similar to: Read and set the UUI in asterisk

Dear All, I have "get full variable" AGI call to get the ANSWEREDTIME channel variable. I have originated the call to one extension, once answered I have called DeadAGI to control the call. I have problem that after hangup the call AGI "GET FULL VARIABLE" returns -1 for ANSWEREDTIME channel variable. What is the problem? Where I made wrong. Please suggest me..

Hi, I have followed the all the steps given, in https://help.ubuntu.com/community/ActiveDirectoryWinbindHowto. to integrate the samba with active directory. I have the following configuration file, [global] workgroup = ASSURANCE security = ads realm = ASSURANCE.LOCAL encrypt passwords = yes winbind separator = + idmap backend = lwopen idmap uid =

Hi there, Is it possible to manage UUI with asterisk and ISDN (T0 Fritz card). Basically, is it possible to send User to User Information using the D-channel, while making a call?

Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. My configurations as follows, [2000] type=peer host=dynamic insecure=port,invite ; (both) context=Testing But when I call '2000', I noticed the following message in Asterisk console, "NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user

Dear all, In Sip.conf file how to setup incoming calls not to use authentication? Please provide some steps to do it.. Thanks... Regards, Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090801/d6c39afc/attachment.htm

Hello List, I need to connect with an Avaya PBX (this part is done), and i would like to get and send back User-to-User Information (UUI) with the call. The UUI need because I need to identify the call based on something witch is available on Asterisk and Avaya too. It is possible, or have a better solution? Anybody did it before? Thanks for the help! Zsotya

Hi, I came thru ISDN UUI (User-User Information) protocol which is defined in this RFC - http://www.ietf.org/id/draft-ietf-cuss-sip-uui-17.txt But I don't understand how to use this with Asterisk. Any idea would be much appreciated. Thanks. Gopal. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, If the special handling in the meg2reg pass is to look for lifetime intrinsics, shouldn't it cast to <IntrisicInst> and then use getInstrinsicID to check for lifetime_start and lifetime_end ? The thing that I don't understand is the following piece of code, which finds all the users and cast it to <Instruction> then eraseFromParent(). How can this guarantee that it only

OK, so I'll do that... Is there any infos I need to know about chan_sip.c (because I suppose it's it that I need to play with)? Does anyone know an interesting website where I can find infos about UUI in ISDN (with CAPI maybe?) ? Thanks for your help.

Can you put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards ---------------------- > >jean-marie.goupil@telintrans.fr wrote: >> OK, so I'll do that... Is there any infos I need to know

Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. " WARNING[8446]: chan_oh323.c:3555

Hello, I've seen that Nuance SIP audio provider supports additional information (parameters and extra headers) in SIP URIs, using the format: sip:user:password@host:port;uri-param1;uri-param2?header1&header2 For example, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide

Dear All, I have installed Asterisk 1.6.1.9 to use Bridge Application in AMI. After inatallation I have tried to connect the AMI via telnet. But it didn't connected. I used netstat to know the listening socket. But it was not available. How to start the AMI server socket. Please any one help me....... Thanks, Velusamy. -------------- next part -------------- An HTML attachment was

right, so far, here is what I've done: I succeed in take in a new variable the UUS1 field sent with the connection request for incoming calls. It was quite simple afterall... (I just had to find where the data CMSG->Useruserdata is coming in chan_capi.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I

----- Original Message ----- > From: "guoqing zhang" <gqzhang81 at gmail.com> > To: llvmdev at cs.uiuc.edu > Sent: Friday, January 30, 2015 4:29:16 AM > Subject: [LLVMdev] About user of bitcast/GEP instruction > > Hi, > > > In PromoteMemoryToRegister.cpp, it seems to rely on the fact that the > only users of bitcast/GEP instruction are lifetime

Hi, In PromoteMemoryToRegister.cpp, it seems to rely on the fact that the only users of bitcast/GEP instruction are lifetime intrinsics (llvm.lifetime.start/end). I did some searching in llvm/test folder, it seems to be true. However, by reading LLVM IR manual, I don't see any restriction stated on the possible user of bitcast/GEP instruction. So my question is who impose the restriction ?

Hi, Asterisk 1.4.17 Sangoma a102DE I'm having some issues sending CallerID Name to a Dialogic based phone app. According to the pri debug (asterisk2a-pri-debug.txt in [3]) you can see that it is sending the CallerID Name "Mike - Budgetone - reachme.com" to the Dialogic card, but it isn't regestering on the Dialogic based system. I can receive CallerID Names from our Paetec,

Hi all, This is Jagan.i have been provided a task of analyzing sales data of a company in R programming...Just wanted to know,how can I pull Tally 9.1 software data into R programming dataframe. Waiting eagerly for your inputs. With Regards,Jagannathan Krishnan Sent from Yahoo Mail on Android On Tue, Aug 22, 2017 at 10:47 AM, jagan krishnan<jagan.krishnan at yahoo.co.in> wrote: Hi

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

Hi, I need help in calculating Java method time-stamp in following fashion. Consider following method example. long method3(long stop) { try { Thread.sleep(1500); } catch (Exception e) { } //////////////////// real CPU intensive operation /////////////////////////// for (int i = 1; i < stop; i++) { stop = stop * stop * i; };