similar to: asterisk-gui +spa3102

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

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)

(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 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

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 -->

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

Hi, I have been considering a purchase of the linksys spa3102 for a couple hours but I would like to know from someone here, wether this device will support faxing on my local asterisk server, I have had success sending and recieving faces with an x100p, and recall that in the old documentation, they mention that if I send/recieve faxes, that it all should be done on the local server for best

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

Hi, I'm connecting a Linksys SPA3102 to 3 different PSTN analog lines. With only one of those, CID is shown. Beside that, everything is working OK. Lines have different providers and/or locations. All are located in France and CID Detection Method is ETSI FSK / Bell 202. If I'm connecting a TDM400-enabled Asterisk system, to one of those 2 non-working lines : it does work. The only

Hi everyone, I'm trying to set up a VOIP mass deployment. To do so, I want to generate a configuration xml fails. I read somewhere that I had to use : http://<phone -or- device ip address>/admin/spacfg.xml but it work with an upper firmware only. My Software Version is 3.3.6(GW) The last firmware versin is 5.1.10(GW) Upgrade Enable: is set to yes on the linksys SPA3102 web interface

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:

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

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

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

Sorry, I am resending this, I tried earlier, but I couldn't see it appear on the archives - apologogies if it appears double! -------------------------------------------------- My Sipura 3000 ATA died on me this morning. I had a Linksys SPA 3102 available which I would like to use as a replacement. Unfortunately, the SPA3102 is not able to register with the asterisk server - I am

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

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

Hello everyone, Day before yesterday (i.e. 07-Nov-2007) i got all source(trunk) for asterisk and asterisk-addons. After compiling everything together. i found that there is no voice between mobile phone and asterisk server. Phone <-->(SPA3102) <--> Asterisk <-- bluetooth (CSR chipset) --> Siemens (S55) i can call anyone on mobile network from phone(analog) call establishes

Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP->PSTN