<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Answers inline:<div><br><div><div>Le 23/08/2010 à 20:35, Ken Gillett a écrit :</div><br class="Apple-interchange-newline"><blockquote type="cite"><div>On 23 Aug 2010, at 15:13, David Ponzone wrote:<br><br><blockquote type="cite">There are 2 simple ways to do that:<br></blockquote><blockquote type="cite">-use only one account on your softphone, and use prefixes: easy to implement, but easy to make mistake for the users too<br></blockquote><blockquote type="cite">-use 6 accounts on your softphone as you used to do, with a 1-to-1 mapping with the external SIP accounts. You have to use a different FS context for each account in order to do that.<br></blockquote><br><br>That's what I am thinking would be the best solution, but could it be 'many to 1'? IOW, could I have several different extensions all mapped to the same external account, so incoming calls on that account go to the grouped extensions and if any of them ring out, it goes out via that same external account?<font class="Apple-style-span" color="#000000"><font class="Apple-style-span" color="#144FAE"><br></font></font></div></blockquote><div><br></div>I am not sure I get it but I think :)</div><div>Let's say you have 3 external accounts and you have 3 users (softphone).</div><div>You want all 3 users to receive calls for all external accounts and also to be able to dial out through them, don't you ?</div><div>Well, of course, you can:</div><div>Softphone1 would have 3 users: user11 in context ext1, user12 in contet ext2, user13 in context ext3.</div><div><div>Softphone2 would have 3 users: user21 in context ext1, user22 in contet ext2, user23 in context ext3.</div><div>etc.. for Softphone3.</div><div>ext1 is a context where calls going out will use external account 1.</div><div><div>ext2 is a context where calls going out will use external account 2.</div></div><div>etc..</div><div><br></div><div>For incoming calls, it's going to depend on what information allows you do distinguish the calls.</div><div>If it's a DID, it's easy.</div><div>All incoming calls will hit the public dialplan.</div><div>You just have to add extensions (FreeSWITCH extensions, so rules) so that if the call is for DID1, then user11, user21 and user31 will be called.</div><div>A such bridge command allows you to dial them all simultaneously or sequentially.</div><div><br></div></div><div><blockquote type="cite"><div>Here's another related question. Is it possible to register more than one SIP client to a single extension, or would you just use a different extension for each and set them up in a (hunt) group?<br><br></div></blockquote><div><br></div><div>Yes you can!</div><div>You have to enable that in the SIP profile with:</div><div><param name="multiple-registrations" value="true"/></div><div>Value can also be "contact", but I don't know the difference.</div><div><br></div><div>For outbound, it's the same than previously, except you only need 3 accounts (one per external account).</div><div><br></div><div>For inbound calls, you will be able to ring all devices at the same time in an easier way with:</div><div><span class="Apple-style-span" style="font-family: monospace; font-size: 13px; line-height: 16px; white-space: pre; "><br></span></div><div><span class="Apple-style-span" style="font-family: monospace; font-size: 13px; line-height: 16px; white-space: pre; "><action application="bridge" data="${sofia_contact($${sip_profile}/${dialed_ext}@$${domain})}"/></span></div><div><span class="Apple-style-span" style="font-family: monospace; font-size: 13px; line-height: 16px; white-space: pre; "><br></span></div><div><br></div><div>Was this helpful ?</div><div><br></div><div><br></div><blockquote type="cite"><div><blockquote type="cite">If you find a nice softphone with programmable keys, you can even assign the keys so you have a line key per account. That would be the easier to use I think.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">David Ponzone Direction Technique<br></blockquote><blockquote type="cite">email: <a href="mailto:david.ponzone@ipeva.fr">david.ponzone@ipeva.fr</a><br></blockquote><blockquote type="cite">tel: 01 74 03 18 97<br></blockquote><blockquote type="cite">gsm: 06 66 98 76 34<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Service Client IPeva<br></blockquote><blockquote type="cite">tel: 0811 46 26 26<br></blockquote><blockquote type="cite">www.ipeva.fr - <a href="http://www.ipeva-studio.com">www.ipeva-studio.com</a><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite">Le 23/08/2010 à 15:26, Ken Gillett a écrit :<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"><blockquote type="cite">Let's say there are 6 SIP accounts to which FreeSwitch is 'registered' and it can make calls using any of them. But which one is used? One of the local extensions (let's say a softphone) needs to receive calls from all of these accounts. This is simple enough and the recipient should see the incoming call information i.e. which account the call is to. But when that extension makes an outgoing call, how can it specify which is the SIP account used to by FS to make the call?<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">This can be very important when each SIP account represents a different company/business. Although one person is dealing with all those businesses, when an outgoing call is made it is imperative that the correct SIP account is used to make that call so that the recipient is correctly informed who is making the call.<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">Currently (no PBX), my softphone registers to each of these 6 accounts and I can choose which account to use to make a call. But if I am registered to FS as a single extension, how can I tell FS which account to use when I place an outgoing call? Is there any way to do this without having to use Dial plans?<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite">On 23 Aug 2010, at 11:46, David Ponzone wrote:<br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">Ken,<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">I am not really sure to understand your issue/question.<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">Can you describe exactly the equipements involved and what you want to do ?<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">Is FS used as a PBX or a a provider softswitch to terminate the trunk coming form the PBX ?<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">Some various information that could help you in the meantime:<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">-most softphones can have several SIP accounts, but you should check that they can register all of them at the same time<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">-if your objective is to have FS sending calls to a specific external VoIP account when it receives a call from a specific internal account, like this:<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">phone1-----> FS-------> Provider SIP Account 1<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">phone2----->FS--------> Provider SIP Account 2<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">you would need to split the outgoing calls one way or another: you could do that based on the caller-id, or you may put your internal accounts in different contexts, so they use different dialplans.<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">There are probably other ways, like using a prefix, but this one is probably a burden for the user and a security issue possibly.<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">A such configuration is really some sort of SBC, when you want to avoid your SIP devices to connect to the accounts provided by your carrier directly, because you are concerned with security or because you want to keep control on the calls to provide more services to your users.<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">Le 23/08/2010 à 09:57, Ken Gillett a écrit :<br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">If one wishes to have use of several VOIP 'lines', but with no PBX, you need to register all those SIP accounts with the client (softphone etc). You should then be informed which account is receiving a call and can pick a particular account from which to make calls. Once a PBX is in use, you can register the client as a single extension of the PBX and direct calls as appropriate to that extension - I assume with the correct caller ID and incoming account information passed to the recipient so they know as much as in the 'no PBX' configuration.<br></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><blockquote type="cite"><blockquote type="cite">But what about outgoing calls. In this scenario, registered as a single extension, how would it be possible to pick the outgoing 'line' (i.e. account) to use? Would it have to be done by dialling a prefix or is there another way? Is it client dependent?<br></blockquote></blockquote></blockquote></blockquote><blockquote type="cite"><blockquote type="cite"><br></blockquote></blockquote><br><br>Ken G i l l e t t<br><br>_/_/_/_/_/_/_/_/<br><br><br><br><br>_______________________________________________<br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br></div></blockquote></div><br></div></body></html>