similar to: asterisk fax in debian

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A

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

I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running over Oracle VM VirtualBox (v 4.1.8). So far I have tried it following two guides. The first is the one from "Asterisk: The Definitive Guide" 4th edition (http://ofps.oreilly.com/titles/9781449332426/asterisk-Install.html) and the one from Billy Chia "How to Install Asterisk 11 on Ubuntu 12.04 LTS"

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

hello everyone. i am concerned about security to the PBX and i would like to discuss different fraud detection methods. Apart from making everything to secure the PBX (latest patches, iptables, firewalls, no outside users, strongs passwds,...) i would like to find out if there are any fraud detection techniques. As for my setup i do have a PBX running asterisk 11.4 and it has 3 sip trunks (over

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

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

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'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'm trying to do smtp authentication using dovecot and exim4 by following this guide http://wiki.dovecot.org/HowTo/EximAndDovecotSASL. for the exim4 part, i got an authenticator dovecot: cannot find authenticator driver "dovecot" ?error after i entered the codes in /etc/exim4/exim4.conf.template and reloaded exim4 with the update-exim4.conf command. i'm running debian

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:

Hi, I have looked at the LLVM code generation quality using small test cases and in general it is better than I thought and in some cases better than gcc. However, there are still some gap in SPEC performance. I have not looked at the root cause of those gaps. Anyone who cares about LLVM performance need to take this seriously. For fair comparison, I used -fno-strict-aliasing in gcc to turn off

https://bugs.freedesktop.org/show_bug.cgi?id=104181 Bug ID: 104181 Summary: [NV50/G94] system freeze on youtube with epiphany 3.26 Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

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)

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

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