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...and give me feedback. This implementation was tested on Windows 7 running Mingw and Ubuntu 10.04. By the way, do you guys have any idea, why my binary on Mingw gets that huge (about 2.5 MB)? I'm using "./configure" and "make", nothing more.. Best, Daniel Schall -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://www.tinc-vpn.org/pipermail/tinc-devel/attachments/20100720/f6aa09e3/attachment-0001.htm> -------------- next part -------------- A non-text attachment was scrubbed... Name: tinc-1.0.13.MULTICAST.tar.gz Ty...

...2012 server running Microsoft?s Hyper-V 2012 server virtualization software and advice on building this on AMD 16 core or Intel 8 core processors where we want to put up a 64 core VM. We have not built this yet, we are seeking advice on whether R will run in this environment. Thank you; Matthew Schall Matthew at Catalysis.com catalysis? 1601 East John Street ? Seattle, Washington ?98112 matthew at catalysis.com ? 206.860.2534 (direct) ? catalysis.com

We have the caller id with name option enabled with our provider, however, our polycom 501 phones will only display the number of the incoming call. Is there a way to see the callerid name from the cli when the call is coming in (like a print in the dial plan)? I'm not sure if the problem is with asterisk or our phones. I did turn on the calleridpres option in zapata, but I'm unsure what

...load the sources. Unfortunately, my enhancements are based on a rather old git-checkout from Guus. The version should run under windows and Debian/Ubuntu. Best, Daniel -----Original Message----- From: folkert [mailto:folkert at vanheusden.com] Sent: Monday, November 22, 2010 6:22 PM To: Daniel-Schall at web.de Subject: local address announcements Hi, Can you please e-mail me the local address announcements patch? Or tell me where to get it? I have a setup in which I can test/use it. Thanks, Folkert van Heusden -- MultiTail er et flexible tool for ? kontrolere Logfiles og commandoer. Med f...

...= Node3 (this line is of course missing on Node3) Device = {.. Windows UUID.. } DeviceType = tap Mode = switch Node adresses are assigned using a DHCP server on Node3. I'd be happy hearing from you guys. Best regards Daniel Schall -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://www.tinc-vpn.org/pipermail/tinc/attachments/20100506/5f1bec65/attachment.htm>

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone

Hello all, I currently have all outgoing calls set to mask the caller id so it will always appear to be coming from our main number. The problem I'm having though, is with both the call detail in mysql and with the automon (recording) feature. It shows the originating number as the number I masked it to, rather than the actual person calling. How can I go about having both the destination see

Hi, One of my users is in trouble with his polycom phone hooked to an asterisk server. The phone works fine for a few days, and then disappears from the registered sip peers in asterisk. The user is able to place outbound phone calls, but can't receive incoming calls until the network plug is unplugged/plugged. Working line XXYYZZAA24/XXYYZZAA24 10.50.5.186 D A 5060

I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type "reload", it resets to "off" again. I've tried setting the clearglobalvars=no as well as just

Hopefully a fairly easy question for the group... I have a queue which should contain about 10 agents (it will be all the phones in the office). This office is remote, so I would like to add their sip phones into the queue remotely. Also, if the system ever gets reloaded or rebooted, I need those agents to remain in the queue. Question: 1) How do you remotely add agents to their respective

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

...in the last months 3) an improved PMTU discovery with predictable packet sizes for the probe packets By the way, I've re-formatted the code to better suit your coding style. Best, Daniel -------------- next part -------------- commit 50a9f9c9d055dbd20d81e7072f4a059e17c68118 Author: Daniel Schall <tinc-devel at mon-clan.de> 2011-01-03 18:46:51 Committer: Daniel Schall <tinc-devel at mon-clan.de> 2011-01-03 18:46:51 Parent: 4b8a5993036fccc2108fcc2550649d9b78fb1ab7 (Update the NEWS.) add iphlpapi and other headers to have.h and configure.in add flags to vpn_packet ---------------...

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars

I'm sure this has been discussed many times, but I have a question about hoteling. My understanding would be this: A phone sitting on a desk. A user hits 9000 and it asks what extension you'd like to become. You type "1001" and then it asks for your password. You type 1234, and it says you're "logged in". You now are accepting calls at your phone and you're

Hi Guus, while checking the source code, I stumbled upon PMTU Discovery. I've got a question regarding the process of sending/receiving PMTU packets. As I understand, the packet flow is like this: 1 .Tinc creates a packet with a specific payload length to send it as an PMTU probe. (The data part is just some random bytes.) 2. This packet gets compressed and sent

Hi all, TL;DR: when Tinc 1.1 release? I plan to use Tinc for my GSoC project which basically simplifies setup of a Tinc mesh providing IPv6 to nodes in community mesh networks. As I'm new to Tinc I don't know it's history and the changes from 1.0 to 1.1, but it seems to have at least a different syntax in some cases. To make and keep it simple for users, I'd like to stick

hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because