similar to: Re: Generate Random Numbers in dialplan

So we've got a TE410P configured as E-1. The PRI is showing up as normal, I have green lights, but d channel doesnt seem to come up and i keep getting this error if i do a "pri intense debug" The carrier swears up and down that there are no issues on their end. Any thoughts? localhost*CLI> > Unnumbered frame: > SAPI: 00 C/R: 0 EA: 0 > TEI: 000 EA: 1 > M3:

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

All we have a deal on Canadian termination. Rate: $0.0039 US Dollars Billing: 1/1 Protocol: SIP or H323 Codec: G729 Terms: Prepaid Only. We have a real-time web interface where you can monitor or download your CDR's. Please e-mail me offlist if you are interested: jweisman@ibell.net Thanks, Jon -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

All, I'm using a Digium TE410P w/ Asterisk 1.2.18. Trying to connect it to our NACT STX switch via PRI, d channel is up, T1 shows normal, but I'm getting crazy errors. I rewired this thing three times, then I connected the same cable from the STX to a Cisco AS5300 (same pri settings as asterisk), and all slip events and frame sync errors went away, so the cable is good. STX --> DSX

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 all, I'm having a problem with an asterisk server being unable to call certain cellphones and answering machines. Anytime the person answers the phone call, everything works well. But when the call goes to voicemail or an answering machine, I get the error message below: ===================================== *CLI> -- Attempting call on IAX2/u92613106/15551234567 for

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

I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe E&M. I have done E&M wink but have no idea about E4 Superframe E&M and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve

Hi, The problem solved After installing new zaptel drivers, we ran the "genzaptel" command to generate /etc/zaptel.conf file,checked with "zttool" command and the card status was "Yellow alarm/Blue alarm/Recovering" and the card LED was blinking red and green. The problem was with the generated zaptel configuration., but not with the pin

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