similar to: Dahdi interface flapping

I am trying to setup Asterisk to act like a PBX connected via a PRI gateway to a voice netowrk where Asterisk is doing outbound overlap dialing for calls that terminate via that PRI. AFter researching through the archives and online dcocs, I thought I had everyting setup right, dialplan configured for '_X!' and the 'overlapdial=yes' in the chan_dahdi.conf file, but when I try and

Hi, Starting in Asterisk 1.8.0, Asterisk supports connected line updates. This is fantastic for SIP. How can I prevent them from being sent to a PRI channel? I'm having problems when a call is answered by an internal SIP extension, then transferred (blind or attended) to another internal SIP extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform APDU and drops the

Hello everyone, I am facing a strange problem on my asterisk box (using isdn lines with pri card installed on it). Normal incoming/outgoing calls are working perfectly fine. When a user dial a wrong/out-of-service number they don't hear back any such message like "The number is wrong or user is switched off" in some cases, and it's just a silence for the user. Now while

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All

Asterisk 1.8.4-rc2 (and 1.8.3) DAHDI Version: 2.4.1.2 libpri version: 1.4.12-beta3 We are having a problem with getting the nationalprefix option of chan_dahdi.conf to work. National calls do not have a "1" added to them when nationalprefix=1. The PRI debug shows the call coming in as a National Call, but the dialplan sees the call without a 1. chan_dahdi.conf: <snip>

Hello, I am trying to diagnose an intermittent error when a call comes in over our PRI lines. The problem appears random, however I have feeling it has something to do with the call volume, as the frequency increases with more calls on the system. I am not an expert when it comes to reading the PRI Span Debug statements but here is a call that had a problem and I bolded, italicized, and

Hi all, Last night I went to replace an Asterisk 1.4 + mISDN + b410p box with Asterisk 1.6.2 and DAHDI BRI - to no avail. I had two servers so copied network setting etc from the working one, moved the card across, ran dahdi_genconf etc and it didn't work. Here's the console output with notices disabled: lucas*CLI> pri intense debug span 1 Enabled debugging on span 1 [Dec 16

I just updated libpri 1.4 on my system to the latest from that branch and my QSIG connection to an NEC SV8300 stopped working. The trace showing the problem is below: q931.c:5640 q931_connect: Call 7168 enters state 10 (Active). Hold state: Idle > DL-DATA request > Protocol Discriminator: Q.931 (8) len=21 > TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent to originator) >

Hi I have done everything richard told to do ECT . below is my trace, anyone can help ? -- DAHDI/i1/09123278669-4 answered DAHDI/i1/88050048-3 -- Native bridging DAHDI/i1/88050048-3 and DAHDI/i1/09123278669-4 PRI Span: 1 Adding facility ie contents to send in FACILITY message: PRI Span: 1 ASN.1 dump PRI Span: 1 Context Specific/C [1 0x01] <A1> Len:11 <0B> PRI Span: 1

Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

Hello, I'm experiencing some weird problems on my server: - 1) The following messages are filling up my logs: [Sep 29 08:24:59] WARNING[7077]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Sep 29 08:24:59] WARNING[7078]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 171 as D-channel anyway!

picture. I know the firmware on the Nortel is old, so I'm guessing that libpri is sending something that the Nortel does not know how to handle. Is there a way to dumb down what libpri sends? From everything I've read PRI is an evolving standard - and older devices may struggle with newer extensions/developments. (This might be very handy for users trying to talk to old pbx's.) Is

handled. So....what do I do? Thanks, MD =1=================================================== !! Invalid Protocol Profile field 0x11 -- Accepting call from '2004000' to '111' on channel 0/23, span 1 -- Executing NoOp("Zap/23-1", "Incoming call from Meridian1") in new stack -- Executing NoOp("Zap/23-1", " From number: 2004000|

I have some more investigation to do on this, but I wanted to see if anyone here had any insight into the issue I've run into. The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one of several systems that have been running without issue since 2010/2011. They have all been running CentOS 4 32-bit with Zaptel 1.4.12.1 (with patch for gen 5 card), libpri 1.2.8 and

Was'nt sure if this mail got through earlier: I have been having a weird issue with my telco's ISDN PRI occasionally resetting on a incoming call, i suspect it to possibly be a timing issue since some of the incoming call work. This problem happens very frequently. I am using asterisk-1.6.0.1 with libpri-1.4.9, the asterisk server is connected viw TDMoE to a Redfone Fonebridge into

Hi, I've set up an Asterisk as voip gatway: VOIP <-> Asterisk <-> hfc-s card <-> NTBA <-> Siemens Gigaset Dect ISDN pbx. Outgoing calls from dect handset to the world are working. Incoming calls don't even ring the handset. I'm using the dahdi driver with the zaphfc kernel module. The hfc-s card is in nt mode. The msn is set at the dect phone/base station

In article <CAHZ_z=w5DMg93gShtC93kuC+fnmraPgV46BS956U5BQXVgyhxg at mail.gmail.com>, Matt Fredrickson <creslin at digium.com> wrote: > On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield <tony at softins.co.uk> wrote: > > I have some more investigation to do on this, but I wanted to see if anyone > > here had any insight into the issue I've run into. > > >

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Hi all, So, we reverted the LibPRI version and tested it, and then tried with the latest version of everything. Still no changes. The BRI line is in PTMP. If we set the configs to PTMP in the genconf_parameters and try it, we get the following: [Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error: Unable to receive TEI from network! If we set it to PTP (which it is not) we