similar to: Supervised call transfer problem

I issued this command to join a Linux computer to a Windows 2000 domain /usr/bin/net rpc join, and I got an error message: [2002/11/06 16:04:50, 1] utils/net.c:net_find_server(229) no server to connect to Unable to find a suitable server [2002/11/06 16:04:53, 1] utils/net.c:net_find_server(229) no server to connect to Unable to find a suitable server ===================== To create the

On Samba 2.2.6 and Winbind I used the following command and all are working: /usr/local/samba/bin/wbinfo -u (it did list all win 2000 users) /usr/local/samba/bin/wbinfo -g (it did list all win 2000 groups) root# getent passwd (it list all Linux users) root# getent group (it list all Linux groups) ============================================== The problem is. When

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. 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

Hi all, I'm trying attended transfer on Asterisk 1.0.7 and AT-320 phone. I met a lot of problems during this steps, while in the blind transfer all works fine. I had this kind of problem: CASE 1: A call B B set on hold A B call C (that is busy for some reason) B try to get the first call with "hook flash" (or pressing the "hold" key) and A stop to work. B

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:

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

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

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 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

I'm trying to get asterisk to pass through a call without requiring supervision on the line. Any thoughts? Thanks, Frank

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

Hi all, I've the current scenario: User "A" - Zaptel call incoming in my Asterisk to my SIP user "B". "B" gets the Call. "A" says : "B" i would like to call PSTN user "C" "B" places a call to user "C" and asks if "C" wants the call from "A". "C" says yes i want, then B needs to

Hi, I am currently find myself selecting manually amoungts several hundreds Google Alerts (GA) texts those that are indeed relevant for my research vs those which are not (despite they are triggered by some relevant seach keywords). Basically each week I get several hundreds GA email such as:

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

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

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

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: