP clients on SER and forward >>>>all the calls to Asterisk? >>>> 3) If NO, do I need to register my SIP clients on Asterisk and >>>>forward all the calls to SER? >>>> 4) Can I use only Asterisk to REGISTER, AUTH, ACC, ROUTE CALLS, >>>>etc, SIP clients? >>>> 5) Could I use extension.conf file to route my calls to my >>>>diferents Cisco PSTN GW? >>>> 6) And how can I use MySQL instead of file? (I have created the DB >>>>and tables but I do not know how to make Asterisk use it instead the >>>>extension.conf file) >>>> 7) I found easy to use only Asterisk, but I have read that it uses >>>>to much CPU and memory, is that true? >>>> 8) Could anyone some me information about how to configure Asterisk >>>>to receive calls through Cisco PSTN GW? >>>> 9) THANK YOU VERY VERY MUCH!!! >>>> >>>> Nahuel Ramos. >>>>_______________________________________________ >>>>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 >> >> >> >> > >
To helpful link!. It answer lot of my question. Thank you very much "Asterisk in Yahoo"... Nahuel Ramos. On Fri, 22 Oct 2004 11:46:46 -0700 (PDT), Asterisk . <asterisk_in@yahoo.com> wrote:> --- Nahuel Alejandro Ramos <nahuelon@gmail.com> wrote: > > > Performace: CPU & MEM per Call. > > I will use SER to route call but I want to know if it is better to > > register SIP clients on SER or Asterisk? > > SER. You will get all the functionalities of a SIP Proxy there. > This link might be helpful: http://www.voip-info.org/wiki-Asterisk+at+large > > > > Regards, Girish > > _______________________________ > Do you Yahoo!? > Declare Yourself - Register online to vote today! > http://vote.yahoo.com >