similar to: Turning off splash ring on PAP2T

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, 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, 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.

Hello. I've got a Cisco SPA303 that I'm trying to provision via http. I noticed that this device looks very similar to a PAP2T, so I used that as a template for my provisioning file. However, the result is less than stellar. Line 1 registers and works. However, lines 2 and 3 also register as line 1, effectively giving me a 1-line phone with 3 buttons. Also, the line name is the

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 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.

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, I'm not sure how this happened, but one of my customers managed to turn call forwarding on on his spa112. I thought I had that turned off in the provisioning file. I have this in the provisioning file: <Cfwd_All_Serv_1_>No</Cfwd_All_Serv_1_> <Cfwd_Busy_Serv_1_>No</Cfwd_Busy_Serv_1_> And I have a similar entry for line 2. When I dial the device, I use this

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

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

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

Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls

Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.

Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them "try out" our dialtone

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

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:

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

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 in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other