similar to: Maillinglist as newsgroup ?

Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1

Hi, Yesterday I finaly got my handytone sip adaptor. It works.... But when dialing to and from ISDN I got echo in both ends, I had tried diff. codecs, but then the GS wont work at all - It can do a call, but after 3 'ring' it disconnect. Any hints ? _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to

Hi, How much bandwidth do I need for 1 conversation ? I know it depends on the codecs, in X-lite I can see a codec called gsm, and the grandstream aha analog/ip converter have a codec called 721. Doc. I have found the asterisk handbook, but only a draft from marts 2003 anything newer ? Guides/howtos are welcome as well. anyone have a php interface to accounting ? /HHA

Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like <tftpserver-dir> <mac-address> firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _________________________________________________________________ Rethink your

Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of IAX2[2109@2109]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA

Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to

Hi, When I make a call from my asterisk and it is passed thru another astrisk eg. iaxtel, I got 'Answered' in my astrisk, and bill-sec is start counting as soon I get connected to the other asterisk, and not if the party on the other asterisk server pick up the phone. So IF the other party are not aswering the call at all, I still get Answerd and billsec in cdr. Whats wrong ? Can I do

Hi! playing with --include, --exclude, and --exclude-from=file I found these not working: xcnlm00s:/etc/adsm/script # rsync -navx --include="*/" --exclude="*" -e ssh newsfeed:/ receiving file list ... done wrote 29 bytes read 28605 bytes 3014.11 bytes/sec total size is 0 speedup is 0.00 xcnlm00s:/etc/adsm/script # rsync -navx --exclude="*"

Why can it not find my object? What am i missing here? Here is my code: require ''rexml/document'' include REXML # classes to represent the objects and relationships in the xml file class Article attr_accessor :id, :post, :archive, :ntype, :head, :blurb, :body, :fblurb, :fimage, :att, :source, :copy, :brand end # the base parser class BaseXMLParser def initialize(filename)

WHY? because most of the Desktop users are intersted in ogg as replacment for .mp3 and not as astreaming alternative to Real (which I think is very important)! because most of the discusion here is concetrated on licencing/implementation and I would hate to add more to it (for example I hate discussions over winamp topics ... since I don't think it is very important to the ogg as

Smörk wrote: > I think I have the same problem. I didn't managed to complie the > "example" streamer, but if I open a url with a browser the connections > hangs too. IIRC the old icecast send a 404 if there is no stream > available. icecast 2 does nothing if I point to > http://localhost:8000/wrongmountpoint, doesn't matter if I use winamp > or a normal web

Hi I got a problem with asterisk 1.0.2 - it drops the calls, both sip<-->sip, and zap<-->sip. The conntions can stay for seconds to several minuttes, and then the connection just cut off. I can't see anything in the logfiles. (or dont know what to look at.) It drops several connections at a time, but not all. Where to start looking ?? /HHA

As nobody seems to answer my request, I simply post it again. Is there any reason why the r-help-mailinglist should not be converted to a newsgroup? These were the advantages of a newsgroup I mentioned earlier: -) you can easily search the archives -) the discussion is faster (I experience that the R-mailinglist has a lag of about 1 to 2 hours (not for everyone!!!). When I ask a question, I get

Hey everyone. I''ve got a best practices question. How are you guys rendering newsfeeds? We have a couple of apps where we send newsfeed items from a backend process. As such, we aren''t in the context of a controller and can''t use the rails template rendering. We''ve tried about 3 different ways to make that bearable, but aren''t having much

Hi, I got this setup. analog phone (ext7) ---> analog pbx ----- (ext 6 analog) voicemodem (ext 3 asterisk) ---- ttyS0/asterisk ---- sipphones q1: I got the voicemodem to work, but oneway only. I can talk from my analog phone, to my sipphone, but not the other way ? I know it only suppose to works in half duplex, but nothing come TO the phone. q2:

How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ? /HHA

Hi, Are there any hardware for ISDN30 ? if yes any problem with this ? is i out-of-box like ISDN2 but with 30 linies ? Do I need more than the cable from my teleprowider and a PCI-card ? /HHA _________________________________________________________________ Find high-speed ‘net deals — comparison-shop your local providers here. https://broadband.msn.com

I'm having an issue with cpu-spikes on the following setup Samba 3.0.24 as PDC with cups as the printing system on a Debian Etch 64 bit version Intel dual-core system My main problem is, that everything works just fine, there are no serious errors in the log files, even if i step the log level up to level 3. What happens is that the processes belonging to the guest-account have occasional

Hi I am trying Manager interface for originate a call. This is what I get --------------- Action: Originate Exten: 555 CallerID: test <6656> Context: local Timeout: 600 Channel: SIP/8782 Priority: 1 Response: Error Message: Originate failed ---------------- What do I do wrong? Thank you Serge _________________________________________________________________ MSN Premium with Virus Guard

Hi I am running firewall/router "brew" made of RedHat, Shorewall, Siproxd and GnuGK on a box that connects through PPPoE to Internet. I run Asterisk on another box behind of it and it seem to work fine for me. I am thinking of replacing the router box because hardware is getting flaky. I do not want to go through pain of assembling all this stuff together again. Does anybody know of