similar to: Wanted: UK-specific hardware recommendations (FXO and FXS)

----- Original Message ----- > Roger Burton West wrote: > > > I want to hook one of them to the PSTN. Given that I am in > > the UK, what is a reasonably easily-available device to > > provide an FXO interface from a Linux box, with a minimum of > > faffing around with drivers? Just one line is needed, though > > in theory two might eventually be useful. My

At the risk of being flamed.... Has anyone had any success get the 'El cheapo' Wildcard W100P clone's (?20 flavour) to work with UK Caller ID? I'm not sure what the status of Asterisk 1.6 is with respect to UK caller ID, being that we have an odd method of sending the FSK ahead of the ring, but I'm guessing I can't be the first to ask this? Keeping in mind that cost is

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 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

We will be connecting our Asterisk server to ISDN 30 and intend using the Sangoma A101 card. The install location is in London (UK). Sangoma card at Voipon http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA I would be grateful to hear if this is the right choice of card. Usage reports would be helpful. Regards Rory -- Rory Campbell-Lange

Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. Thanks, Gavin.

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

Not FAX over VoIP, but testing 2 FAX machines back to back ... The scenario is that a client has one of my asterisk VoIP only systems which they're happy with, but need to test FAX machines, so rather than plug in a TDM card with FXS ports, I'm suggesting something like a PAP2T device with 2 analogue ports registering to their asterisk box to use to send test faxes from one fax

Folks, I have a TDM400 with an FXS module. I'm having some trouble ringing some phones attached to the device. I have a real el cheapo handset that cost me about 9UKP. It works fine, I plug it into the Wildcard and call the channel and it rings. I have tried plugging in my BT Diverse 3016 cordless phone but it will not ring, although I can call out from it. I have also tried my GPO 332L

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, 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.

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

Hello, Some time ago, I noticed that whilst I had upgraded to Xen 3.1 (kernel 2.6.18-xen), I''d forgot to upgrade the paravirt DomU .cfg kernels from 2.6.16-xen to reflect this. So, I upgraded the DomU kernels & ran into issues with mounting devices, and the ext3 module not being available. After a bit of faffing about, I managed to sort this out and load the ext3 module.

Hi, I am looking for a good ATA recommendation, ideally something: 1) with one FXS and one LAN port (so it's as inexpensive as possible) 2) That can be provisioning using FTP (configuration and firmware upon reboot, ideally remote reboot from a sip notify) 3) Supports T.38 Nice to have would be: a) PoE powered and AC powered (my choice) b) Small size-wise I have been

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 am having a hell of a problem trying to get a linksys pap2t to register with my asterisk from outside the LAN. I have tried every combination of NAT, outbound proxy, stun, specify external IP address etc and it just won't work. Here are the relevant details. In asterisk I have set the following. externip=my.ip.address localnet=192.168.0.0/255.255.0.0 nat=yes bindport=5060 here is the

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

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

On 04/12/2018 05:24 AM, Richard W.M. Jones wrote: > I don't think we have nbd-server in RHEL, and in any case wouldn't it > be better to use qemu-nbd? > > You just start a new qemu-nbd process instead of faffing around with > configuration files, kill the qemu-nbd process when you're done, and > qemu-nbd supports qcow2 already. That, and qemu-nbd supports