similar to: Originate and ForkCDR()

Hi, I'm working with asterisk 1.4.35 and found an issue regarding codecs negotiation when T38 is enabled (t38pt_udptl=yes). In particular if the INVITE sdp contains no allowed codec the call is not rejected with "488 - Not acceptable here" but it goes through and the 200 OK SDP is as follows: v=0 o=root 27285 27285 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0

Dear members of the list; I am writing in the hope to get some help with a very peculiar problem with my new asterisk 1.6.0.1 installation. The same code runs on version 1.2 without problems, but it seems the behaviour has changed (also on 1.4.7, which I tried). Please consider the following extension: exten => 1213,1,Answer exten => 1213,n,Set(counter=X) exten =>

Hello, I'm looking for a way to restore old behaviour (before Arkadia patch #0010668) of ForkCDR application in 1.4.18 I've done some research directly in the code (cdr.c & forkcdr.c), but can't find any flag. I am just f*c*ed or do you have something to suggest ? :) Thank you for help. Mathieu

Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to

Hello, In my 1.6.1.18-powered system, I've got the following dialplan (in extensions.ael) : Dial(SIP/foo,15); if (${DIALSTATUS}="NOANSWER") Dial(SIP/bar,15); When SIP/baz dials peer SIP/foo which do not answer, I've got a single CDR entry like this: SIP/baz SIP/bar time_when_foo_started_to_ring time_when_bar_ended_talking ANSWERED How can I get two CDR entries :

Hi Guys, The ForkCDR is doing pretty well, except for the call disposition. The time stamps are great, but the disposition always reads answered, even when the call is busy or not answered at all. It should read BUSY and NO ANSWER. I can send a snippet of the CDR record with forked calls for an example if needed. I think the record is not forking properly, but is maintaining the disposition

Gang, I'm having this error pop up when I do a ForkCDR, and I'm not sure how to get around it. Here are a few log lines: Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing ForkCDR("Zap/49-1", "") in new stack Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a CDR The scenario occurs like this: I use a .call file to generate a call on

I've been Googling all morning and searching voip-info.org but not quite finding what I'm looking for. I've read that you can modify the billing/account information on a CDR via AGI but I can't find an example or a "how to". I'd like to then assign specific accounts in the CDRs. Possible? Thanks, PB -------------- next part -------------- An HTML attachment was

Hello, CDR fans! I'm looking at some issues brought forward over time: 12726/10668: someone wants me to revert the changes I made via bug 10668, last Sept; (that's they are messing him up. And I didn't do the change suggested in ForkCDR, for fear of lousing up folks depending on current behavior. Which probably sparked: 11721 :

Hi all, I'm installing Centos 5.0 (fresh install) x86_64 via NFS. I've downloaded just the DVD iso via bittorrent, but when it starts the graphical installation phase an error message appears telling me that anaconda can't find the CD iso in the remote server. Seems that's looking for Centos-5.0-x86_64-bin-1of7.iso, I'm downloading it right now to go ahead but... I think

It appears that my X100P card is only answering after two rings. Ideally, I'd like it to answer on the first ring. Here is the incoming section of my extensions.conf file: [incoming] exten => s,1,Answer exten => s,2,BackGround(demo-congrats) ; Play a congratulatory message exten => 1234,1,Goto(jgunther,1234,1) exten => 4321,1,Goto(mgunther,4321,1) exten =>

Hello, we are running a Asterisk (1.2) installation with about 80 snom phones (300,320,360). Now have the demand for a special manager - assistant setup for a few extensions. Since Shared Line Appearance is not available in 1.2 I?m wondering how to realize this... What we need is that the manager can decide whether he wants to get calls or not. If not he must have the possibility to redirect

Hi, I had Asterisk threads consuming 100% CPU at times since last week. Of course, last week an extra card was installed (we had a 1PRI, a 4PRI was added) so search concentrated on that, but to no avail. Today, I installed DDD on the machine and quickly found out that it was looping because cdr->next->next == cdr in ast_cdr_setapp(). I patched this up with some simple code in

Dear all, How may I configure my extensions.conf so that only the boss's secretary can call the boss through his extension, all others when dial his extension only makes the boss's secretary phone ring, not his. If she wants, she can transfer the incoming call to the boss dialling his extension. I've tried the following, but it doesn't work: exten =>

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR

I have created an asterisk.org blog entry: http://www.asterisk.org/node/48358 to describe what I will shortly be committing to trunk to correct the weaknesses of CDRs, that asterisk users and developers have been complaining about for quite some time. Highlights: Restructuring the code and philosophy of CDRs. Plans to eliminate the ForkCDR() application Plans to create

Hi, Using 1.6.2.13. I'd like to know how I can force Asterisk to fork a call. To simplify things, Let's say I have an out context (for outbound calls) and an in (for inbound). If person A wants to call person B, and both are on my servers, I don`t want to send the call out. I want all this to happen internally on my server. The problem is if I use some condition to send

Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent.