similar to: OT - SPA3102 - Provisioning with config file

2009/12/15 Olivier <oza-4h07 at myamail.com> > > > 2009/12/15 Steve Howes <steve-lists at geekinter.net> > > >> On 15 Dec 2009, at 10:42, Olivier wrote: >> > Unfortunately, it seems macro expansion doesn't occur in Line1 tab : >> > when I type $A or $(A) or ${A} or $GPP_A or $UID1 in User ID field >> > (in Line1 tab), asterisk

Hello, I've got an analog phone which is currently receiving unsollicited FAX calls from PSTN. For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would let voice calls come in and out and translate incoming FAX calls to TIF files (forwarded through email)). My target setup is : PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->

hi I am trying to install symbian in Redhat Linux machine.I am following the steps given in the link http://gnupoc.sourceforge.net/HOWTO/ for Nokia 9210.i have win98 as dual os and configured wine as mentioned. while running helloworld example,helloworld.armi is having a command like -------------------------------------------------------------------------- wine --

Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I

Here is another crash we've been seeing recently in rc15 on Solaris 10. (gdb) bt full #0 0xff1c12a4 in ?? () No symbol table info available. #1 0xff140040 in ?? () No symbol table info available. #2 0x000786a8 in t_buffer_alloc (size=688976) at data-stack.c:346 __PRETTY_FUNCTION__ = "file %s: line %" #3 0x00078190 in t_pop () at data-stack.c:149 frame_block = (struct

I have an ATA186, a tech just told me to set UID0 and UID1 to the same username, and PASS0 and PASS1 to the same password. In my mind, this would seem to have the unit registering twice under the same account, which Asterisk wouldn't support. When a call comes in, it should go to the last line to register. So to me, this means the call could sometimes come in on Line 1 and sometimes on Line

Dovecot-1.1.14 on Mac OS X. Panic: IMAP(user): file cmd-append.c: line 266 (cmd_append_continue_parsing): assertion failed: (ctx->count == uid2 - uid1 + 1) Trivially reproducible: $ telnet mailserver 143 a login user password b append inbox Backtrace: 0 libSystem.B.dylib 0x00007fff8458aa92 __kill + 10 1 libSystem.B.dylib 0x00007fff84606a1c abort + 83 2 imap

Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I get this error now and then, but only for users, that share or use shared mailboxes. See this snippet: 2015-05-22 02:27:09 IMAP(<uid1>) [28776]: Info: Disconnected: Logged out in=21450 out=76933 2015-05-22 02:27:09 IMAP(<uid1>) [28774]: Info: Disconnected: Logged out in=3769 out=16379 2015-05-22 02:27:09 IMAP(<uid1>) [28770]:

Op 2/16/2018 om 3:36 AM schreef @lbutlr: > Before I spend a lot of time trying to replicate a procmail script that automatically sorts list mail into mailboxes based on the List-ID header (and possibly some other data) I thought I'd check if someone had already done this for sieve. > > Basically, what I do now in procmail is > > 1. Get the listname from the List-ID header (or

Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on

Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)

Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. -------------- next part

Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've read in this mailinglist archives some notes related to Linksys SPA3102 provisioning but I couldn't find there the answer I'm looking for. Is it possible with this box (mine is unlocked) to store its config file(s) in a TFTP server, and have this(these) file(s) reloaded at boot time, for instance ? In embedded web server, there is a Provisioning tab full of settings but none

Hi - I got a SPA3102. I've set it up without to many problems. If the unit looses power, calls to the PSTN are bridged which is nice. However, if the Asterisk server is unavailable (I turned it off to test), calls out are not bridged to the PSTN. I've rebooted the SPA3102 with the asterisk server off, but still it gives me no dial- tone. Under the configuration, Auto PSTN

Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? Thanks. --

I'm faxing from stand alone fax machine via linksys SPA3102 but most of the time only half or quarter page goes through. Did anybody have any experience like this? -- #Joseph

Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura -- #Joseph GPG KeyID: ED0E1FB7

Hello, I got a SPA3102 and everything works fine except calling from voip to phone on fxo port. The phone ring but doesn't get any sound. I connected SPA at my asterisk server and i want to call from asterisk through SPA to fxo port where i have a regular phone. Thank you for support. -------------- next part -------------- An HTML attachment was scrubbed... URL: