similar to: SIP Remote-Party-ID

Hi to all, to manage properly a call center for multiple companies is possible to let the X-lite/X-Pro softphone to display the number or context called from PSTN to let operator answer with the correct name of the company?? I explain better. If a call come from PSTN to Number A for company A i want the operator recognize it and answer "Good Morning, I'm Operator of company A"

Hi, i read the * support ringing multiple devices at the same time, i inserted this line on my configuration on default context: exten => s,1,Dial(SIP/260&SIP/261&SIP/262&SIP/263|30) exten => s,2,Voicemail,u260 exten => s,3,Hangup And i have both 4 clients in sip.conf . The problem is that if i call it fall immediately in the Voicemail if the client 260 is not registered .

Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to

Hi to all, we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi, notice that i have Grandstream phones and i have the problem if i activate the Send Anonymous function on them. If i do not activate that option the ACCOUNTCODE is correctly populated. SO i think it may be a bug of asterisk. I'm using Asterisk CVS-HEAD-10/07/04-18:07:25 . Thanks, Bye, Marcello

Hi to all, can you suggest to me the best way to avoid problems in the CDRs for anonymous sip calls? I have some peoples that set Send Anonymous : Yes in their Grandstream phones and i don't receive the username as phone number that i use to make billing. It is empty. The only place where there is the phone number is in the peer name where it write the name of the peer that in this case is

Hello, I'm using Asterisk 1.2 with MySQL support. I use sip_buddies table for SIP clients definition. My problem is that I can not define CLIR. Sip.conf docs says that restrictid = yes hide caller identification. The problem is that definition of sip_buddies field named restrictid is char(1). I tried to set restrictid = y, = 1 - no results. I changed definition of the filed to restrictid

Hello list, We have some kind of a problem with our Asterisk installation. We want to be able to publish different caller id when placing outbound calls through the PSTN. We have Asterisk with TE410P and T1 from FDN Communications. The problem is that all our outbound calls show our main number, regardless of what we set with SetCallerID, even using CallingPres with all possible

I've developed a web application in PHP and R my script is <?php ... exec("R CMD BATCH --silent /home/marcello/R_in/myfile.bat /home/marcello/R_out/myfile.out"); ... ?> This script execute in R batch mode and write the myfile.out. On Win2000 the similar script is ok, but on linux I've a problem. I suppose is a permession problem because the same script on shell

2011/11/21 Tobias Grosser <tobias at grosser.es>: > On 11/20/2011 04:36 PM, Marcello Maggioni wrote: >> >> Sorry for the noobish question, but what kind of subscripts generate a >> SCEVCouldNotCompute  from the SCEV engine? >> I tried for a while but I wasn't able to trigger that > > Hi Marcello, > > the SCEV returns SCEVCouldNotCompute in case it

Hi Jean-Marc, well spotted! The patch provided fixes the issue for me. Nevertheless, in my code (and I would suggest doing the same in libopus) I am going to replace the function with one that accumulates on 64 bits and then calculates the shift, for at least 4 reasons: - It is less and simpler code - The result is likely to be slightly more accurate in case big numbers come early in the

On 11/18/2011 01:36 PM, Marcello Maggioni wrote: > 2011/11/18 Marcello Maggioni<hayarms at gmail.com>: >> The patch is attached. >> >> The patch to correct the test runs on OSX will be posted shortly after >> this one (I preferred to separate the two so that a problem with >> either one of the two wouldn't give problems to the other and also to >>

Hi Marcello, Thanks for your reply. I will try to pass down the mask! I have one more question. In my backend I return CSR_RegMask in getCallPreservedMask and return CSR_SaveList in getCalleeSavedRegs. Is that a correct setup? I dumped the regmask and found that callee saved regs are marked 1 and non-callee saved regs are 0. Thanks, Xiaochu On Wed, Aug 26, 2015 at 5:58 PM Marcello Maggioni

On 11/21/2011 12:44 PM, Marcello Maggioni wrote: > 2011/11/21 Tobias Grosser<tobias at grosser.es>: >> On 11/20/2011 04:36 PM, Marcello Maggioni wrote: >>> >>> Sorry for the noobish question, but what kind of subscripts generate a >>> SCEVCouldNotCompute from the SCEV engine? >>> I tried for a while but I wasn't able to trigger that >>

We have a scenario where we wish to present a toll-free caller id, yet have our calls rated based on our billing-telephone-number. Is it possible to present a number in the sip header for billing and another number in the header for jurisdicional call rating? Whereas today, all of our calls are billed at the highest rate (intra-state) because we're presenting a number that isn't in the

Hi Jean Marc, please find attached the audio file (mono 16khz). I shortened it to about 10 seconds. I also add 2 patches that worked for me. Further info that might help: - The problem seems to be related to silk_burg_modified not reaching the maximum gain, so the actual filter order is 16 rather than 2 (which is what would be expected with a sine wave). - The problem seems to happen when

Something like, keeping the compiler alive for a many compilation sessions, using the same LLVMContext, but without the problem of accumulating data in the LLVMContext that would slowly fill up the memory. This as much I can enter into details that I can :/ Probably this is also a quite common use case scenario too. Marcello 2014-06-24 18:52 GMT+01:00 Eric Christopher <echristo at

Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the

Hello all Im using a linux box running CentOS 4.1 as a proxy server with user auth with an AD Its working for a long time, but suddenly this weekend the users cant authenticate anymore looking on logs i obtain this Oct 10 08:29:59 sol (ntlm_auth): [2005/10/10 08:29:59, 0] utils/ntlm_auth.c:get_require_membership_sid(237) Oct 10 08:29:59 sol (ntlm_auth): Winbindd lookupname failed to resolve