similar to: SPA-3000 disconnect tone

Hello everybody, I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip installation. I actually have a problem when a PSTN user calls and hangs up. The disconnect tone is not detected by the SPA, the the call continues and, for example, leaves an empty message on the voicemail before hanging up (because * hangs up). How could resolve this problem?. I set, Detect

Hi, We are running SPA 3102 in multiple places and the one that we have problem with is with China telecom. Does anyone know the correct disconnect tone setting for China? Thanks in advance. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080604/23458301/attachment.htm

Hi, I have a SPA-3000 and would like to use the 911 recipe from http://www.voip-info.org/wiki-Asterisk+tips+911. So I took the simple recipe and modified it slightly: exten => 911,1,ChanIsAvail(SIP/potsoutbound) exten => 911,2,Dial(SIP/potsoutbound/911) exten => 911,3,Hangup() exten => 911,102,SoftHangup(SIP/potsoutbound) exten => 911,103,Wait(1) exten => 911,104,Goto(1) Now,

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

Well, I thought and hoped my issue of random hangups on our TDM400P were related to busydetect=yes in zapata.conf. The behavior of a call being hungup has not changed, however, since setting the busydetect option to 'no'. Again, the only affected user is my loud talker... What are some causes/solutions to seemingly random call disconnects on Zap channels that people have seen? I have

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

Hi , Thanks danny nicholas. Finally we get the things done with following. If i specify busypatten=500,500 then asterisk does not recognize hang up signal. After removing it only all are working fine. I choosed 2nd option as per your suggestions. working chan-dahdi.conf: ==================== signalling = fxs_ks busycount = 3 busydetect = yes callprogress = yes progzone=in usecallerid=yes

Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS

Hello, This is not exactly an Asterisk question, but I was encouraged to seek advice here anyway. The kindness of the * open source community is legendary :) I am getting going with an Asterisk 1.2 box, and I'm having trouble getting good quality transmit sound using handsets with VoIP phones. I'm primarily trying to focus on SPA-941, but also experimenting with Aastra 9113i and Uniden

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

I have a SPA-3000 that is failing to hanging up pretty often; almost every day now. The weird thing is that an almost identically configured (same FW, different HW rev) second unit right next to it isn't having the same problem. Swap the lines and the problem stays with the unit. I've been going round and round with Sipura and their latest message told me to use a procedure detailed in

I recently lost my setup (bad drive) and I'm now trying to get my setup back. I have Asterisk setup to a BT100, a Cisco 7960 (7.2 SIP) and an SPA-3000. I can call the phone extension, I can call from the phone on the SPA to other extensions and I can call out to the PSTN. What I can't do is to call from the PSTN to through the SPA to Asterisk. It rings twice then I get fast busy. I

Hello, I've been experimenting with Asterisk 1.0.6 and a Sipura SPA-3000, and I've run into a couple of questions I haven't yet found clear answers to: It appears that the SPA-3000 has no call progress on it's FXO interface? Asterisk considers a dial() as answered when the SPA-3000 has dialed the number on the PSTN line, not when someone has answered a phone on the

I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com

Hi All, I'm on of those brave souls who bought into the preproduction beta of the Sipura SPA-3000 FXS/FXO adapter. I've had the unit a few days and am exploring it's workings. I really want it mostly as a straightforward FXO adapter, to replace an X101p. Let me be clear, I'd love to support Digium in every way possibe, and will likely buy a TDM40 card shortly. But, the X101p has

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk "mainmenu" context (or ext I guess). Currently the

Hi, I have my Sipura SPA-3000 setup with Asterisk as follows: [spa3k_line1] type=friend context=home secret=PASSWORD host=dynamic dtmfmode=rfc2833 dissallow=all allow=ulaw When an incoming call comes in, I have a Zap interface in Asterisk which just does a Wait,15 then answers with voicemail. The SPA-3000 detects the PSTN call and makes Line 1 ring - so I can answer the phone if

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000