similar to: Hangupcause to handle Called party disconnect ? PSTN----E1----OldPBX---E1--Asterisk

Hi all, My scenario is this one: LandLine------------------E1---------------|-------------| |-------------------| |OLDPBX|-------E1-----------|Asterisk1.2.5|-----VoIPusers GSMGateway---------Analogue------ |-------------| |-------------------| What is happening: 1- SipUserAgent "A" Dials a call to a Local Extension "B" in the OldPbx 2- "B" , the called party

Hi all, i have: out side PSTN-------->OldPBX---------Analog----->Asterisk (X100P) ^ | Local Ext What is happening is: Calls from Local ext goes to Asterisk and everything is fine. Calls from Out side PSTN reach the OldPBx and are bridged do Asterisk, but i don't listen firs audio

Hi , Today I have the following sharing in "Samba version 2.2.12 for Solaris 2.9": [produto] write cache size = 384000 comment = AREA DE DADOS DE PRODUTO path = /usr/produto read only = No valid users = @produto where exists the users cs02929 pertaining to the group @produto. And everything

Hi, I really need help(for God). After that I installed the version 3.0.x (actual: 3.0.23d) I am have many problem. when I create an file inside of exported area of samba, where exist ACL?s fo group of the Solaris, the samba do not respected these ACL?s. I can to create files, edit it, but I do not can to rename or to erase/cancel this file. (always said problem acess denied) Please, help me.

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

Hi, I am using Asterisk: 1.6.1.20 LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (A104d) I handle some calls with my own PHP-AGI-Script. After a dial-command I use "GET FULL VARIABLE ${answeredtime}" or "GET FULL VARIABLE ${dialstatus}" and get valid information. Sometimes "dialstatus" has the value

Hi, I have the following code that operates when a channel is hung-up: [record-hangupcause]exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})exten => s,n,Return() Before the dial a hangup handler is registered: Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1) The routine is called and the variables are being set, however not on the channel's CDR which made the call. I believe this

I am trying to include the ${HANGUPCAUSE} in my mySQL cdr tables. I have a field called cause_code but it won't write. I belive it is because the record has already been written by the time I hit the h section of the code. How might I get this info into the CDR. I need this info for Quality of Service as well as route checking. Any ideas would be apperciated. Here is my dial line and my h

Today I get the remark that a call got disconnected after 10 minutes. This what my VERBOSE-logfile tells me : [May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing [00493516426 at intern:1] NoOp("SIP/51-b76023b8", "Gesprek naar GSM-nummer via Telenet") in new stack [May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing [00493516426 at intern:2]

Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris

Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful.

Hi, I'm trying to get HANGUPCAUSE on my cdr the problem I'm facing is that this option: endbeforehexten=yes is not working at least on asterisk 1.6.0.1, so if I put yes o no I cant set CDR value with that value. It seems to finish the CDR record before h is executed. I'm using cdr_mysql. Any idea?? Thanks!! -------------- next part -------------- An HTML

** THIS IS NOT WHERE YOUR REPLY BELONGS ** On Wednesday 25 Mar 2015, Salaheddine Elharit wrote: > tnaks for your response but the number dialed exist and i can call this > number when i configure the trunk directly in x-lite and i call call also > this number from my cell phone . > any help > thanks and regards Make sure you are sending the number in the correct format, when you

Hi List We want to be able to reject some pjsip calls with a 'temporary' failure so that the PBX of the caller plays an announcement in the language of the caller, that the call can temporarily not reach the destination. https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings Case 41 seems right... Haugup(41) causes the call to be rejected with '603' causing the

Hello list, Hope all doing well! I've been checking some cases when a Dial fails and dialplan execution continues to handle this. I am finding it a little confusing how we should handle the DIALSTATUS and the HANGUPCAUSE in this situation.... More specifically, I am facing a case in version 13.6.0 where I am getting a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP error. Seems

Dear R Community, I am pleased to announce that a new version of the mefa R package is available at the CRAN. mefa is a package for multivariate data handling in ecology and biogeography. It provides object classes to represent the data coded by samples, taxa and segments (i.e., subpopulations, repeated measures). It supports easy processing of the data along with relational data tables for

Dear R Community, I am pleased to announce that a new version of the mefa R package is available at the CRAN. mefa is a package for multivariate data handling in ecology and biogeography. It provides object classes to represent the data coded by samples, taxa and segments (i.e., subpopulations, repeated measures). It supports easy processing of the data along with relational data tables for

Hello useRs, MBESS (Methods for the Behavioral, Educational, and Social Sciences) has recently been released and should be on all of the mirrors by now (with binaries for Mac and Windows: http://cran.r-project.org/src/contrib/Descriptions/MBESS.html). The major contribution of MBESS is confidence intervals for noncentrality parameters (t, F, and chi-square) and standardized effect sizes (e.g.,

Hello useRs, MBESS (Methods for the Behavioral, Educational, and Social Sciences) has recently been released and should be on all of the mirrors by now (with binaries for Mac and Windows: http://cran.r-project.org/src/contrib/Descriptions/MBESS.html). The major contribution of MBESS is confidence intervals for noncentrality parameters (t, F, and chi-square) and standardized effect sizes (e.g.,

Dear list, The new package 'multivator' is now available on CRAN. This presents a multivariate generalization of the emulator package. The corresponding JSS article is: Robin K. S. Hankin (2012), "Introducing multivator: A Multivariate Emulator", Journal of Statistical Software, 46(8), 1-20. URL http://www.jstatsoft.org/v46/i08/ best wishes Robin -- Robin Hankin