similar to: Polycom Attended Transfer

I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -------------- next part

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

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

On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

I know this is not a Polycom support forum, but I also know there are a lot of you with a great deal of Polycom experience. Is there anyway to remove the "Attended Transfer" but keep the "Blind transfer"? Or better yet, just swap the two soft buttons locations? I know you can remap the "Hard" buttons, but what about the soft buttons? The reason I need this is my

Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the

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

Hello! I have a Polycom phone and sometimes I need to transfer calls without picking them up to local extensions. Polycom has a transfer button which sends SIP 302 packet to asterisk. Another local extension, receiving the call, sees not the original number but the local number that was transferring the call. I would like that the original number is shown. I am stuck at this point. I see messages

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!

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

I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any

Hi All, Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Potentially using a mix of phones would create confusion in a user base, any ideas on attended transfer or how to achieve this / mods to dial plan etc would be greatly appreciated. I have been on an almost vertical learning curve with Asterisk and Linux for 6 months this is just

Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in & we answer & transfer, everything works fine. But if we call out to a customer & then transfer to another internal extension, that

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

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

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

On Tuesday 25 June 2019 at 16:49:23, Doug Lytle wrote: > We have Polycom phones (I'm using a VVX601, the destination is a VVX301). > We're also on Asterisk 13. > > I forwarded my call to the VVX301 and then dialed my phones DID. Surely that is "call forwarding", which is quite different from either a blind or attended transfer? A transfer involves a call coming

Just out of curiosity, What version of CVS and Polycom SIP software are you running happily? Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0? I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with poor results. Transferring did not work as expected. Using the # key to do blind transfers after a call was on hold did not work. Just curious. Thanks, - Brent

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