similar to: Re: asterisk-users Digest, Vol 33, Issue 102

Hi, Few subjects cause as many arguments as "which SIP client works best?" on IRC #asterisk, voip forums, and probably the -users mailing list. I have tried most of the SIP clients available in the last 5 years, both with Asterisk and other platforms such as OnSIP.com, IConnectHere.com, ZipDX.com and the venerable old FWD (in the days when that almost worked). Speaking of ZipDX, we

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List (Again) Thanks to some answers on this list and another I now have a MultiTenant system/setup working the way that I want it to, So now my next job is to find a SIP SoftPhone that I can brand to my own company images and so on. Again an OSS would be preferred, Even though X-Lite is bar far one of the best free SIP clients I have used and it

Hi, In the aftermath of Digium's and Counterpath's Bria for Asterisk announcement, we're happy to chat with Todd Carothers, Counterpath Product Manager today at 1 PM EST. For more info, http://vuc.me Join us on IRC #vuc on Freenode.net or use the web client at http://vuc.me/irc Call in starting at around 12 Noon EST: sip:200901 at login.zipdx.com Hear you there! /r

Thanks. In app_realtime, it is very easy to get a value of a field by only applying the realtime application. However, in func_realtime, we need to get the key-value pair according to the position of it by using function CUT. After that, we need to apply another CUT to get the value. It will cause the following problems. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+realtime 1.

I am new to SAMBA and I have what I'm not even sure is an issue. I am aware of the difference in time stamps between *nix and Windows. What I don't understand is this: I used touch to modify the time stamps of a large number of files on the file server from the server side to match the time in the file name. They were video files from my Digital Video Cam and the import program used

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.

We installed na Asterisk System whith 400 Softphone users (Eyebeam 1.5 from Counterpath). As far as we know, Asterisk don't support yet IM (Instante Message) feature,instead Eyebeam have this feature. Is that true? Is there any new version from Asterisk that supports IM? > Eduardo R. Assis > Soluziona Ltda > Consultor S?nior - TELECOM > Al. Tocantins, 125 - 290 andar -

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

Rob Hillis wrote: >> The solution for the problem of an IAX client is a SIP client. >> >That's not a particularly good solution if you have a NAT between your >client and Asterisk. IAX is still *much* easier to get working through >a firewall. It's working fine here (Twinkle/Ubuntu over NAT/Netscreen). Didn't have to change any settings on the firewall either.

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten

Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set

hello, James FitzGibbon: thank you for your help. i am very new to arm-linux and embedded linux. i think what you said is right. i am not very sure the steps i taken are correct. i post it here and please give me some help. it might be help other arm-linux users too. i installed all necessary libraries in my linux. if i just install asterisk under my linux. there is no problem. but when i

On Fri, Aug 24, 2007 at 12:27:14PM -0500, Russell Bryant wrote: > James FitzGibbon wrote: > > Let me ask a question myself: what kind of regression test does * undergo > > before release, and what level of traffic gets put through stuff like > > app_queue? I assume it's not real-world scale, else these hard to pin down > > concurrency issues we're seeing would

> local({a <- CRAN.packages() + install.packages(select.list(a[,1],,TRUE), .libPaths()[1], available=a)}) trying URL `http://cran.r-project.org/bin/windows/contrib/1.8/PACKAGES' Content type `text/plain; charset=iso-8859-1' length 16079 bytes opened URL downloaded 15Kb trying URL `http://cran.r-project.org/bin/windows/contrib/1.8/fdim_1.0-2.zip' Content type

I am having problems with Eyebeam when the user dials a number that is not available. This problem exists with both Asterisk 1.4 and 1.6 using Eyebeam or Xlite. The problem seems to be that when the soft phone receives the 503 Unavailable response it will not be able to dial another number for a few minutes. Anything you dial will say that the number is unavailable and it will not even send the

I'm chasing down some DTMF interop issues would like to hopefully rule out Asterisk in the following configuration: RTP path is: Linux/PC/Mac SIP clients -> [MediaProxy as needed] -> Asterisk 1.8.7 -> SIP termination provider(s) DTMF is strictly RFC2833 with no in-band. Asterisk stays in the media path for application reasons and is "Locally bridging SIP/foo and SIP/bar"

I have installed several call centers in the netherlands with the eyebeam softphone (from the counterpath guys) It is not free, but very stable, and pretty easy to use. It works great with asterisk (specially the presence option, so agents can see whether somebody is actually ready to take a call). In combination with sennheiser headset CC series, I have had no complaints. We also use a tapi