similar to: Asterisk (meetme) and SMP/HT OK?

Hi, I have a grandstream HT 503 ATA which seems to support TLS/SRTP. I generated the keys and certificates for both the server and the device as mentioned at https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial I then copied the grandstream.crt and grandstream.key over to the ATA. Unfortunately, the device is refusing to register and in asterisk cli i see SSL certificate ok ==

Hello there, I just wasted some time setting up a Grandstream HT-488 to be used with Asterisk, so I thought I'd share the experience by writing a small tutorial at: http://astrecipes.net/index.php?n=338 Most tutorials I came across were for old versions of the firmware, and I spent too much time trying to figure out what the Forward to VoIP option wanted.... :-( I hope this helps,

Just to tease people I'll be post a URL to my review of the HT-286 by thursday..

Hi List, Just received a HT-286 Analog Telephone Adapter. This device is allows the user to take a standard analog telephone set and connect it to a VoIP/SIP based gateway. A low cost way of having PSTN devices make use of VoIP/SIP services. For example you could take a credit card processing machine and punch it into a location where you have broad band access. You could take a FAX

Hi, Can I use HT-486 as VoIP Gateway together with Asterisk? Are there any success experiences? -- Best Regards, Miroslav Nachev

Does anyone know if the HandyTone 486 has the option to turn the two ethernet ports into either a switch/hub, or does it have to do NAT ? Thank you, Steve Maroney

I am trying to get Asterisk 1.6.2.5 working with a Grandstream HT-503 ATA. The FXO part is giving me fits. Every call I try to make to the FXO port outbound I get 403 Forbidden coming back. I've been through every configuration setting I can see, and Uncle Google is not helping me much. I updated the firmware to the current version, and that didn't help. If anyone has this working, I

Hi, I have HT Feature available on my Dell Server R720. Please help me understand if HT is enabled on the server, will there be a performance impact. What are the deciding factors to enable HT on the servers. Regards Kaushal

I would like to build a package for the HT HG-U133+ PM arrays from affy, but I can't find any good documentation on how to go about it. Naively using makecdfenv's make.cdf.package() causes R to seg-fault. I'm unfamiliar with the CDF format as such, but I'm guessing that it's changed somewhat because the PM arrays no longer have P/A and mismatches. I'm looking to build

Hi, Do we need HT enabled on CentOS release 5.5 (Final) and Kernel Version :- 2.6.18-194.el5 Please help me understand the pros and cons of having HT enabled or disabled on the Server Dell R 710. cat /proc/cpuinfo -> http://fpaste.org/K2dT/ Do let me know if anyone needs more information. Regards, Kaushal

I have 2 Grandstream HT-286 devices and an Asterisk server. The * Server is not using NAT and has port 5060 opened up. One HT-286 is using traditional NAT and the other HT-286 is behind a residential DSL router/firewall. I have the HT-286 setup as the "DMZ Host" in the router/firewall so that all incoming connections are forwarded to the HT-286. HT-286-1 ====== NAT FW ====== * Server

Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator

At wiki page: http://wiki.xapian.org/Omega I added a comment that ht://Dig looks like dead. Does anybody really use it? >From brief glance at docs I had a feeling it is not easy to configure. Maybe better crawler is GNU wget? Mature, stable, maintained? -- Peter Masiar

Dear R-help, Could someone please try to explain this paradox to me? What is more likely to show up first in a string of coin tosses, "Heads then Tails", or "Heads then Heads"? ##generate 2500 strings of random coin flips ht <- replicate(2500, paste(sample(c("H", "T"), 100, replace = TRUE), collapse =

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

Hi there! I just fresh-installed Xen on new server, Xeon E5-2620, and noticed following thing: node1 ~ # xl info -n host : node1 release : 3.8.8 version : #1 SMP Tue Apr 23 20:54:17 MSK 2013 machine : x86_64 nr_cpus : 6 max_cpu_id : 5 nr_nodes : 1 cores_per_socket : 6 threads_per_core

Hi All, I'm still having trouble trying to figure out if it is possible to define (in the dial plan) a context for meetme? I'm using 1.4.4 with dialplan logic of: exten => 123,1,Meetme(,Msa,) This defaults to conferences defined within the rooms context of meetme.conf Is it possible to specify another context as with voicemail? Or can any one think of another

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

hi@all I have the following problem: With asterisk 1.09 the grandstream's registers fine with both ports, with version 1.2.1 (the newest port on freebsd) I get "Unauthorized" SIP messages from the 2nd port. The ports are configured identically, the only difference is the sip and rtp port. On the first port the sip port is 5060 on the second 5062. The rtp on the first 5004 on the

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17