search for: jasomi

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...worried about NAT Tranversal problematic, he is thinking on focus the service on SER, so use SIP clients, but he would like to be able to migrate every user to IAX in a a near future. I have questions about a solution that is NAT Transversal, what beneficits/problems will give me products as JASOMI (why are better than STUN), STUN installation considerations....etc. Also.. Should I consider SIPFOUNDRY instead SER ? If anyone is interested, please send me your hourly rates as well as details about your implication with large scale proyects, with Asterisk; SER &STUN, etc, so I can eva...

...o #define USE_POSIX_THREADS when building --with-pam defined? The autoconfig stuff doesn't test for pthreads libraries, so by default you get threads simulated by Unix processes. Presumably there ought to be an "official" way to specify this. Thanks. -- Bob Bramwell Jasomi Networks (Canada) | This space Ph: 403 269 2938 x155 #310 602 11th Ave SW | intentionally FX: 403 269 2993 Calgary, AB, T2R 1J8 | left blank.

Hi List! does anyone played around with the LCS and Asterisk? Because the LCS is doing no RFC compliant SIP, i wonder if it can work. Google couldn't tell me. If someon heared about that, please let me know. The fact i figured out is that the Border Controler from Jasomi can be used as a gateway from MS-LCS-SIP to regular SIP. But that is not really handy and expensive too. Thank you Volker

We're on track for a release of Asterisk 0.4.0 soon. I'd like to try to see to it that we have squared away our SIP implementation by then, and after that point, try to keep it in tip top shape. In general, I find that SIP is extremely fragile, and every time I try to fix one bug, I end up creating another somewhere. What I need are strategies for verifying that the SIP implementation

...ut_userauth_info_response to be a little more forgiving would that cause any grief, open any security holes, or whatever? Would anyone like to suggest a suitable approach to a fix? Does this sound like a good idea? Constructive criticism appreciated. Cheers, Bob. -- Bob Bramwell Jasomi Networks (Canada) | This space Ph: 403 269 2938 x155 #310 602 11th Ave SW | intentionally FX: 403 269 2993 Calgary, AB, T2R 1J8 | left blank.

One for the gurus.. I have seen there has been a lot of discussion about using SER with Asterisk.. This to me seemed like an over kill becasue it would basically be doing most of what Asterisk is doing anyway unless you create some weird and wonderful config in SER.. Anyway, I decided to go and have a quick read through the SER docs and in the section about NAT they say that the best way to

I know that NAT is something no one really likes to talk about, but does anyone know how work with it elegantly? There are many providers which deal with it on a daily basis in fact they cater to it, is this possible to do with asterisk or does it require other exotic setups? I even know of a provider which uses asterisk with many different types of devices, and they handle all NAT config on

Hi asterisk list, this is a bit off topic, but can anyone explain the point of the commercial far-end solutions floating around (jasomi, for example)? or are the far-end things just hyped up media proxies? They claim to be b2bua devices but that's a very wide category and only implies that the media stream passes through it - exactly what can be done with fairly simple OSS stuff. In short, what advantage does such a setup hav...

...? Because the LCS is > > > doing no RFC compliant SIP, i wonder if it can > > work. Google couldn't > > > tell me. If someon heared about that, please let > > me know. > > > > > > The fact i figured out is that the Border > > Controler from Jasomi can be > > > used as a gateway from MS-LCS-SIP to regular > SIP. > > But that is not really > > > handy and expensive too. > > > > > > Thank you > > > Volker > > > _______________________________________________ > > > Asterisk...

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

We are interested in how other people are handling NAT problems. We have several customers all of which have some sort of firewall/NAT device at their location. For simplicity sake, all customers' internal networks are 192.168.*.*. Our asterisk box is on public IP not blocked by any FW/NAT. I use QUALIFY=yes on all our customers' phones and I feel that sending out 80-something

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??

...t; >>> > >>> STUN is helpful, but as I understand it analyzes the situation and > >>> reports > >>> the configuration of a NAT. It doesn't help you keeping the NAT > >>> session open, > >>> as SER module nathelper or the FWD/Jasomi solution. > >>> Check here http://www.voip-info.org/wiki-SER+module+nathelper > >>> It's ugly, but what it does is sending UDP packets from the outside > >>> to the > >>> NAT to keep the ports open for incoming calls. NAT is an ugly thing, > &...

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP