Hi Michael,
> I meant that attended transfer doesn't work (at least for me) when I'm trying > to transfer call to a different device too. It works for any other type of IP phones, except ATA186. Tested on Cisco 7960 and X-Lite. > Let's say I dial from ata186 another h323 endpoint. Put it on hold. Then dial my > cell phone (ata -> asterisk chan_h323 -> h323/pstn gateway). Then if I hang > up on ata the call to my cell phone drops. This is a normal behavior, as both calls where originated from ATA. > What interesting is that asterisk > redial my cell on itself in a second or so and then I get connected to h323 > endpoint. Unfortunatelly it does it in less than 1 second and this is the reason that ATA cannot handle attended transfers. It consider this delay as flash and remain ini a busy state. > If before hanging up I press flash on ata186 to have 3way conference > call, it works fine - 3 phones get connected, but then if I hang up on ata then > 2 other parties don't stay connected but both get dropped. This is normal. See before. > In the latter case > asterisk doesn't redial either phone. > I think I've seen in the development maillist that asterisk doesn't support > attended call transfer yet (at least on voip channels). It would be nice > if someone of the gurus confirm (or better disprove ;-)) this. If the redial dis done in more than 1s, then it can work on ATA too. I think this is something very easy to be done by someone with greater experience in the Asterisk source. Best regards, Dan > > Michael > > On Monday 28 July 2003 01:00 pm, Dan wrote: > > Hi, > > > > It works, bot ONLY when I try to transfer the call to another type of phone, > > like X-Lite or Cisco 7960. > > If the destination is an ATA too, it does not work because hanging-up is > > considered as a closed call only after 1 second in ATA (if less than 1s, the > > it is a flash function), but the transfer function in Asterisk tries to > > recall the first extension in less than 1 second, so during this short > > period of time, ATA based phone is bussy and cannot accept calls, so the > > call is redirected to the voicemail. > > One way to make this attended transfer work with ATA too, is to enter a > > minimum delay of 1 second in th transer function, but I don't know how to do > > it. > > > > Look at the ATA186 specification for extended SIP functions, at the address: > > http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/ataadmn/ata88sip/supp.pdf > > or as HTML ast: > > http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administration_guide_chapter09186a0080150e5a.html > > > > It is stated that the attended transfer is done like that: > > > > Step 1 Press the flash button on the telephone handset to put the existing > > party on hold and get a dial tone. > > Step 2 Dial the telephone number to which the existing party is being > > transferred. > > Step 3 When the callee answers the phone, you may consult with the callee > > and then transfer the existing party by hanging up your telephone handset. > > > > It works for me on ATA if the final destination is not an ATA too. > > > > Best regards, > > Dan > > P.S. I'm interested in the attended transfer. The unattended one works > > perfect. > > > > > > ----- Original Message ----- > > From: "Michael Ulitskiy" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Monday, July 28, 2003 7:41 PM > > Subject: Re: [Asterisk-Users] Call transfer on ATA186 > > > > > > > On Monday 28 July 2003 12:24 pm, Dan wrote: > > > > Hi Iain, > > > > > > > > > The basic call transfer functions, set with the T and t options to the > > > > dial > > > > > application and triggered by pressing a # work fine for me. > > > > I have T and t options in dial application, but how can '#' be used for > > > > transfer. > > > > Escuse my ignorance... > > > > > > > > > Make sure that > > > > > you have set the DialPlan on the ATA 186 so as not to grab the # (ie > > look > > > > > for any ># character pairs and change the second character or remove > > it). > > > > Where to do that? In the extensions.conf file? > > > > > > > > Now I have used Flash key to put the other part on hold and then dial to > > the > > > > new extension and after this one answer, I close the phone. > > > > It works in that way only if the last party is anything else, but not > > > > another ATA186. > > > > > > Does it really work this way for you? I thought asterisk cannot bridge > > together > > > 2 channels if originating party hangs up. I mean if I press flash button > > to put > > > one party on hold, then dial another extension and then hang up the two > > other > > > extensions do not get connected but both calls get dropped. Only "blind > > transfer" > > > with # key works for me. > > > If it really works for you, would you mind to show your configuration? > > > Thanks. > > > > > > Michael > > > > > > > Thanks for your support, > > > > Dan > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
