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<h2><a href="https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google?focusedCommentId=18416017#comment-18416017">Calling using Google</a></h2>
<h4>Comment edited by <a href="https://wiki.asterisk.org/wiki/display/~mdavenport">Malcolm Davenport</a>
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<h4>Changes (3)</h4>
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<tr><td class="diff-deleted-lines" style="color:#999;background-color:#fdd;text-decoration:line-through;">I am trying to connect to setup Google Voice calling: <br></td></tr>
<tr><td class="diff-changed-lines" ><span class="diff-deleted-words"style="color:#999;background-color:#fdd;text-decoration:line-through;"> </span> <span class="diff-added-words"style="background-color: #dfd;">edited...</span> <br></td></tr>
<tr><td class="diff-deleted-lines" style="color:#999;background-color:#fdd;text-decoration:line-through;">In *gtalk.conf* file <br> <br>[general] <br>context=local <br>allowguests=yes <br>bindaddr=0.0.0.0 <br>externip=216.208.246.1 ; SERVER IP?? <br> <br>[guest] <br>disallow=all <br>allow=ulaw <br>context=local <br>connection=asterisk <br> <br>*jabber.conf* <br> <br>[general] <br>autoregister=yes <br> <br>[asterisk] <br>type=client <br>serverhost=talk.google.com <br>username=your_google_username@gmail.com/Talk <br>secret=your_google_password <br>port=5222 <br>usetls=yes <br>usesasl=yes <br>statusmessage="I am an Asterisk Server" <br>timeout=100 <br> <br> <br>sip.conf <br> <br>[malcolm] <br>type=peer <br>secret=my_secure_password <br>host=dynamic <br>context=local <br> <br>*extensions.conf* <br> <br>exten => s,1,Answer() <br>exten => s,n,Wait(2) <br>exten => s,n,SendDTMF(1) <br>exten => s,n,Dial(SIP/malcolm,20) <br> <br>I am running asterisk 1.8. On the SIP client first i am trying to connect to user malcom, secret password and asterist server IP (without port number). I tried to connect using Xlite 4, i am getting a 408 timeout error after attempting to connect for a long time. <br> <br>So, i gave up try the Google voice connection. I just wanted to test my SIP connect to my asterisk server <br> <br>For the *sip.conf* i added additional users <br> <br>[12345] <br> <br>type=friend <br>secret=blah <br>auth=md5 <br>nat=yes <br>host=dynamic <br>reinvite=no <br>canreinvite=no <br>qualify=1000 <br>dtmfmode=inband <br>callerid="Test" <1111> <br>disallow=all <br>allow=gsm <br>context=theflintstones <br> <br> <br>in extensions.conf i added context <br> <br>[theflintstones] <br>exten => _[123456789]XXXX,1,NoOp("call for "${EXTEN}) <br>exten => _[123456789]XXXX,2,Dial(SIP/${EXTEN},60,tr) <br>exten => _[123456789]XXXX,3,Congestion <br> <br>Why is Xlite or other SIP client cannot connect to the 12345 extension? Where can i find out about SIP connection logs? It would be great if someone explains where and how to trouble shoot SIP by looking at the log file. I read somewhere that if your SIP client is located in different machine from asterisk server i need to configure *sip_nat.conf* <br> <br>nat=yes <br>externip=10.0.0.2 ; Is this SIP clients IP? explain <br>localnet=asterisk_server_ip/255.255.255.0 <br> <br>I tried with and without the sip_nat.conf still my SIP client (Xlite) cannot connect to the asterisk server assuming if i get connected in the CLI> prompt it will show up. Can someone have a test commands to see if asterisk server is running properly. <br> <br>Please help. Please, I really appreciate if you could give my some pointers. I am getting frustrated with Asterisk unable to get basic SIP connections to work. <br></td></tr>
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<p>edited...</p>
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