Topic: Dect hardware ?

[fpausp]

Hi All,

I am searching a possibility to connect my smeserver and sail with a siemens gigaset dect base station (c470ip).
The server should act as a dect client and work like a dect-phone. I would like to have a dect-trunkline.
Do you know hardware who works out of the box with sme74 or sme8b5 ?

Best

[SARK devs]

All of the Gigaset DECT SIP phones work with Asterisk AFAIK.  The model numbers and ranges seem to change from country to country so its not always easy to compare the units internationally.

Gigaset DECT SIP phones currently have no provisioning capability so you have to set each extension up manually.  Other than that, they just work.

I'm not sure what you mean by a DECT trunk.  The DECT base unit can be plugged into your network (using standard ethernet) and also into an existing analogue phone line.  The phone can make calls either over SIP/Asterisk or over the analogue line.

Kind Regards

S 

[gippsweb]

I use a C470IP with my box here. It's by far the best cordless handset I've used here. We only have it connected via sip though.

The one thing you cant do with it is call transfer from analog to sip.

If I understand your question correctly you won't be able to do what you want. You'll need a card or ATA to give you the analog trunk through sail.

We use the Digium TDM400p and a Linksys SPA-3102 for this.

[fpausp]

I dont like to use the tdm400p anymore because kernelupdates are difficult to handle.

Code: [Select]

The one thing you cant do with it is call transfer from analog to sip.

Yes thats the main problem.

Maybe i should use an external box for incomming analog calls (spa-3102) ?
Do you use the spa-3102 with provisioning ?

[SARK devs]

Do you use the spa-3102 with provisioning ?

3102 provisioning has issues in SAIL.  You'll need to provison it manually until we get 'round to fixing it. 

It's an awkward thing to auto-provision because it can be either (or both) a trunk and/or an extension and both have the same MAC address.

To set it up manually just use general SIP for the trunk and extension and treat them as separate entities.  Works just fine.

ind REgards

S 

[fpausp]

Code: [Select]

To set it up manually just use general SIP for the trunk and extension and treat them as separate entities.  Works just fine.

Ok, I will buy an spa-3102 and give it a go. Are there special things I should know about for austria (pots).

[SARK devs]

There almost certainly will be different setup requirements for your national phone system, but there are lots of forums dedicated to the little spa devices so I'm sure you'll be able to find setup information.

Let us know how you get on with it.

Kind Regards

S

[fpausp]

Hi,

I got my spa-3102 and after a firmwareupdate I played a bit, the spa has the extension 5010.
How should I set it up with sail to be able to call to and from the pstn ?

Code: [Select]

Product Information
Product Name: SPA-3102 Software Version: 5.1.10(GW) Hardware Version: 1.4.5(a)

Line 1 Status
Hook State: On Registration State: Registered Last Registration At: 6/11/2010 21:31:39 Next Registration In: 25 s
Message Waiting: No Call Back Active: No Last Called Number: 5010

PSTN Line Status
Hook State: On Line Voltage: 64 (V) Loop Current: 0.0 (mA) Registration State: Registered
Last Registration At: 6/11/2010 21:31:40 Next Registration In: 25 s


[SARK devs]

If you look here

http://sarkpbx.com/twiki/bin/view/Main/DocChapter253

It's a little out of date but the SPA info hasn't changed.

You'll need to define it as a regular GeneralSIP trunk as far as SAIL is concerned. Once you've done thart and set up your spa you should be able to send and receive calls just as you would over any SIP trunk. 

Kind Regards

S

[fpausp]

Thanks, I´ll try it.

Best
fpausp

[fpausp]

Hi,

50% are working, incomming calls are working but I cant call out via spa-3102 ?

Code: [Select]

    -- Executing [02652XXXXX@internal:1] AGI("SIP/5000-00000030", "selintra|OutCluster|02652XXXXX") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
    -- AGI Script selintra completed, returning 0
    -- Executing [02652XXXXX@qrxvtmny:1] AGI("SIP/5000-00000030", "selintra|OutRoute|PSTN") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Set) Options: (__filename=1276510767-02652XXXXX-5000.wav)
    -- AGI Script Executing Application: (Set) Options: (__channame=SIP/5000-00000030)
    -- AGI Script Executing Application: (Monitor) Options: (wav|1276510767-02652XXXXX-5000|mb)
    -- AGI Script Executing Application: (Dial) Options: (SIP/00432652XXXXX@02616XXXXX||)
    -- Called 00432652XXXXX@02616XXXXX
    -- SIP/02616XXXXX-00000031 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script Executing Application: (Playback) Options: (beep)
    -- <SIP/5000-00000030> Playing 'beep' (language 'en')
    -- AGI Script Executing Application: (Playtones) Options: (congestion)
    -- AGI Script Executing Application: (Congestion) Options: ((null))
  == Spawn extension (qrxvtmny, 02652XXXXX, 1) exited non-zero on 'SIP/5000-00000030'
    -- Executing [h@qrxvtmny:1] Hangup("SIP/5000-00000030", "") in new stack
  == Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/5000-00000030'
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected

Best
fpausp

[fpausp]

I think I got it, I changed the sip port from 5060 to 5061 and now I am able to call out.

Code: [Select]

PSTN Line

SIP Settings
SIP Port: 5061 EXT SIP Port: 5061

I will do more tests, by.

Best
fpausp

[fpausp]

Hi All,

I´d like to know if it is possible to clone the config from one spa-3102 to another ?

Best
fpausp