similar to: Caller ID, Attended Transfers, Polycom

Greetings list, Have there been any further developments recently regarding presenting the original caller's caller ID to SIP devices after an attended transfer? I've googled around on the topic, but most of the threads I've found (some from this very list) are all dated back in mid-2006 and I wondered if there have been developments on the issue. To recap, the desired behaviour

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

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

Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and the From: should be the party doing the attended transfer. This seems like a bug. - Doug.

Hello list, I'm sending calls to a queue in the attended way, that is, *1.* the original call is put on hold, *2.* a second line is open to call the queue, *3.*after an agent is connected the original call is transfered to its final destination. 1. Zap/1-1 <--> SIP/agentA-tag1 2. SIP/agentA-tag2 <--> SIP/agentB-tag 3.

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

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:

Hi, I have been trying to enable attended transfer for callee. When the callee pressed *2, DTMF tone was heard by the caller. But when the caller pressed *2, attended transfer started. It's strange. I used two SIP phones. My Asterisk version is "Asterisk CVS-HEAD built by root@router on a i686 running Linux on 2005-06-27 06:07:18". In features.conf, I have: [featuremap]

Hi, I'm having problems with sip remote attended transfer using 2 asterisk boxes (same version, latest 1.4.X). Whenever I transfer from a call from box A to a call on box B, one call leg of the transferring phone is not disconnected (the one that is normally dropped by server side, phone disconnects the other one). The same situation works perfectly with local attended transfer. Is anyone

We have two Asterisk servers running CVS-HEAD (06/02/05 and 06/28/05). Most of our calls are either incoming or outgoing to external (PSTN or non-Asterisk) numbers, and only our internal users can initiate the transfer. Only half of the attended transfers work. It goes like this: 1)Extension 8123 calls number 19876543210 2)During the call, extension 8123 dials *2 to do an attended (non-blind)

Well I'm sure this issue has been bean up a few time since it's one of the only ones I can't find a real "simple" answer to. I'm trying to find away to do attended transfers through the manager interface, for a pc switchboard / Agent client solution, but so far coming up short. The action Originate is part of the solution, but what really I want is the phone being taken

Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

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'm trying to set up outboud caller ID on 3 polycom ip 500 phones. here is how i have them set up. each phone is using 2 lines, all registered to * phone 1 line 1 is registered using the DID, and the next 5 lines reg to 03152 03153 03154 03155 03156 from my understanding, the way the polycom configs are set, i have to put the 03152 in the reg.x.address field. and it seems that is where

Greetings, I'm wondering if it's possible to display line breaks with caller ID display. I have the Polycom ip500 phone, and what I am trying to accomplish is instead of the phone saying 'Incoming call from: name/number' i want it to appear on the phone like this Incoming Call from: Menu Context last in Name Number I tried using \n and \\n between the variables (${VAR} \\n

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

Hi, all. When I get calls on my SoundPoints, I only see the number -- is there a way to get the alpha portion of the CID, as well? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

I need attended transfers, but I do not have time to talk to another extension and see if they accept the transfer, my features.conf is: [general] parkext => 700 ; What ext. to dial to park parkpos => 701-720 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in parkingtime => 220 ; Number of

Recently I asked a question about possibly unwanted calls due to extended transfer rights after attended transfers using DTMF sequences (http://lists.digium.com/pipermail/asterisk-users/2013-September/280536.html). Obviously, transferring with SIP INVITEs (hold + transfer keys) is not immediately affected by the this, but it is not always possible to enforce this. Meanwhile I have changed the