FAQ Overview Digital Line Interface Cards

 

How can I quickly check whether the card is working correctly without configuring Asterisk thoroughly?

In the directory sirrix-pci/asterisk/examples are sample files: sirrix.conf.pnp, extensions.conf.pnp, which can simply be copied into /etc/asterisk as sirrix.conf and extensions.conf respectively (save old extension.conf previously!).
Start Asterisk with "asterisk -vc" in console mode.
At the connected telephone (2nd port on 1st card (0001)) you should be able to hear the dial tone.
With 600 you should get to the echotest.
With 0 you should get the exchange (1st port on 1st card (0000)).

In the directory sirrix-pci/asterisk/examples/route-through you can find an example for a configuration of an Asterisk server in between an existing telephone system and the external link.
See sirrix-pci/asterisk/examples/route-through/README.txt and comments in the configuration files for further details.


What are the default values in the sirrix.conf?

All default values are listed in the file sirrix-pci/asterisk/examples/sirrix.conf.sample available in the driver package.


Why are my groups in sirrix.conf disabled at the start of Asterisk?

Deactivated groups can occur if the chosen hardware ports are not existent or “disabled = yes“ is set.


Sometimes sending faxes through the Sirrix.PCI cards causes problems. Why does that happen?

These problems can be caused by the following to points:

  1. The reference clock of the public net is not used. The setting „master = yes“ must be used on TE ports.
  2. Echocancellation is activated which should be disabled after the answering:
    exten => 123,1,Answer
    exten => 123,2,SrxEchoCan(0)
    exten => 123,3,Dial(...)

How can an ISDN participant use diversion ISDN features with Asterisk (*XX-key sequence für call forwarding, etc.)

The following information is also contained in the file README.config included in the driver package:
All three types of call forwarding (CFU, CFB und CFNR) are integrated in the Sirrix channel driver. With the appropriate menu in your telephone you can program the call forwarding.
The channel driver inserts entries in the following way into the Asterisk DB:

TYPE/1234 5678

TYPE = {CFU, CFNR, CFB}
1234 = Asterisk source extension as defined in sirrix.conf for the telephone (who is rerouting)
5678 = Asterisk destination extension (where the call should be going to)

The evaluation must be done in the dialplan for example with a makro. In the same way the special extensions (*123...) can be used to manipulate the entries in the Asterisk DB.


What happens, if I put an NT parameter into a group configured in TE mode?

If a TE-/NT-only parameter is set in a NT-/TE-group a warning in Asterisk-log is displayed and the group is created with the remaining parameters.

The following parameters are only valid in NT groups:
mailbox, vmexten, aocd, aoc_unit, cfnotify, cfu, cfb, cfnr, cd, 3pty, busyonbusy, callwaiting, implicit_ct, implicit_ct_rev

The following parameters are only valid in TE groups:
providetones, master


What happens, if I accidentally set for a port in one group "mode=NT" and in another group on the same port "mode=TE"?

For a port only the first setting is programmed. If there are inconsistencies a error message is displayed. This holds for the parameters "mode" and "ptp".


If I use “echocancel” in the sirrix.conf file, the analogue fax stops working. I have tried to use the “SrxEchoCan(0)” option in the fax context that I have running in Asterisk but this doesn’t seem to be turning off the echo cancellation. What can I do?

For disabling the echo cancellation with fax, please use the following order:
[fax]
exten => s,1,Answer
exten => s,2,SrxEchoCan(0)
exten => s,3,Dial(...)
This is only relevant for software FAX application which are switched through software (connections received in Asterisk via software or rerouted via VoIP). Connections that are incoming and outgoing on an ISDN port are switched in hardware. With hardware switching, echocancellation is not needed.


Is it possible to receive incoming ISDN calls for which only the base number was called?

Yes, to get calls without extension number, set  "dialtimeout=yes" and create a rule

exten => t,1,DoSomething

Calls without extension number are activated after a timeout of a few seconds (see README.config for more details).


What do I have to consider if I want to use the Sirrix card in Austria or Switzerland?

If you set "number = +" and "numbertype = unknown" there shouldn't be any problems.

In Austria only the digits of the phone extension are transmitted. To receive calls without phone extension please set also "dialtimeout=yes" and create a rule "exten => t,1,DoSomething".


Why is 'busy indication' not working in Switzerland?

The NTs released by Swisscom have an analog adapter. These adapters usually responds on all numbers and don't stop ringing when Asterisk indicates 'busy'. Please configure analog ports so that they don't respond to incoming calls.


Can I turn off immediate dial ?

Yes, simply add a line with a s-extension and set the DigitTimeout and wait for a new extension:

exten => s,1,DigitTimeout(3)
exten => s,1,WaitExten(30)

This will handle dialed digits after a 3 second time-out Here, Asterisk will not wait longer than 30 seconds for a new extension.
In TE mode you also need to set "force_start=yes" in your sirrix.conf. In NT mode force_start is enabled by default.


I have compiled Asterisk with AGGRESSIVE SURPRESSION in the zconfig.h on to get rid of echo. It doesn't work. Is there anything else I can do?

As the Sirrix.PCI4S0 drivers are independent of Zaptel, you have to enable the echocancellation by setting e. g. "echocancel = yes" in the corresponding group in "sirrix.conf". See README.config for details.


What does ptp=yes/no in sirrix.conf mean?

The following information is also contained in README.config:
"ptp = yes" means Point-to-Point (PtP) mode on Layer 2 (TEI 0 for calls).
"ptp = no" means Point-to-MultiPoint (PtMP) mode on Layer 2 (TEI 127 + TEIs for each TE (Terminal Equipment = phone / PBX / ...)).
On PtP you may only connect one TE to a NT port, on PtMP you can connect up to 8 TEs to one NT port. If in doubt ask your local Telco what he provides for external lines. For internal lines, if you want to connect phones, you usually set "ptp = no".


When I start or reload Asterisk, my BRIs go out of service into an DEACTIVED state, but they never assign a TEI and won't accept any calls. Furthermore, dchdump shows packets like this:

1149029394.953823 (0x0000) HEX:
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



Probably, your Telco configured your access to PtP but your setting in sirrix.conf is "ptp = no" effectively configuring the port to PtMP. Try to change that setting to "ptp = yes".


How can I auto load the sirrix drivers (modprobe sirrix_base, pfic and bch) and start-up Asterisk during boot?

You can use CRON with a "@boot" crontab entry calling a script performing that.


Every time I reboot computer I'll need to "make dev". Is there any way I could make them start automatically before Asterisk has started?

Probably, you are running UDev on your system. Please take a look at
sirrix-pci/tools/udev/README.udev.