similar to: Linksys PAP2 / RT31P2 + multiple G.729 calls

Hi All, I have installed chan_unicall and MFC/R2 successfully, and is runnign fine. But I noticed that once unicall is installed, asterisk CPU usage as reported by 'top', jumps to 99% every few seconds. I have no incoming calls, and I have even removed the E1 lines from card and I tried almost everything possible but I was not successful in determining the cause of this high cpu

Hi all, I'm trying to set up a test case for an ISP featuring an asterisk server and a couple of linksys rt31p2-na routers registering on it. Instead of using dsl lines, i'm trying to plug the * server and the routers on a cisco switch, just to test their functionality. I have created a vlan and a subnet on the switch and set up the ip addresses of the routers in that subnet. When i plug

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652&scid=29 -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, hoping that experienced hands will quickly show me the right way: after a fruitless web search i am turning to this list with my rather elementary question: is there any other way to communicate with the IAXy besides using special utility software that needs to be compiled under UNIX? Here is the story: about two months ago, after some not very satisfactory attempts at using SIP (my phone

Hello Has anyone been able to get the Soyo G668 phone to work with Asterisk? I've been playing with it for two days and I haven't had any luck. I've got a soft IP phone(xten) to work fine. If anyone has a sample config page that would be great. Thanks Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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 >

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

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...

I've got a Linksys PAP2 on my Vonage account with unlimited usage, but my softphone-addon account only has 500 minutes. Anyone ever try to setup their * to mimick the Linksys PAP2 ? Any comments or suggestions on what problems I might run into if I try? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? Thanks, Jose

Hi, I have a Linksys PAP2-NA connectd to my asterisk. I would like the device to add 2 characters in front of the dialled number always when it send the call to my asterisk. I dont know how to do that. I will summarise my requirement. My friend dials 1-210-1234345, i want the asterisk to get 55-1-210-1234345. Can someone help me to add this dialplan. Thanks in advance Dan -------------- next

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 had just found a Dlink DVG-1120 on ebay and I'm curious if anyone has used you it with asterisk. They were only $65. I have tested with the Linksys Pap2 and found that box to be fairly nice except for a lot of backgound/white noise. I was wondering if any else had experienced that? Let me know if I've wasted $65 on the Dlink and also if you had similar experience with white noise on

Hi! Can I use my Linksys PAP2 with asterisk and an analog CLIP phone to show the Caller number on the phone. There's a "Caller ID Method:" option on Regional settings, but I tested all options, and my CLIP phone never shows the Caller number... :( Any idea?

I have a PAP2 with 2 phone ports. When I call them everything works fine until I hung up the call. There is about 30-40 seconds until I can call to that extension again. Before that it gives me busy messages. Extension config: exten => 199,1,Dial(SIP/199,30) exten => 199,102,Hangup Any suggestions? Thanks

My asterisk box doesn't recognize DTMF from my analog phone, plugged into my ATA(linksys pap2 version2). I can make/receive calls fine... it's just that, for example, I cannot login to my asterisk voicemail. Softphones (such as x-lite) are fine. I've turned up a few articles via google where some people have this trouble, but have not seen suggestions on how to fix. I presume

Hey, I'w got a problem (bug maybe?). I have recently got my Asterisk to work perfect and I'm not trying to setup some dial routes and get the system working as I wan't it to. Yesterday I was installing Festival and also did a "aptitude upgrade" on my Debian Unstable installation. After that the problem started. After some serious testing yesterday night and today I have