Topic: Subsequent calls are engaged ?

[d6hq]

We are having some weird problems routing calls on an Asterisk box where the only real handsets are Siemens S450's. The screenshot shows the way that Asterisk is configured extensions wise. There are 4 physical handsets attached to the same base station - hence the identical IP. Each of the 4 handsets has been allocated its own extension 5001-5004. Ignore ext 5000 it is irrelevant to the problem. ext 5002 is configured as the Operator.

We have a basic IVR and the intention is that if no extension is selected the call will go to 5002. That works fine on the first call in to the PBX.  However any subsequent call hits the IVR greeting but if no extension is selected instead of going to 5002's voicemail (as the Operator is busy with the first call) we get a busy tone which is presumably being generated by the Siemens base station.

The problem is compounded as the client really wants all extensions to ring on an incoming call and only if it is unanswered should it go to the operator voicemail. With the IVT set to a timeout to a ring group containing all numbers with an outcome of *5002 doesn't seem to work - only 5002 rings. You can set the Siemens handsets to answer calls for more than one extension but that doesn't get round the second inbound call getting an engaged tone.

[SARK devs]

I believe the group ring problem is due to a limit on the Siemens Base unit.  I'm pretty sure it can only handle one SIP call at a time.  You are sending it 4 invites (one for each extension) which it was never designed to  handle.   

To understand why you are getting busy tone on the second and subsequent calls through the IVR...  It isn't clear whether you mean second concurrent call or second single call (i.e. - the first call cleared).   If you mean concurrent, the Siemens base unit limit on concurrency may be the issue.  Otherwise, you are going to have ti Tshark it too see what is going on.

I suspect maybe you are asking a little too much of these phones.  They were adapted from a simple original DECT unit and built down to a price.

S
   

[gippsweb]

Haven't seen the 450's, are they similar to the c470ip.
We have one of the 470's, they can supposedly run up to 6 handsets. But only 3 sip lines.

[SARK devs]

450 is a less functional version than 470 as I understand the range, but Siemens complicate things by using different numbering across countries

S

[d6hq]

Thanks

The S450 is designed to handle 2 calls concurrently. It has capability to connect to 6 SIP accounts and one landline and can handle either 2 SIP calls or 1 SIP and 1 landline. It was discontinued by Siemens sometime last year and has been replaced in the UK by the S475 range. It contains the ability to transfer calls between handsets without involving Asterisk.

There is now a C460 range but that only has the capability of 1 SIP / 1 landline and you can't transfer SIP calls using it.

@Selintra - yes I mean concurrent calls. The first call is handled correctly the second call gets an engaged tone which must be generated by the Siemens base unit. The S450 does contain a setting to allow call forwarding with one of the options being "on busy" so that is my next direction for testing.

[d6hq]

Have made some progress on this. Firstly the client was giving me duff info - the ring alias was working as intended. Secondly the busy on subesquent concurrent calls was indeed being generated by the Siemens base station. However when the handset call forwarding was set to *xxxx on busy the system is now working as they want it - almost. One thing I have noticed though is that *xxxx is reported as deprecated - a lot of Sail is set up to use this method of dropping to voicemail. What is the recommended alternative method and how will it be implemented in Sail ?

[SARK devs]

Deprecated where?  We changed it slightly in a recently release but it isn't deprecated.  We'd be strung up!  Everyone uses it 

Best

S

[d6hq]

[Mar 20 09:03:04] WARNING[6800] app_voicemail.c: Prefixing the mailbox with an option is deprecated ('su5002').
[Mar 20 09:03:04] WARNING[6800] app_voicemail.c: Please move all leading options to the second argument.

This is where the outcome to a ring alias is declared as *5002

[SARK devs]

ah - OK

The * prefix is ours - it tells us that we are dealing with a mailbox.  The su prefix is Digium's (from their dialplan language).  The two are not directly related, although they do both concern voicemail.

Kind Regards


S