search for: viop

...This has to go to: - modem that operates in half bridge mode (no nat) to a linux firewall (does natting ip is 192.168.0.20) that has the ports above forwarded to the staitc ip of the asterisk box (192.168.0.21 packaged version for ubuntu hardy). This phone works fine with a commercial provider of viop (via asterisk), but I can't get it to work with my install of asterisk in my remote network! ngrep-ing traffic on the firewall shows asterisk continually sending invites to the public ip of the ip phone. I would be very grateful for any pointers of where to start. If you need sip debug or ngr...

...service. Typical environment: --------------------------- Incomming Lines ISDN 2 Channel From BT (yes im in the UK) (Do I need some type of ISDN Interface Card?) Extensions 10 Users require (Can I use a computer to answer and field calls?) VOIP Phone Numbers Do we need to register some type of VIOP telephone number? (are there differnt standards or are the VOIP number accessable by all?) Cost What is an average Hardware cost for this type of system? -- Regards Phil Brooks Technical Support Team Brooks Computer Solutions 0115 468333 -- This message was scanned for spam and viruses by B...

...t connecting and registering with them according to 'sip show registry', I can't dial out through it, but it does dial out through my regular phone line. I'd like to set it only to dial 911 through that line and have all other calls go over voip. I've checked out a bunch of viop-info pages, anyone already setup with Broadvoice that can help me out? JD

We're got a T1 from Sprint that we use for internet. During VIOP calls, if you download something, the VOIP calls break up. I found some info at Sprint for adding 'class of service', and I also have some information on configuring our Cisco routers. I've read the relevent pages on the wiki, but it seems vauge what's required and what's r...

...all, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this correctly. The handsets are Linksys SPA922 The issue we are getting is in transferring calls, which happens like this :- 1. Incoming call from pstn/viop provider 2. Call is answered by a user 3. Call needs to be transferred 4. Xfer button is pushed, other user is called, answered, and they speak about the call 4b. The incoming call is held, listening to MoH 5. Xfer is pushed again, 6. <SIP Debug Output> 7. MoH stops, 8. Office user gets no a...

...seems better to use a good linux firewall like it was intended. ******************************************************************* Redhat 9 scripts - basic firewall rules for SIP forwarding: file: /etc/rc.d/init.d/firewall In the Services section, add this: SIP=your.internal.ip.here # VIOP SIP Add the following code amongst the service scripts toward the bottom: #----------------------------# # SIP # #----------------------------# function S...

...can also recieve a call from the Ericsson and transfer it to another IP phone without a problem. What I can't do is receive a call from the Ericsson on an IP phone and then transfer it back to another Ericsson extension or a PSTN number. Watching the call progress with Ethereal (really neat VIOP tracker in the latest version), I can see the call setup take place, the receiveing phone actually rings, but once I answer the receiving phone the call setup never completes and the trace shows a continuous stream of invites, acks and ok's. Here is a sample: SIP Request: ACK sip:6215...

Hi, I am having a problem with inbound calls (from a SIP VIOP provider). When caller ID information is not available, the calls go straight to voicemail. We are using a mix of either Sipura 841 phones or SPAs. When the call is passed to the phone/SPA, Asterisk reports "Got SIP Response 406 "Not Acceptable" back from..." I have searc...

I use Voicepulse as a VIOP provider, the line comes in via a Sipura 3002 box. That's connected to the Asterisk box via a TDM422B POTS card. I'd like to add a virtual phone number to my VOIP service so that I can direct calls to a home business to a different voice mail than calls to the home phone number. Is there...

...849]: app_meetme.c:465 build_conf: Unable to open pseudo channel - trying device Mar 23 15:55:49 WARNING[19849]: app_meetme.c:468 build_conf: Unable to open pseudo device I presume, being the Debian binary package, that I don't have the ztdummy driver installed. On the otherhand I read in viop-info.org that that wasn't necessary with Linux 2.6 (I am running 2.6.18). Where do I go from here? -- Alan Chandler http://www.chandlerfamily.org.uk

...d, most the time , we need to listen voice only. also, we use to share desktop screen. But as far as I know SIP is the standard for video telephony. SIP can handle video +Audio. now, I am thinking that, I can give solution like selling a server which has Asterisk over it. AFAIK, Asterisk can handle VIOP calls. Now system will load some GUI application from there he can add remove users. Now at the same time I want to give small device which has Wifi + LCD (for video) + Android + webcam+Sipdroid or IMSdroid. China can make such device in less then 100 dollar. now, every customer will be given with...

...etween DataCenter and User Site Admittedly I didn't read your SIP debug (on the mobile), but do you have reinvite=no set for the extensions and SIP trunks (providers)? This sounds on the surface like a classic case of the Mondays. Erm reinvites I mean. <snip> 1. Incoming call from pstn/viop provider 2. Call is answered by a user 3. Call needs to be transferred 4. Xfer button is pushed, other user is called, answered, and they speak about the call 4b. The incoming call is held, listening to MoH 5. Xfer is pushed again, 6. <SIP Debug Output> 7. MoH stops, 8. Office user gets no au...

It would be nice just for once to actually use a mailing list with people who are a little more sympathetic to the fact that your not a rocket scentist or molecular biologist and that you might actually need some help, without being made to feel like your completely useless and should be cleaning toilets for a living. "Ahhh man not another stupid newbie question! are these people completely

Hi, I am currently trying out the asterisk@home (version 1) release of Asterisk, and I want to configure it as follows: Calls from regular telephony network (PSTN) come in through my VoIP provider over SIP and outgoing calls to the PSTN should be routed through the ViOP provider onto the PSTN network. I thus have no direct PSTN connection, but only a SIP connection. Incomming calls work fine. No problems with this. I have found numerous documents (on this list & on the voip-info.org website) that desribe how to dial out to PSTN numbers via the VoIP p...

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These

I am trying to get a queue to do more than just play music and hold calls. Specifically, making some "comforting" voice announcements would be nice. Below is the queues.conf file relevant portions. Member phone number is munged to protect the guilty. We shouldn't need the announcement source info, but I have been trying everything. The problem is with the member busy, we get no

...ive way to priority. So if any VOIP traffic is sitting in the queue, it jumps the queue ahead of all other traffic. If and when there is no VOIP in the queue, Interactive traffic gets to jump the queue and when there is no VOIP or Interactive traffic then all else just gets FIFO access. And yes, VIOP traffic can completely fill the link an starve all other traffic out, and yes, if there is no VOIP, Interactive can completely starve all of the "All other" traffic out. But one writes their priority lists with this in mind. The way I see it, if VOIP needs 100% of my bandwidth, it shoul...

...nd share it again please, this way we make everyone happy. ? My goal is to have some good and simple guidelines to install samba4 without compiling anything, since I dont want any compiling software on productions servers. ? Why like this, and not fully samba4 domain. In my case due to a windows VIOP solution, i must used 2 windows server, to bad, but so be it. I'm still working and testing my setup, i will go transfer the FSMO roles, setup replication of netlogon and sysvol also, but thats not ( yet ) in this setup. ? Ok, lets start...? ? ------- the howto ------- I installed Ubuntu 12...

Hello all, Saturday night, after a couple of shots of bourbon, I realized that I had an old PC sitting in the garage that I could use as an Asterisk gateway if I just blew the dust off it and reloaded it with a modern Linux distribution. In my characteristically impulsive manner, I grabbed it and started cleaning it up so that I could put it in my office without my wife having a fit. The

Mandi! Rowland penny via samba In chel di` si favelave... > Who was this 'someone' ? [...] > Yes, stop listening to spurious people who have never done the upgrade and > follow our documentation ;-) I'm 'someone'! ;-) And, as you know, i've correctly migrated/merged 4 NT domains in an AD domain some year ago, following also hint from this list. ;-) > I