similar to: cdr on call transfer

hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2.

Hello, I was looking into cdr (w/ mysql) before enabling dialout through CAPI, found out that transfered call are not reported as such. For instance, let's say we have 2 local SIP extensions A and B and C is a GSM number. If A calls B and B transfers to C (automatic or after picking up) the CDR shows A calling C. IMHO, It should record A called B then B transfered to C, so that the cost of

hi ! I have asterisk with Licensed G.729 codec enabled. Whenever I make a call using this codec a warning apears as, WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames

hi! I wanna do some arithmatic operations (addition and substraction -integer operation) inside extensions.conf. Is there a simple way to do this. If I do yy = ${xx} + 1 // say "xx" is initialized to '0' the resulting "yy" will show "0 + 1" Obiviously not the result I need. Any help !!!!! denzel.

Hi All, What the support that asterisk has to send/receive Faxes? Can I plug a FAX machine in the a FXS extension and send out Faxes? What's the codec I need to use? g.711? Also can we receive a FAX into a FXS extension in Asterisk PBX? Also I need to know if we can send/receive FSK data from/to an extension plugged into Asterisk PBX? For example if there's a phone model which can send

Hello, I have an application that needs to record outgoing calls. It's running on Asterisk 1.4.18, with CDR data stored in MySQL. Outgoing calls are recorded based on their uniqueid. When outgoing calls are placed, there is a line like this on my extensions.conf: exten => _.,n,MixMonitor(/var/spool/asterisk/monitor/${UNIQUEID}.gsm) For regular outgoing calls, this works fine. The

Hi, There's a quite complex dialplan scenario and I found out that CDR of main channel is flushed right after hangup on Local channel. I will try to simplify my scenario: [incoming] exten => 555,1,Noop(do something before using local channel, fill some variables, play IVR menus and so on) same => n,Dial(Local/555 at office/n,,g) same => n,Noop(Notice the option "/n" and

Hello I have a problem with bad CDR's after transfer of call. This is an example: I've called from 616222820 to 616222821. Next I've called from 616668020 to 0600340000 and then I've transfered the call. I think, that I should received two CDR's where: in first CDR source=616222820 and destination=616222821 in second CDR source= 616222820 and destination=0600340000 but

<P>hi,</P> <P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried&nbsp;editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P> <P>Thank you inadvance,</P> <P>Surajee</P> <P>&nbsp;</P><br> --------------This mail sent

Hello, I'm using digium E1 cards and serving SIP users at Asterisk. After the following call (see below) CDR shows two records. First looks as outbound call, but the second - as inbound call. Is it a bug or intended behavior? Call flow: SIP (ext: 100) -> ZAP (national number) SIP (ext: 100) transfers to SIP (ext: 200) SIP (ext: 200) -> ZAP (national number). In CDR it looks like

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hi, I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer. My extensions.conf: [globals] __TRANSFER_CONTEXT = transfer [common] exten => 123,1,Playback(demo-congrats) exten => 123,n,Hangup() exten => _0X.,1,Dial(SIP/${EXTEN}@PSTN-GW,60) exten => _0X.,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() [transfer] exten

Hi, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango

Hello, Sorry for a bit of a newbie post but we all had to start somewhere right .. I'm wondering if someone can briefly explain the difference between blind and attended transfers and why they would generate two very different CDR entries.? From my own research, it seems that transfers are both ultimately a SIP REFER and thus seeing two different CDR entries just confuses me further.

I have searched for some time but I have not found an asnwer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extensi?n the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a

When you use a product which version number is 11 or even 12, you might go with the assumption all big bugs are fixed and then you find there is a huge, important, expensive bug still running in the code we are relaying upon... The problem is simple. If you transfer a call, that dialing will be not reported in the CDR, so no billing will happen. This is a simple example: Extension 100 calls

hi! I've got cisco 7960G working with * box. Calls could be Blind Xfered through the phone but not the supervised transfer( Message on the phone: Transfer failed). Even when I put the caller on hold and resume it later, I can't hear the other side but the otherside can hear me. (It shows as the line is connected though. Yet the respective caller entry blinks.) Any suggestions most

Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/xxxxxxxxxxxx ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound

Hi I'm having 2 strange problems (or rather I am doing it wrongly!) ..need some help.I am working on trixbox 2.6 with Asterisk 1.4.22 1. I've a Queue with dynamic agents (ampextensions=deviceanduser mode in amportal.conf), some of agents when they punch in the wrong UserID - they still get logged onto the Queue as a result many of them complain that they don't receive calls on

I'm attempting to link calls together in my CDR and would like to try to do it via the userfield. Is there any way to copy the userfield between calls when doing an attended transfer? I can't seem to find anything about it searching Google. -Jon