similar to: SPA-3000 caller ID and KPN

Hello all. I am just getting back into Asterisk and I am setting up my Linksys SPA3102. I have incoming calls working fine, as is the phone plugged into the unit. My problem is I cannot get the SPA3102 to dial a phone number automatically. I can call the extention of the PSTN and I get a second dialtone, and I can then manually dial. I'd like to be able to have Asterisk pass the

Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA

Hi, I'm using a ISDN-30 E1 line from KPN Belgium. The challenge is to get a correct CallerID on outgoing lines. When I put this in my dialplan: exten => _0.,1,Set(TEMPVAR=${CALLERID(num):1}) exten => _0.,2,Set(CALLERID(num)=144622${TEMPVAR}) exten => _0.,3,NoOp(${CALLERID(num)}) exten => _0.,4,Dial(Zap/g1/${EXTEN:1},,) The resulting CallerID is accepted by the telco, but on

Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I

Hi all, My problem boils down to these errors: ... Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time This is triggered by lines in extentions.conf such as: exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W) The system is CentOS v5.2 with Asterisk 1.4.23 (druid-asterisk-1.4.23.1-2), a Sangoma A104

Hi list! I am installing an * box that will be installed on a site with KPN BRI ISDN in The Netherlands. I am using bristuff fron Junghanns. Does anybody know the correct settings for this? I will not have internet access there which makes it harder to google around on location. switchtype = euroisdn is pretty obvious but what about these settings: signalling = bri_cpe_ptmp ; p2p TE mode

Hello all, Recently I purchased an QuadBRI card from junghanns.net after some playing around, reconfiguring dialplans etc with the exception of 1 thing everything seems to work: I seem to be unable to set the outbound callerid. The dutch telecom operator (KPN) provided me with 4 MSN's on 1 BRI interface. In the past years I'm more then used to setting the MSN without the leading 0, this

Hi list, I have a server running a simple hfs isdn card running with chan_misdn. the problem is, I can't get asterisk to pick up the phone, outgoing calls work fine. when running asterisk with -vvvvvvvvvvvc I get the following "log" when I try to dial the isdn server. P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's' extension P[ 1] MGMT: SSTATUS:

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

Hello, I would like to put 1 asterisk box in Country A and 1 asterisk box in Country B. Let's assume : - Asterisk box in country A = GWA - Asterisk box in country B = GWB - Calling party number (located in country A) = CgPNA - Called party number (located in country B) = CdPNB - Second Called party number (located in country B) = sCdPNB - PSTN in country A = PSTNA - PSTN in country B = PSTNB

Hi list, Has anyone been able to get ISDN-BRI support to work reliably on Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro, kernel, modules, versions, config files). I've tried to get it to work on a Debian etch system with an HFC-PCI card and the zaptel package (v1.4.7, also from xorcom.com), but with no luck: all three channels that are created when the

Hi everybody. I''d like to announce that 3.2.11, 3.1.10, 3.0.19, and 2.3.15 have been released. These releases contain two **extremely critical security fixes** so please update **IMMEDIATELY**. You can read about the security fixes by following these links: * [CVE-2013-0155](https://groups.google.com/group/rubyonrails-security/browse_thread/thread/b75585bae4326af2) *

Hi everybody. I''d like to announce that 3.2.11, 3.1.10, 3.0.19, and 2.3.15 have been released. These releases contain two **extremely critical security fixes** so please update **IMMEDIATELY**. You can read about the security fixes by following these links: * [CVE-2013-0155](https://groups.google.com/group/rubyonrails-security/browse_thread/thread/b75585bae4326af2) *

I have a problem with Asterisk-bristuffed using a zaphfc card. I am located in the Netherlands, so I have an ISDN line from KPN. When I start Asterisk, and plug in the ISDN line, everything works perfectly for about 5 minutes. And then the ISDN line is down for 1 minute, and after that minute, the line comes back up and works for another 5 minutes. Every time the line goes down I get the error

Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,

Hi to all, I have a question about the callerid (msn number) off calls comming FROM a queue. This is my setup: - ISDN using zap (zaphfc) - Incomming calls arrive in a queue - One of the members of the queue is my cell phone... member => Zap/g1/XXX73XX19 (X to protect my privacy ;-)) The problem is the call from the queue to my cell phone. In a dailplan i can set the outgoing msn using

Hey all, I recently noticed that DTMF caller-ID was implemented in CVS, so I requested the service from my telco (the Dutch KPN) and tried to get it going in Asterisk (current CVS), without success so far. This system has 1 X100P, 2 TDM400P's with 4 FXS-modules each and 2 HFC-PCI ISDN-cards (zaphfc-driver) in it. The analog line I'm trying to get caller-ID working on is obviously on the

I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? -- #Joseph GPG KeyID: ED0E1FB7

On Tue, Jan 09, 2018 at 06:53:51PM +0000, Kaylor, Andrew via cfe-dev wrote: > I think we're going to need to create a new mechanism to communicate > strict FP modes to the backend. I think we need to avoid doing anything > that will require re-inventing or duplicating all of the pattern > matching that goes on in instruction selection (which is the reason >

>The standard argument against trying to introduce "scope-like" mechanisms to LLVM IR is inlining; >unless you're going to prevent functions that use stricter/laxer FP rules from being inlined >into >each other (which sounds disastrous), you're going to need to communicate strictness on an >instruction-by-instruction basis. If the backend wants to handle that by