similar to: Help with inbound dialplan

Hi My setup is Internet -> firewall -> asteriskbox -> spa3102a -> spa3102b the spa's can talk to the firewall directly. The firewall does NAT. The current asterisk flow for outgoing calls is phone => spa3102 => asterisk => vsp and vis versa for inbound calls. can I use re invite for outbound calls such that the spa3102

Hello all, I am running Asterisk 1.4.17. I have 2 Linksys SPA3102's and one PAP2-NA (I have a second on order). They have G729a built into them. This is supposed to be compatable with G729. I was trying to have them use that codec when they talk to each other, but it seems they always switch to alaw or ulaw (depending on my sip.conf file). Shouldn't they be able to use G729a in

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 have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

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 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

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'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

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

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 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

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, 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:

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

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

Lukasz Sokol wrote: > but have you considered a web-managed config-builder such as FreePBX? > Instead of building your dialplan from scratch ? I've never used FreePBX, but, after having looked at its website, I think I have a general understanding of what it can do. What I don't understand is how FreePBX answers my question about the Linksys SPA3102 being good for a mission

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Hi, I'm rather new to this domain so I may be doing stupid things without being concious of that. I've got a Patton MATA I'm trying to setup as T.38 fax adapter. Whenever I connect a fax machine (Dell MFP1815dn) or a phone to it, I can successfully send a fax or talk to the other end. Whenever I connect a fax modem (Dell Inspiron 6400 laptop), I keep getting "No signal. Line is

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