I'm pretty sure only hardphones (Aastra, polycom) support this feature. At least I do not know of any softphones that do.
On Sat, Jul 24, 2010 at 5:21 AM, <unsero...@aol.com> wrote: > Hi, > > i just tried to use the CONNECTEDLINE() feature but it does not work, at > least with my softphones (zoiper, 3CX, Xlite) > > in sip.conf under general I have: > trustrpid = yes > sendrpid = rpid,pai > rpid_update = yes > > in extensions.conf I have: > exten => 2000,1,Set(CONNECTEDLINE(number,i)=98) > exten => 2000,n,Set(CONNECTEDLINE(name,i)=test) > exten => 2000,n,Set(CONNECTEDLINE(pres)=allowed) > exten => 2000,n,Dial(SIP/2000,20) > > It seems to be executed correctly > > -- Executing [2...@default:1] Set("SIP/1000-0000002e", > "CONNECTEDLINE(number,i)=98") in new stack > -- Executing [2...@default:2] Set("SIP/1000-0000002e", > "CONNECTEDLINE(name,i)=test") in new stack > -- Executing [2...@default:3] Set("SIP/1000-0000002e", > "CONNECTEDLINE(pres)=allowed") in new stack > -- Executing [2...@default:4] Dial("SIP/1000-0000002e", "SIP/2000,20") > in new stack > -- Called 2000 > -- SIP/2000-0000002f is ringing > -- SIP/2000-0000002f answered SIP/1000-0000002e > -- Remotely bridging SIP/1000-0000002e and SIP/2000-0000002f > == Spawn extension (default, 2000, 4) exited non-zero on > 'SIP/1000-0000002e' > > > but neither the number is changed on the calling softphone nor the name is > displayed. > > Did anyone successfully test this, maybe with a hardphone? > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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