similar to: Multiple Sipura 3000

How can I dial *67 on a Sipura 3000 if I dial from a SIP phone connected on an asterisk server. I always get a message saying that authentication failed for INVITE for sip221@192.168.1.6. If I dial a number without doing *67 it's working fine... sip 221 being the extension of my Cisco phone and 192.168.1.6 being the IP of my asterisk server... I have my outgoing context configure

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

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

Can anybody show me a working Sipura 3000 setup please? I need to setup one to my * box, ... What are the variants you can setup? Advantage - disadvantage. bye Ronald Wiplinger

Anyone know what I need to do to get the FXO port on the SPA 3000 to forward calls to Asterisk? My Asterisk is running on port 5061 and I set the dial plan on the device to forward to s@asteriskip:5061 but Asterisk is not picking it up. I can see on tcpdump traces that the Invite packets do go to through to the asterisk machine on port 5061, but it's not picking them up. sip debug does not

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

Hi there everyone! I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of

Does anyone have a Sipura 3k running, and using the FXO? I've got things working right, but if I try to toss a *67 in the dialplan, it seems the sipura is throwing a 403 forbidden back. For example: exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/${EXTEN:1} works fine exten => _91NXXNXXXXXX,3,Dial,SIP/sipura1pstn1/*67${EXTEN:1} does not (even if I toss a couple Ws in) I can't

My apologies for the off-topic post ... No matter what settings I try, when I dial in to the SPA-3000 on the PSTN line, it picks up the call and immediately gives me a fast busy tone then hangs up. The info tab says under PSTN Line status: Last PSTN Disconnect Reason: PSTN Disconnect Tone which seems to indicate that the SPA thinks the caller has hung up. Since I am in Japan, it is possible

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

Does any body know if it is possible to group SIP channels just like it is possible with Zap channels? I have a group of FXO gateways (Sipura 3000's) and I would like to treat them as a group the same as I would Zap channels. Does anyone know if this is this possible?

I'm experiencing a very unusual problem with a few of my sipuras. They keep over heating after minimal usage. I unplug them and let them sit for 15 minutes and it starts working fine. The other way around is that I put a fan on it or something to keep it cooling. Has anyone else experienced this or are they just defective units? Regards, Mohammed Salim EZZI Telecom, Inc.

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

First of all, I'm not knocking Sipura/Linksys. I have heard very good things about their products. I'm just wondering if they are the only quality shop on the market. I know about the zoom 5801 where you can't dial out the FXO from SIP, only from the FXS port. And I have heard similar about the HT-488 also. I want to know if anyone else makes ATAs where all of the features work

I'm new to asterisk and already a fan. Please forgive me if my questions are covered by some FAQ and thanks in advance for any pointers anyone can give me. The basic problem that I'm having is that sometimes outgoing calls result in a busy signal when the outgoing line is free. I'm thinking that the channel is timing out or something but haven't figured out how to debug or gather

I'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret=1111

This is a wierd one. Can't figure it out. I have an SPA-3000 at the house handling my incoming line. It's setup to direct the incoming call to asterisk. Works great 99% of the time. A few times a day, I'm getting calls that ring once internally and are then hungup. I managed to get a detailed log [1] of what's happening today and it looks to me that the SPA is acting wierd.

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 need to place a call using a "pin code". To access an external line, the host PBX (a Ericsson MD-110) will require that I dial *72*pincode#phone_number to complete any (trunk) call. When I send the number, my Sipura 3000 will reject the call with "Forbidden - wrong password on authentication for INVITE" (see below). All other calls sent to the Sipura box without the

Hi in order to keep the dial tone after pressing 9 for 'outside line' I have this in my extensions.conf [localpstn] ignorepat => 9 exten => _9[123456789]XXXXXXX,1,Dial,${PSTN}/${EXTEN:1} exten => _9[123456789]XXXXXXX,2,Congestion this is properly included in the handsets' context but the dial tone disappears after pressing 9. am I missing something? thanks in advance