<font size="2">
<div>What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/ directory, asterisk loads 144 of them, omitting only chan_gtalk.so and res_jabber.so.
</div>
<div> </div>
<div>
<p>Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371)<br>Verbosity is at least 3<br>foo*CLI> module load chan_gtalk.so<br>[Mar 7 10:23:07] WARNING[9407]: loader.c:362 load_dynamic_module: Error loading module 'chan_gtalk.so':
libiksemel.so.3: cannot open shared object file: No such file or directory<br>[Mar 7 10:23:07] WARNING[9407]: loader.c:614 load_resource: Module 'chan_gtalk.so' could not be loaded.<br>foo*CLI></p>
<p><a href="mailto:user@foo:/etc/asterisk">user@foo:/etc/asterisk</a># ls /usr/lib/asterisk/modules/ | grep gtalk<br>chan_gtalk.so<br><a href="mailto:user@foo:/etc/asterisk">user@foo:/etc/asterisk</a>#</p>
<p><a href="mailto:user@foo:/etc/asterisk">user@foo:/etc/asterisk</a># ls /etc/asterisk/ | grep gtalk<br>gtalk.conf<br>gtalk.conf.old<br><a href="mailto:user@foo:/etc/asterisk">user@foo:/etc/asterisk</a>#</p>
<p><a href="mailto:user@foo:/etc/asterisk">user@foo:/etc/asterisk</a># ls /usr/local/lib/ | grep libiks<br>libiksemel.a<br><a href="http://libiksemel.la">libiksemel.la</a><br>libiksemel.so<br>libiksemel.so.3<br>libiksemel.so.3.0.0
<br><a href="mailto:user@foo:/etc/asterisk">user@foo:/etc/asterisk</a>#<br></p>
<p>Is it possible I am having an issue with libiksemel? I can't figure out what the problem can be with Asterisk.</p>
<p>-Nick</p><font size="2">
<p>-----Original Message-----</p>
<p>From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [</p></font><a href="mailto:asterisk-users-bounces@lists.digium.com"><u><font color="#0000ff" size="2">mailto:asterisk-users-bounces@lists.digium.com
</font></u></a></div></font><font size="2">] On Behalf Of Klaverstyn, David C
<p>Sent: Tuesday, March 06, 2007 8:44 PM</p>
<p>To: Asterisk Users Mailing List - Non-Commercial Discussion</p>
<p>Subject: RE: [asterisk-users] gtalk2voip and Asterisk</p>
<p>After upgrading to Asterisk 1.4.1 from 1.4.0 it just worked for me.</p>
<p>There must have been a bug in 1.4.0. I have successfully connected to a Gmail and MSN instant message client.</p>
<p>-----Original Message-----</p>
<p>From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a></p>
<p>[</p></font><a href="mailto:asterisk-users-bounces@lists.digium.com"><u><font color="#0000ff" size="2">mailto:asterisk-users-bounces@lists.digium.com</font></u></a><font size="2">] On Behalf Of Mani Sridhar
<p>Sent: Saturday, 3 March 2007 8:44 PM</p>
<p>To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a></p>
<p>Subject: Re: [asterisk-users] gtalk2voip and Asterisk</p>
<p>hi,</p>
<p>i was able to get this working with google talk.</p>
<p>i entered <a href="mailto:myusername@gmail.com">myusername@gmail.com</a> using the <a href="http://gtalk2voip.com">gtalk2voip.com</a> website's "invite" </p>
<p>box, and as a result, saw a request from <a href="mailto:service@gtalk2voip.com">service@gtalk2voip.com</a> to be added as a buddy in my google talk contact list. i accepted the request.</p>
<p>in my asterisk dialplan, i have this entry...</p>
<p>exten => 3501, 1, Dial(<a href="mailto:SIP/myusername_at_gmail.com@gtalk.gtalk2voip.com">SIP/myusername_at_gmail.com@gtalk.gtalk2voip.com</a>)</p>
<p>this allows any extension in my asterisk box to dial 3501, and <a href="mailto:myusername@gmail.com">myusername@gmail.com</a> receives the call on the google talk client. the call is established, and voice quality is good.
</p>
<p>to allow a call from google talk client for <a href="mailto:myusername@gmail.com">myusername@gmail.com</a>, i opened a chat window to the buddy <a href="mailto:service@gtalk2voip.com">service@gtalk2voip.com</a> and typed "call
<a href="mailto:ABC@mysipdomain.com">ABC@mysipdomain.com</a>", and this made extension ABC on my asterisk box start ringing. again, the call was established, and audio was ok.</p>
<p>as far as asterisk is concerned, this is a SIP call. bottomline - it's a</p>
<p>good alternative to using the native jabber/jingle library in asterisk</p>
<p>1.4 .</p>
<p>in fact, i haven't been able to get asterisk to successfully set up a call to googletalk using the chan_gtalk module . i am inside a NAT-ed LAN, and audio works in one direction only for the asterisk (SIP) - gtalk call.
</p>
<p>anyone else got asterisk - googletalk using chan_gtalk working?</p>
<p>></p>
<p>>Message: 10</p>
<p>>Date: Fri, 02 Mar 2007 19:07:41 +0200</p>
<p>>From: Cosmin Prund <<a href="mailto:cosmin@adicomsoft.ro">cosmin@adicomsoft.ro</a>></p>
<p>>Subject: Re: [asterisk-users] gtalktovoip and Asteirsk</p>
<p>>To: Asterisk Users Mailing List - Non-Commercial Discussion</p>
<p>> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>></p>
<p>>Message-ID: <<a href="mailto:45E859DD.8080103@adicomsoft.ro">45E859DD.8080103@adicomsoft.ro</a>></p>
<p>>Content-Type: text/plain; charset="iso-8859-1"</p>
<p>></p>
<p>>I don't think it works. I tried calling my own yahoo messenger ID with</p>
<p>>no success: it rings a number of times and then it goes to some sort of</p>
<p>>voice mail.</p>
<p>>And I did "invite" the user they specified to my yahoo list, I also</p>
<p>>entered my yahoo id into the registration form on the site.</p>
<p>>I used a extensions.conf command like this for the try:</p>
<p>></p>
<p>>exten => 641,1,Dial(<a href="mailto:SIP/mabojambofake@yahoo.gtalk2voip.com">SIP/mabojambofake@yahoo.gtalk2voip.com</a>)</p>
<p>></p>
<p>>(and yes, that's one of the yahoo ID I tryed with, and I don't think it</p>
<p>>exists! )</p>
<p>></p>
<p>>Klaverstyn, David C wrote:</p>
<p>> ></p>
<p>> > Has anyone managed to get gtalktovoip working at all? If so please</p>
<p>> > explain.</p>
<p>> ></p>
<p>> ></p>
<p>> ></p>
<p>> > </p></font><a href="http://www.gtalk2voip.com/faq.shtml"><u><font color="#0000ff" size="2">http://www.gtalk2voip.com/faq.shtml</font></u></a><font size="2">
<p>> ></p>
<p>> ></p>
<p>> ></p>
<p>> ></p>
<p>> ></p>
<p>> > *2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP</p>
<p>?*</p>
<p>> ></p>
<p>> > A: This is a major feature of our gateway and it is very easy.</p>
<p>> ></p>
<p>> > o GTalk: <a href="mailto:user@domain.com">user@domain.com</a> can be reached by calling to</p>
<p>> > <a href="mailto:sip:user_at_domain.com@gtalk.gtalk2voip.com">sip:user_at_domain.com@gtalk.gtalk2voip.com</a></p>
<p>> ></p>
<p>> > o MSN: <a href="mailto:user@domain.com">user@domain.com</a> can be reached by calling to</p>
<p>> > <a href="mailto:sip:user_at_domain.com@msn.gtalk2voip.com">sip:user_at_domain.com@msn.gtalk2voip.com</a></p>
<p>> ></p>
<p>> > o Yahoo: <a href="mailto:user@yahoo.com">user@yahoo.com</a> can be reached by calling to</p>
<p>> > <a href="mailto:sip:user_at_yahoo.com@yahoo.gtalk2voip.com">sip:user_at_yahoo.com@yahoo.gtalk2voip.com</a></p>
<p>> ></p>
<p>> ></p>
<p>> ></p>
<p>> ></p>
<p>------------------------------------------------------------------------</p>
<p>> ></p>
<p>> > _______________________________________________</p>
<p>> > --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --</p>
<p>> ></p>
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