similar to: Sipura SP3000 question

Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 22, 2006 8:26 PM To:

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

Arrggghh. Tearing my hair out here. I'm trying to set up the spa3000 in the UK for my home, and want * to control the dial plan I've googled to no avail. I've read the manual to no avail. Can someone, please let me know what the parameters is the spa and * are to a) receive a call from the pstn b) make a call to the pstn from the phone attached I can make sip to sip calls (i.e. I

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Hi, all I am building a new server. Have installed FC 6 and put in TDM400 card. Checked out latest asteriusk code, run make install in zaptel directory. So far all is fine. Now I am trying to install the drivers. # modprobe zaptel FATAL: Module zaptel not found. Fair enough, no zaptel driver is found on the system. Is there are any known problems with FC6? I did not have much trouble running

I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. /carmi

Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as s@127.0.0.1. (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID("SIP/phone2-22c3", ""CID Name" <CIDNUMBER>") > in new stack > -- Executing

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI> exit No such command 'exit' (type 'help' for help) *CLI> quit No such command 'quit' (type 'help' for help) *CLI> Any other ideas? I started asterisk with -cvvvvg option. Same problem if use asterisk -r to connect. Can not exit. Any

Hi, a simple question.. is add dataset not part of zonecfg ? global# zonecfg -z myzone (OK) zonecfg:myzone> add dataset (fails as there is no dataset option) zonecfg:myzone> add zfs (fails as there is no dataset option) Basically how do I add a dataset to a zone ? Thanks Roshan please cc me pererar at visa.com

Hi, all I am still battling to connect * and voipbuster. What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or IAX traffic when using their client. VoipBuster client connects to connectionserver.voipbuster.com on port 11112 for authentication. Call itself is placed on different server. I have tried to connect using SIP and IAX and it seems that no authentication is

Hello, I'm configured Sipura-3000 to forward IP calls to PSTN number on no answer (In User1 tab Cfwd No Ans Dest: 123456@gw0) IPPhone ---IP---> Sipura-3000 ---PSTN---> PSTN User Generally it works fine, but Sipura sends back SIP OK to IPPhone just prior to dialing to PSTN number. How to configure Sipura to detect that the remote side on PSTN picks up the phone and only then to

Is it possible to emulate a unionfs with zfs and zones somehow? My zones are sparse zones and I want to make part of /usr writable within a zone. (/usr/perl5/mumble to be exact) I can''t just mount a writable directory on top of /usr/perl5 because then it hides all the stuff in the global zone. I could repopulate it in the local zone but ugh that is unattractive. I''m hoping

Hi, Could someone please let me know if I can get the pre-built binaries for Dovecot 1.0 for Debian 6.0? If I do "apt-get install dovecot...", I am getting Dovecot v1.2.5 installed, but it doesn't like the existing Dovecot 1.0 configuration I have, and due to some urgency, I am trying to avoid migrating the configuration to the new structure / requirements. Regards, Roshan

This post is simply documenting a spouse-friendly way of using the spa-3000 as both a fxs and fxo port for basic soho environments in the US, allowing asterisk to participate as needed/wanted. All home phones are connected _only_ to the spa-3000 fxs port. The incoming home pstn line is connected _only_ to the spa-3000 fxo port. Defined Line 1 (fxs) to register with asterisk via sip (extn

Hello all, I''m somewhat new to networking, and I''m having trouble masquerading connections that are coming over a bridge. The bridge only has a single port for now, but I''m going to add more ports later. I''m basing my configuration on the two-interface quick start guide. I''m using Shorewall 3.0.4 on Ubuntu Dapper. My network looks like this: * The

Hi, I''ve been successfully using shorewall in our K12 school since the 2.x days initially on Mandrake and now on Debian. Because of that my config has got quite complicated. The firewall has a working MultiISP setup with four interfaces (I''ve renamed them with udev to easy their identification): lan-if, dmz-if, snt-if and dnt-if (one of the providers (the one on dnt-if) is a DSL

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a