similar to: chan_dahdi.conf parser question

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

Hi, I've connected Asterisk with 4 PRI to a Siemens HiPath 4000. For CALLERID(name) feature I wanna use Q.SIG as switchtype. Cause Siemens PBX orders Channels logical I need the parameter qsigchannelmapping=logical. Here is my chan_dahdi.conf trunkgroups] [channels] language=de context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes

Hello, I'm running Asterisk@home 2.5 asterisk 1.2.4 zapatel 1.2.2 libpri 1.2.2 on a Dell Poweredge 2850 (1 CPU) with a TE210P I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able to receive inbound calls on all channels and can only make outbound calls on channels 25-48. Attempting to make an outbound call on channels 1-23 results in congestion.

Hi list, Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we only have the first 6 plugged in right now). Everything works fine until we fail the primary D channel (D's are on 24,48) the secondary then picks up and outbound calls do not work, if we reboot Asterisk the D on 48 comes up and it

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

I have NFAS setup on several quad port T1 cards (Sangoma). It mostly works well with the exception that calls coming in on channels 48,72, and 96 have no audio. I tried removing these channels from zapata.conf with hopes that the channels would not come up or be used. Now I get "Ring requested on unconfigured channel". How can I busyout these these channels so that incoming

Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example:

Hi, I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here my settings: chan_dahdi.conf [trunkgroups] [channels] context=default switchtype=national facilityenable=yes rxwink=300 ; Atlas seems to use long (250ms) winks ; where the ring cadence is changed *after* the callerid spill. usecallerid=yes hidecallerid=no

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

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

I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

Have you tried to load the driver with ec disable? Last time (long time ago) when I was working on a quad card, we weren't able to get ec to work with hardware ec on. ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brian Alexander Sent: Thursday, September 27, 2007 10:59 AM To: Asterisk Users

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

Did you compile and install libpri *before* Asterisk? I had same problem (among others) b/c I didn't install in the correct order. Try the awesome asterisk_update.sh shell script. Are you trying to emulate CPE or NET? Try signalling=pri_cpe Check for whitespace behind the statement, zapata.conf seems bitchy about whitespace. hth -----Original Message----- From: Steve Totaro

I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. cat chan_dahdi.conf cat chan_dahdi.conf [trunkgroups] [channels] language=en ;internationalprefix = 00 ;nationalprefix = 0 context=from-pstn switchtype=national

Any ideas? 51] logger.c: [chan_zap.so] => (Zapata Telephony) Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing

Hello list i have an issue with my dahdi_channels.conf in span 1 when i use it like below i can do my outband calls without issue ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 17-31 context = default group = 63 but when i add in channel 1-15 like: channel => 1-15,17-31 i receive all

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

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

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX* A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity