similar to: Asterisk, PAP2T and 2Wire DSL router

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

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

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

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

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

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.

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.

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.

We have an installation with around 50 sip phones but only 5 of those are hardware. There are three Grandstream phones, one Snom and one PAP2T. We are running Asterisk 1.2.8 with an E1 (R2). Only the hard phones are having problems which are either echo or distortion. The softphones all work fine and no one is reporting any problems. They are using 3Com switches which are fairly new. I

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

I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter "localnet" defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have

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

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.

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.

I have a weird problem with a client. I recently upgraded to Asterisk 1.4.22 and Zaptel 1.4.12.1 on their server and now there is a problem when a fax call is received. Basically when faxdetect=incoming is set in zapata.conf the call comes in and the fax extension dials a Linksys PAP2T where a fax machine is connected. The fax answers and almos immediately it hangs up and I get this: --