With all honor and respect you deserve, Do I need your permission to express my point of view on community forum ?<div><br></div><div>also it would be quiet helpful for us if you understand well the requirement of post <div>
<br>Regards</div><div><br><div>On Fri, Aug 20, 2010 at 1:34 PM, Sherwood McGowan <<a href="mailto:sherwood.mcgowan@gmail.com">sherwood.mcgowan@gmail.com</a>> wrote:<br>><br>> Paddy,<br>><br>> I believe I have a solution, let me sober a bit ;) and rum it through<br>
> (typo not intended but funny) my test server to doublecheck<br>><br>> Sent from my iPhone<br>><br>> On Aug 20, 2010, at 12:20 AM, "Paddy Grice" <<a href="mailto:paddy@wizaner.com">paddy@wizaner.com</a>> wrote:<br>
><br>> > Hi Sherwood<br>> ><br>> > I actually do want "dynamic" CLID as I tried to make clearer<br>> ><br>> >>> I don't know if this makes it any clearer -<br>> >>><br>
> >>> An internal call from Ext123 should send 123 as the CLID to SIP/<br>> >>> Ext400<br>> > but should<br>> >>> send 442071110123 to SIP/TheWorld but an external call from<br>> > 44123455667788 should<br>
> >>> send the received CLID 44123455667788 to both.<br>> ><br>> > So over the provider connection the CLID will be different for<br>> > different<br>> > calls. Setting the main office number in sip.conf is fine as a<br>
> > default but<br>> > as the code/dialplan needs to set cli for some calls I actually set<br>> > CLID for<br>> > all calls. This setting and onward transmission by provider works<br>> > fine.<br>
> ><br>> > So what I am trying to do is call 2 different sip endpoints AT THE<br>> > SAME TIME<br>> > presenting different AND VARIABLE CLIs. If Nasir's trick is not<br>> > recommended<br>
> > what is the best way to achieve this.<br>> ><br>> > As a newbie to Asterisk advise and best practice gained from user<br>> > experience<br>> > is always welcome.<br>> ><br>> > Paddy<br>
> ><br>> ><br>> ><br>> ><br>> > -----Original Message-----<br>> > From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>> > [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Sherwood<br>
> > McGowan<br>> > Sent: 20 August 2010 04:58<br>> > To: Asterisk Users Mailing List - Non-Commercial Discussion<br>> > Subject: Re: [asterisk-users] Calling Line Identity - any ideas<br>> ><br>
> > On Thu, Aug 19, 2010 at 7:46 PM, Nasir Iqbal<br>> > <<a href="mailto:nasir@ictinnovations.com">nasir@ictinnovations.com</a>><br>> > wrote:<br>> >> Hi,<br>> >>> there's still no conceivable reason<br>
> >> What can be? except performance! (as asterisk has to create one<br>> >> additional leg and bridge it) Which is very conceivable to those who<br>> >> are dealing with high load traffic.<br>> >> And what will be the option, if other outgoing call requires<br>
> >> different<br>> >> custom CLI while using the same trunk?<br>> >> Regards<br>> >> --<br>> >> Nasir Iqbal<br>> >><br>> >> ICT Innovations<br>> >> <a href="http://www.ictinnovations.com/">http://www.ictinnovations.com/</a><br>
> >><br>> >><br>> >> --<br>> >> _____________________________________________________________________<br>> >> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a><br>
> >> --<br>> >> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> >> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br>> >><br>
> >> asterisk-users mailing list<br>> >> To UNSUBSCRIBE or update options visit:<br>> >> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
> >><br>> ><br>> > First, the reason is, why use a BAD IDEA when there's perfectly good<br>> > solutions in front of the user.... There was no mention on this ONE<br>> > call<br>> > going outbound over the trunk needing a different CID...the request<br>
> > was as<br>> > follows:<br>> ><br>> > Client needs to call an INTERNAL extension, where the INTERNAL<br>> > CallerID will<br>> > be used, and at the SAME TIME, a call to an EXTERNAL number (which<br>
> > would<br>> > necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL<br>> > CallerID....<br>> ><br>> > Now, p-lease tell me how just configuring the damned trunk's<br>> > outbound CID is<br>
> > NOT more sensible, efficient, and just friggin' COMMON SENSE TO START<br>> > WITH...over using a Local channel call, which would require slightly<br>> > more<br>> > typing, and using something that I've almost NEVER found a good<br>
> > reason to<br>> > use, and if you'd care to search the damn archives, you'll see that<br>> > I was<br>> > pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk<br>> > and the<br>
> > RealTime addiiton (which was experimental)...<br>> ><br>> > For the love of whatever you find holy and good and true...don't<br>> > come at me<br>> > like that...I'm really not in the mood anymore...I put 3-4 solid<br>
> > years of<br>> > helpjng newbies figure out why shit didn't work, reporting REAL bugs<br>> > and<br>> > issues to thew developers and even assisting with some of the<br>> > fixes....I<br>
> > feel entitled (yes, I know that's an asshole thing to say) to a little<br>> > common respect....<br>> ><br>> ><br>> > Now...anyone for a pint? I'm off to vent some frustration with<br>
> > people who<br>> > jump on the WRONG bandwagon and try to take over....<br>> ><br>> > Sherwood Mother-F'in' McGowanb...<br>> > Telecommunications and Tattooing....<br>> > You konw anyone else who combines those two professions? I'd like to<br>
> > buy<br>> > that guy a drink!<br>> ><br>> ><br>> ><br>> > --<br>> > _____________________________________________________________________<br>> > -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>
> > New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> > <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br>> ><br>> > asterisk-users mailing list<br>
> > To UNSUBSCRIBE or update options visit:<br>> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>><br>> --<br>> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br>
><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br><br><br>--<br>Nasir Iqbal<br><br>ICT Innovations<br><a href="http://www.ictinnovations.com/">http://www.ictinnovations.com/</a><br><br></div></div></div>