similar to: GS Handytone Echo-problem

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, I was thinking if it was possible to get this list as news ? It would be much easier that 'hotmail-account' /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

There seems to be a problem related to the Grandstream HandyTone-286. When a call is placed through the adapter, the call can be transferred. However, when a call is received through the adapter, the call cannot be transferred. The problem does not exist with a BudgeTone-101 (1.0.4.23) using the same Asterisk configuration and Dial() settings (Ttm). I tried all of the firmware on their BETA

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 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 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

this is in reply to a different message i posted a couple days ago , im not sure how to specifically reply to it sorry :( heres the message: thank you very much vinod this does give me a nicer idea :) but i still have a couple questions well firstly, speex_bits_write is still blocking forever, i can never seem to get past that line (i am using the armv4 build on ppc2002, but i havent found

Does anyone have concern about this? What if Redhat stops giving SRPMS for new releases and updates in public? If you buy a subscription to RHEL AS and they have to give you the SRPMS because of GPL agreement can that person provide those SRPMS to public again because of GPL? They claim they are premium OpenSource company but they are still a company and for them money is bottom line.

In an earlier post I made about learning how to use speex someone nicely responded to me with a Speex Wrapper they had written.. however I am having an issue now where all I seem to hear is a ticking noise after encoding and decoding with speex. It is definitely getting proper data, as if I hit the mic somewhat I hear some extra crackling noises. Now, my code was nearly the same... but...

Hello, I bought a Grandstream Handytone 486 to forward incoming calls from our old analogue PBX to the asterisk server. My first test was connecting an analogue phone to the Handytone and calling a sip phone - worked. Now I used the same cable to connect the line port of the Handytone to the analogue pbx. When I call the number of the analogue PBX I hear a clicking inside, but the call

Hello List, I am an Asterisk newbie, and I got a question about Asterisk Background command's option "noanswer": What is required from the user agent, such as a SIP phone, to be able to hear the playback without Answer()? I'm asking this because when I used X-Lite, I could hear the the audio file but when I used a hardware phone (an ATA in fact) I couldn't hear it. The

Hi I have a analog phone connected to my Gateway Handytone and registered to Asterisk 1.4 I have configured my HandyTone 488 (in the section FXS Port) for make and receive calls, however I can not transfer a call when it come via PSTN. But, when a call come from via IP I can transfer it. [phoneanalog] type=friend secret=XXXXXXX context=local nat=no qualify=yes host=dynamic dtmfmode=rfc2833

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

Hello, I'm trying to configure Asterisk and my Handytone 488 to pass incoming calls coming over PSTN through the FXO port to Asterisk, which will process the calls with voicemail, or some such service. I point the Handytone 488 FXO port configuration to 192.168.0.2 (the machine that is running Asterisk) and have configured a catchall extension to receive the call: [from-pstn] exten =>

For some reason, this didn't go through the first time, maybe because I had JUST signed up. Hello, I'm trying to configure Asterisk and my Handytone 488 to pass incoming calls coming over PSTN through the FXO port to Asterisk, which will process the calls with voicemail, or some such service. I point the Handytone 488 FXO port configuration to 192.168.0.2 (the machine that is running

>From: "Park Heesob" <phasis@nownuri.net> >To: "Shashank Date" <sdate@everestkc.net>, <djberg96@hotmail.com> >CC: <win32utils-devel@rubyforge.org> >Subject: Re: win32-service ideas >Date: Sun, 25 Jan 2004 16:41:32 +0900 > >Hi, > >In my computer, I can''t reproduce timeout error. :-) >I have committed service.c adding

Park, Shashank, It turns out I was doing something stupid with regards to Dir.chdir in testing, which appears to be the reason it was failing. It seems to work now. Hooray! Shashank, I''ll take a look at your errors and see if I can nail them down. The RPC message is one that I got from time to time, but from what I can tell, it''s bogus or a system config/permissions

Sorry guys. I was in a hurry and forgot to include the url. Just click <a href="http://wxruby.rubyforge.org/wiki/wiki.pl?Frames_(Part_1)">here</a> or enter in this url: http://wxruby.rubyforge.org/wiki/wiki.pl?Frames_(Part_1) Ugh! Robert _________________________________________________________________ Stop worrying about overloading your inbox - get MSN Hotmail Extra

Kevin, Very cool. I initially tried to figure out how to do it without those evals, but I failed (thanks for picking up the slack :) Your changes definitely improved the code, and made it much cleaner. Thanks. By the way, what is pickaxe? I''m not familiar with that term. _________________________________________________________________ Free up your inbox with MSN Hotmail Extra