similar to: SPA3000 to Asterisk Server - Asterisk server not answering calls

Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0<:1000@10.0.0.101> my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid="PSTN GW" <3000>

I tried to setup an PBX System with the following parties: 1. SIP Phones 2. One AVM ISDN Card with CHAN_CAPI for Outbound dialing and receiving external calls 3. One HFC-S PCI Card in NT-Mode with ZapHfc for physical ISDN Phones. 1+2 work perfect I have problems with part 3 Card is working, drivers loading fine, asterisk initializes chan_zap without problems. Demo did work. My problem is

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 A quick question. I have SPA3000 and trying to get call waiting to work. I do receive call waiting tone, however hook flash does not seem to work. I think, I set up SPA3000 correctly. Basically, doing HF 2 (switching calls in Australia) does not do anything. Is there any examples on how to setup hook flash operation in Asterisk? Thanks, Rudolf

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

Dear all, Can someone from Netherlands who has SPA3000 send me the Regional & Line1 settings. I'm not sure what is wrong but I can call from asterisk to phone attached to spa300 FXO, but not the other way. I tested three phones: siemens gigaset, tiptel 160 and hpoj k80(fax). Only tiptel 160 can call to the asterisk, other two just dial but nothing happens. I guess this has something with

Hi all, Can anybody explain what these values exactly mean. As you all know its the dialtone value on an SPA3000 of linksys. 350@-19,440@-19;10(*/0/1+2). Can anybody help me how to write this code for a dialtone of frequency 425 which is continous. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a little situation here :( I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor my router have DynamicDNS. So, if I

Hi, I'm using a SPA 3000 as FXO and FXS termination connected to my * box. I'm using the caller ID prefix trick explained here: http://www.voip-info.org/wiki-Sipura+3000. All seems to work really fine, there is only a problem when I hangup my IP phone after a conversation and the other party don't: in this case the PSTN line remains connected until the other party hangs up. Any

Hey, all... If this is too off-topic, I'd be grateful for directions to a more appropriate mailing list. I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000 which is registering twice with Asterisk - once for its FXS/Line1/VoIP1 and once for its FXO/PSTN/VoIP2. My eventual goal is to have inbound calls on its FXO ring four times on its FXS and then fail over to

Hi, I'm using a Sipura SPA 3000 to receive calls from a PSTN line and to send them to Asterisk. But on Asterisk, I don't see the PSTN caller ID! I only have User ID of the SPA 3000 as caller number. The caller number is present on the PSTN line. I'm in france, maybe the SPA 3000 is not compatible? If you have an idea, please tell me! Thanks, Micka?l

Hi, I have a little situation here :( Perhaps somebody can give me a hand with it. I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor

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

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

I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu --------Dial 1 for support | Dial 2 for special | Dial 3 sales

Hi, I'm trying to setup a bridge with a bonded device (2 links, balance-rr). The problem is that after attaching the bonded device to the bridge the network throughput drops down from 110MB to 100KB. This seems to be due to the MAC address of internal devices of the bridge being seen on the external ports, where the bonded device exists (see also the URLs below). An arp packet from some

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hello everybody, I?ve been trying to solve a problem for several weeks now but it really beats me. There are several hard phones connected to an Innovaphone 3000 VoIP gateway. On the other side I have a SIP softphone connected to Asterisk. The problem I have is that on incoming calls (hardphones to softphone) I only have outgoing audio (from soft to hardphone); everything is OK when I call the

All the aforementioned techniques need change everytime on the dialplan. I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote: