similar to: Polycom SIP 501 Transfer Question

Howdy - > The transer button on the polycom phone does not seem to transfer/park the > call properly. I have to use the # -> 700 to park the call. If I recall, using the Polycom transfer, you have to make sure it is done as a blind transfer. The Polycom attended transfer (default) option does not work. > Furthermore the # -> 700 only works on incomming calls. If I

When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

Guys, this is a weird question but has anybody disabled the blind button that appears on polycoms or know if you can disable the use of blind transfers on polycoms to make any transfer attended? Thx!

I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my

Hi Andrew - > On Monday 13 March 2006 10:20, Noah Miller wrote: >>> The transer button on the polycom phone does not seem to transfer/park >>> the call properly. I have to use the # -> 700 to park the call. >> >> If I recall, using the Polycom transfer, you have to make sure it is done >> as a blind transfer. The Polycom attended transfer

I am doing some experimenting with Asterisk 1.0.9 and Polycom IP501's. I have the extensions setup, and everything is working well up to this point. Now, I want to setup my system so that a user at an extension can start a recording on demand. I have tried various Google searches, but am coming up empty. Could someone point me in the right direction, or have some sample Polycom/Asterisk

Hi, I'm using Asterisk@home and am having trouble using the conference bridge that comes built in. We're using Polycom phones. When we transfer the first person into the conference room (e.g. 8101) , they get into the room fine. When we try to transfer a second person into the conference room, they get dropped as soon as we finish the transfer. This is using Polycom SoundPoint 301

I cannot seem to get any transfers to work at all. The console show I have #1 amd #2 set up for Blind and Attended Transfer, but when I hit these buttons on my handset nothing happens (other than I hear the dtmf tones on the other end of the line). roo*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8

Hi, In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind Transfer when transferer hangs up before callee answers : - in Blind Transfer, caller (ie transferee) is hearing Ringing tone when callee's phone is ringing - in Attended Transfer, caller (ie transferee) is hearing Music On Hold when callee's phone is ringing - in Attended Transfer, if callee don't answer

Hi gang, When I try to park a call using blind-transfer (#1), the caller hears the lot instead of the transferring party. Attended transfer and blind transfer from the phone buttons (Polycom 501) work fine, so this isn't a showstopper, just a "WHY??". Thanks for you attention. Danny Nicholas -------------- next part -------------- An HTML attachment was

Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

Hi - > I am not sure what I did but blind transfers do not work. The Polycom does > not allow me to dial the extension of the person I want to transfer to after > I hit: > > transfer -> blind I would strongly suggest getting the latest firmware, and using the sample configuration files with that firmware to set up your phone. This SHOULD work. If it still does not work

This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not

PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to

OK I have just gone live with asterisk in a new office with approx 40 Polycom 501 handsets. I have a few questions: 1) Call Parking: I am able to park calls using the standard Asterisk call parking system (transfer to ext *70 etc...) I would like to make this a little easier for my users The Polycom 501s w/ SIP 1.5.2 seem to support some type of standard call parking, however I don't

There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

Hi, we use Polycom SP IP 501 phones. We use the standard key/softkey configuration to do attended transfers. The only thing we miss is the CID info of the original caller after the call is transfered. This behaviour is different from the blind/direct transfer. With blind transfer method the original CID info is displayed. We already opened a call (in 2006) with Polycom JIRA. This is what they

I'm using asterisk blind/attended transfer feature on a queue (also tried with sip phones feature), and both type of transfers work fine. The problem is that attended trasfers doesn't get logged to queue_log, but blind transfers are logged just fine. Anyone knows if this is the correct behavior? -- Regards, Miguel Paolino -------------- next part -------------- An HTML attachment was

*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * *0 Dial option is tTwWr I tried to call from 601 to 615 601 keys in *0

Hey, all. I've got an office set up with Asterisk, and forwarding's got a bit of a glitch: When they forward, they listen for the remote phone to ring, then hang up. If the remote phone doesn't connect, it goes to the original phone's VM. Is this Polycom's "fault," or Asterisk's? I've been reading up on blind/supervised forwards, and, honestly, have