I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets and 400 analog units. For the analog units I have quotes for 9 ADIT 600 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used neither. Which is the best choice? The price difference is not that great. I am looking at Citelinks 24 port Handset Gateway for the Nortel Digital units. (Any other suggestions would be appreciated). Also how many Asterisk servers would I need to handle 200 IP units in addition to the the above referenced legacy units? How do I size the server? Do I put voice mail on a different box? Your comments much appreciated. -------------- next part -------------- A non-text attachment was scrubbed... Name: carey.mould.vcf Type: text/x-vcard Size: 308 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050322/7ecc468b/carey.mould.vcf
On Tue, 22 Mar 2005 19:36:26 +0000, cmould <carey.mould@e2team.com> wrote:> I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets > and 400 analog units. For the analog units I have quotes for 9 ADIT 600 > 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used > neither. Which is the best choice? The price difference is not that > great. I am looking at Citelinks 24 port Handset Gateway for the Nortel > Digital units. (Any other suggestions would be appreciated).I'm actually trying to accomplish the same thing. 360 analog units, just hung up the phone with carrier access tech support, they where very helpful. plus this: http://lists.digium.com/pipermail/asterisk-users/2004-December/077099.html looks like I'm going with Adit. But instead of T1 from the Adit to * I plan on using CMG02 cards with the Adit 600, that gives me 9 Adit boxes, each one will have 5 FXS cards (5*8=40) and one CMG card, 9*40=360 FXS ports. That will make the Adit handle the bulk of the transcoding, and hence the CPU eat up.> Also how many Asterisk servers would I need to handle 200 IP units in > addition to the the above referenced legacy units? How do I size the > server? Do I put voice mail on a different box?This is only a problem if you will be doing lots of transcoding (Zap < -- > SIP/G729 < -- > G711), if however you will be staying strictly VOIP and no codec transcoding (thats why I'm going with the CMG cards above, although it has to convert from MGCP to SIP, it doesn't eat up as much as from G711 to G729, or Zap to SIP), then you should't have a problem using one Dual Xeon box. If you must use telco provided T1s, you can either use another Adit 600 with a CMG on it, and hand it off to asterisk that way, or you could have one asterisk box just for the handling of the T1s, however asterisk with 4 T1s using a Digium quad T1 card, might (this is from experience, some people do have and others don't) have some echo problems. The other solution would be to have the 200 IP units connected to one box, and the analog ones connected to the other, and then use IAX from box to box, but I'm not sure it is better. I for myself am thinking of going with Quad Xeon boxes, an overkill? maybe. But I've never seen anybody crying for getting a better system than they need. Putting VM on a different box I don't think will accomplish anything, maybe make it even worse, since you will need the phone connected asterisk to bridge the call and open a stream to the voicemail box, maybe I'm wrong, but this is what I think. Also don't forget to look at this: http://www.voip-info.org/wiki-Asterisk+dimensioning Hope this helps, what ever your decision please put it on the list so others know about it. I plan on putting my installation on the wiki when it is done and running (another 3-4 months).> Your comments much appreciated. > > > _______________________________________________ > 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 > > >
Sure u can. Buy nokia phone, buy nokia serial cable, use gnokii software to trigger script on sms received. Script would write sample.call file to asterisk queue directory.
I have such setup in testing. SER as SMS gateway and callback through Asterisk. W ----- Original Message ----- From: "Cristian T" <cristian@redux.com.mx> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Saturday, March 26, 2005 12:53 AM Subject: [Asterisk-Users] make a call based on SMS request> Hola > I have a costumer whit this idea: > > > I am looking for a solution that will make a call based on SMS request. > Can > you solve this problem with Asterisk? > > Let me know if you have the solution and what exactly it does. > > > This is posible??? > _______________________________________________ > 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 >
On 3/26/05 11:45 AM, "Wojciech Tryc" <wojtek@tryc.ca> wrote:> I have such setup in testing. SER as SMS gateway and callback through > Asterisk. > W > ----- Original Message ----- > From: "Cristian T" <cristian@redux.com.mx> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Saturday, March 26, 2005 12:53 AM > Subject: [Asterisk-Users] make a call based on SMS request > > >> Hola >> I have a costumer whit this idea: >> >> >> I am looking for a solution that will make a call based on SMS request. >> Can >> you solve this problem with Asterisk? >> >> Let me know if you have the solution and what exactly it does. >> >> >> This is posible??? >> _______________________________________________ >> 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
How are you connecting to the SMS gateway, are you calling an external script to send a http request or something, or is there a way of using SMPP Iqbal On 3/26/2005, "Wojciech Tryc" <wojtek@tryc.ca> wrote:>I have such setup in testing. SER as SMS gateway and callback through >Asterisk. >W >----- Original Message ----- >From: "Cristian T" <cristian@redux.com.mx> >To: "Asterisk Users Mailing List - Non-Commercial Discussion" ><asterisk-users@lists.digium.com> >Sent: Saturday, March 26, 2005 12:53 AM >Subject: [Asterisk-Users] make a call based on SMS request > > >> Hola >> I have a costumer whit this idea: >> >> >> I am looking for a solution that will make a call based on SMS request. >> Can >> you solve this problem with Asterisk? >> >> Let me know if you have the solution and what exactly it does. >> >> >> This is posible??? >> _______________________________________________ >> 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 > >
At this point (proof of concept) I am using a GSM phone connected over the serial cable. The only problem is that all incoming/outgoing SMS messages are using the same number assigned to the SIM card on this phone. W ----- Original Message ----- From: "Iqbal" <iqbal@gigo.co.uk> To: <asterisk-users@lists.digium.com> Sent: Saturday, March 26, 2005 5:00 PM Subject: Re: [Asterisk-Users] make a call based on SMS request How are you connecting to the SMS gateway, are you calling an external script to send a http request or something, or is there a way of using SMPP Iqbal On 3/26/2005, "Wojciech Tryc" <wojtek@tryc.ca> wrote:>I have such setup in testing. SER as SMS gateway and callback through >Asterisk. >W >----- Original Message ----- >From: "Cristian T" <cristian@redux.com.mx> >To: "Asterisk Users Mailing List - Non-Commercial Discussion" ><asterisk-users@lists.digium.com> >Sent: Saturday, March 26, 2005 12:53 AM >Subject: [Asterisk-Users] make a call based on SMS request > > >> Hola >> I have a costumer whit this idea: >> >> >> I am looking for a solution that will make a call based on SMS request. >> Can >> you solve this problem with Asterisk? >> >> Let me know if you have the solution and what exactly it does. >> >> >> This is posible??? >> _______________________________________________ >> 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
> Couple observations. > > Adit with CMG uses MGCP vs SIP. Not sure how extensive the * support is > for this. I believe Carrier Access is working on a SIP release but not > sure how complete it will be. CMG/CMG2 are nice cards. Also have a nice > license builtin for G729. If you use though you will need a matching > one for the * server. However there is also a call limit of 12, I > think, for the CMG and 24 for the CMG2 cards. Depending on application > you could easily exceed this. So take the 8 T1 capacity with a grain of > salt, you will not be able to have that may calls up at a time.I researched this before I posted, I was on the phone with Carrier Access Tech Support for a good hour about the limitations and capabilities of the CMG cards, the CMG02 will take up to 48 simultaneous calls, unless G729 is used (in which case you are right and only 24 can be used), since this is all on LAN there is no need for G729. When I asked them about SIP support, they told me that no plans as far as he knows for SIP support, but he told me don't quote me on it (I guess if I add this I'm allowed to quote him), when I asked him about configuring it with Asterisk he told me that he had many happy customers that configured it with asterisk, although he doesn't know asterisk, but that much he knows that ppl have gotten it to work with asterisk, and they were happy.> Also one of my favorite applications is connecting an Adit directly via > T1 to a Digium card. Then using another T1 port on the Digium card to > connect to PRi from PSTN. This is called a traditional TDM switch. No > IP in the patch, no headaches. Great for such things as FAX and ALARM > circuits which are very problematic with IP in the path.I agree on this one, but still doesn't justify using all TDM, you could take just one Adit 600 and split half FXO, and the other half FXS, just for such applications (the FXO for backup and 911).> It is not always about up front costs but about capabilities and > support costs down the line. >Right here again, thats why we are having this post.