Voxilla VoIP Forum - Other Hardware/Software

Drishti Computer Telephony Integration (CTI) System

vijaysharma — Mon, 22 Jun 2009 10:27:47 GMT

A wide range of technology systems is used to facilitate communications across an enterprise. Some of such technology systems are media servers for different channels such as voice, email, web, office applications, workforce management applications and many more. How well the disparate systems are integrated affect the efficiency of customer relationship management. Better the integration, better the business performance.

Computer Telephony Integration (CTI) from Drishti helps you to keep your business processes under the control of business rules. Loosely integrated disparate systems lead to high operational costs and severe revenue losses. If all systems in the enterprise operate as a single unit, it will lead to smooth business functioning and high profitably. With Drishti's CTI, you can integrate your enterprise resources with different business communication platforms in order to coordinate customer interactions.

Drishti's CTI capabilities provide you with the following functionalities:

- Linking agent desktops with customer information database for screen pops
- Queuing and routing of multimedia interactions

The pre-integrated database & CRM allows you to maintain and manage multiple client relations in a more efficient manner. Screen pops allows the agents to access customer information in real-time mapping it into a GUI based interface through the database. With such CTI capabilities your agents can service faster and more efficiently.

If you have any questions about any of the above technologies or are interested in setting up a contact center, please feel free to get in touch.

Visit: drishti-soft dot com

Feedback on Sangoma U100 USB device?

pmlco — Sun, 14 Jun 2009 11:14:32 GMT

I have two. Once they work, they seem pretty stable. I have three issues with them:

1. Sometimes - perhaps one in ten reboots, the device is not recognised, i.e. doesn't show up in lsusb, or it does but wanrouter can't find it. If this happens, the only solution is a complete power down of the system and manual reboot, which is not very convenient as one of the systems is at a remote site and often there's nobody there to restart it. A couple of times, the system has frozen when doing lsusb and the only thing to do then is pull out the power cord.

2. These is a lot of echo. I have enabled echo cancellation in Asterisk (mg2) and that helps, but does not eliminate it. I will try oslec, but I have not got around to it yet.

3. Not a big deal, but the installation procedure is only half converted to dahdi. For example, the file /etc/wanpipe/scripts/start produced by the installation procedure looks for /dev/zap - you have to edit it by hand to change all references to zaptel to dahdi.

All in all, I'd say the product had promise, but there's still some work to do.

Regards
Ian

Dial Plan : the equivalent of for Audiocodes MP112/MP118

uozzap — Thu, 11 Jun 2009 07:34:29 GMT

Hello,
I need to know how can i automatically let to MP112 compose the '0' to exit from PBX

For the Linksys i have the string , where it works putting <:0>

Thanks in advance :)

D-Link DVG-1402S problems

nzschooltech — Tue, 09 Jun 2009 22:23:21 GMT

I have an unlocked DVG-1402S that I purchased which came with nothing programmed into it. I used to use it with my first VOIP provider, iTalk here in NZ. It worked reasonably well except that every few days it would disconnect and I would have to go into the web interface and tell it to re register. As the registration timeout was set for 1 hour it was not failing to re register every single time but about once every week or two weeks it would somehow lose connection to the server because the Query button would produce some sort of error or other message indicating it was not registered.

Anyway I decided to switch providers to 2talk here in NZ and this has brought more problems because now I cannot make any outgoing calls on it but it receives incoming calls fine. Both of DLink Support and 2talk say there is a compatibility problem, the most obvious sign of this is that the outgoing proxy server cannot be set by FQDN, only by IP address.

I have heard that the DVG-1402S is a bit of a lemon, it is not very common in NZ, there is no newer firmware available that I can see and the providers here seem to know very little about it.

Part of the reason I bought it is that I changed from a DSL provider to a cable provider and the modem changed from one with full router capabilities to a basic cable modem with limited capabilities. My old DSL modem had a DHCP server and router in it and with a network switch connected to its output port it would generate IP addresses for connected computers and route them through to the Internet. The cable modem doesn't do any of this stuff so I was compelled to buy a new router to plug into it to add the same capabilities that I had originally. I switched to VOIP at around the same time and looked for a VOIP capable router that could connect my old cordless phone as well as the other capabilities mentioned and this D-Link was a good price and it does all that stuff as well.

Firstly, what do other manufacturers produce that is equivalent. For example Linksys (1 or 2 analog phone connections, WAN and Ethernet connections, DHCP server and router)

Second: Here are the instructions for setting it up that my current provider have put out:
http://blog.2talk.co.nz/dlinkdvg1402s.html

Third: Here is the response from DLink Support to the query I raised about the connection problems:
"You may find that the DVG-1402S is not 100% compatible with your new VoIP Provider as per why you are unable to complete a SIP Call.

Could you please provide reference to broken tone, do you refer to engaged signal? What is the Router in front of the DVG-1402S, does it have SIP ALG enabled and have you tried to disable it and setup SIP Port Forwarding to the WAN IP Address on the DVG-1402S? Whom is the new provider you are using and have they tested the DVG-1402S with their infrastructure? What are the setting which you are using with your new provider?"

Callers Can't Hear Me on DPH-541

mel2000 — Mon, 08 Jun 2009 04:00:53 GMT

I can hear and be heard on my DLink DPH-541 WiFi SIP phone under all conditions except when someone calls me. I can hear them but they can't hear me. Nothing changes after I forward numerous relevant VOIP, RTP and STUN ports.

Interestingly, I had the same problem using the same SIP credentials on the X-lite softphone, but it was remedied by deleting all audio codecs except G711 aLaw. Unfortunately, when I selected only that codec on the DPH-541 it didn't make a difference. I tried all the other DPH-541 codecs but none of them did the trick either.The firmware has already been upgraded. Is there a fix or workaround for my situation? The phone works perfectly in all other respects. Any assistance gratefully appreciated.

Siemens A58H auto reboot,when use handfreee key

Jaing — Sun, 07 Jun 2009 14:06:54 GMT

Hi~
I have a siemens gigaset A580IP + A58H Additional Handset *2

The a58H additional handset will reboot, when use handfree for callout/answer both
Please inspect that the video

YouTube - P1040635

" title="YouTube - P1040635" target="_blank">YouTube - P1040635

YouTube - P1040636

" title="YouTube - P1040636" target="_blank">YouTube - P1040636

This video for normal handset

YouTube - P1040641

" title="YouTube - P1040641" target="_blank">YouTube - P1040641

Anyone have same problem as me?
Can I repair the problem by way of the firmware update?
Perhaps I need return to the purchase shop.

Regards,
Jaing