similar to: Call retard from a softphone to a hardphone

hello List i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the

Here is some info that may allow some "locked" PAP2 and PAP2-NA units to be used with Asterisk: I have a PAP2-NA (from a provider other than Vonage) for which I did not know the admin password, though the "user" pages were accessible to me. The provider had set it up to fetch at startup, its configuration file by HTTP from a numeric IP. It was running 2.0.10(LSc). A search

I have a handful of Linksys PAP2-NA's all talking nicely to Asterisk using standard telephones. I've been running them for the better part of this year. No complaints whatsoever. We chose the PAP2-NA's mainly due to cost and especially the ease of provisioning. In an effort to inexpensively bridge our office PBX (InterTel Axxess) to our VoIP network, we've opted to connect

Hi, I have 3 PAP2 connected to *, they work fine but there are some things which I would like to improve, some of them are: - double ring tone when placing a call (I hear two tones it seems like the PAP2 is generating it's own tone) - some kind of noise (like glitches or something) when I pick up the phone (seems like some polarity thing) - I'd like to keep the tone after

Hello, We use Asterisk with PAP2 and today we connected the FXS ports of PAP2 to CO ports of our Panasonic KX-TD1232. Problem is that Panasonic doesn't ring - that is doesn't ring every time the PAP2 is ringing. When we reset either Asterisk or the PAP2 it usually rings, but after couple of minutes it stops and only the automatic operator is answering - after 2 rings. We tried changing

The best I can do so far (which appears to be a bit of a hack) is (<:0>S0), which says to add a '0' to the start of the string and dial immediately. This gives asterisk an extension dialled of '0', which isn't the 's' that i'd hoped for, but is a good start! (S0) by itself doesn't work, nor does (<:>S0). Any other suggestions? Thanks James >

hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked

I am in the process of building up an Asterisk-based voice network using PAP2-NA SIP clients from Linksys. Our network consists of several disconnected private networks (unaware of each other), and are all proxied out to the Internet via a Linux server. Our Asterisk PBX lies on the Internet on yet another network. I'm hoping to get all SIP clients (PAP2-NA's) to register at the central

I am using asterisk (2 week old CVS) am for the first time have been starting to experiment with busy and congestion. At this point I am only using sip endpoints PAP2-NA devices. All testing of this is being done on a local network. my test extension looks like this: exten => 7777,1,Answer exten => 7777,2,busy(35) exten => 7777,3,Hangup Or like this: exten => 7777,1,Answer

Is there a way to configure the linksys pap2 to automatically connect to asterisk on lifting the handset (presumably into the 's' state)? Asterisk would then be listening for DTMF tones to figure out what to do rather than having to put a dial plan into each pap2. I think the pap2 is pretty much the same inside as a few of the sipura boxes so the same thing might work if anyone knows...

Hello All! I have a problem with my calls, that drops after 20 - 30 seconds. I got a piece of PAP2-NA log and Asterisk log and there's an error 603 - call declived, as showed. Thanks for any help. McCoy *********** PAP2-NA LOG *********** Feb 9 09:00:56 192.168.4.205 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060

I am using my first PAP2 device from linksys. Used many polycom phones... I configured the PAP2 device with asterisk. I have the registration, thought I was good to go. Plugged in my Valcom 2924 public address analog connection, called the extension and I got busy... very strange I thought. I then looked at the status page of the PAP2 and it says the following Reg online and hook state OFF.

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

Has anyone made this work? For me everything is fine until I switch canreinvite form no to yes. What happens is that asterisk hangs on "attempting native bridge" ... from what I understand "attempting native bridge" means that the RTP is routed through asterisk (just without any codec translation) But it shouldn't do that ... right? ... canreinvite is set to yes ...

>Message: 7 >Date: Sun, 29 Oct 2006 22:00:22 +0100 >From: "Jose Limeres" <jlimeres@gmail.com> >Subject: [asterisk-users] Linksys PAP2: calling tone stops after 5 > tones >To: asterisk-users@lists.digium.com >Message-ID: > <2b3431b20610291300u420116e5scf9103d7dac54321@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed >

Hi All, I have my Asterisk box running with 6 extension all connected to CAT5 Grandstream phones. I'm trying to connect 2 extension on a different office across the hall by WIFI bridge using SMCWEBT-G configured as Ethernet client. If I connect the Grandstream to that box on the other office it works fine. If I connect the PAP2-NA, both extensions register with no problems with the Asterisk

My setup is this: Analogue phone attached to a Linksys PAP2 | Asterisk | VoIP provider I have put the PAP2 in 'batphone' mode where when you pick it up it immediately dials the 's' extension in the pap2_incoming context in Asterisk, where asterisk answers the call and does a DISA(no-password, internal). I do this because it means I can centralise all of my dialplan logic in

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the