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PlugPBX Admin
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« Reply #1 on: January 28, 2011, 04:48:34 AM » |
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Nice find!
I will put my 1.5 efforts on hold, dust off the 1.8 based build scripts and add the patching logic and give this a whirl. Perhaps asterisk 1.8 can go back in woho!
Cheers!
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-Greg
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toolbox123
Newbie
Posts: 16
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« Reply #2 on: January 28, 2011, 09:44:05 AM » |
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Did you do this with 1.8.2.2? |
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twinclouds
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« Reply #3 on: January 28, 2011, 09:53:18 AM » |
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Actually, I did it on 1.8.2.3 (1.8-current).
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toolbox123
Newbie
Posts: 16
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« Reply #4 on: January 28, 2011, 01:49:02 PM » |
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What do I need to do to remove the python script and use the built-in Gtalk of A1.8? I have built and installed A1.8.2.3 on my DockStar.
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twinclouds
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« Reply #5 on: January 28, 2011, 03:23:05 PM » |
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Are you using FreePBX? I am using Asterisk 1.8 alone without FreePBX. For Asterisk 1.8, you can read Mario's postings and his conf files (mainly jabber, gtalk, extensions and sip). You can find information here: http://supermario-world.blogspot.com/2010/11/asterisk-18-and-native-google-voice.htmlhttp://supermario-world.blogspot.com/2010/12/outbound-google-voice-dialing-proper.htmlIt is very simple. For FreePBX you might need to do some more digging because I have not try that yet. Maybe you can get the information from Ward's website, e.g. this one:http://nerdvittles.com/, or googling around to find instructions. I will let you know if I come to it but I really don't need this. For my family use, Asterisk 1.8 alone is good enough (for making gtalk calls and some voip service providers, e.g., call with us.) |
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twinclouds
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« Reply #7 on: January 28, 2011, 04:16:10 PM » |
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If it is the case, you don't need anything about FreePBX. Assuming you have already compiled and installed Asterisk 1.8.2.3. You have already done the hard part. (However, in my how-to, I don't think I compiled asterisk but just did a apt-get asterisk, right? So you must have done the complete installation.) Now, with Asterisk 1.8.2.3 installed, you just replace the four files that I mentioned before with the corresponding files by Mario. Edit the files with the information of your gtalk and that is it. You can use your sip phone to login and making calls. At least that was I did. BTW, I assume you have checked during menuconfig that the modules chan_gtalk and res_jabber are there. If not, you probably should do an "apt-get install libiksemel-dev" before that.
When I get chance this weekend, I can write another how to to help others. It may take more than 20 minutes but is actually simpler than my previous how-to.
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toolbox123
Newbie
Posts: 16
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« Reply #8 on: January 28, 2011, 05:18:47 PM » |
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I am lost. Which 4 files? Don't I need to keep sip.conf as is so that my ATA can be registered with sipgate?
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twinclouds
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« Reply #9 on: January 28, 2011, 05:53:46 PM » |
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Four files are jabber, gtalk, extensions and sip (.conf). No you don't need sipgate any more unless you want to use it for other purposes, but not gtalk. If you need to keep Sipgate, you can simply merge the two sip.conf files. I think you can keep the old sip.conf file as long as the context is right.
One possibility to see if gtalk works is you can save your previous 4 files and replace them with Mario's temporary to test out. Then you can copy them back or merge them. Just make sure you know what you are doing.
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« Last Edit: January 28, 2011, 05:59:34 PM by twinclouds »
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toolbox123
Newbie
Posts: 16
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« Reply #10 on: January 28, 2011, 11:20:58 PM » |
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Thanks. Looks like I have it working now. Will test it some more but it is great so far.
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« Last Edit: January 29, 2011, 10:07:33 AM by toolbox123 »
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twinclouds
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« Reply #11 on: January 29, 2011, 10:59:29 AM » |
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Great! I am working a complete how to for installing Asterisk 1.8 now. Hope I will complete it this weekend and it can be helpful for others.
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« Last Edit: January 30, 2011, 11:43:44 AM by twinclouds »
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toolbox123
Newbie
Posts: 16
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« Reply #12 on: January 29, 2011, 01:16:42 PM » |
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I did another installation on my pogoplug and is not 100% working. Outgoing calls are fine but no incoming call. For incoming call, I got: -- Executing [MyGV@gmail.com@google-in:1] GotoIf("Gtalk/+1AAANNNXXX-b055", "0?bridged") in new stack
-- Executing [MyGV@gmail.com@google-in:2] NoOp("Gtalk/+1AAANNNXXX-b055", "Callerid +1AAANNNXXX@voice.google.com/srvres-xxxxx==") in new stack
-- Executing [MyGV@gmail.com@google-in:3] Set("Gtalk/+1AAANNNXXX-b055", "CALLERID(num)=+1AAANNNXXX") in new stack
-- Executing [MyGV@gmail.com@google-in:4] Set("Gtalk/+1AAANNNXXX-b055", "CALLERID(name)=") in new stack
-- Executing [MyGV@gmail.com@google-in:5] Dial("Gtalk/+1AAANNNXXX-b055", "SIP/101, 180, D(:1)") in new stack
[Jan 29 12:12:17] WARNING[16753]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [MyGV@gmail.com@google-in:6] Bridge("Gtalk/+1AAANNNXXX-b055", ", p") in new stack
[Jan 29 12:12:17] WARNING[16753]: features.c:5528 bridge_exec: Bridge failed because channel does not exists or we cannot get its lock
-- Auto fallthrough, channel 'Gtalk/+1AAANNNXXX-b055' status is 'CHANUNAVAIL'
UPDATE: Don't know what happen. Now incoming calls are working. |
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« Last Edit: January 29, 2011, 01:25:59 PM by toolbox123 »
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