similar to: rtcp to cdr for calls from dahdi to sip

I have a bunch of CDR records in the mysql database "asteriskcdrdb" on a FreePBX system. There is a DAHDI trunk defined in FreePBX which uses the "gN" identifier to make calls. So in this setup the trunk is roughly equivalent to a DAHDI logical group. I want to know, given a CDR, which logical group (and therefore, trunk) was used to place the call, or was used to receive the

Hi list, I need some help to improve my cdr, now in my company are asking me how to know which of my phone numbers are most used when receiving calls from the PSTN and incoming the IVR was thinking about using userfield field, and I'm trying to do, I have at the moment 4 channel DAHDI ; DAHDI CHANNEL 3=23XXXXX6 context=in callerid=asreceived group=1 signalling=fxs_ks channel => 3

I am not sure if its just me, but i am able to get only local channel variables containing RTCP QOS values. The Version is 1.8.14. I want to store values of bridged channel in CDR. Phone is Cisco 7941 SIP and with sip show channelstats i see all the relevant information (jitter,packet loss) i want to get. It even calculates packet loss in %. But i am not able to store it to CDR. Asterisk 1.4

Hi, Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. ! ael agent agi cdr channel cli config console core database devstate dialplan dnsmgr dundi features file group hangup help http iax2 indication keys

05.03.2015 11:42, Dmitry Melekhov ?????: > 05.03.2015 11:29, Dmitry Melekhov ?????: >> Hello! >> >> Just installed asterisk 13.2.0 and see many such messages in log, I >> see them in console during calls, really something like this: >> >> >> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", >> "SIP/6166 at

Hi all, I need 2 CDR Records of below 2 numbers for outgoing calls, detail is given as below: *96XXXXXXXX88XXXXXXXX* *=> Call file : outbound call generate through below file* Test.call ====== Channel: local/s at outgoing/n WaitTime: 45 Context: outgoing_ivrs Extension: s Priority: 1 Set: contact_no=96XXXXXXXX extensions.conf ============ [outgoing] exten => s,1,NoOP(----- First LEG

Hello gang, We just got MaBell to turn on our callerid. I tested the capability with a southwest bell box and a plain phone, so I know the line is sending the signal. I'm running Asterisk SVN-branch-1.4-r204834 using a TDM400P card. Here is my dahdi_cfg -vv output: dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.3 Echo Canceller(s): MG2

Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve

Hi, The ami manager call out with an originate through dadhi to a local number (A). If this call is answered, then the ami manager redirect this call to a dial command. This dial command calls through dadhi to another local number (B). Number B answers this call and number A en B are connected. If number B and number A hangs up, there is will be no CDR be written If the dial command is commented

This article below came up on the newwire. The RTCP XR RFC was published. Will Asterisk be supporting this function in a future release? Does anyone know if any phone vendors are going to be supporting it? Thanks Lee Goodman Our Technology Update this week is about one of those mechanisms. Known as RTP Control Protocol Reporting Extensions (RTCP XR), the technology defines a standard way to

On Tue, Mar 10, 2015 at 5:00 AM, Dmitry Melekhov <dm at belkam.com> wrote: > 05.03.2015 11:42, Dmitry Melekhov ?????: > >> 05.03.2015 11:29, Dmitry Melekhov ?????: >>> >>> Hello! >>> >>> Just installed asterisk 13.2.0 and see many such messages in log, I see >>> them in console during calls, really something like this: >>>

I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting asterisk i am getting this error on console. func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered custom function

HI, I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26

Hi, My costumers are logged in on my Asterisk PBX through XLite Softphone (SIP). My server is connected to PSTN. Problem is when SIP phone calls ordinary phone via dahdi I get DAHDI/1-1 ANSWERED SIP/number-number and billsec field from cdr is start counting. Is it normal behavior ? Can I change that ? So channel gets in ANSWERED state and billsec starts as soon as line starts to ring even if no

http://bugzilla.netfilter.org/show_bug.cgi?id=757 Summary: SIP connection helper not setting RTCP conntrack expectation Product: netfilter/iptables Version: linux-2.6.x Platform: i386 OS/Version: Ubuntu Status: NEW Severity: normal Priority: P5 Component: ip_conntrack

Hi, Asterisk 11 doc says CDR(src) value is read-only (see https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). For various reasons, I would appreciate to change its value so that it my own presentation rules instead of telco rules. Very often, I'm connected to telcos through DAHDI (and ISDN). For instance, telco presents calls with 123456789 while I would prefer a normalized

Hi 1. What is the best way to convert wav (44000 Khz) to gsm format for asterisk ? I;ve tried sox command but the outcome is not satisfying...The built-in gsm files shipped with asterisk are simply superb ..How do i create gsm files of similar quality ? Can anyone help me out ? if sox is the only way can anyone tell me the exact command ? 2. Can Freepbx 2.5 installed above asterisk 1.6.0 or

I used my cell to call in and create a CDR record here from asterisk 1.4.43: "","317XXXXXXX","s","default","""GEIS JERRY "" <317XXXXXXX>","DAHDI/23-1","","BackGround","SM_ATTENDANT","2012-01-05 18:12:09","2012-01-05 18:12:10","2012-01-05

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

Hello, We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls. With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up already by the caller. It seems that when a BYE is lost, Asterisk has no mechanism to check whether a call is still active. Is there a way to activate a RTCP call control,