similar to: pap2 Dial plan

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3050 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/3396d198/smime.bin

HI I have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ? i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help,

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 >

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

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 all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

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 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

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'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:

I'm having a real problem with one of my linksys pap2. On outgoing calls the callee will ring, but caller (pap2) will not here it ring When the callee answers, no audio is transmitted either way. Asterisk reports the call connected and bridged correctly. Now the kicker is that sometimes it works and other times it doesn't. I have had the most luck calling land lines, but sometime

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

I am having an odd problem with a linksys pap2 ata and asterisk... Asterisk won't detect digits from it until I issue a 'sip debug'. As soon as I turn on sip debugging, everything works perfectly (classic heisenbug)! Asterisk is latest Debian 'etch' packaged 1.2.13. sip.conf looks like: [mc_ext01] type=friend secret=ext01 context=mc_ata_in host=dynamic dtmfmode=rfc2833

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

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 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

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