similar to: Incoming PSTN calls not routing to Asterisk? (using Sipura 3000)

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

Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also seems to not notice any of the line state changes on the PSTN when the remote party terminates the call. It only recognises the offhook signal which gets sent much later. Chris

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

The options are not seperated by commas. exten => s,1,Dial(SIP/50,23,r,d) should be exten => s,1,Dial(SIP/50,23,rd) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of John Klimek Sent: Thursday, June 22, 2006 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

Hello Gurus I've installed my Asterisk server for testing on the company I work the setup or the approach let's call it is: 1 Asterisk Server fully configured and with some SIP extensions setup on two cities A and B. 2. One local PSTN line connected thru a x01p card to call local phone numbers numbres on city A. 3. A Sipura 3102 Gateway on city B connected to a city's B PSTN line. I

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

I have a working Asterisk 1.2 and Sipura SPA3000 combo but I would like to have more control over incoming PSTN calls to the Sipura. Right now such calls come in over the fxo port (sipurafxo1 on the PSTN tab) and are routed by dial plan 8 to the fxs port (sipurafxs1 Line 1 tab) I'm not even sure if the calls are getting to the fxs port and they sure are not going to the specified context.

Asterisk seems to register just fine with BroadVoice (asterisk -r, and then sip show registry shows sip.broadvoice.com is "Registered") ...but when I try to call my broadvoice number (from a cell phone), it rings one single time and then says "The party you are trying to reach is not available to take your call." This doesn't seem to be an Asterisk message but seems to be

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

We have issues with all of the SPA-2000 ATAs we have where incoming calls from only one of our Asterisk servers do not complete. Details: 1- On the CLI we see that when the call is pushed to the ATA it shows Busy/Congested 2- We can make calls to this same server just fine 3- We can receive calls from other Asterisk servers running older CVS versions of Asterisk with the same exact ATA

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten => s,1,Dial(SIP/50,23,r) exten => s,2,VoiceMail(u50@default) exten => s,3,Playback(vm-goodbye) exten => s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are

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

Hi, the topic says it all really. Does the Sipura 3000 detect and report UK clid correctly? thanks Mike

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

Hi all, Anyone able to remotely reboot their password protected Sipura SPA-3000 from command line. I am trying: Sipura SPA-3000 from command line: # wget http://admin:mypassword@192.168.1.55/admin/reboot The strange thing is it works fine when I go to http://admin:mypassword@192.168.1.55/admin/reboot with my web browser... Thanks....

Hi We are getting more and more email from Mac users asking how they can connect their MacOSX based Asterisk server to a PSTN phone line. This has led to two ideas ... 1) Mid term: set up a donation fund to sponsor the development of Zaptel drivers for MacOSX Note: if everyone who downloaded the Asterisk installation package for MacOSX during the last two months since its release would donate

Hi all, We have several sipura 3000's working well for outbound calls, however the issue we have is that when calls are sent to the Sipura with Dial(SIP/${EXTEN:0}@sipura1) the Sipura does a SIP answer immediately and then proceeds with the call "in band" therefore sending dialing sounds back to the caller. Other SIP gateways we have notably the Vegastream and others do not do a SIP

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