similar to: dahdi_pri_error No more room in scheduler

Hi, I'm getting the following error from an asterisk 1.6.0.9 installation: [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: Asked to delete sched id -1??? [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: No more room in scheduler This repeats a few times, then asterisk crashes. I can't seem to locate any info on this error at all. I'm using

Starting from today i am receiving the following errors on asterisk.. What can be the reason for it? [May 7 11:45:16] ERROR[14885]: chan_dahdi.c:10515 dahdi_pri_error: ACK received for '0' outside of window of '3' to '4', restarting [May 7 11:45:16] WARNING[14885]: chan_dahdi.c:3347 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel

getting some errors on my test system. this is 1.4 (Asterisk SVN-branch-1.4-r214194) with a 4 port T412p card. Three of the ports are connected: Span 1 to the PSTN on a 10 channel bearer line, ports 2 and 3 are cross-overed (!) to each other. Port 4 is not plugged in. This has been working fine for several months. I updated a few days ago to the latest 1.4 branch. However, now I cannot dial into

HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and libpri-1.4.11.4 When dial, when 492131 answer, in console appear some error messages -- AGI Script Executing Application: (DIAL) Options: (DAHDI/g1/492131|60) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/492131 [Dec 4 11:15:59] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

I'm running 1.8.2 rc1 on a Centos box with dahdi-linux-complete 2.4, using the wct4xxp module. All operations appear normal however I noticed an error repeating occasionally on the console. [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 ROSE RETURN ERROR: [Jan 10 13:53:05] ERROR[6906]: chan_dahdi.c:13691 dahdi_pri_error: 1 INVOKE ID: 11 [Jan 10 13:53:05] ERROR[6906]:

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

Hi, I running into the following problem on my Asterisk setup: --snip-- [Sep 3 01:40:59] NOTICE[9170] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep

Hi, I noticed that Dahdi starting producing these error messages: ERROR[29250] chan_dahdi.c: No more room in scheduler ERROR[29250] chan_dahdi.c: Asked to delete sched id -1??? during which time I could not send any calls or receive calls on at least one of my Dahdi spans. The only way to clear the problem seemed to be to restart Asterisk. It appears to start after the following message

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

Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even

I'm using QSIG between Asterisk and an NEC SV8300. Whenever I make a call from the SV8300, I see: [Oct 4 21:02:49] ERROR[5729]: chan_dahdi.c:12226 dahdi_pri_error: !! < Unknown IE 50 (cs5, len = 3) I see an IE 50 in the Q.932 specification, so I don't understand why this error is occuring.

Got this in the log, with no calls active. Is it a problem with my isdn line, or * ? [Mar 15 11:36:18] ERROR[29161]: chan_dahdi.c:8735 dahdi_pri_error: ACK received for '0' outside of window of '39' to '40', restarting [Mar 15 11:36:18] == Primary D-Channel on span 1 down [Mar 15 11:36:18] WARNING[29161]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available!

[Dec 23 17:58:49] ERROR[3091]: chan_dahdi.c:8413 dahdi_pri_error: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX I am seeing the above error on DAHDI 2.1.0, asterisk 1.4.22 and libpri 1.4.7 I am using a TE120P card. I am also getting this VERY frequently: -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'DAHDI/0-1' Versus a normal hangup:

Hi, i managed to connect to Ericsson MD110 with PRI at last. And made a successful call thru asterisk to ericsson. But when i try to call from ericsson to asterisk i got an error on asterisk side. And i couldnt figure out why. Here's my extensions.conf about incoming calls. [DID_span_1] include = DID_span_1_timeinterval_all,${timeinterval_all} DID_span_1_timeinterval_all] exten =

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

Hello, I just tried to upgrade to 1.6.1.0 from 1.6.0.9 and i had problems in registering users. As i see from debug it successfully reads from users.conf but later,when a user tries to logon it say peer not found.... And there were an error msg about mysql about the username field..Smthing changed in mysql tables??? Now i downgraded to 1.6.0.9 again and everything is working..

Hello, I have a default installation of asterisk 1.6.1.9-2 When i create a user in users.conf via asterisk-gui, calls, voicemail etc works. But if i create a user realtime (and my realtime caching is available too) i can see the realtime user with sip show peers. But, my local dial rules does not work. I can call from realtime user to static users(the ones in users.conf) and if they are not

Hi, I'm experiencing some strange problems with out Digium card. First the details abount hardware and software: Digium, Inc. Wildcard B410 quad-BRI card (rev 01) Asterisk 1.6.0.20 dahdi-linux-complete-2.2.1-rc2+2.2.1-rc2 libpri-1.4.10.2.tar.gz The problem now is that there are a number of clients that I call and suddenly the connection drops. Just a few seconds later the clients calls