similar to: Configuring Linksys spa8000 devices through xml

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, Has anyone compared SPA8000 vs. GXW4008 especially in terms of firmware and hardware stability (the feature sets are apparently similar)? Vieri

Hello everyone. I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing. Voice is working great, but I never configured anything using T.38 in Asterisk so I'm kinda lost. On my googling I found out that would be best letting the Linksys SPA8000 (for those that don't know, it's identical to the PAP2 but with 8 ports instead of 2) and the telco negotiate with each

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: 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 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'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

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Hi, I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to both a local Asterisk server and also with a trunk directly to a VOIP provider. Everything works great, except I'm having a problem setting the outbound caller ID to a value different from the SIP username/authname. The SPA8000 has SIP setting for Display Name, User ID, Password, and Auth ID, as well as a "Use Auth

Add a xen command-line parameter, sched_credit_tslice_ms, to set the timeslice of the credit1 scheduler. Signed-off-by: George Dunlap <george.dunlap@eu.citrix.com> diff -r 4a4882df5649 -r 782284c5b1bc xen/common/sched_credit.c --- a/xen/common/sched_credit.c Wed Aug 31 15:23:49 2011 +0100 +++ b/xen/common/sched_credit.c Thu Sep 01 16:29:50 2011 +0100 @@ -41,15 +41,9 @@ */ #define

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

Dear R-gurus, Just supose I have a dara.frame that looks like myDF<-read.table(stdin(),head=T,sep=",") codID,namesp,k1,k2,k3,k4 1,spA,2,5,6,3 2,spB,4,5,4,6 3,spC,2,1,5,6 4,spC,5,4,3,2 5,spD,1,2,3,4 6,spE,2,4,3,1 I need to update the columns k1-k4 with the namesp, but considering the math between Kx and codID. My desired output must looks like: codID,namesp,k1,k2,k3,k4

Found my answer for those who would like to know: Profile Rule: [--key $A]http://your.addre.ss/$B/$MAC.cfg GPP A: urtopsecretultrasecureaesencryptionkey GPP B: OddBallDirectory123098 Hope that helps someone! Curt -----Original Message----- From: Curt Shaffer [mailto:cshaffer@gmail.com] Sent: Wednesday, February 07, 2007 11:35 AM To: 'Asterisk Users Mailing List - Non-Commercial

I've got a user that wants an Access database shared out. I've got a decent machine serving out the directory (300 MHz Pentium II) running RedHat 5.0 (kernel 2.0.34) with Samba 1.9.17p4 (RPM is samba-1.9.17p4-3). I've created a simple share entry in the smb.conf file: [global] debug level = 1 browsable = yes comment = Samba under Linux (RedHat 5.0) domain master = yes

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

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

Hi, I am currently analysis Raman spectroscopic data with the hyperSpec package. I consulted the documentation on this package and I found an example work-flow dedicated to Raman spectroscopy (see the address : http://hyperspec.r-forge.r-project.org/chondro.pdf) I am currently trying to remove outliers thanks to PCA just as they did in the documentation, but I get a message error I can't

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

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