similar to: Okay can somebody explain this...

Hi, I have a really oddball time problem. When I check the server time using 'date' it is correct. When I review the time in Freepbx (under time conditions) it is correct. When I look at the time stamp in the CDR it is correct. When I review the time displayed for a voicemail in a web browser it is correct. When I hit *98 and then my extension the CID says a time that is some 6 hours

On Sat, 2009-02-07 at 15:51 -0500, Alexander Lopez wrote: > > > -----Original Message----- > > From: Steve Murphy [mailto:murf at digium.com] > > Sent: Saturday, February 07, 2009 1:59 PM > > To: Alexander Lopez > > Subject: RE: [asterisk-dev] 1.4 and CDRs -- The Breaking Point > > > > On Fri, 2009-02-06 at 12:28 -0500, Alexander Lopez wrote: >

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.

Hi Steve, > .. I'll try to sort all this out, and then I'll attack this > problem. Hopefully, I get it all into svn before the next release of > 1.4...! Just wondering if any new CDR functionality made it into the 1.4.16.2 release? I have looked through the ChangeLog for the 1.4.15 and 1.4.16.2 releases but didn't spot anything to do with changes in CDR handling. I for one

Hi, I am getting repeated messages in my logs with the following: ********************* Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: 70975D7A-CF70-4C05-8F21-56B06195995F@10.0.0.40 Feb 23 07:56:12 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23

Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on

Hi, I cannot do the following: telnet 127.0.0.1 5038 I get connection refused and this is preventing AMP from installing. I had this working when I was using FC3 but I had to upgrade to FC4 for another application. So I am running PHP5, MYSQL 4 with FC4 and asterisk is running (I had this problem before with FC3 and it turned out asterisk was not running) I am using 1.2.0 beta1 Asterisk

I'm using all sip, no zap, or iax, so I don't know if the same problem exists when using zap, or iax. When device 1 calls device 2, and device 2 blind transfers to device 3, the following happens: The cdr record shows that 1 called 3, and ignores completly the fact that 1 called 2 and 2 called 3. In normal billing I think that the duration of the entire call should be billed to device 1

Greyman-- I'm taking this discussion to the list. Folks, what we are talking about here, is me trying to get a grasp around Greyman's (Aaron's) request for a bare-bones CDR generation that describes just total connect time for channels, stripping out all the details. Who cares about xfer, park, hold, etc.? So in the following is our discussion about what *should* be there, and in

Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way:

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

Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks

Hi, I am having an issue under a specific circumstance with Asterisk 1.4.26.1, using blind transfer. If my SIP phone dials a number (so I am the caller, happens on both Polycom phones and eyeBeam softphone), do a BLIND transfer to another nuber (internal or external) ${CDR(accountcode)} is NULL fo the rest of the dialplan. My dialplan logic depends heavily on knowing the accountcode.

correct for a transfer. In the traditional Telco World the src (or A Number) field tends to be both the callerid of the customer and an identifier that ties the CDR to the customer for billing purposes. With Asterisk and a lot of other modern day softswitches there's usually a field called accountcode or similar which can be used to tie a CDR to a customer. The src field is then only

Hi all, I''ve the following test method in my controller_test which increment the ''qty'' attribute in a has_many association, ''sa_report_products''. The ''item_count'' method returns the sum of all qty''s in ''sa_report_products'': def test_add_product_to_report sa_rpt = SaReport.find(1) assert_equal 3,

On Tue, 31 Jan 2017 10:22:35 +1300 Kosala Atapattu via samba <samba at lists.samba.org> wrote: > Hi All, > > We're implementing a fully integrated Samba setup with the Active > directory on IBM AIX. From AIX level we have established the single > sign on against Windows AD 2012R2. Currently the following user > accounts and groups exists on the AD domain. > > #

Hello! I have been doing some tests with call transfers and I have been looking at the CDRs that Asterisk generates. Scenario 1: A calls B B answers and does a blind transfer to C (using # key) C answers and talks with A Scenario 2: A calls B B answers and does an attended transfer do C (using the phone's transfer key) C answers, B hangs up, and C talks with A For scenario 1, the CDR shows

I am having a hell of a time with transfers. First the Snom issues: The transfer button on the Snom 220 does not work. I have read about setting break key off in the advanced page of the web config but the Snom 220 has no such option. At the moment I am having to use the # transfer hack which makes this phone look really stupid to have buttons on it that cannot be used. Anyone know how to

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hi Rowland, Thanx for the response. For certain configurations idmap would be suitable, in our case we cannot use idmap, as the OS users are AD users, where UIDs and GIDs are mapped through Unix Attributes from AD and Samba mix up the GID permissions with idmap from the tdb backend end and map incorrect GIDs. I do not think the problem we have is related to the IDMAP, in fact the GIDs and UIDs