similar to: Looking for business-grade SIP Softphone

We've been using RNK Telecom for terminations for our SIP service, but their billing is questionable, they've been in breach of contract multiple times, and when I brought it to their attention, Russ Man, our 'friendly' account manager told me, and I quote "If you are having that much of a problem..please find another carrier." With customer service like that, I've

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? I'm looking for something for a call center that I can provision from a central location by generating config files. If the phone has "soft keys" (yes, I know they're all soft - but you know what I mean; programmable buttons whose

Hi, I checked the ODG (Objective Difference Grade) scores for a few reference vectors using the PQEvalAudio Tool and found that some of them show ODG scores as high as -3.5 If we look at the range as described in the link below, it looks unacceptable. http://www-mmsp.ece.mcgill.ca/documents/Software/Packages/AFsp/PQevalAudio.html Am I missing something or are these scores valid? Thanks and

Hi Rhishi, PQevalaudio is very unreliable and buggy. I have compared to PEAQ and - as a result - now I am not using it anymore. With best regards, Christian Hoene Von: opus-bounces at xiph.org [mailto:opus-bounces at xiph.org] Im Auftrag von Rhishikesh Agashe Gesendet: Freitag, 4. Oktober 2013 12:35 An: opus at xiph.org Cc: Rasmi Mishra Betreff: [opus] ODG (Objective Difference

Hi, Yes I aware to the NAT possibility, But let's assume that this is the issue, there is no reason that 30 listeners >From the same country will connect and disconnect at the same time range... I'm pretty sure that its individual listener/IP. I deleted the Access log files, but in the next time that I will catch similar situation again, I will complete the investigation. Nobody from

Is anyone aware of a R package that uses the Grade of Membership model for data? Thanks

The side effect of <audio> is also a bandwidth usage? linear to the amount of "fake" listeners? From: Marcin Lewandowski [mailto:marcin at saepia.net] Sent: Tuesday, July 05, 2016 4:31 PM To: Christoph Zimmermann; icecast-dev at xiph.org; Yaniv Sharon Subject: Re: [Icecast-dev] multiple connection (be careful with carrier-grade NAT) As I have written before, I’ve seen that

Hi, folks. We've gone where no man has gone before. On HP-UX, rsync bombs at about 1.75 million directories, files and links (combined sum) in a single transfer. Is there a professional-grade alternative on HP-UX for folks willing to pay for it? It wouldn't even need to be network-aware, just from single-system areas to the same box, but with the nifty delete and update features that

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

We currently have an Asterisk system with a PRI and 6 POTs lines for backup. We are looking to add service such as Voicepulse Connect as an extra level of redundancy and a cost saving alternative to PRI calls. VP Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times that to support our call center. Also, in looking through the archives, it seems like VP has had their share of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I need some user grade windows-based IMAP stress tool. I have a situation with a lot of "disconnect" error on the user side (outlook, thunderbird and outlook express) but no sign of error on the Dovecot server. Between clients and server there are a LAN of dubious performance and a ZyWALL. I have already removed every kind of packet

Hi Folks, Had a quick search through the archives for softphones and cannot see any recommended ones. My question is what recommended free softphones are out there that can be used with Asterix? I don't really know how many are out there. Is anyone currently using a softphone with Asterix and if so which one and how do you find it? I'm only interested in ones that I can download and use

Lately I have been getting many complains that Eyebeam crashes when you dial a number that does not exist. This happens in both R2 and ISDN PRI lines. The softphone stops working and has to be restarted. The response I got from tech support was: the actual issue is that asterisk should not be sending a 503 service unavailable when a particular softphone is not online. The soft phone stops

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come that call goes to Line 2(same extension used by Line 1) and 3rd call goes to 3rd line and so on. But i

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --------------- Wanna buy a duck?