i4b-L2 ... i_queue full or no fifo translator (Asterisk
chan_capi)
Hans Petter Selasky
hselasky at c2i.net
Thu Sep 11 19:35:22 UTC 2008
Hi,
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
On Thursday 11 September 2008, Thomas Zimmermann wrote:
> Hi all,
>
> 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 at 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 at 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
>
>
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