Here is my situation: I have an office with around 10 users. Inbound calls will come in via 4 PSTN lines. Outbound calls will be routed across a maximum of 10 SIP trunks. How can I set up a "group" of outbound trunks which will rotate use dependant on how many outbound calls need to be made. There will be no discrimination or routes based on outbound calling, like a certain trunk for international calls, another for local calls, etc... Only a group of 10 SIP trunks to be rotated for all outbound calls. For example: Customer Support person 1 makes an outbound call on trunk1 (selected randomly by asterisk). Tech support person 1 needs to make an outbound call but for some reason is getting routed to trunk1 instead of to the next available open SIP trunk. Can anyone offer any suggestions, links, websites, or conf files that I could refer to in order to make all of this work. Thanks in advance.
On Tue, 15 Nov 2005, Pikoro wrote:> There will be no discrimination or routes based on outbound calling, > like a certain trunk for international calls, another for local calls, > etc... Only a group of 10 SIP trunks to be rotated for all outbound calls.Can you explain what you mean by a "SIP trunk"? SIP just has addresses - sometimes slightly hidden away in sip.conf behind a SIP peer. So if you Dial(SIP/remotehost/number), a SIP invite is sent to the host IP address defined in the SIP peer in sip.conf. If you Dial(SIP/number@hostname) then the invite is sent to the host "hostname". Normally it makes no difference to either side how many other calls may already by in progress between the two sides. Steve
> > > On Tue, 15 Nov 2005, Pikoro wrote: > > > There will be no discrimination or routes based on outbound calling, > > like a certain trunk for international calls, another for local calls, > > etc... Only a group of 10 SIP trunks to be rotated for all outboundcalls.> > > Can you explain what you mean by a "SIP trunk"? >I took it to mean different accounts or providers.
---- Original Message ---- From: <steve@daviesfam.org> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Tuesday, November 15, 2005 2:27 PM Subject: Re: [Asterisk-Users] Multiple Outbound SIP Trunks> On Tue, 15 Nov 2005, Pikoro wrote: > >> There will be no discrimination or routes based on outbound calling, >> like a certain trunk for international calls, another for local >> calls, etc... Only a group of 10 SIP trunks to be rotated for all >> outbound calls. > > > Can you explain what you mean by a "SIP trunk"? > > SIP just has addresses - sometimes slightly hidden away in sip.conf > behind a SIP peer. So if you Dial(SIP/remotehost/number), a SIP > invite is sent to the host IP address defined in the SIP peer in > sip.conf. If you Dial(SIP/number@hostname) then the invite is sent > to the host "hostname". Normally it makes no difference to either > side how many other calls may already by in progress between the two > sides.Some providers allow only one outgoing call at a time. Leif
My question is the same/similar. I want to test a VOIP provider. I only want one LD call to that provider at a time so that I can check with the users on the quality, etc. I want the first LD call to go to the VOIP provider, if one session to that provider is in use, I want to use ZAP for any additional LD calls. Preferably I want to be able to change it from 1 session in use to 2, then 3 etc. until I reach a level of quality vs. savings. If I switch over completely, then the day that we make 15 simultaneous LD calls will ruin our quality. Zaptel seems to have this functionality built in, where in a group of 5 trunks, asterisk will use the next unused trunk. But SIP and IAX do not seem to get tagged as in use as far as I can see. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. --- - --- - - - - - - - -- - - - --- - ------ - - --- - - -- - - - -- - - - "Pikoro" <webmaster@psphacks.net> wrote in message news:4379DF67.2090303@psphacks.net...> Here is my situation: > > I have an office with around 10 users. Inbound calls will come in via 4 > PSTN lines. Outbound calls will be routed across a maximum of 10 SIP > trunks. > > How can I set up a "group" of outbound trunks which will rotate use > dependant on how many outbound calls need to be made. > > There will be no discrimination or routes based on outbound calling, like > a certain trunk for international calls, another for local calls, etc... > Only a group of 10 SIP trunks to be rotated for all outbound calls. > > For example: > > Customer Support person 1 makes an outbound call on trunk1 (selected > randomly by asterisk). Tech support person 1 needs to make an outbound > call but for some reason is getting routed to trunk1 instead of to the > next available open SIP trunk. > > Can anyone offer any suggestions, links, websites, or conf files that I > could refer to in order to make all of this work. > > Thanks in advance. > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 >
By "trunk" I mean each trunk is a different account on the same SIP provider. Yes, they only allow one call per account. We are an internet provider so I can obtain as many trunks(accounts) as I need. Cheers asterisk wrote:> > >>On Tue, 15 Nov 2005, Pikoro wrote: >> >> >> >>>There will be no discrimination or routes based on outbound calling, >>>like a certain trunk for international calls, another for local calls, >>>etc... Only a group of 10 SIP trunks to be rotated for all outbound >>> >>> >calls. > > >>Can you explain what you mean by a "SIP trunk"? >> >> >> > >I took it to mean different accounts or providers. > >_______________________________________________ >--Bandwidth and Colocation sponsored by Easynews.com -- > >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 -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051115/5e4b210b/attachment.htm
Hi all, In case you have a number of trunks there is a software named astbill (www.astbill.com) in which you can configure the trunks and decide their costs and it will automatically choose the most suitable trunk. Thx MAG Pikoro wrote:> By "trunk" I mean each trunk is a different account on the same SIP > provider. Yes, they only allow one call per account. We are an > internet provider so I can obtain as many trunks(accounts) as I need. > > Cheers > > > asterisk wrote: > >> >> >> > On Tue, 15 Nov 2005, Pikoro wrote: >> > >> > >> >> There will be no discrimination or routes based on outbound >> >> calling, >> >> like a certain trunk for international calls, another for local >> >> calls, >> >> etc... Only a group of 10 SIP trunks to be rotated for all >> >> outbound >> >> >> calls. >> >> > Can you explain what you mean by a "SIP trunk"? >> > >> > >> I took it to mean different accounts or providers. >> >> _______________________________________________ >> --Bandwidth and Colocation sponsored by Easynews.com -- >> >> 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 >> >> >> > ---------------------------------------------------------------- > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 >-- Thx MAG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051116/15026189/attachment.htm
> How can I set up a "group" of outbound trunks which will rotate use > dependant on how many outbound calls need to be made.You could do this by writing a simple (but laboriously long) macro to try the accounts in order, dialing via the first available one. There would be a dial() command followed by a test of DIALSTATUS with jumps to the next account. You could also break the users up into two or three groups affected to a smaller number of accounts, the number being equal to the number of users in the group. The most obvious solution on the same provider would be to just have one account per phone and dial out on that.
I have 3 sip trunks registered with an outside provider, however asterisk always seems to work when going out the third trunk. Any way to round-robin this so that we can make more than one outbound call at a time? Thanks in advance, Aaron