similar to: send Q931 information element keypadfacility ?!

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

Hi Guys, We have a problem with Caller ID not being displayed. I want to test everything to see where the problem is with the incoming Caller ID and why it's not displaying. I am tracking this down to "Presentation prohibited of network provided number" even though the Caller doesn't use *67 and even though they haven't asked their provider to block their CLID for outbound.

Hi Guys, Can q931.c be re-compiled using gcc or something else without the need to re-do the whole libpri? Some changes were made to q931.c and I want those to be reflected in .a .o .so .lo files as I think those are the files read by Asterisk vs the .c file. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Please help me, i need transfer a call in asterisk to other telco number and free the channel. Can i do with any q931 function?. Thanks a lot Aris... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090528/2bcd93ae/attachment.htm

Hi, does anyone know if its feasible to run asterisk with a PRI card but not run any q931 signalling.. basically push calls down the PRI and tell asterisk in some other way to pickup a particular Zap channel? Steve

Dear list, I would like to generate random numbers from a Voigt distribution, hopefully in a way as simple as getting random numbers from a normal distribution with 'rnorm'. Is there any package to do this? Speed is an issue in this application. Or, as the Voigt distribution is a convolution of a Gaussian and a Lorentzian, can I simply combine random numbers from rnorm and rcauchy in some

Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and

Dear expeRts, first of all I'd like to thank you for the quick help on my last which() problem. Here is another one I could not tackle: I have data on an absorption measurement which I want to fit with an voigt profile: fn.1 <- function(p){ for (i1 in ilong){ ff <- f[i1] ex[i1] <- exp(S*n*L*voigt(u,v,ff,p[1],p[2],p[3])[[1]]) } sum((t-ex)^2) } out <-

I have a server running Asterisk 1.4.14, Zaptel 1.4.6, Libpri-1.4.2 on a CentOS 5 server. The server has a single TE110 card connected to a provider called Alestra in Monterrey, Mexico. Since we installed everything we have been having problems dialing certain numbers, those numbers always fail when dialed from Asterisk but if you dial from your cell phone they always go through. I once has a

- I've just installed 0.6.2, & I would like to see the q931 messages going back & forth. I turned on debugging with "h323 debug toggle", which the README says is "very verbose", but I don't see much. Is there a way for me to see more debugging information, like the "debug isdn q931" of IOS? Or am I missing something? Thanks, Glen IAS

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

While testing a problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon this problem: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1210770512 (LWP 11346)] 0x002e3fe1 in pri_release_timeout (data=0x93a0478) at q931.c:2589 2589 q931.c: No such file or directory. in q931.c q931.c is in libpri, function

Hello! I've got a question regarding the Q.931 Setup-field Calling Party Number. It contains five things: Type of number, Number Plan, Presentation and Screening indicators and the actual number. Our provider uses some of those to decide if the numer should be presented or not to the outside world. I've done a crude hack in our GnuGK to always change those so that our numbers are

Hi Guys, I am calling out 416-999-1111 on Channel 1 of PRI and then calling 416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/4169992222 -- Zap/2-1 is proceeding passing it

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16.

Hi, My configuration is pri----*(te405p)---iaxclient. My * version is 1.0.7 running on tyan dual opteron board. I have several problems. 1) inbound echo For outbound call(iaxclient-->pri), there is almost no echo. But for inbound(pri-->iaxclient), I can hear distinct echo. Can Sangoma a104 or digium te406p help this problem? 2)Today i received te406p. I know T1/E1 jumper. But how can i

Hi, how may I define in a callfile the CallerID presentation to be used for the requested call, eg. set it to prohibited? TIA, Bruno -------------- next part -------------- A non-text attachment was scrubbed... Name: Bruno.Voigt.vcf Type: text/x-vcard Size: 270 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050913/fcb5c595/Bruno.Voigt.vcf

Dear List, Consider the following example x=c(1,2,3,4,5) y=c(2,4,6,8,10) linearmodel=lm(y~x) To predict a y-value if you know the corresponding x value is very easy with the command predict. predict(linearmodel, newdata=(x=1.5)) The other way around, to predict an x-value with a corresponding y-value, doesn't work unfortunately. Is there another function that can do that, or do I need to

Hi all, I've programmed on a SNOM function key the destination "**77" which in my extension plan reprograms/toggles an asterisk DB variable which I use in the extension plan for some call routing decisions. I would like the SNOM extension light to permanently reflect the current toggle status of my application logic/asterisk DB variable. Is something like that possible to do in