similar to: SIPPROXD for SIP thru NAT

Program your locked ADSI phone not by unlocking it but by using the feature slot FDN download descriptors of ADSI SEC security code telco provider lock number key from database list. ch33s3 First reset phone. Set clock to Jan 1 12:00AM (options-2?). exit to main screen. press options, (mute/flash/speaker/?) until you see firmware version. press # to reset. google more info, bam ch33s3d ;

Hi, I have been looking for info on * and the BT Easicom 1000 without much luck when i found a post to this list from Andy Powell saying that he had the phone working quite well. Before i go buy a shedload of these things I would like to know what problems/sucesses people have had with these phones and * in the UK. What they can/cant do with * Also does anyone know of any good ADSI

Hi people! Asterisk PBX (version 1.6.5): I have Asterisk behind a NAT (192.168.1.2) SIP Phone: A client behind NAT (192.168.1.3) Softphone: One other client somewhere in the internet (also behind an NAT). they want to speak with each other, and if they do, there is no sound. if softphone in the internet is no more behind a NAT router, it can hear SIP Phone but SIP Phone is not able to hear

Any suggestions? The Aastra 480 & 390 seem popular along with the CybioLink. Does anyone use these phones (or others)? Are they compatible with atsterisk's ADSI? If so, how are people programming these phones? Searched thru archives, lots of previous talk but no soild info. I'd like to get a couple of ADSI phones to play with, just hate to waste the money if they won't work with

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

I have a question hoping someone can help me with. I just purchased a 350 from http://lktelecom.zoovy.com/product/HPT350. I have been reading some posts and heard that these phones won't work with Asterisk.. Is this true? I had a previous Nortel Phone working in this enviroment. Well anyway here's my problem. The phone keeps ringing.. It won't stop.. Which makes the phone

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 was wondering if anyone has had any experience with PowerTouch 350 & ADSI. I am able to configure the phone with ADSI.. Which is pretty cool in it self. But I have a question regarding the Service feature. I have to choose the service button and choose the Asterisk PBX ADSI. But let's say I make a call it does not default back to this. I have to go back into Services and choose it

Hi, I have a meeting with an important customer in a couple of days and I am aware that most of their questions are going to be related about scability of Asterisk. We want to propose this customer to integrate Asterisk with SER, but I have a loot of complex doubts that I would like to known before this meeting. I would like to contact with a busines that has experience with large

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

I just need to buy 5-6 ADSI Phones. I wondering which of these models to choose - aastra 390 - I don't know is this an ADSI phone at all. Is there versions with and without ADSI - aastra 350 - I'm sure that it have ADSI. If there is some other good working model, it will be great, if someone points me. Thanks

Let me refine my previous question: > Can anyone provide some insight into why the auth-pam module uses a fork in > pthread_create (auth-pam.c)? This completely breaks the ability of one PAM function > to pass data to others via the pam_set_data/pam_get_data functions. Can anyone tell me how to #define USE_POSIX_THREADS when building --with-pam defined? The autoconfig stuff

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

Greetings All, I am trying to get sshd to authenticate using PAM in a situation where there is no password entry (as found by getpwent et. al.) for a user. Setting: AllowUsers * UsePAM yes causes the right PAM stuff to be invoked, but as soon as the PAM module tries to have a conversation with the (illegal) user (in order to get the password) sshd throws out the authentication context.

Dear Pals One customer has a Panasonic PBX KX-T336 with 60 ext. and a E1 (R2) for Trunks working perfectly now, This customer has 10 wireless links to his branches, wireless working great now, no voice at the present MY IDEA : T1Card into the Panasonic (additional to the E1) connected to a T1 Digium Card into Asterisk as far as I know the T1 Card can be configured as E&M to act as

Hi, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

I know I heard people looking for inexpensive ADSI phones a while back. Can't vouch for these guys, but this looks like a reasonable deal for what perhaps are new phones. No experience with ADSI, myself, but thought I'd pass it along. These folks also carry the Nortel analog units that do intercom and 2-line, as has been discussed in prior threads. JT ---------------------

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

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?