search for: jooray

Hello, I see, that Rails is quite english-centric. I am developing some webs, that are not primarily in English. I have a few questions: - besides turning of plurals, what should I take care? How to use utf-8 for all data and converting it from local charsets to utf-8? - how do I make my page multilingual (i.e. adding english support later)? Is there something like gettext support? Is

Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there

Hello, I would like to implement a home GSM gateway using asterisk. What would you recommend me as a low-cost hardware for creating a gsm channel? I found 2n gsm gateway, that supports sip and chan_blue for bluetooth connections. Any recommendations? Basically, I want to end calls to some GSM number in my sip telephone and for some prefixes dial out using that same sip telephone. Also

Hello, I would like to use Asterisk with fayn.cz service. They should be using a standard H.323 protocol, but I have no more information about this. They provide a softphone and/or rebranded H.323 telephone, but I don't know any H.323 settings nor if the firmware in the phone is modified. Has anyone tried this successfully? They provide a Prague telephone number reachable from

Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar.

Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client "subscribes" to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome.

Hello, I added queuing support (based on SQLite database to store the queue) for my SIP Messaging patch. Works with eyeBeam, probably lots of bugs, but it's at least something. I created page about installation on the tips and tricks of voip-info.org: http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+Messaging Any bugfixes are welcome. Yes, it's a huge hack and supports

Hello, has anyone seen or created a Debian Sarge package for 1.2beta1? I saw some for Sid, but I prefer not upgrading glibc right now, as this is a production server (Asterisk on it will be for testing). Thanks, Juraj.

Hello, I have a polycom ip600 and eyebeam. When I call from polycom to eyeBeam, everything, including audio works. When I call the other side (from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows the same codec: g711u. Also sip show channels shows ulaw codec for both sides and correct addresses. I have canreinvite=no. I don't know if it's important, but asterisk

Hello, I would like to ask, if there's a way to transfer a call from some external program? I would like to build something like Asterisk Flash Operator Panel, with the ability to transfer a call using drag and drop. So I would like to connect to asterisk command line interface and transfer one side of a call to someone else. Is this possible somehow? Thank you, Juraj.

Hello, I was again asked to try to add support for presence (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions: a.) are there any, at least partial projects, patches, anything, that at least partly implements presence and/or IM to chan_sip? I don't care about presence on other channels, I have one SIP client per user. I've read this list's archive several times and