similar to: SIP/IAX softphones for use in callcentre environments

I've been working my way through the softphones listed on voip-info over the last few weeks and I've not really found anything to fit the bill. Has anyone had more luck? The environment is a small call centre of 5 users. Operators often need to be able to transfer calls to other operators with different specialties, so the softphone needs to be easy to use and quick to transfer calls.

I have installed the sipTAPI from http://sourceforge.net/projects/siptapi/ when I use user names like joash.herbrink in Asterisk, it is not working when I change the sip username to my internal extension, like 1006, it works fine. Anybody any idea as to why this is? met vriendelijke groet, Joash Herbrink Technical Consultant "Control the flow" De Kahuna groep

Hi there, I have bought 70 sipura 841 phones for a customer of mine. When following the mass provisioning guide in the admin manual for the sipura, I see it download the spa841.cfg file from my tftp server Sometimes the phone also downloads is phone specific file via tftp, and it works okay then. But, after a reboot of the phone, it is very very likely that it won't startup

Why is using ulaw or alaw an unlikely scenario? I wouldn't use anything but ulaw\alaw. The Bells can compete on price and will if they have to. Where they CAN'T compete is quality. If there were something better than 711, I'd offer that. Well, there is 722, but not many things support it. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original

Hi all A slightly off topic post, but one that I think someone out there in the NUT community could probably help me with. Summer is coming, and my office at home is getting hot... helped in part by the APC 3KVA UPS I have sitting under my desk. It kicks out heat like a bar-heater, even when there is almost no load on it (idle load is one PC, one switch, one wireless AP, and thats it). It is

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays

Hi group! How to set language for queue? I have several queue's. In every queue, agents speaks different language. I need to announce queue-youarenext and similar on different languages. This is what I have in my extensions.conf and it does set language, but when calls enters queue, it doesn't use that language. exten => 313,1,Answer exten => 313,n,Set(LANGUAGE()=de) exten =>

Hey Guys We are replacing a BM4 with a trixbox (asterisk) virtual numbers as the customer wants to move the callcentre. They are asking for an equiv to the ipview I gather HUD may be or the panel view The problem is that we need to see (a) total calls in the queue (b) calls for specific DID - How can you give 1 DID preference to another DID ie DID 61740410001 = Fred Electrical DID

Hi guys, I'm playing around with Samba3.0-alpha17, and it seems that I'm having some odd problems with file visibility and access. I've run other versions fine in the past, with similar smb.conf files (which I'll post, if necessary, but right now it's a bare minimum!), but for some reason I now can't see all my files! For instance, if I go into a subdirectory

Dan Liew <dan at su-root.co.uk> writes: > On 7 July 2015 at 15:05, David A. Greene <greened at obbligato.org> wrote: >> I'm learning the cmake build. >> >> Is there a way to disable the build of a subdirectory in llvm/projects? >> I'm getting a build error on one project and don't want it to hold up >> the build of everything else. >

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote: > Does anyone have a way to do wake calls? > > > > Jordan Novak > > Communications Technician > > Logistics Health Inc. You could use cron and /var/spool/asterisk/outgoing scripts to dial numbers, etc... > Can you elaborate, I am fairly new to Linux and a phone guy to boot. I am looking for a way for the

To deal with google dropping xmpp for voice, I've gotten a callcentric number. The cc number connects to asterisk, and all works fine. Then I set up the cc number as the gvoice forwarding number. If I'm on the gvoice site, I can make a call and it will ring my cc number and then the outside number. That also works fine. BUT, when an outside call comes into gvoice it forwards the call

Asterisk Users, I am looking for Sip Providers for my Asterisk 1.2.13, running Debian Etch system with McLeodUSA's T1 service. Has anybody ever used Callcentric for their Sip Provider? Any service issues with Callcentric? Best Regards, John _________________________________________________________________ Messenger Caf? ? open for fun 24/7. Hot games, cool activities served daily.

Hello Everybody I got some little issues with the sound. Logged in as my normal user - I've the testsound (winecfg) on the internal device. I can't change it. If I log in as root I've the testsound on my external USB speakers (the way I would prefer it works). I can't Imagine me - why there is a different between root and my normal user. Please find below some details. Code:

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

Hello everyone, I'm evaluating the Celt in terms of robustness against bit errors. So what I have done is encoding a wav file using the \tools\celtenc.exe and then manipulated the wav file according a desired bit error rate or burst error length. But the Celt decoder stops decoding with the first bit I manipulated even if this bit is not in the header. So is there a way to do the test

X.Org Security Advisory: October 25, 2023 Issues in X.Org X server prior to 21.1.9 and Xwayland prior to 23.2.2 ===================================================================== Multiple issues have been found in the X.Org X server implementation published by X.Org for which we are releasing security fixes for in xorg-server-21.1.9 and xwayland-23.2.2. The first issue (CVE-2023-5367) can be

I have configured winbind on a Linux file server, connecting to a Samba PDC. When I run wbinfo -g, I can see the group "domain users". On the other hand, when I run getent group, I do not see this group. Apart from a few other groups, all groups are visibile in both wbinfo -g and getent group. When running for the first time wbinfo -u, getent passwd and wbinfo -g, I got the results