similar to: Announcement: YAACID (Caller ID for Asterisk)

Hello, I'm trying YAACID ( http://www.shatterit.com/opensource/yaacid/ ) for incomming call notification on PC (and open url with callerid), but it does not display/pop anything :-( my config is very simple... (yaacid is successfully registered as manager in asterisk) thanks PJ * dialplan: '953' => 1. NoOp(${CALLERID}) [pbx_config]

Does anyone know if the 1-800 iaxtel gateway is down? I've been trying to use it all day today and asterisk says it's ringing: Channel (Context Extension Pri ) State Appl. Data IAX2[iaxtel]/1 ( s 1 ) Ringing AppDial (Outgoing Line) SIP/2201-a253 (home 18888476626 1 ) Ring Dial IAX2/XXX:YYYY@iaxtel.com/18888476626@iaxtel But I

hey all, from a lot of great feedback we found some bugs in YAACID that appear in *@home and the stable asterisk versions and older CVS versions. we're putting the fixes in and will have a new one tomorrow on the web site.. Thanks, Mark

Hi - I am going to open port 5038 on my firewall so that I can use YAACID to spawn browser popups on an incoming call. My question is, under manager.conf, what are the suggested settings so that I can get the browser popups only? I'll be at different IPs so I can't lock it down that way.. I guess I don't need any write access? [managername] secret=secretword

For redundancy I would like to write the CDRs on tow mysql servers. cdr_mysql.conf accept only one configuration [global], how to add a second host? Thanks Rosario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050609/9eacbae2/attachment.htm

After an extensive conversation with Mediatrx 's sales department , I stand corrected and so does the salesman who spoke to me. My apologies to Voip Supply. I understand now you never knew about the CD. Garrett Smith wrote: > I though I would post an update for everyone on what DOES and DOES NOT > come with every Mediatrix product. > > > > Every Mediatrix product,

Hey Everyone, We are in the process of reviewing headsets for use with our GXP-2000s. I'm looking for some feedback as to which headsets people are using, the pros and cons of those headsets, and if they would recommend them to someone else. Any help would be appreciated... - Jason -------------- next part -------------- An HTML attachment was scrubbed... URL:

Are there any other wireless IP phones out there other then the Cisco 7920?? -- James Moran <jmoran@potentialtech.com> Potential Technologies

Hi, I am generating following code: %base = getelementptr inbounds %ravi.CallInfo* %6, i32 0, i32 4, i32 0 %7 = load %ravi.TValue** %base %8 = bitcast %ravi.TValue* %7 to i8* %9 = bitcast %ravi.TValue* %5 to i8* call void @llvm.memcpy.p0i8.p0i8.i32(i8* %8, i8* %9, i32 16, i32 8, i1 false) %10 = load %ravi.CallInfo** %L_ci %base1 = getelementptr inbounds %ravi.CallInfo* %10, i32 0,

Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has

On 22 February 2015 at 22:54, Hal Finkel <hfinkel at anl.gov> wrote: >> I tried setting the module's DataLayout to the engine's DataLayout. >> Don't see any improvement. >> The memcpy() is to perform a struct assign, so I tried replacing that >> with member by member store. >> But even then the loads are not being eliminated so I guess the >>

Hi Guys, Hope fully somebody out there will have experienced this and can shed some light on how it was overcome. Current setup includes asterisk 1.6.2.11, GNU GK and a Quintum Tenor CMS on the same lan. Earlier I was unable to make a sip call from the CMS back to a sip client registered on my asterisk box. So I moved onto passing the call from the Quintum CMS to a Quintum Tenore DX which is also

On 22 February 2015 at 20:58, David Jones <djones at xtreme-eda.com> wrote: > Not sure if this is your problem, but it was mine: > > You must create (or obtain) a DataLayout *and install it into the Module*. > > It is possible to generate machine code for IR and not install the > DataLayout into the Module. Rather, the DataLayout is used locally at the > point where code

The parts of klibc specific to the parisc architecture. Signed-off-by: H. Peter Anvin <hpa at zytor.com> --- commit 078d6614054391efe17093f8d70340e2c0644ffb tree 63a4bf899e5ca2ef3c0a8e9ef3098273012f7a33 parent ebd2860ad3dc19cb11fd5b9cc235cab54e9165f4 author H. Peter Anvin <hpa at zytor.com> Sun, 25 Jun 2006 16:58:36 -0700 committer H. Peter Anvin <hpa at zytor.com> Sun, 25 Jun

I'm implementing this unusual gcc form of mips16 hard float. I ran into one unexpected wrinkle. It effects efficiency as opposed to correctness but I'd like to figure out the best way to resolve it. The signatures for the libcall routines have floating point arguments in at least some situations. I have not fully examined if it's all of them. So in callinfo while I'm lowering

What happens here is that the code that is doing the "softening", generates a libcall On closer inspection, llvm is doing half of this correctly. The signature for the soft float function for extending a float to double is of the form: i64 xxx_extendsfdf2(float); But it should be i64 xxx_extendsfdf2(i32) This particular paradigm is duplicated many times in the the float type

Hello, Maybe some of you tried the SIP support recently introduced by Quintum in their AS devices. I have one Asterisk machine connected to PSTN via E1. It works properly. On the other side I got an ADSL line, with NAT and few devices behind it, like computer with X-Lite client installed or mentioned Quintum device. It works great - calls initiated from there are OK, as well as PSTN originated

Hi, I am using the LLVM release_37 branch and have a strange issue. Please see the two IR outputs I have saved below: https://github.com/dibyendumajumdar/ravi/tree/master/ravi-tests/comp_issue The original.ll is the code I am generating. The aftercompiling.ll is the output from LLVM with opt level 0. The issue is that LLVM is deciding to remove the blocks with labels updatei and updatei.64

Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() application by way of setting the CALL_INFO variable. For example, the following macro can be used to dial up a single

I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13) and noticed a setting marked 'paging' under supplementary services on the Phone settings page on the advanced admin login. Anyone know how it might be used? Could it be like the Snom - exten => 10,1,SetVar(VXML_URL=intercom=true) exten => 10,2,Dial(SIP/testuser) Craig