tag:blogger.com,1999:blog-7734469747818641247.post5550523079166535700..comments2023-06-11T08:24:45.383-07:00Comments on PC Problems Blog: HOWTO: Use Google and Asterisk For Free Home Telephone ServiceUnknownnoreply@blogger.comBlogger24125tag:blogger.com,1999:blog-7734469747818641247.post-70823941393373640372011-10-21T12:01:20.528-07:002011-10-21T12:01:20.528-07:00How could this be implemented into an home alarm s...How could this be implemented into an home alarm system?Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-81295365582751890272011-09-10T08:21:25.991-07:002011-09-10T08:21:25.991-07:00I followed this excellent guide and set up my aste...I followed this excellent guide and set up my asterisk server working great for calling US numbers and receiving calls. But I always got busy tone when I dial international numbers. How should I change the settings? In addition, I noticed that after I dial a number, there is a ~8 seconds silence before I hear a ringer tone. Can we shorten this? Thanksmdhttps://www.blogger.com/profile/14103165318350429282noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-46390757391672618532011-08-26T10:26:44.243-07:002011-08-26T10:26:44.243-07:00Alright, so I did some of my own troubleshooting a...Alright, so I did some of my own troubleshooting and went back through the instructions again. I noticed using my Debian 6.0.2.1 install disk and software manager I was only using Asterisk 1.6 so I upgraded to 1.8. By the time it was installed and the .conf files were updated again it worked like a charm! Very excited to have free home phone service using an old linux box and the ATA part you suggested. Thank you very much for the tutorial. Well done.Anonymoushttps://www.blogger.com/profile/13945267813376205301noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-84499873248305970092011-08-22T20:02:13.820-07:002011-08-22T20:02:13.820-07:00I followed all the instructions, the Grandstream i...I followed all the instructions, the Grandstream is registered and I doubled checked all the instructions but when I try to dial a number I hear a dial tone, dial the 1-xxx-xxx-xxxx number and I don't hear anything. No ringer, no busy tone, etc. What did I do wrong? I'm sure I'm close but can't figure out why it won't dial out. Any help would be greatly appreciated!Anonymoushttps://www.blogger.com/profile/13945267813376205301noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-37312471257968157422011-07-09T23:53:42.310-07:002011-07-09T23:53:42.310-07:00I was getting the following error in my /var/log/a...I was getting the following error in my /var/log/asterisk/messages file and in the asterisk console.<br /><br />db.c: Unable to open Asterisk database '/usr/lib/asterisk/astdb': Permission denied<br /><br />I don't know if anyone else has run into this problem, but changing the group on the /usr/lib/asterisk folder to asterisk fixed it for me.Jeremyhttps://www.blogger.com/profile/09252455428870735907noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-75030676330007877692011-07-08T11:44:33.120-07:002011-07-08T11:44:33.120-07:00everything works great thanks for the guide. I was...everything works great thanks for the guide. I was wondering how to get voicemail working. When I set up asterisk to handle the voicemail it goes back to the googlevoice voicemail. How I want it to work is to use asterisk as the voicemail service so I can check it from the phone.l3ioHazardXhttps://www.blogger.com/profile/14850952236948916571noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-75754784557598985562011-06-26T21:41:31.237-07:002011-06-26T21:41:31.237-07:00I've gotten to the end and have Registered: Ye...I've gotten to the end and have Registered: Yes Unfortunately, when calling out I get a busy tone no matter the number. When calling in the phone does not ring and I have to leave a message on google voice. Any ideas?Robertnoreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-27580411971340925382011-06-10T17:05:14.879-07:002011-06-10T17:05:14.879-07:00Thank you so much for this article. Worked great ...Thank you so much for this article. Worked great on Ubuntu 10.04. To let other readers know, you can simply use this PPA for Asterisk, and it still works perfectly fine. No need to compile from source: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-APT%28Debian%2FUbuntu%29dbray925https://www.blogger.com/profile/15401227917688595065noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-26883286275757100302011-05-19T11:08:53.288-07:002011-05-19T11:08:53.288-07:00I got this setup and can dial out fine. When I re...I got this setup and can dial out fine. When I receive a call, I get Google Voice asking me to press 1 to accept the call, and no matter what I hit it won't allow the connection. I don't have any other devices tied to this voice account, is that required? Can you point me to something to resolve this?havok3114https://www.blogger.com/profile/03495381532304306853noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-84388855773743630292011-03-29T18:50:00.611-07:002011-03-29T18:50:00.611-07:00Thanks! I did struggle until I noticed a setting o...Thanks! I did struggle until I noticed a setting on my grandstream box- I had to switch NAT transversal from "Yes, STUN server is..." to "No"<br /><br />For a guy who didn't know squat about Ubuntu- you've done VERY well with your instructions paired with the others that you referenced. thanks!Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-27605870178381900912011-03-25T00:17:49.713-07:002011-03-25T00:17:49.713-07:00TSR.
Thank you for that helpful info. I did this ...TSR.<br />Thank you for that helpful info. I did this in my sip.conf as well now, and it works great.HeadSheezhttps://www.blogger.com/profile/16363491691041497770noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-6104744336605196982011-03-24T09:42:52.532-07:002011-03-24T09:42:52.532-07:00If you want a simpler way of getting the "Pre...If you want a simpler way of getting the "Press 1 to answer..." prompt to work add the line<br /><br />dtmfmode=inband<br /><br />to sip.conf. This will also allow the voice menus of a business you call to work, which didn't for me under your setup.TSRnoreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-36071504168510775852011-03-23T12:39:43.290-07:002011-03-23T12:39:43.290-07:00Can Asterisk run on a router with openwrt installe...Can Asterisk run on a router with openwrt installed? Here is a link on how to install Asterisk on openwrt: http://lestblood.imagodirt.net/archives/83-Asterisk-on-OpenWRT.htmlioanhttps://www.blogger.com/profile/03015305588473245376noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-83706436366091102172011-03-22T16:06:30.774-07:002011-03-22T16:06:30.774-07:00Anon,
I doubt it. The obihai is the only closed b...Anon,<br />I doubt it. The obihai is the only closed box I have seen that can do it, and Cisco is notorious for intentionally limiting their products for lock-in purposes.HeadSheezhttps://www.blogger.com/profile/16363491691041497770noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-78141132673711721332011-03-22T10:27:24.857-07:002011-03-22T10:27:24.857-07:00This looks interesting. Do you think this would al...This looks interesting. Do you think this would also work with a Cisco Call Manager router instead of asterik and a PBX?Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-65970103624824743852011-03-20T13:08:01.840-07:002011-03-20T13:08:01.840-07:00Brad,
You can set up 2 lines using 2 separate goog...Brad,<br />You can set up 2 lines using 2 separate google accounts and then assigning the lines to separate extensions. I recall googling and finding some posts on setting up your extensions.conf file to handle this. I think that gtalk allows for more than one line on a single gtalk account and the obihai obi110 device can handle this somehow, but I have no idea how to do this on Asterisk.<br /><br />Anon,<br />This guide is actually a bit different than the MaximumPC one. For one thing, I am not pushing any front end or pbx-distro, and I give a specific guide about how to set up a specific ATA. Also, the majority of the guide is content on forum posts other people made. I specifically did not want to use PBX In A Flash, FreePBX, etc. I wanted to do it natively in Asterisk so I could get my hands a little dirty and learn something about it.HeadSheezhttps://www.blogger.com/profile/16363491691041497770noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-12702161310305121502011-03-19T10:05:02.114-07:002011-03-19T10:05:02.114-07:00You got this from Maximum PCYou got this from Maximum PCAnonymousnoreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-21824687175869946932011-03-18T21:49:50.678-07:002011-03-18T21:49:50.678-07:00Is it possible to set this up with 2 or more googl...Is it possible to set this up with 2 or more google voice numbers to have multiple lines? I am also interested in setting it up so if a client calls into my business number it picks up right away and can play a prerecorded message and music while "finding me" or basically forwarding to my cell.Bradhttps://www.blogger.com/profile/04169514717539677808noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-60727716994419227912011-03-18T19:36:34.747-07:002011-03-18T19:36:34.747-07:00Love you for this. Amazing writeup. I'm goin...Love you for this. Amazing writeup. I'm going to be doing this but instead of an analog adapter, plugging the information into the new SIP capabilities of Gingerbread on my Android :)<br /><br />Free cell phone calls! :)ThantiKhttps://www.blogger.com/profile/01941608833503343973noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-7894665308214040242011-03-18T10:09:06.812-07:002011-03-18T10:09:06.812-07:00Roman,
The reason is that your gtalk account does ...Roman,<br />The reason is that your gtalk account does not provide a standard SIP line. Asterisk logs in to your gtalk account and pretends like it is a computer using gtalk as far as google knows, and it pretends like it is connected to a SIP line as far as your ATA+phone or IP phone knows. There is an ATA device (obihai obi110) that can do all this internally, but it is not common.<br /><br />David,<br />I'm not sure, but I don't think that would work. The ATA has to have onboard functionality to talk to your Asterisk system being used as a SIP proxy and handle interaction between your phone and the box. For example, your ethernet and router have no idea what it means when you press "2" on a phone wired via ethernet. The box has to handle this interaction. A typical ATA+phone or IP phone is what you need.<br /><br />Unununium,<br />I suggest an unused account because it logs into gtalk/gchat (and even posts a status message about Aterisk). It may cause problems for it or you when both are logged in simultaneously. I haven't had any problems with it myself, but it is probably a good precaution.HeadSheezhttps://www.blogger.com/profile/16363491691041497770noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-73608448055842236282011-03-18T09:35:00.123-07:002011-03-18T09:35:00.123-07:00"1 google voice account that you don't ty..."1 google voice account that you don't typically log into for gmail or chatting"<br /><br />Why does it have to be an otherwise unused account?Unknownhttps://www.blogger.com/profile/16195080548109393707noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-65792784466356988842011-03-18T08:16:36.188-07:002011-03-18T08:16:36.188-07:00I too would very much like to know the answer to R...I too would very much like to know the answer to Roman's question if you would be so kind :).<br /><br />Also, what required features does the HandyTone provide when used in conjunction with the Asterix server? Is is possible to use a product such as this: http://www.amazon.co.uk/Socket-RJ45-Secondary-Telephone-Adapter/dp/B0021JNY38/ref=sr_1_3?ie=UTF8&qid=1300461112&sr=1-3 to achieve the same result?ZeroCool42https://www.blogger.com/profile/08918009744213440893noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-18611126138041622782011-03-18T07:31:51.330-07:002011-03-18T07:31:51.330-07:00could you explain why asterisk is needed? can'...could you explain why asterisk is needed? can't you point ata directly to gv?Romanhttps://www.blogger.com/profile/16751445884237136619noreply@blogger.comtag:blogger.com,1999:blog-7734469747818641247.post-68753316825444472212011-03-17T06:17:15.962-07:002011-03-17T06:17:15.962-07:00http://redd.it/g5bbnhttp://redd.it/g5bbnm-p-3http://www.reddit.com/user/m-p-3noreply@blogger.com