similar to: Problems with sipura 1001's and 2002's

I hooked up a SPA-1001 with asterisk yesterday and all works well except the phone doesn't ring. The phone I'm using has a LCD display so I can see the call come in. (with caller id info) I can answer and complete the call but it's just not ringing. The phone rings if pluged into a POTS line so it's not the phone that's the problem. I've used the SPA-1001's web

The first release candidate of the 9.1-RELEASE release cycle is now available on the FTP servers for amd64, i386, and powerpc64. The MD5/SHA256 checksums are at the bottom of this message. The ISO images and, for architectures that support it, the memory stick images are available here: ftp://ftp.freebsd.org/pub/FreeBSD/releases/ISO-IMAGES/9.1/ (or any of the FreeBSD mirror sites). Current

anyone else have the above issue? this is today's CVS. thanks.

I've been looking out for the Uniden UIP1868 for a while now, but I haven't seen it anwhere that I'm used to buying things from. According to froogle, a couple of places (that I've never heard of) have a small number in stock (small = 10 in this case). I'm doubly suspicious because even uniden's own online store doesn't have them available yet, not to mention

Hi All, I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd PAP2-NA units all hooked up to Asterisk. As you can imagine, setting them up took a while, and changing settings on them also takes a while. In order to prepare for future deployments, I'd like to use XML provisioning (or any kind of remote provisioning). I figured since Sipura/Cisco won't release the utility

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

Greetings, I'm RP for a fairly large chunk of IP real estate. I carved out a /27 segment for my home network. Which is currently running over a cisco 837 GW (adsl/router). I'm not really keen on it (the router/modem). So I thought to myself that it couldn't be /that/ hard to build a box with FBSD that could replace it - am I crazy? Wouldn't it be possible to upload a minimal build

Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will

The FreeBSD Project has begun the switch of its source code management system from CVS to Subversion. At this point in time, FreeBSD's developers are making changes to the base system in the Subversion repository. There is a replication system in place that exports our work to the legacy CVS tree on a continuous basis. People who are using our extensive CVS based distribution network

Hi, I've Sipura 1001 phone adapter. In the settings it has separate Line 1 and Line 2 tabs, which apparently means it can control two separate phone lines. I've Asterisk@Home server and want to setup two different extensions for two phones, i.e. 201 and 202. After doing all this, I can see in Info tab that both lines are registered but only one phone gets the dials tone. Am I doing

Ok, I've done some more research and I don't think I want an FXO box... What I'd like to do is use BroadVoice (with their BYOD plan) and then run Asterisk on my WRT54G router. I'd also like to use my regular home phones without having to use a special "SIP" phone... (eg. I like my Vtech normal cordless phones) What do I need to buy to get this working? It sounds like I

I have setup as per the monitor example configuration on the wiki site and all works well for an extension dialing 8 then the number. However, if I dial from an AGI script the recording stops after a few seconds. I see an extra answer in the console and suspect that is the reason. Could any kind soul help me to get around this? Extensions.conf.. exten =>

Hello. Has anyone run into problems accessing voicemail with the Sipura SPA-2002's? Our SPA-2000's work fine (registers fine, able to make and receive calls properly & also able to access voicemail). We've configured the 2002's exactly the same way. However, with the SPA-2002 we're unable to access the voicemail system (though it does register fine and is able to

I have a couple of devices that need an analog modem to communicate outside of our Asterisk system. Most FXS gateways don't seem to support this... I have a stack of Sipura 2002's that are, AFAIK, worthless for this purpose. I've heard that Digium's IAXy FXS will work with modems, but I can't find any reference to that in their documentation. There is also the

I've been having problems getting MusicOnHold to work, so I've dumbed down my setup to as simple of a setup as I can. Asterisk 1.0.9. SIP ATA's (Sipura SPA-2002's) <SIP ATA 1> <---> <Asterisk> <---> <SIP ATA 2> Both ATA's have public IP's. No NAT'ing going on here. Reinvites are allowed so the media stream bypases Asterisk once a call

Hello, in light of the recent CVE-2016-0777, I came up with the following idea, that would have lessened its impact. Feel free to ignore or flame me, maybe its stupid or I missed something :) - private key material should only ever be handled in a separate process from the SSH client. ssh-agent (maybe slightly extended) seems the logical choice. - in places where the client currently reads

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

Hello, Our 500+ company is slowly moving away from our hosted conferencing solution to one I built a few weeks ago with Asterisk and MeetMe. When our Q3 conference call comes around, we will have the need to have approximately 300-400 users in this call. Obviously, all would be 'listen only' mode and only 1 or 2 two would be speaking as marked/admin users. Our conference hosting

Hi, recently we (oVirt) have started discussing whether the way virt-v2v handles import from OVA files is good. And I would be interested in ideas how it can be improved. It is likely somebody already gave some thought to this problem. TL;DR: Extracting the OVA before import is a problem for large VMs (in sizes of TBs). Can we change something to prevent the extraction and work directly over

Hi all, Today's jam-packed sessions include the security theme for the first hour or so, then a debate about hosted vs local VoIP services. Hour one guests are Sjur Usken, telecom consultant who has been working with VoIP since 2002 and helping companies migrate to an all IP world and Sandro Gauci, a security researcher and consultant based in, author of VoIP security tools SIPVicious,