Hi, First of all apologies because this isn't strictly a purely asterisk question. I am quite new to asterisk and actually to voip/telephony as a whole. I currently have sip calls working through asterisk. The asterisk server is behind a linksys router. I would now like to connect calls to the pstn. I have researched into several ways to do this but because I am not very knowledgeable about telephony I am now quite confused. This is what i understand so far. If this is incorrect or if anybody has any ideas as to I could implememnt this in a better/more scalable fashion, I would really appreciate it. I could put a fxo card in my asterisk server and connect this to a telephone line. This would enable sip to pstn calls but only one call at a time. To connect analog phones from the inside network(i.e. the asterisk network) going out I would need an fxs card. Now the problem with the above scenario is that only one call would be allowed at a time. I know I could get an fxo card with a few ports but that would still only allow a few calls. To implement a network where several calls are possible then do I need a pbx with a PRI interface?? Also where does all the digium cards come in all this??Where do they fit in?? I would be extremely grateful if somebody could shed some light on my currently very hazy understanding of voip telephony with asterisk Thanks again, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
Ashling O'Driscoll wrote:>Hi, > >First of all apologies because this isn't strictly a purely asterisk >question. > >I am quite new to asterisk and actually to voip/telephony as a whole. >I currently have sip calls working through asterisk. The asterisk >server is behind a linksys router. I would now like to connect calls >to the pstn. I have researched into several ways to do this but >because I am not very knowledgeable about telephony I am now quite >confused. This is what i understand so far. If this is incorrect or >if anybody has any ideas as to I could implememnt this in a >better/more scalable fashion, I would really appreciate it. > >I could put a fxo card in my asterisk server and connect this to a >telephone line. This would enable sip to pstn calls but only one call >at a time. To connect analog phones from the inside network(i.e. the >asterisk network) going out I would need an fxs card. > >Now the problem with the above scenario is that only one call would >be allowed at a time. I know I could get an fxo card with a few ports >but that would still only allow a few calls. To implement a network >where several calls are possible then do I need a pbx with a PRI >interface?? Also where does all the digium cards come in all >this??Where do they fit in?? > >I would be extremely grateful if somebody could shed some light on my >currently very hazy understanding of voip telephony with asterisk > >Thanks again, >Aisling. > > >-------------------Legal Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users > >__________ NOD32 1.931 (20041123) Information __________ > >This message was checked by NOD32 antivirus system. >nod32.com > > > > >Hi, I'm pretty new to asterisk myself and we haven't gotten the system running yet, but this is what I found out so far about ISDN and PRI (I'm really new to phone technics, so I hope I'm not telling rubbish here): - There are ISDN Cards that support BRI-interfaces (Basic Rate ISDN) (2 simultanous calls per line) with up to 4 lines. e.g. AVM C4 - From Digium and other manufactures you can find 1-4 line PRI (Primary Rate ISDN) cards (each line supporting 24/30 simultanous calls, depending whether you are in europe or america) e.g. TE405P I think the prices for those two types are about the same (execpt that the telecom provider will charge a lot mor for the PRI-Connection). Have a look at Wikipedia for some basic information. philipp
In simple terms ISDN is a digital interface to the PSTN as opposed to the analogue RJ11 phone connectors your used to at home (POTS - Plain Old Telephony System). ISDN lines (Integrated Services Digital Network) typically comes in two configurations BRI (Basic Rate has 2B+D channels ie. 2 speech 1 data ) and PRI (Primary Rate , (Europe)30B+D - 30 speech + 1 data). Because ISDN is digital the interface it's more advanced and supports a much wider set of functions and services than POTS. In Ireland Eircom market ISDN as 'hi-speed', you may be familiar with this (this is not ADSL, your not always online). The main suppliers of this type of service would be Eircom, Esat and COLT. Line rental for PRI will be in the order of ?3K/month plus a setup fee (yes - that puts the cost of the card in perspective). To conect your * box to PSTN with BRI/PRI interface you'll need one of digium's cards or an equivalent CAPI card. Br /Kev/ ----- Original Message ----- From: "Ashling O'Driscoll" <ashling.odriscoll@cit.ie> To: <asterisk-users@lists.digium.com> Sent: Wednesday, November 24, 2004 4:17 PM Subject: [Asterisk-Users] asterisk and pstn Hi, First of all apologies because this isn't strictly a purely asterisk question. I am quite new to asterisk and actually to voip/telephony as a whole. I currently have sip calls working through asterisk. The asterisk server is behind a linksys router. I would now like to connect calls to the pstn. I have researched into several ways to do this but because I am not very knowledgeable about telephony I am now quite confused. This is what i understand so far. If this is incorrect or if anybody has any ideas as to I could implememnt this in a better/more scalable fashion, I would really appreciate it. I could put a fxo card in my asterisk server and connect this to a telephone line. This would enable sip to pstn calls but only one call at a time. To connect analog phones from the inside network(i.e. the asterisk network) going out I would need an fxs card. Now the problem with the above scenario is that only one call would be allowed at a time. I know I could get an fxo card with a few ports but that would still only allow a few calls. To implement a network where several calls are possible then do I need a pbx with a PRI interface?? Also where does all the digium cards come in all this??Where do they fit in?? I would be extremely grateful if somebody could shed some light on my currently very hazy understanding of voip telephony with asterisk Thanks again, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users
Thank you very much for the reply. That has made things a good bit clearer. Am I correct in my current understanding: So basically if I want to support approx 100 calls, I would have to purchase a digium PRI card and then pay eircom (or whoever my service provider is) approx 3000 a year for the PRI ISDN connection?? My other option is to have a BRI connection which could support approx 8 calls ("calls perline, 4 line card) and pay my service provider alot less for a bri connection. I could also use an fxo card in * connected to my hpine line but that would only support one call at a time. Are these the only main implementation options? Has anyone come across the Skype Voip gateway (VTA1000) or where does that fit in the scheme of things?.....are there new options emerging?? Sorry if these questions seem a bit dense but I am doing this as part of a research project and since I am a student who has has no experience working in industry I am clueless when it comes to how such a service would be implemented in practice and the cost of it. I am appreciative of any knowledge passed on from others who have practical know how. Thanks again, Aisling. ---- Original Message ---- From: kevin.brennan@redsquared.com To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] asterisk and pstn Date: Wed, 24 Nov 2004 21:11:02 -0000> > >> In simple terms ISDN is a digital interface to the PSTN as opposed >to the >> analogue RJ11 phone connectors your used to at home (POTS - Plain >Old >> Telephony System). ISDN lines (Integrated Services Digital >Network) >> typically comes in two configurations BRI (Basic Rate has 2B+D >channels >ie. >> 2 speech 1 data ) and PRI (Primary Rate , (Europe)30B+D - 30 speech >+ 1 >> data). Because ISDN is digital the interface it's more advanced and >supports >> a much wider set of functions and services than POTS. In Ireland >Eircom >> market ISDN as 'hi-speed', you may be familiar with this (this is >not >ADSL, >> your not always online). The main suppliers of this type of service >would >be >> Eircom, Esat and COLT. Line rental for PRI will be in the order of >?3K/month > >Sorry that should read ?3K/annum > >> plus a setup fee (yes - that puts the cost of the card in >perspective). >> >> To conect your * box to PSTN with BRI/PRI interface you'll need one >of >> digium's cards or an equivalent CAPI card. >> Br /Kev/ >> >> ----- Original Message ----- >> From: "Ashling O'Driscoll" <ashling.odriscoll@cit.ie> >> To: <asterisk-users@lists.digium.com> >> Sent: Wednesday, November 24, 2004 4:17 PM >> Subject: [Asterisk-Users] asterisk and pstn >> >> >> Hi, >> >> First of all apologies because this isn't strictly a purely >asterisk >> question. >> >> I am quite new to asterisk and actually to voip/telephony as a >whole. >> I currently have sip calls working through asterisk. The asterisk >> server is behind a linksys router. I would now like to connect >calls >> to the pstn. I have researched into several ways to do this but >> because I am not very knowledgeable about telephony I am now quite >> confused. This is what i understand so far. If this is incorrect or >> if anybody has any ideas as to I could implememnt this in a >> better/more scalable fashion, I would really appreciate it. >> >> I could put a fxo card in my asterisk server and connect this to a >> telephone line. This would enable sip to pstn calls but only one >call >> at a time. To connect analog phones from the inside network(i.e. >the >> asterisk network) going out I would need an fxs card. >> >> Now the problem with the above scenario is that only one call would >> be allowed at a time. I know I could get an fxo card with a few >ports >> but that would still only allow a few calls. To implement a network >> where several calls are possible then do I need a pbx with a PRI >> interface?? Also where does all the digium cards come in all >> this??Where do they fit in?? >> >> I would be extremely grateful if somebody could shed some light on >my >> currently very hazy understanding of voip telephony with asterisk >> >> Thanks again, >> Aisling. >> >> >> -------------------Legal >Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >only >for >> the person to whom it is addressed. Its contents may be protected >by legal >> and/or professional privilege. Should it be received by you in >error >please >> contact the sender at the above quoted email address. Any >unauthorised >form >> of reproduction of this message is strictly prohibited. The >Institute does >> not guarantee the security of any information electronically >transmitted >and >> is not liable if the information contained in this communication is >not a >> proper and complete record of the message as transmitted by the >sender nor >> for any delay in its receipt. >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> lists.digium.com/mailman/listinfo/asterisk-users >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> lists.digium.com/mailman/listinfo/asterisk-users >> > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users > >-------------------Legal >Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >-------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.
Again, thanks for the pointers. Much appreciated, Aisling. ---- Original Message ---- From: kevin.brennan@redsquared.com To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] asterisk and pstn Date: Thu, 25 Nov 2004 15:10:43 -0000>If this is to gain knowledge a good source of background information >is the >IP telephony cookbook >informatik.uni-bremen.de/~prelle/terena, >you should find some answers there. > >One solution you did not mention is the use of a 3rd party >VOIP-PSTN/PLMN >gateway - ie. you connect using H.323/SIP/IAX/whatever and they have >the >PSTN/PLMN interface hardware. > >Br /Kev/ > >----- Original Message ----- >From: "Ashling O'Driscoll" <ashling.odriscoll@cit.ie> >To: <asterisk-users@lists.digium.com> >Sent: Thursday, November 25, 2004 11:13 AM >Subject: Re: [Asterisk-Users] asterisk and pstn > > >Thank you very much for the reply. That has made things a good bit >clearer. Am I correct in my current understanding: > >So basically if I want to support approx 100 calls, I would have to >purchase a digium PRI card and then pay eircom (or whoever my service >provider is) approx 3000 a year for the PRI ISDN connection?? > >My other option is to have a BRI connection which could support >approx 8 calls ("calls perline, 4 line card) and pay my service >provider alot less for a bri connection. > >I could also use an fxo card in * connected to my hpine line but that >would only support one call at a time. > >Are these the only main implementation options? Has anyone come >across the Skype Voip gateway (VTA1000) or where does that fit in the >scheme of things?.....are there new options emerging?? > >Sorry if these questions seem a bit dense but I am doing this as part >of a research project and since I am a student who has has no >experience working in industry I am clueless when it comes to how >such a service would be implemented in practice and the cost of it. I >am appreciative of any knowledge passed on from others who have >practical know how. > >Thanks again, >Aisling. > > >---- Original Message ---- >From: kevin.brennan@redsquared.com >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] asterisk and pstn >Date: Wed, 24 Nov 2004 21:11:02 -0000 > >> >> >>> In simple terms ISDN is a digital interface to the PSTN as opposed >>to the >>> analogue RJ11 phone connectors your used to at home (POTS - Plain >>Old >>> Telephony System). ISDN lines (Integrated Services Digital >>Network) >>> typically comes in two configurations BRI (Basic Rate has 2B+D >>channels >>ie. >>> 2 speech 1 data ) and PRI (Primary Rate , (Europe)30B+D - 30 >speech >>+ 1 >>> data). Because ISDN is digital the interface it's more advanced >and >>supports >>> a much wider set of functions and services than POTS. In Ireland >>Eircom >>> market ISDN as 'hi-speed', you may be familiar with this (this is >>not >>ADSL, >>> your not always online). The main suppliers of this type of >service >>would >>be >>> Eircom, Esat and COLT. Line rental for PRI will be in the order of >>?3K/month >> >>Sorry that should read ?3K/annum >> >>> plus a setup fee (yes - that puts the cost of the card in >>perspective). >>> >>> To conect your * box to PSTN with BRI/PRI interface you'll need >one >>of >>> digium's cards or an equivalent CAPI card. >>> Br /Kev/ >>> >>> ----- Original Message ----- >>> From: "Ashling O'Driscoll" <ashling.odriscoll@cit.ie> >>> To: <asterisk-users@lists.digium.com> >>> Sent: Wednesday, November 24, 2004 4:17 PM >>> Subject: [Asterisk-Users] asterisk and pstn >>> >>> >>> Hi, >>> >>> First of all apologies because this isn't strictly a purely >>asterisk >>> question. >>> >>> I am quite new to asterisk and actually to voip/telephony as a >>whole. >>> I currently have sip calls working through asterisk. The asterisk >>> server is behind a linksys router. I would now like to connect >>calls >>> to the pstn. I have researched into several ways to do this but >>> because I am not very knowledgeable about telephony I am now quite >>> confused. This is what i understand so far. If this is incorrect >or >>> if anybody has any ideas as to I could implememnt this in a >>> better/more scalable fashion, I would really appreciate it. >>> >>> I could put a fxo card in my asterisk server and connect this to a >>> telephone line. This would enable sip to pstn calls but only one >>call >>> at a time. To connect analog phones from the inside network(i.e. >>the >>> asterisk network) going out I would need an fxs card. >>> >>> Now the problem with the above scenario is that only one call >would >>> be allowed at a time. I know I could get an fxo card with a few >>ports >>> but that would still only allow a few calls. To implement a >network >>> where several calls are possible then do I need a pbx with a PRI >>> interface?? Also where does all the digium cards come in all >>> this??Where do they fit in?? >>> >>> I would be extremely grateful if somebody could shed some light on >>my >>> currently very hazy understanding of voip telephony with asterisk >>> >>> Thanks again, >>> Aisling. >>> >>> >>> -------------------Legal >>Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and >intended >>only >>for >>> the person to whom it is addressed. Its contents may be protected >>by legal >>> and/or professional privilege. Should it be received by you in >>error >>please >>> contact the sender at the above quoted email address. Any >>unauthorised >>form >>> of reproduction of this message is strictly prohibited. The >>Institute does >>> not guarantee the security of any information electronically >>transmitted >>and >>> is not liable if the information contained in this communication >is >>not a >>> proper and complete record of the message as transmitted by the >>sender nor >>> for any delay in its receipt. >>> >>> _______________________________________________ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> lists.digium.com/mailman/listinfo/asterisk-users >>> >>> _______________________________________________ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> lists.digium.com/mailman/listinfo/asterisk-users >>> >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> lists.digium.com/mailman/listinfo/asterisk-users >> >>-------------------Legal >>Disclaimer--------------------------------------- >> >>The above electronic mail transmission is confidential and intended >>only for the person to whom it is addressed. Its contents may be >>protected by legal and/or professional privilege. Should it be >>received by you in error please contact the sender at the above >>quoted email address. Any unauthorised form of reproduction of this >>message is strictly prohibited. The Institute does not guarantee the >>security of any information electronically transmitted and is not >>liable if the information contained in this communication is not a >>proper and complete record of the message as transmitted by the >>sender nor for any delay in its receipt. >> > > > >-------------------Legal >Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended >only for >the person to whom it is addressed. Its contents may be protected by >legal >and/or professional privilege. Should it be received by you in error >please >contact the sender at the above quoted email address. Any >unauthorised form >of reproduction of this message is strictly prohibited. The Institute >does >not guarantee the security of any information electronically >transmitted and >is not liable if the information contained in this communication is >not a >proper and complete record of the message as transmitted by the >sender nor >for any delay in its receipt. > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users > >-------------------Legal >Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >-------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.