similar to: call back on failed transfer or dial?

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,

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

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 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

Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd

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

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

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, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Am I doing something wrong? I can't get this dial command to timeout.... Dial(Zap/g1/xxxxxxx,20) -- Gary White admin@netpathway.com Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hiya. I've been using * for a while now and everything is working fine (but then I'm only using basic features :) However I have to make a change to the setup to allow a new feature for the users. Basically I have a few people on the * box using the current agent / queue system where I create a call file into the spool directory so they can

Hi, I am using a number of snom190 phones, and an asterisk "gateway" server, and recently started experimenting with call transfers. The snom phones provide support for attended and un-attended call transfer, so I would rather use that than call-parking. I have found that un-attended transfer works fine, and that attended transfer works fine if the originating phone call is NON-SIP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code,

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

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

Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file ("Please waite while you are being transfered") before transfering the call to the cell phone. How can I do this? Andre

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hi, I'm unable to pick up parked calls after they are transfered. I get the "transfer" message when I press # and then I'm told "701" The extension I'm dialing goes to the on hold music. I'm disconnected, I hang up, dial "701" and I see this message on the console "Everyone is busy/congested at this time" I just have the default

It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active. Does anyone know any workarounds for this problem? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection