similar to: Instant messaging (yeah, again)

well, My fulltime job is to add (i.e code if have to) some instant msging to our asterisk pbx. I have no idea, where to start. i want contribute write code, test etc. pls advice. there is SIP extension method called "MESSAGE" which get IM done. Does Asterisk support both SIMPLE and XMPP. thanks

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

Hiya, SIP Messaging is implemented in asterisk-10... >The only documentation I can find talks about a patch and is pretty old:>http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging <http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging> Like anything on voip-info.org it's horrible outdated. I think there's a documentation for the message-routing in docs.... Regards

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 guys, is there a comment from digium on the license of chan_skype? I could not find the GPL_KEY in the precompiled module, and they don't release the source. So i'm guessing, they'd need a commercial license...? Regards, Andreas _________________________________________________________________ Become a fitness fanatic @ http://xtramsn.co.nz/health

I'm trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I'm only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the "call" hangs up without executing further instructions. Anything

Hi, i have some spare (read: Boss get's a new one every few month ;)) Android Phones laying around. Does someone know a way of using them as a mobile gateway for asterisk? I could not find any SIP-Gateway in the Market, and i don't think it's possible to use the GSM Audio directly with something like chan_datacard... Regards, Jay -------------- next part -------------- An HTML

Hi people, I have Asterisk 1.2.13/DebianEtch as my VoIP server, using SIP. I need to use IM (instant messaging) among X-Lite clients, but when I send a message to any other client I get the error "Error: method not allowed". I read Asterisk does not support instant messaging, so.....What's the best way to have instant messaging with Asterisk ??? Thanks a lot. Alejandro

Just want to know if anyone has used instant messaging using Polycom and Asterisk.

Just wondering if anyone has gotten instant messaging working between two polycom phones/and or MSN messenger with asterisk in the middle? I tried a few things but couldn't get it to work... Regards, Adam

Hello, I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I see plenty of online help for chan_sip, but nothing for chan_pjsip. I imagine there is both pjsip.conf configuration and extensions.conf

I am running Debian on a LAN. I am trying to send popup messages(Winpopup protocol) but just get the same error over and over. We have a WinXP machine named "Mindelessdemon"(He is actually a very nice guy, don't get foolled by the demon stuff), a Win2k machine called "Whitehead" and a win98 machine called "Wyrm". I cant connect to the WinXP or Win2k machine

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 --------------

I have been working with distributed device states in Asterisk using XMPP attached to an OpenFire server. I have it working well across two servers and want to roll it out across every server in my company. All servers are Asterisk 11.6.0. I am running into a problem that seems like it should be a bit easier to solve than it is seeming to be. On the third server I am rolling into this

Hi all, I was wondering if someone could explain the relationship of Asterisk, SER & Jabber (XMPP) to me. I understand that there are facilities within Asterisk to use jabber to notify of incoming calls via XMPP clients, however I'm trying to work out exactly where the SER server would sit in all of this and what it's actual role would be. This might seem like a silly

Can anyone explain the Polycom Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to it? I've seen vague references to MSN Messenger, and somehow that's mentally disturbing. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

I'm trying to get my Asterisk 11 test box set up with XMPP, having troubles with JabberSend(). My jabber.conf file is as follows: [general] debug=no autoprune=no [testaccount] type=client serverhost=my.jabber.server username=myaccount at my.jabber.server secret=mypassword port=jabberport usetls=yes usesasl=yes xmpp show connections gives the following output from the console: