similar to: Missing CDR data

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial("SIP/line1-74ac",

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

Hello, I'm having a problem I can't seen to figure out. In a nut shell, I have asterisk running with 4 accounts configured. All accounts work fine for local calling to each other and voicemail. However, only 1 account can make outgoing calls. All the others fail with the following error. If anyone can shed some light on the possible problem or where to look for more info it

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

I'm working on an IVR that gives the users the option (number 5 in the main menu) to dial by extension: exten => 5,1,Set(TIMEOUT(digit)=5) ; Dial Extension exten => 5,2,Set(TIMEOUT(response)=10) exten => 5,3,Background(LCL/prompt-60) exten => 5,4,WaitExten(15) When going option 5 you can dial some extensions such as 2802, it goes to the extension (all extens start with 28 on the

Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working. Firstly the asterisk version is: Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily

I have written up an guide on how to do bulk provisioning of the Linksys phones and ATAs. http://voipspeak.net/index.php?option=com_content <http://voipspeak.net/index.php?option=com_content&task=view&id=73> &task=view&id=73 Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 -

Hello all, I'm looking for software for my asterisk logs that will compile the information into nice web-based charts and graphs. Something that works similar to webalizer for apache. I want to be able to spot trends of usage, call volume levels, disconnect/failure levels, and basically see exactly where my system has been at over the past day/week/month, etc. . I would prefer for

We have just posted a review of the Sipura SPA-3000 with a complete setup guide for using the device as a trunk to Asterisk. It is easy to setup and works as good as any Zaptel setup and includes an ATA. It really is one cool device! http://www.geekgazette.com/index.php?option=com_content <http://www.geekgazette.com/index.php?option=com_content&task=view&id=28>

The client's needs are the mother of invention. We have a client that currently uses a Cisco Call Manager and one of the features they love was the Locate-Me function (or follow-me, or find-me, whatever you want to call it) which basically rings their desk phone a few times then plays a short message, and then rings their other remote phones and cell phones. The customer wants this same

Hello, A question on channel originating (call files and AMI Originate): How can I change the CALLERID(num) var (because of the E1 provider needs), but having another n?mber (the original one) stored on the "clid" CDR field on the database? A channel agnostic solution would be the best one, without having to deal with the problem based on what type of Tech used for the outgoing

Hi everyone, i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal: http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 ). Does anyone know a better and more documented management interface for * ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am already using the unique in both servers, but both generate different id?s, but i am unable to get the original one from the gateway box to store it in the final CDR? -- Rui Mota Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday 26 June 2015 at 14:52, Tech Support wrote: > Check out the ?uniqueid? parameter in cdr.conf and cdr_adaptive_odbc.conf. > John V. >

1.8 or higher. Att, H?lvio Junior SafeId - Gest?o de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br helvio.junior at safetrend.com.br On 29/06/2015 14:43, Abdul Basit wrote: > Hi Helviom > > I am interested to evaluate your product. > > What asterisk version you build this product around? > > -- > regards, > > abdul basit | p: +92 32 1416 4196 | o:

hi,all for a long time, i cant understand the difference between ${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)} i know ${CDR(start)} mean when a call is start. and ${CDR(answer)} means when a call was pick up. but what's ${CDR(calldate)} mean? Could you help me ? Thansk a lot! -- Thanks for your supporting, have a nice day. Sucan

Dear all, I am having an issue with CDR logging. What I want to do is log jitter variable from RTPAUDIOQOS module into Master.csv at the end of each call. I am using asterisk version 1.4.26. For CDR purposes, I am using cdr_custom, and the content of my cdr_custom.conf is the following: [mappings] Master.csv =>

Hi, In my application I jump to different extensions For example: [begin] exten => s,1,Goto(starts,s,1) [start] exten => s,1,Play(welkom) ..... exten => h,1,Goto(end,s,1) [end] exten => s,1,Macro(end_call) exten => s,n, Hangup When I look at my CDR record I see three different CDR's in my record. Is there a way to use one CDR on every call, and not

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

We have just implemented cdr-custom. Works fine minus the timestamps that appear in the CDR. The system's timezone is GMT. In the configuration usegmtime=yes is set. However, all of the CDRs in the Custom CDR comes as GMT-5. Another oddity is that the standard cdr/Master.csv is using GMT. Please advise. Thanks, kw -------------- next part -------------- An HTML