similar to: call back on failed transfer?

hi, i'm under the impression that this feature is not available in asterisk, consider this scenario: - you are the operator. you answer a call from outside and you want to transfer it to one of the extensions. after you transfer, if the person you transferred the call to, doesn't pick up or if his line is busy, the call is transfered back to you, you can speak to the caller and tell him,

We have set up an IP telephoney system hosted by Asterisk and its working pretty well. We primarily use SIP and hardware IP phones. We have the ability to transfer calls to another SIP phone using either the "Transfer" button on the phone (these phones are Grandstream BudgeTone 100s) or using the "#" key (the T/t flag must be set in the Dial command in asterisk for

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do

I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

Recently I put callwaiting=yes in zapata.conf because customers want to speak to the operator in person, not leave her a voicemail, when she's busy with another caller. But now she can't transfer either of the calls (which she can do when there's only a single call). The operator has an analog phone connected to a TDM400B FXS line. The calls are coming from PSTN lines connected

Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/f220d3ae/attachment.htm

I get this message at CLI. what does it mean? - shabanip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040925/dbcd8d80/attachment.htm

I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based

Hi, How do you transfer (using say blind transfer) a call to an extension that is currently busy on another call? You don't want the call to be transferred to voicemail, it must stay in 'hold' until the extension becomes available, and then immediately ring that phone. Thanks, Thomas

hi i have a question. is there an easy way to remember transfered files to a specific destination host. little example: 1) host A: with files a, b, c 2) rsync-transfer them to host B 3) add a new file d to host A 4) delete file c from host B 5) again a rsync transfer, but file c get skipped, d get transferred maybe as a feature request: add an check for an extended attribut of the files which

Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G

Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a

Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like " I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him "wait a sec" and push "Flash" and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to"...

I have a customer that needs to record all calls coming in and out. The problem I am having is when a call comes in to the operator and it is transferred to another extension. The first mixmonitor begins recording when the operator picks up but the recording stops when the call is transferred. I need to have a single recording for the complete call no matter how many times it is transferred.

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm wondering why when I use --size-only on the same file, except the modification time, rsync transfers the modification time? I was under the impression that --size-only caused rsync to transfer files based upon differences in the size of the file. For example: #cp Untitled.pdf Untitled-2.pdf #touch Untitled.pdf #rsync -ani --size-only Untitled.pdf Untitled-2.pdf .f..t......

Hi all, Is there a way to tell which extension transferred a call in a blind transfer? Sorry if it's a basic question, but I haven't seen an answer. ${CALLERID(num)} still holds the outside party caller id (which it should), but I'd like to the extension number of the extension that transferred the call. Any suggestions? Thank you, -- Warm Regards, Lee

Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer.... also can