similar to: AT-320 + supervised transfer

Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to

The procedure that will do asterisk is very nice ;) but whe it was available ? Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why. Here my sip.conf for the phone, can u say me if there is somethingh wrong ? [2391] type=friend username=2391 secret=2391 language=it host=dynamic context=intern dtmfmode=rfc2833

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

No...maybe i don't explain u well. After that B call C andC not answer (go in timeout), B hear first the beeperr and then, together A the busy tone. Now i can't re-take the call :| Thanks Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: mercoled? 1 giugno 2005 12.34 A:

Hi list, This wll be my first post, so I want to thank all the developers for the great product they have created. Now, the question, I have installed asterisk 1.05 on debian sarge (binary package) with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100) This all works fine, exept for som echo on the ISDN channel, but I'll replace the I4L card with an AVM-C4 card next

Hi all, I'm quite new on this mailing list, and I discover the asterisk world. I m experimenting a PBX with SIP phones, grandstream budgetone (not expensive for tests) All the features I need work just not one : the supervised call transfers. I know there are a lot of posts about that, but none gave me the correct answer (unless I missed it). Here is my config : 2 sip phones BT102 with

PLEASE RESPOND IF THERE'S A SOLUTION I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the

Hi, We are testing new Asterisk 1.6.0.1 because we would like to use the Attended Transfer feature and we are trying to use the new action Atxfer developed for AMI. As far as we know, it is suposed to be in this release as it can be read in Digium's changelog /New command: Atxfer. See doc/manager_1_1.txt for more details or manager show command Atxfer from the CLI/ But, when we try to

I am trying out the new atxfer feature from CVS-HEAD. I set atxfer equal to *7 and it seems to work OK. I am having a problem getting it to work the way a receptionist would want. If an extension calls me, I hit *7 and I hear the voice say "transfer". I dial another extension. If the newly dialed extension goes to voicemail, I can't figure out how to get the original call

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

Hi. I was wondering if there's a way to do supervised transfers on the budgetone 102 . Blind works ok, but can't do supervised. I thought that with the flash button that could be possible, but seems I'm wrong .... matteo.

I've seen a lot of traffic on the list recently about which phones can do supervised transfers and which cannot, and I have to admit that I'm a bit puzzled. Our existing PBX, which is software based, handles the transfer functions for our call center -- the agents never touch their phone, and instead use software. We can plug any old phone into it, and it'll work just the same. So

Is there a way to keep the incoming CallerID from the PSTN and pass it onto the sip phone receiving the supervised call transfer? The receptionist receives the PSTN callerID, performs a supervised transfer, we get her local SIP callerID, not the original callers. The main reason we would like the true callerID is for asterisk monitor to name the file correctly for call records. Is this

Hey all, I've been trying to get supervised transfers working without success. I'm currently running 1.0.7-stable and think it might be a version problem. Is the supervised transfer feature available in 1.0.7 or do i need to suck down a new version from CVS? Otherwise, apart from setting up features.conf, is there anything else i'm missing? TIA, Jamie. -- Jamie Carl

Luca: 1. We are not a consulting service. We *help* with R pogramming issues. Users are typically expected to make an effort by providing R code and, if appropriate, small data sets that illustrate their difficulties. 2. SEARCH! e.g. on "text processing R" or some such; or try Rseek.org with such searches. R has extensive text processing capabilities, e.g. via regex's. 3.

Hi all I am having problems with atxfer if I do the extact same thing with blind xfer it works fine when I hit press #2 (defined in conf for atxfer) i get "transfer" I dial the number I want and i get the following on the console -- Playing 'pbx-transfer' (language 'en') -- Executing Dial("Local/18005558355@jesnjer-f97a,2", "/18005558355")

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at

Hi, Is there any package that implements semi-supervised clustering through 'must-link' and 'cannot-link' constraints? thanks! __________________________________________________ [[alternative HTML version deleted]]