similar to: CDR on channel 'IAX2/u92613106-3' already started

On Fri, 2006-10-13 at 12:52:38 -0400, Jon Weisman <jweisman@ibell.net> wrote: > Hi All, Anyone know how to generate random numbers in the > dial plan? I've tried using the RAND function but it doesnt > work. Basically I need to generate a random 5 digit number > everytime a particular extension is dialed and then save that > into

https://bugzilla.netfilter.org/show_bug.cgi?id=1328 Bug ID: 1328 Summary: Please allow ipset add and del via the /proc/net/xt_ipset mechanism Product: ipset Version: unspecified Hardware: x86_64 OS: All Status: NEW Severity: enhancement Priority: P5 Component:

Hi everyone, I am facing lots for problem with CDRs in 1.6 and above versions,its shows wrong records when I do transfer(caller side and calee side),callforward,call parking.Is the present CDRs in 1.6 is enough for Complete billing.?What I need to do to make it proper.Please help me on this. Thanks Nikhil

Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples:

Hello! Oh, wise ones, ponder with me over two of the surprises that populate the universe! I have a phone, that I sometimes cannot reach, connected via pjsip. It can call other extensions just fine, it can call out over a trunk to my cell, all is well, but getting a call? Forget it most of the time. Here is all the config relevant to that phone: [murftest12] type=aor qualify_frequency=1992

On 7/5/19 10:50 AM, Doug Lytle wrote: > On 7/4/19 6:40 PM, hw wrote: >> This has again, and for no reason, ceased to work again after >> restarting asterisk.  No matter what I try, I can't create a >> certificate asterisk >> would verify. > > Have you considered using LetsEncrypt for a valid certificate? > > Doug > > What would be the point

On Sat, 2006-10-14 at 12:00 -0700, asterisk-users-request@lists.digium.com wrote: > Steve, > > Is RAND available in the latest trunk or do I need the 1.4 > beta? > > If I do show function RAND it says its not available. > > Thanks, > Jon Jon-- Forgive me, you didn't say which version you

Dears: Really, later I faced problem in the asterisk system which is : Message is shown when the unique id which is generated with each caller reach 9000 and something: No more room in scheduler Asked to delete sched id . . after I restarted the server this message is not shown again till now (after 2 week) >>> My question: What is the reason of this error and how can I solve the

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is

Hi, Asterisk does not post CDR when dial status is CHANUNAVAIL. Can someone tell me what are the conditions under which CDR is not posted? Thanks Jim

On Wed, Oct 29, 2014 at 7:10 PM, sean darcy <seandarcy2 at gmail.com> wrote: > On 10/29/2014 08:06 PM, Matthew Jordan wrote: > >> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? >>> >>> >> codec_silk for Asterisk 12 will most

Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

Is there any code that I can cut/paste that will allow me to receive an SMS text on Asterisk? and, where can I capture the incoming text?

Hello-- In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? murf

Hi, With 1.6.1.7-rc2, doc says: select*CLI> -= Info about function 'SPRINTF' =- [Syntax] SPRINTF(<format>,<arg1>[,...<argN>]) [Synopsis] Format a variable according to a format string [Description] Parses the format string specified and returns a string matching that format. Supports most options supported by sprintf(3). Returns a shortened string if a format

Hi all, using Skype for Asterisk I have the following problem. In my dialplan I need to have a CID matching (example.skype) over a DID (test.skype) : [example] exten => test.skype/example.skype,1, NoOp(nothing) exten => test.skype/example.skype,n, Hangup() Where test.skype and example.skype are Skype business account. In this case, when I get a : CLI> show dialplan example I get: [

On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote: > Hi Steve. > > Thanks, but unfortunately, I can't be involved in that. We are > running Asterisk in a production environment and we're using > 1.2, not 1.4. I don't have the resources to work with 1.4. > Last time I filed a bug against 1.2 I got told off. >

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

Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO - PRI - ASTERISK - VODAVI(pbx). Thanks, Dave *CLI> -- Starting simple switch on 'Zap/69-1' -- Executing Wait("Zap/69-1", ".1") in new