similar to: How to configure a Linksys PAP2T ATA to connect an analog fax machine to Asterisk

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support

Hi all, Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl.

i have the problem in the hardware linksys pap2t, I am install asterisk with asterisk-gui and work fine but the hangup the phone (linksys pap2 t), no tone and sound like tu,tu, tu , tu , tu , tu , tuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuu what is the problem with phone ??? add param special??? Note: i am mark number phone and wait ... sesonds and call. thank you. -------------- next

Here is Mexico the phone company uses a DSL router from 2Wire which in my opinion is quite bad. I am having problems getting PAP2T adapters connected to Asterisk using these routers. They connect fine but after about 5 minutes I get a message on the Asterisk console that the ATA is unreachable. So far the only way I have found for the ATA to stay connected more than five minutes is to put it in

Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070607/3f90695c/attachment.htm

I'm trying out a Linksys PAP2T-NA. Calling out works great, no problems there. Calling in, though, the phone doesn't ring. Caller ID shows up, I can pick up the phone, and the call is connected, but no ring. I've tried it on two analog phones, same behavior. Suggestions? Asterisk SVN-branch-1.2-r31555. - James Moore

Hi Folks, I know the PAP2T-NA has a jitterbuffer.... however, it seems to be adaptive, which is fine for most situations... however, is there some way I can either: A) Specify how long it waits before it starts to shrink? B) Specifiy a fixed sized jitterbuffer? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all. I received a PAP2T-NA from a potential customer to see if I could get it configured for testing. I plugged it into my network and plugged a phone into it and attempted to do a factory reset from the handset. I pressed "****" and got NOTHING! Just silence. So, is this TA a brick? Or is there additional voodoo that I'm not aware of? Mike.

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

Hi all: when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. _________________________________________________________________ Windows Live?: E-mail. Chat. Share. Get more ways to connect.

Hi all, I'd like to know how I can turn off the "splash ring" voicemail waiting indication on a PAP2T from the provisioning XML file. I can do it from the web interface, but I need to do it on "a lot" of machines.... TIA, -- Take care and have fun, Mike Diehl.

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

OK this is the RTFM question of the day but I need a sanity check. I have 2 Astra 51i phone and a linksys PAP2 on a single DSL connection. 2 Aastra 51i---------| |-NAT on dsl moden--(Internet)--Asterisk PAP2t----------------| The DSL modem/router which has QOS set for the src and dest to the * box the PAP2 has both lines registered @ ports 5060 and 5061 and work like a charm. one of the

Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: <Proxy_1_> _sip._udp.example.com </Proxy_1_> However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine.

Hi all, I've got a customer who has a complaint about the splash ring that tells him that he's got voicemail waiting. My understanding this that this signal is being generated by the PAP2 and not astrisk. Anyway, he'd like to change the ring tone so that he can distunguish it from a regular call. Can this be done? -- Take care and have fun, Mike Diehl.

Hi all! I am fitting a Poisson model, using the following command: > fit2<-glm(canc~id1+year1+time+lnpa,family=poisson) where 'id1', 'year1' and 'time' are factors. I defined them with: > id1<-C(factor(id1), treatment) and 'lnpa' is a continuous variable. The 'summary' function gives me all the effects estimates, that is, for id1, I

Hi, I was wondering if I might be able to ask some advice about doing residual plots for the lmer function in the lme4 package. Our group's aim is to find if the expression staining of a particular gene in a sample (or "core") is related to the pathology of the core. To do this, we used the lmer function to perform a logistic mixed model below. I apologise in advance

Anyone have an example XML file for the PAP2T? Cheers, j