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For those tired of deleting old phone calls, texts, etc. from their GV account by hand via the web interface, here is a patch to add a deleteForever() method to pygooglevoice: https://github.com/rxcomm/pygooglevoice/commit/62c4130c902812f7e0cb6bd3ec1afef7b89d6bbc You can then do the...

To register your vanity sip id with your Asterisk box, you will need to create two trunks (one custom), an outbound route and an inbound route. These are very similar to those required for a sip2sip.info registration. I will describe the differences below. First, create a SIP trunk named...

First post in a long time... be gentle, guys. I gave the cloaked vanity sip id a try. It worked with some clients and not others (as noted above). I really wanted a vanity sip id that worked all the time. I also wanted one that did not require me to accept anonymous inbound sip calls and did...

Either way should work. Simplest would be to just replace the string, but changing the variable would work as well. According to Polycom's recommended approach, you are supposed to do this in a sip_local.cfg file that is loaded after (and therefore overwrites) sip.cfg. The reason is, they do...

Edit the SNTP tag in your sip.cfg file. Replace the line: tcpIpApp.sntp.gmtOffset="0" with whatever your offset should be in seconds. The parameter is negative for offsets to earlier local times than UTC. Dave

I am having an interesting problem with the Pirelli LP-D10 phone and OpenVPN. First - the Pirelli works fine when I register with Asterisk (both 1.4 and 1.8) outside of OpenVPN. A bit about my OpenVPN network: I have a tunneled (not bridged) network with a variety of clients connected to my...

There's a good discussion of the Asterisk AGI interface in the Asterisk book 2nd ed. You can download a (free) copy here: http://astbook.asteriskdocs.org/ . Chapter 9 has several examples of how to do this - of varying complexity and scripting language. It's pretty well written, too. You can...

You can do this with iptables This actually works very well and can be done using iptables. You can find instructions for internet connection sharing with iptables at https://help.ubuntu.com/community/Internet/ConnectionSharing . See the section on Ubuntu Internet Gateway Method (iptables)...

How's the battery life? Dave

I don't believe there is a way to do this. You can seamlessly integrate GV voicemail instead of asterisk as your voicemail provider per the link in the first post, however. On another note, here is a vm.php file that implements click-to-dial for returning calls on polycom phones. <?php...

You need to install a few packages (I always forget what I have previously installed by hand!). yum install curl curl-devel php-xml php-pear php-devel Then build php-json (not sure why this isn't a package in CentOS) pecl download json pear install json-1.2.1.tgz And finally, add the file...

Displaying the GV vm transcripts is not an asterisk thing - it is web-based. So it will work with your PIAF server of any version (1.4, 1.6, or 1.8). Of course, the GV/asterisk integration will only work with asterisk 1.8. Dave

http://humbuglabs.org Haven't used them, but the price is right and they've got a nice demo on their page. Dave

Merlyn - I'm guessing one of two things is wrong. 1) Make sure you have created a user ipphone with password 15354all for your ftp server. 2) Since the phone uses passive mode for the ftp transfer, it uses random ports for the data transfer. You need to make sure none of those ports are...

Be sure each sip phone operates on a separate port. Then the NAT router won't get confused as to which device it needs to send packets to. Dave

Displaying GV voicemail transcripts on my phone's microbrowser is something I have been interested in doing for a while. It turns out that aaronpk has written a nice php API for GV that can be used for this application quite easily. Here's how: 1) Get the Google Voice PHP API from here. 2)...

Hi Merlyn. These were my instructions originally, so if they aren't clear, its probably my fault. The basic idea is to copy the FirmwareUpdateRequest.acfg file from the newer directory to the older directory. You've already made the backups (FirmwareUpdateRequest.acfg.old) in each directory...

Actually - the second mailbox (702) in the above example will work if you symlink it to the 701 mailbox. That way, you can eliminate line 2 of the crontab instructions and reduce your banging on Google Voice a bit. You get seamless integration of all aspects of GV voicemail - nice. Dave

Google Voice MWI with PIAF I went through and got this working with multiple GV trunks and PIAF. Thought I'd post a HOWTO. Note that for this to work, you need to have your dialplan set up per this thread...

Thanks Fredl! This works great. Dave