similar to: How does call routing actually work with SIP?

I've got soft phone that allows me to dial SIP URI's. I'd like to route these calls through a provider to be completed, because I'm beind a NAT box and doing it directly doesn't work. Right now I've got an extension defined like this: Dial(IAX2/${FWDUSERID}:${FWDPASSWD}@${FWDSERVER}/**356<username>) This will connect a call to FWD and call a user at FWD. It works

Well I just need to say I got my phone last week. Here is my quick review of the phone and hope that someone has a possible fix for it or I will be sending it back. First the phone is nice looking in my view and it's heavy so it feels like a real desk phone. But it has these stick, gummy or I really don't know how to describe the bottoms on the phone. There good size but when you press

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

I'm looking to give the SPA-3000 a whirl as I'm having too much difficulty with the irq sharing thing inside the box. I'm reading the book but without having one in-hand to play with it appears a little obtuse at this time. Before I drop down my money can someone with some hands-on with one of these confirm if the SPA-3000 can: a) detect inbound distinctive ring (this looks to me like

Incoming calls from my Sipura 3000 don't seem to be correctly routing to Asterisk (or something?) Here is my Asterisk configuration for my incoming PSTN line: Code: [1000] type=friend host=dynamic context=incoming secret=6769 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very Inside of extensions.conf, I have this: Code: [incoming] exten => s,1,Answer( ) exten =>

Folks! I discovered some serious problem with several Sipuras 3000 but I don't know if the problem is with them or Asterisk. Basically, if I call a Sipura PSTN line, when there is a call already in progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I am able to get through and connect to dialed number. The other call gets disconnected but the originator of the

I have two Sipura 3000s, one for our main phone line the other for our fax line. I think I need to handle each device in seperate context sections. Both contexts use the s extension and things are not working as it was before I added the second Sipura for the fax line and additional context. Is it a problem to have two contexts with s extensions? What is the proper way to handle this senario?

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

Hi list, Well I really didn't want to see things get to this point, but Sherman at Sipura along with their President Jan F. leave me no other choice. SIPURA has been provided a letter from our attorney for Breach of Contract and damages. They have yet to respond. A quick background. 1. Sherman (SIPURA's Director of Marketing), stated that we would do a join press release for the Oct

I have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I have to make a config change to any of these