similar to: Supervised/Attended transfers

Hi all. I`ve installed a X100P on my box and is working well with incoming and outgoing calls as a trunk with one PTSN line. I want to connect the X100P to my Panasonic 616 EASA-PHONE as an internal extension to permit to users to make calls to SIP devices from analog phones, the problem is when I dial the ext number where the X100P is connected I get busy tone. What config I need to change to

Hi all, I'm thinking buying a UPS Mustek PowerMust 600 USB to shutdown a Linux box on case of power failure. Can I use nut with this UPS without problems? I checked the page http://www.vamos-wentworth.org/upsinfo.html and was only something (incomplete) about Powermust 400 (information from November 2005). Could this be already changed? Thanks in advance.

Apologies if this is in the archive - did look but couldn't find it. Does anyone have any experience of migrating from CommunigatePro to Dovecot? We currently run CGP 5.3.4, supporting a small system (20 or so users, one domain). We've been using it for years, and have a mixed bag of MailDir and mbox folders accessed via IMAP clients. Some users have large mail accounts (15GB total).

Hey all. I've got a weird problem with the grandstream budgetone101 and asterisk that I'm having no luck finding any info on. I'm positive it's a grandstream problem but i'm hoping someone here can at least point me in the right direction. Basically, (and it's a simple problem) if a user taps the hook switch quickly they get dialtone again but it does not hangup the

Hi, I have several Linux machines at home and I'm considering buying a UPS for them. I see that many models are supported by NUT (great!) but since I'm new to the UPS world I have some questions about UPS features and NUT support of them I couldn't find an answer for: 1/ are there UPS models who report/meter power usage through USB? per UPS socket or globally? 2/ are there UPS models

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

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used

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 folks, Everything is working great and I am not having any issues with the three domain controllers that I currently have set up. We are migrating from Puppet to Ansible for configuration management, and I decided to create a playbook that will do all the things necessary to set up a DC and join the domain. I have found that in the domain joining process, an error stops replication from

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.

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 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

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.

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

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 all. I have currently a server: cat /etc/redhat-release CentOS release 5.7 (Final) uname -a Linux host.domain.com 2.6.18-274.18.1.el5 #1 SMP Thu Feb 9 12:45:44 EST 2012 x86_64 x86_64 x86_64 GNU/Linux I have there a filesystem mounted: /dev/vg0/paczki /home/paczki-workdir ext4 defaults,noatime 0 0 on which df gives strange output: LANG=C df -h

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: