i4b-L2 ... i_queue full or no fifo translator (Asterisk chan_capi)
Thomas Zimmermann
I use Asterisk with CAPI on FreeBSD 7 and Nagios only for sending and (later) receiving SMS alerts over ISDN to and from mobile users. Receiving SMS is an option to acknowledge or delegate alerts. Now I have stumbled across three issues.
1. Sharing Asterisk on the same NT with an existing PBX:
We have an ISDN line with 4 NTs (8 BRI channels) and 100 telephone numbers for DDI. The protocol is point-to-point.
Since connecting Asterisk behind the existing PBX is not possible, I share one of the NTs with the existing PBX.
I assume that the existing PBX listens to all DDI Numbers. How can I configure Asterisk to listen to one number without interfering with the other PBX?
2. I am still confused as to how I should setup capi.conf and extensions.conf correctly for an ISDN line with DDIs! All documentation I have found (readmes and Google) mainly focuses on ISDN with point-to-multipoint (msn).
Although my configuration works, I see strange behavior. After running Asterisk for about 15 minutes, the /var/log/messages log file and the console fill up about five times every second with the following message:
i4b-L2 dss1_pipe_data_req: unit=0, pipe=0, i_queue full or no fifo translator!!
I can stop this message using '/usr/local/etc/rc.d/asterisk stop'.
You can see my configuration files further down in this mail.
Do I have to set more parameters in my configuration, or should I replace my (passive) ISDN interface with an active one?
3. Sending SMS, the command above runs successfully, and I receive a message on the mobile phone. However, it displays the wrong ?Calling Line Identification (CLIP)?: It shows the main number instead of the DDI extension.
smsq --motx-channel=?CAPI/ISDN1/0622100000' 079xxx8690 'Hello World? (valid for Swisscom in Switzerland)
Versions:
- Asterisk 1.4.21.2 (build from the ports tree)
- i4b and chan-capi (svn rev. 850) from Hans Petter Selaski
- FreeBSD 7.0-RELEASE-p4 amd-64
ISDN Line:
4 basic lines / 4 NTs
100 DDI extensions (extensions are 2 digits)
Using ISDN and Channel Driver from Hans Petter Selasky. i4b and chan-capi (svn rev. 850).
#pciconf -lv
ihfc0@pci0:6:0:0: class=0x028000 card=0x2bd01397 chip=0x2bd01397 rev=0x02 hdr=0x00
vendor = 'Cologne Chip Designs GmbH'
device = 'HFC-S PCI A ISDN 2BDS0 ISDN HDLC FIFO Controller'
class = network
#isdnconfig
controller 0 = {
Layer 1:
description : HFC-2BDS0 128K PCI ISDN adapter
type : passive ISDN (Basic Rate, 2xB)
channels : 0x3
serial : 0xabcd
power_save : on
dialtone : enabled
attached : yes
PH-state : F7: Activated
Layer 2:
driver_type : DRVR_DSS1_P2P_TE
}
# cat capi.conf
[general]
nationalprefix = 0
internationalprefix = 00
language = de
rxgain = 1.0
txgain = 1.0
[ISDN1]
isdnmode = DID
incomingmsn = 1234500
defaultcid = 1234578
controller = 0
group = 1
softdtmf = on
relaxdtmf = on
accountcode=
context = capi_in
holdtype = local
echocancel = no
devices = 2
# cat extensions.conf
[default]
include = capi_in
include = capi_out
[capi_out]
exten => _X.,1,Dial(CAPI/ISDN1/${EXTEN}/bl,60)
exten => _X.,2,Hangup
[capi_in]
exten = _678,1,Dial(SIP/78)
exten = _678,2,Hangup
noc# cat sip.conf
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 10.10.10.16
srvlookup=yes
[78]
type = friend
context = capi_out
callerid = 012 123 45 78
host = dynamic
secret = timbuktu
nat = no
canreinvite = yes
dtmfmode = info
call-limit = 1
mailbox = 7878@sip
disallow = all
allow = alaw
callingpres = allowed_passed_screen
Thank you for any input.
regards,
Thomas Zimmermann
Alpnach Dorf, Switzerland
+41 41 670 39 90 Telefon/VoIP
+41 79 341 86 90 Mobile
+41 41 670 39 89 Telefax
Hans Petter Selasky
There is a tool called "isdndecode", which you can run like:
isdndecode -u 0 -i -o -x
It will dump all the contents of the D-channel. Maybe you will find some error
messages there.
If you are using P2P you should maybe add the "power_on" feature to your
config:
isdnconfig -u 0 power_on
It will disable ISDN power saving.
Does the following message only appear when you re-start asterisk ?
i4b-L2 dss1_pipe_data_req: unit=0, pipe=0, i_queue full or no fifo
translator!!
--HPS
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Thomas Zimmermann
I found out the message above appears only if asterisk is running and calls
are established with the other PBX mostly during office hours.
i4b-L2 dss1_pipe_data_req: unit=0, pipe=0, i_queue full or no fifo
translator!!
If the message appears I will check again with:
isdndecode -u 0 -i -o -x
asterisk -rvvvvvv and 'capi debug'
How can I set the commands permanently?
isdnconfig -u 0 -p DRVR_DSS1_P2P_TE
isdnconfig -u 0 power_on
Do you have some hints for capi.conf, extension.conf and smsq?
regards, Thomas
Thomas Zimmermann
In the meantime, I noticed there is no correct way how to operate two ISDN
devices (asterisk + pbx) on one ISDN line in p2p mode. Of course its
point-to-point and not point-to-multipoint! Special thanks to Mathej
Ondrusek and David Wetzel for this information.
more details:
Thread: CAPI and ISDN DDI Configuration
http://lists.digium.com/pipermail/asterisk-bsd/2008-September/003437.html
http://lists.digium.com/pipermail/asterisk-bsd/2008-September/003438.html
http://lists.digium.com/pipermail/asterisk-bsd/2008-September/003439.html
regards
Thomas