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Hi List, When a user is on a call, sometimes they hear digits dialing as if the other end is randomly pressing the keypad with their face...but they aren't. It has happened while I've been on calls also, very odd and annoying. Has anyone come across this on Asterisk before? TIA, Skyler

Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to start looking

Hi, I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" in the PAP2s. However, there's sitll no callwaiting on the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have a bunch of PAP2s, and using the # to send immediately does not work as described in the manual. The PAP still waits for the "Interdigit_Short_Timer" to expire before sending the dial string. In addition, the dialplan does not cause the string to be sent immediately as it should. Here's the dialplan I'm using:

Hi List, Has anyone heard of an ATA device that supports TCP & TLS? Not much comes up in searching, thought to check here for some device suggestions. TIA, Skyler

Hello. I am trying to write an interactive function that asks the user for a vector of observations. Unfortunately, if a user inputs a vector, R treats the vector name as a string instead of a variable. Here is an example: vector.input<-function(){ k<-as.integer(readline("Input number of vectors: ")) obs<-as.integer(readline("Input number of observations per vector

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that

Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99 retail. They have a version with a router for $89.99. We picked the non-router version up and it seems to be a rebadged Sipura SPA-2000. The box has a Vonage service package inside as well, but it does work with other services. The box also has a "User Guide" meant for end-users that is very well written [no

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

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

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

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 >

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In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec ) before the call gets dialled i assume that the problem would be due to the dial plan in PAP2 if so please help me changing it

I'm currently using the Linksys PAP2, and since there's a shortage I'm looking for different devices. I'm mainly looking at the Sipura SPA sets since they are the base of the pap2. Anyone else have experience using them, and which one? Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Basically, I am looking to set up an extension which will be used as a "help-line". I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an

Has anybody else experienced problems with the Linksys PAP2-NA's? I've now had two of them fail unexpectedly, with no apparent rhyme or reason, having gone into a RED power LED, with a solid blue ethernet LED. No response from the device either on the network or from the phone.... To make matters even crazier, the one that now failed was the one I received as a replacement for the

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

Hi list! I have an Asterisk server (1.2.14) connected to a E1 line via a TE410P, and some PAP2NA connected to it. The PAP2 DTMF configurations is set to INFO and Asterisk to INFO too. At first, is INFO method different from RFC2833?? Well, I have two problems. The first is that when I place a call to outside, via E1 trunk, sometimes I get some DTMF tones and I'm sure nobody hit any key. Seems