freebsd-isdn Digest, Vol 229, Issue 1
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Today's Topics:
1. i4b-L2 ... i_queue full or no fifo translator (Asterisk
chan_capi) (Thomas Zimmermann)
2. Re: i4b-L2 ... i_queue full or no fifo translator (Asterisk
chan_capi) (Hans Petter Selasky)
3. Re: i4b-L2 ... i_queue full or no fifo translator (Asterisk
chan_capi) (Thomas Zimmermann)
----------------------------------------------------------------------
Message: 1
Date: Thu, 11 Sep 2008 14:55:34 +0200
From: Thomas Zimmermann <tzimm...@mac.com>
Subject: i4b-L2 ... i_queue full or no fifo translator (Asterisk
chan_capi)
To: freebs...@freebsd.org
Message-ID: <158099542962619693611080...@me.com>
Content-Type: text/plain; charset=ISO-8859-1
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@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
------------------------------
Message: 2
Date: Thu, 11 Sep 2008 20:37:09 +0200
From: Hans Petter Selasky <hsel...@c2i.net>
Subject: Re: i4b-L2 ... i_queue full or no fifo translator (Asterisk
chan_capi)
To: freebs...@freebsd.org
Message-ID: <200809112037....@c2i.net>
Content-Type: text/plain; charset="iso-8859-1"
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@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
>
>
> _______________________________________________
> freebs...@freebsd.org mailing list
> http://lists.freebsd.org/mailman/listinfo/freebsd-isdn
> To unsubscribe, send any mail to "freebsd-isdn...@freebsd.org"
------------------------------
Message: 3
Date: Thu, 11 Sep 2008 23:39:35 +0200
From: Thomas Zimmermann <tzimm...@mac.com>
Subject: Re: i4b-L2 ... i_queue full or no fifo translator (Asterisk
chan_capi)
To: <freebs...@freebsd.org>
Message-ID: <C4EF5CB7.BD75%tzimm...@mac.com>
Content-Type: text/plain; charset="US-ASCII"
Thank you Hans Petter!
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
> Von: Hans Petter Selasky <hsel...@c2i.net>
> Datum: Thu, 11 Sep 2008 20:37:09 +0200
> An: <freebs...@freebsd.org>
> Cc: Thomas Zimmermann <tzimm...@mac.com>
> Betreff: Re: i4b-L2 ... i_queue full or no fifo translator (Asterisk
> chan_capi)
>
> 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@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
>>
>>
>> _______________________________________________
>> freebs...@freebsd.org mailing list
>> http://lists.freebsd.org/mailman/listinfo/freebsd-isdn
>> To unsubscribe, send any mail to "freebsd-isdn...@freebsd.org"
>
>
------------------------------
End of freebsd-isdn Digest, Vol 229, Issue 1
********************************************