similar to: How to enable T.38 between SPA3102 PSTN Line port and ReceiveFAX app ?

Hello, I could successfully played with General Purpose Parameters (GPP_A, GPP_B) and a TFTP server : whenever I change a GPP value in a configuration file, my SPA3102 automatically updates the corresponding value its web server shows. My config file is : <flat-profile> <GPP_A>myid </GPP_A> <UID1>myid </UID1> </flat-profile> I though I could use this

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 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

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:

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

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

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

I'm looking to build a VoIP solution for 100+ service vehicles that have WiFi hot spots installed (with cellular uplinks). Currently we are trying out Skype wireless handselts and Majick Jack. I'd also like to consider an Open Source solution that can bring the calls back to our data center [possibly integrated without our existing BCM 3.x VoIP PBX]. For hardware someone on the IRC

Hello list, this is my first post on this list. I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones. I have configured the SPA PSTN line as trunk to receive and send calls. I can call outside from SIP phone throw the PSTN line and all is OK, the problem is when I receive a call from the PSTN, on the out caller phone there is a demo playback. I want to redirect the call to a

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

I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? -- #Joseph GPG KeyID: ED0E1FB7

I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of "softfax" that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling

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

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

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

I'm missing something simple I think: I have an spa3102 for which I want asterisk to use the incoming pstn callerid when it sends the call to a local extension (207). callerid works fine for the internal phones (between each other) The spa3102 is picking up the PSTN callerid and displays it in its own status pages Asterisk however, doesnt see the callerid at all. The spa3102 is set to:

Hello all. I am just getting back into Asterisk and I am setting up my Linksys SPA3102. I have incoming calls working fine, as is the phone plugged into the unit. My problem is I cannot get the SPA3102 to dial a phone number automatically. I can call the extention of the PSTN and I get a second dialtone, and I can then manually dial. I'd like to be able to have Asterisk pass the