similar to: Re: call waiting not working on PAP2 (Andy Kuo)

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:

Greetings list, We are having a puzzle with * (asteriskathome 0.5) and SIP phones (SPA2000 ATA's). If callwaiting is enabled, everything (including call waiting) is normal. If callwaiting is turned off, the phone will not accept incoming calls and the call goes straight to whatever is programmed for the busy voicemail response. It doesn't matter whether reinvite is on or off, or

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:

Hey guys, can you shead some light on this? I will copy my mgcp.conf and post below, but here is the problem. I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick

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?

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

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

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

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 >

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

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 know this is slightly off topic, but I was wondering if anyone can help with a problem getting my PAP2's to connect to Asterisk. I use a provisioning file, and I recently re-wrote the files for each PAP2. I had a small typo and the PAPs logged it as a corrupt file. I corrected the file, however, Line 1 on both of the PAP2's now wont register. Line 2 works fine though. I've done the

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.

We have some extensions in our dialplan that start with a star. We can dial them from Zap phones and SIP phones, but not from phones connected to a PAP2. After the user presses star follwed by two digits (our extensions are dialed with star followed by three digits) he hears a fast-busy that comes from the PAP2, not from Asterisk. This also happens with the builtin *8 (call pickup). In

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

I am using mysql sipfriends and can't seem to get the MWI to work. From what I've read it seems this is not supported with that dynamic system, and probably never will be. I was thinking of just setting a cron job or something to check every minute for voicemail and set our sip NOTIFY messages as needed. Also, the PAP2-NA has the ability to reboot via a sip notify and I would like to be

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

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