search for: 3102

...lpam -ldl -lselinux PAM is enabled. You may need to install a PAM control file for sshd, otherwise password authentication may fail. Example PAM control files can be found in the contrib/ subdirectory =============== root 3100 23936 0 14:58 ? 00:00:00 sshd: kiblerj [priv] kiblerj 3102 3100 0 14:58 ? 00:00:00 sshd: kiblerj at pts/2 root 23936 1 0 14:31 ? 00:00:00 /usr/local/sbin/sshd =============== > lsof -p 23936 > COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME > sshd 23936 root cwd DIR 9,1 4096 2 / > sshd...

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? T...

Hello I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite on XP as an SIP client: http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png The problem is that by default, Asterisk doesn't rewrite the CID name + number in incoming calls, so that XLite displays whatever name I used in the 3102 and...

Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. >From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I close? if so, where does one get the SIP im...

Dear Expert, I am stuck when trying to register SPA-3102 on AsteriskNow .. could any body please advise .. where can I find the article for doing this? .. I googled but got nothing.. Regards bie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071203/6bbbec...

...t: <sip:3101@123.45.67.89> Proxy-Authenticate: Digest realm="asterisk", nonce="23e26a38" Content-Length: 0 to 98.76.54.32:5060 ast1*CLI> Sip read: REGISTER sip:123.45.67.89 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK667022457 Content-Length: 0 To: Port 3 <sip:3102@123.45.67.89> From: Port 3 <sip:3102@123.45.67.89>;tag=f8e5152d35870bf Call-ID: 9a610ba9dca9c942d8e2b12e89939fd3@123.45.67.89 CSeq: 1913617706 REGISTER Contact: Port 3 <sip:3102@0.0.0.0> User-Agent: MxSipApp/4.4.10.60 MxSF/v3.2.6.24 9 headers, 0 lines Using latest request as basis...

I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. I can make calls from it through asterisk I am at a complete loss to know what to try next...

...stalling DCOM98.exe, the Desktop software installed, and it runs. Only problem is it doesn't detect the device connected to the serial port ttyS0. The port belongs to group uucp, and I'm in that group, I even tried it as root. It's giving this: $ wine Desktop fixme:comm:SetupComm insize 3102 outsize 3102 unimplemented stub err:dosfs:DOSFS_CreateCommPort Couldn't open device '/dev/ttyS1' ! (check permissions) fixme:dosfs:DOSFS_OpenDevice device open L"COM2" not supported (yet) err:file:CreateFileW Couldn't open device L"COM2"! err:dosfs:DOSFS_CreateCo...

Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ? From a quick test (got mine yesterday), seems like it is not recognizing Caller ID from PSTN/FXO port.. Using the same configuration as a Sipura 3000 (to be sent to mother-in-law POP :-), no Caller ID at all, (I've even extended the PSTN delay to give it more time, but n...

I've read alot of comments on the SPA-3000, many if not all saying they had echo issues, but I've not seen anyone comment on the SPA-3102. Does anyone have any comments or issues with these? Tim

Hi to everybody, I have an spa 3102 where i connected an analog phone (in the fxs port) and the pstn line (in the fxo port). This is my problem: the incoming call doesn't arrive to asterisk. In the spa web page i configured this dialplane: (<:line01@192.168.1.220:5060>) where line01 is the context in sip.conf, 192.168....

...sterisk server for testing on the company I work the setup or the approach let's call it is: 1 Asterisk Server fully configured and with some SIP extensions setup on two cities A and B. 2. One local PSTN line connected thru a x01p card to call local phone numbers numbres on city A. 3. A Sipura 3102 Gateway on city B connected to a city's B PSTN line. I wnat to be able to call from city A to city B PSTN phone numbers from city A using Internet and vice-versa. What is the proper config on Asterisk and the SPA-3102 so that I can call SIP extension on that device plus PSTN phone lines. Than...

Hi to everybody, I need some help in configuration of the spa 3102. I created an account for line 1 (user 208, sip port 5061) correctly registered in asterisk, then i create an account in sip.conf like this: [general] register = line01:pwdsipura:line01@192.168.1.222:5060/095377078 [line01] username = line01 fromuser = line01 secret = pwdsipura host = 192.168.1....

Can anyone point me in the right direction to learn how to make the 3COM 3102 phone work with Asterisk. I have looked everywhere that I can think of and I am not able to turn anything up. Do I need to have a login to 3COM's site to obtain a SIP image? Any help would be GREATLY appreciated. Thanks, Ben Bush

Hi, We are running SPA 3102 in multiple places and the one that we have problem with is with China telecom. Does anyone know the correct disconnect tone setting for China? Thanks in advance. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users...

Is anybody using Call Waiting works on Linksys 3102? Does it work? If I'm on the phone, I can hear a notification 'beep', but when I put first caller on hold the line is busy. Linksys registers to Asterisk. -- #Joseph

Good morning, I've been having some problems getting the SPA-3102 working properly in India. Specific problem is that calls from the Asterisk server out the FXS port is failing. When trying to make calls, I'm getting this message: [Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call from '' to extension '66200' rejecte...

Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in advance Regards B...

Has anyone had any luck getting this phone up and running on an asterisk server, most noticeably a Trixbox installation? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110325/40e0fea9/attachment.htm>

...tch) sitename_fetch: Returning sitename for DOM.DE: "Hamburg" [2014/02/26 16:08:16.066139, 5] ../source3/libsmb/namequery.c:211(saf_fetch) saf_fetch: Returning "hhhgadsprd01.dom.de" for "DOM.DE" domain [2014/02/26 16:08:16.066201, 3] ../source3/libsmb/namequery.c:3102(get_dc_list) get_dc_list: preferred server list: "hhhgadsprd01.dom.de, *" [2014/02/26 16:08:16.066269, 5] ../source3/libsmb/namecache.c:165(namecache_fetch) name DOM.DE#1C found. [2014/02/26 16:08:16.066416, 5] ../source3/libads/sitename_cache.c:105(sitename_fetch) sitename_fetch:...