similar to: Error saving cdr at h exten in Asterisk13

Hi, We are try new Asterisk13 and was noted it don't execute h exten priorities inside macros. We have a macro where we make all our call processing, and we use h exten inside it for billing (updating CDR(vars)). If context where that macro is called have some h extens, asterisk execute them. So, I wonder, h exten inside macros was deprecated? Thanks in advance. Atenciosamente,

2013/3/29 Julian Lyndon-Smith <asterisk at dotr.com> > check out the endbeforehexten option in cdr.conf > > this needs to set to "yes" > > Julian > Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the "saving to file" part. Then my issue is I can't update CDR value is hangup exten. Here is a

With endbeforehexten=no I actually get two CDR entries. One for the call and a second one for the "h" extension. "","13","10","sip-locals","""13"" <13>","SIP/13-00000006","SIP/10-00000007","Dial","SIP/10","2015-08-04 06:28:44","2015-08-04

Ok I can't get my CDR values to set from the h extension in either 1.6.2 or 1.8 What is wrong? Here is what I found in the cdr.conf ; Normally, CDR's are not closed out until after all extensions are finished ; executing. By enabling this option, the CDR will be ended before executing ; the "h" extension so that CDR values such as "end" and "billsec" may

Hi, I'm trying to migrate from Asterisk 1.8 to Asterisk 13 and can't figure this one out. I'm pretty sure the question has been already asked, but I failed to find a solution. Can you modify CDR values in an h-extension? My cdr.conf contains: [general] enable=yes unanswered=yes endbeforehexten=yes initiatedseconds=no batch=no The diaplan contains a simple "h" extension

Hi, Reading this old thread, may I ask if keeping hangup handlers from updating CDR values still enforced in Asterisk 15 ? If positive, would it be very complex to add in Asterisk, a configuration option allowing a system administrator to list in cdr.conf, the CDR fields allowed to be updated in hangup handlers ? I'm planning to store some RTCP stats. Saving them in CDR(userfield) would be

According to the instructions given at https://www.asterisksounds.org/de I converted and installed German prompts successfully and for numbers, I can successfully listen to a German female voice counting or telling the date/time. But unlikily, somehow the voicemail prompt is still English, although my general language settings are "de". I use pjsip.conf, not sip.conf. In

Scenario: Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisk's wiki, the transport section of pjsip.conf is configured as follows: ; Transport via UDP [transport-nat-udp] type= transport

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

Can someone tell me what I am doing wrong? Why doesn't CDR(duration) or CDR(billsec) return the correct values? cdr.conf endbeforehexten=yes extensions.conf [macro-Dial] ; ${ARG1} - Dial String exten => s,1,Dial(${ARG1},,M(post-dial)) exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long, billed for ${CDR(billsec)} seconds) The log shows: -- Executing [h

This was always possible in the past, however does not work in the current release. I believe this is a bug. To: asterisk-users at lists.digium.com From: cervajs at fpf.slu.cz Date: Fri, 9 Oct 2015 10:04:47 +0200 Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR search in archives save the records to another table like cdr_extended Dne

I have two version 1.6 Asterisks running. One is a small hobbyist thing just at home, and the other is handling calls for several customers. On both, I have added the line exten => h,1,Set(CDR(hangupcause)=${HANGUPCAUSE}) to all relevant contexts. On my little hobbyist box this works perfectly; all calls have their hangupcauses recorded with cdr_adaptive_odbc and cdr_custom. On the

Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR

I am using the new 1.8.5 and I just found out that Asterisk won't record the call if the call just hangup. I did a test like this: exten => 1009, 1, Hangup() Then I called 1009: == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [1009 at init-1005:1] Hangup("SIP/1007-0000003c", "") in new stack == Spawn extension (init-1005, 1009, 1)

Hi, I'm having problems with CDR's and Queues in Asterisk 1.6.1. Heres three examples: Normal call: User A calls in to asterisk, gets a PlayFile, and hangs up. This gives 1 CDR as expected. Call to a Queue and then a playfile afterwards: User A calls into asterisk, goes into a queue, asterisk calls user B. When user B hangs up a CDR for User A is generated. (no CDR for user B)

So... I'm trying to access CDR(duration) and CDR(billsec) inside h... I keep getting 0. Can I access the CDR function inside a hangup extensions? Asterisk 1.4.13 Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Hello, does some people here use https://voicetrading.com which is a Dellmont service from Netherlands. At the high begining they were also known as Finarea (CH and DE mixed Co) Anyway, after moving from Asterisk13/chan_sip to Asterisk16/PJSIP our callerID is no more seen by them. We use Set(CALLERID(num)=+331234356789) and Set(CALLERID(name)=Co name) or equal to CALLERID(num). We tried

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers I was getting this : [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf after fix global issue