similar to: NAT working outbound with Asterisk and ATA-186 phones

So, Tilghman has put a particularly useful patch in the bugtracker: streaming music-on-hold is now supported. You can now specify .mp3 streams to be played back as MOH in the various places where MOH is used. Hopefully, Mark will install into the main CVS tree shortly. http://bugs.digium.com/bug_view_page.php?bug_id=0000413 This allows you to use the very sophisticated mp3 streaming audio

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

http://www.cisco.com/univercd/cc/td/doc/product/voice/ata/atarn/atarn3_0.htm LOVE IT.. call transfers work better but not totally as expected yet. I'm still tweaking the configs. Also bitaid will help alot. bkw

I must be doing something incorrectly, or something is wrong with ATA-186 reINVITEs in SIP. Perhaps someone more enlightened than me can lend me a hand. I have been attempting to get two SIP phones to reINVITE to each other, and I've been unable to think of or uncover the correct method. The calls always go through the Asterisk server, no matter what I try. I've simplified things

When I try to connect to * using a Cisco ATA 188 configured with a MGPC firmware (v3.1.1), I just keep getting this message every 30 seconds or so : May 10 10:08:21 NOTICE[7913]: chan_mgcp.c:1509 find_subchannel: Gateway '192.168.1.27' (and thus its endpoint '*') does not exist Using tcpdump, I have checked that the ATA188 (with IP 192.168.1.27 and port 2427) sends UDP packets to

Dare I ask? I'm new to Asterisk. I like what it has to offer, however, I'm having a hard time finding documentation to configure it correctly. Can anyone tell me where I can get good Asterisk documents? Here's what I have put together: http://www.simplifiednetwork.com/asterisk. Seng -------------- next part -------------- An HTML attachment was scrubbed... URL:

toy buy my first cisco 186 but when i read this page http://www.djernes.org/~shawn/ata186.htm i cant find in my dev page some parameters just like " UseSIP " what i need to do to show this parameters Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030626/42e66005/attachment.htm

Hi, You need to set the DialPlan parameter to allow the proper number of digits to be collected, for all types of numbers used in your system. I believe that the factory default value would work for long numbers beginning 0011, but your unit was probably previously configured for a different environment or country. Below is an extract from the example in my H.323 firmware; I believe that

I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry

The Wine development release 4.0-rc5 is now available. What's new in this release (see below for details): - Bug fixes only, we are in code freeze. The source is available from the following locations: https://dl.winehq.org/wine/source/4.0/wine-4.0-rc5.tar.xz http://mirrors.ibiblio.org/wine/source/4.0/wine-4.0-rc5.tar.xz Binary packages for various distributions will be available

After browsing "Remote timing attacks are practical" (Boneh & Brumley, <http://crypto.stanford.edu/~dabo/abstracts/ssl-timing.html>), I wonder if it might be a good idea to add calls to RSA_blinding_on() before the OpenSSL RSA decryption routines are invoked. The issue is not a LAN-only issue, BTW. Packet delay variation is usually higher in LANs than in WANs. -- Florian

I wonder if someone out there could loan me a peek at their sip.conf? I have conflicting advice, for instance, about whether or not to use "nat=1" and also whether or not the ATA should be registering with the instance of asterisk it is going to be using to dial out. Thanks in advance. B.

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An

Hi guys, sorry to be iterating this on the list once more, but I'm not able to get this stuff to work as I'd expect. So far, I've always managed to keep it out of NAT environments :-> My home LAN is NATed by a simple Draytek router. In the home LAN is an ATA186 with SIP. On the internet (public) is an Asterisk server. I have nat=yes in the sip.conf and the connectmode is set

Hi all,

Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??