similar to: chan_skype license?

Hi, i can see AOC messages on the asterisk console. Can i sendtext() them to the caller or put them in cdr? Regards, Andreas. _________________________________________________________________ Need a new job? Check out XtraMSN Careers http://xtramsn.co.nz/careers

Hi Guys, is there a client which can be used on the SonyEricsson P800/P900...? IAX would be cool, but i take anything that can connect (via bluetooth) to an asterisk-server ;-). The phone is Symbian, and can also execute java-stuff... Greez Andreas _________________________________________________________________ Find your perfect match @ http://personals.xtramsn.co.nz with XtraMSN

This suddenly started appearing and I'm not sure why. Any ideas? asterisk*CLI> module load chan_skype.so Unable to load module chan_skype.so Command 'module load chan_skype.so' failed. [Sep 15 11:08:25] WARNING[12274]: loader.c:429 load_dynamic_module: Error loading module 'chan_skype.so': /usr/lib/asterisk/modules/chan_skype.so: undefined symbol: sfa_send_chat_message

Hi *, does someone has a directory that works with the Cisco 7960 and astdb or mysql/ldap? Regards, Andreas _________________________________________________________________ Gaming galore at http://xtramsn.co.nz/gaming !

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

Hi, on ISDN there are the numbering plans that indicate if it's an national or an internation number. Is there something similar on SIP? How should i set a callerid to an internation number? complete e164, with, without an intl prefix (ie +, 011, 00 etc)...? How to a national number? Regards, Andreas. _________________________________________________________________ Discover fun and

Greetings, First off let me say that Ogg Vorbis kicks butt. Great work to all involved! I apologize for fuzziness in my wording here. There are many details I am not allowed to reveal at this point for fear of upsetting development partners. That being said: I have compiled the beta4 source (without modifying it) and wrapped it into a plugin for a commercial multimedia collaboration

Goodday, I'm an opensource fanatic and I have already installed asterisk in my mandriva linux. Actually, I'm also planning to install the asterisk management portal for GUI of asterisk. If anyone could help me guide in installing this. Thanks a mill for the help.. -Rommel-

I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine. But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work. I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and

Anyone got any thoughts on this and how it compares to the chan_skype that's due soon ? "OpenSky is a free service provided by Gizmo5 which allows *any* mobile phone, web browser or IP aware phone network (SIP, asterisk, etc) to communicate with Skype users. OpenSky supports sending text messages and voice calls." http://www.gizmo5.com/pc/opensky/ Julian

Hi, I want to recieve calls to my Skype account and forward them to a SIP/FXS line. I searched for chan_skype for asterisk (v11), but found it only available for asterisk 10 I know that Digium gives no support for this module, but I am sure that someone somewhere did write some tool to allow such connectivity. Do have any idea if I can use Skype with my asterisk v11 ? Thanks --------------

Hi, does someone knows a IAX2 provider that allows to freely set the USER_NUMBER to the german pstn? It's no problem if setting the NETWORK_NUMBER is restricted, i'm using this for call-forwarding with the correct number, not something illegal... (i'm not sure about the terminology here, but i think USER_NUMBER is callerid and NETWORK_NUMBER is ANI in the US, correct?)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 found this today, I am not a skype user but have read on chan_skype and don't like aspects of how it is implemented. My thoughts on it are only theoretical as I haven't used it I just cringe at adding X to a server. Anyhow there is a new project called sippyskype that appears to do a similar sort of thing with a couple differences. 1. Its

Hi, I wrote a quite useful perl script (and I am still working on it) that does all the renaming and tagging of my MP3 files automatically (which is very useful because I am an order fanatic and everything has to have MY format). I'm not quite done with ID3v2 support but that will probably work soon. As I am planning to start using Ogg Vorbis instead of MP3 in the future I am already now

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> Hi all,<br> I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last version).<br> Everything was going fine, but yesterday I've got this messege when I've tried to restart asterisk

http://www.engadget.com/2010/06/22/skypekit-beta-sdk-adds-skype-to-any-application-or-device/ Great! Finally a change to get a chan_skype without beeing a**-raped by the copyprotection (which is the sole reason i didn't buy it), and maybe even more than the absolute basic features (like Silk and video)... J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, over the year, the "IM" question popped up here and then, the answer was allways something along the line "yeah, there's an experimental patch, but it's not yet ready"... Are there any plans to change this, to ie. allow messages between xmpp and sip or iax? Especially with the chan_skype (that's hopefully is going to be released in my lifetime ;)) a messaging

Hello all. My company currently has an older Executone PBX system that we are outgrowing. Rather than wait until the last minute to make a hasty decision, I thought it would be a good idea to do some research and compare options first. My expertise is in computers and networking, and telephony systems are mostly foreign to me. What we currently have are 5 incoming POTS lines and 25

i installed skype for asterisk i can send and recieve calls normaly how can i receive messages from another skype user i Succeed to send only using for example: exten => 2233,1,SkypeChatSend(fSkypeBcp,User,message text) how to receive messages using this code SKYPE_CHAT_RECEIVE(<account>,<from>,<timeout>),and where and how I should add this code in extensions.conf

Hi, I got lots of "undefined reference" when building vmkit. Anyone can help to give a solution? I download vmkit with "svn co http://llvm.org/svn/llvm-project/vmkit/trunk/", followed the instructions of "http://llvm.org/svn/llvm-project/vmkit/trunk/README.TXT", and built vmkit on ubuntu 13.10 x64 with sunjava 1.6.0_45 + llvm 3.3 + classpath 0.99 + gcc 4.8.1.