Topic: SOLVED: Polycom ip320 & Siemens C450 - a couple of issues (EDITED)

[d6hq]

Have just set up a new SAIL install with a couple of Polycom IP320's and a single Siemens C450 thanks to Jeff at Selintra (or is it Aelintra ?) who provided the necessary pointer we have got the Polycom's working but there are now 2 issues to resolve. Can anyone provide further pointers ?

The Polycom's are setup to auto configure from the server. After a reboot the phone's do not find the correct directory file (macaddress-directory.xml) but instead revert to the default set within the provisioning section. If I add contacts via the phone interface then the file is correctly written but on each reboot it is overwritten. Do I just need to remove that section of the provisioning script ?

The minor issue with the Siemens C450 is that it seems to require some alteration to the FLASH times to get call transfers working from it. Does anyone have any experience of how to set up the Siemens so that call transfers are correctly handled i.e. press R xxxx and the call is transferred ?

[David Harper]

The Polycom's are setup to auto configure from the server. After a reboot the phone's do not find the correct directory file (macaddress-directory.xml) but instead revert to the default set within the provisioning section.

That shouldn't be happening. Can you please do as follows:

1. Log on to the console as root using Putty
2. tail -f /var/log/messages
3. Reboot the phone (pulling the power works best)
4. Read the output carefully as the phone boots

What should happen is that the Polycom gradually picks up a bunch of stuff via TFTP, including $macaddress-directory.xml. Does this occur correctly?

If I add contacts via the phone interface then the file is correctly written but on each reboot it is overwritten. Do I just need to remove that section of the provisioning script ?

No, this is something that should work as advertised. Let's see if we can resolve the issue.

[SARK devs]

Polycom -

We'll try to replicate here to understand what is happening.  WE don't have a lot of experience with Polycoms but we have a couple here that they gave us to use for SARK integration.  They work just fine but we really haven't done much with them other than prove that we could auto provision them.

C450

You don't need hookflash with the C450, it can do true SIP reinvite transfers.  Instead of pressing the "#" hookflash, press the left-hand soft navigation key.  This will give you a list of options, one of which is transfer.  Using this is much better than asterisk's nasty old hookflash mechanism.


Best

S

[SARK devs]

Polycoms

...Guilty as charged

Sam was quite tricksey when he wrote the provisioning sub-system, particularly in the area of extensibility.  Upshot is, you can provision more than one file at a time from the provisioning window.  I had forgotten this over time, because it is very rarely used.  However, we did indeed use it for the Polycoms (which you had obviously spotted and I had forgotten about).

Here is the default Polycom provisioning window...

Code: [Select]

["$mac.cfg"
<?xml version="1.0" standalone="yes"?>
<!-- $Revision: 1.14 $  $Date: 2005/07/27 18:43:30 $ -->
<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="$mac-phone.cfg, polycom-locals.cfg, phone1.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY=""/>
]

["$mac-phone.cfg"
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- $Revision: 1.73.6.2 $  $Date: 2006/07/17 21:46:42 $ -->

<phone$ext>
  <reg
      reg.1.displayName="$desc"
      reg.1.address="$ext"
      reg.1.label="$ext"
      reg.1.auth.userId="$ext"
      reg.1.auth.password="$password"/>
  <reg
</phone$ext>
]

["$mac-directory.xml"
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- $Revision: 1.2 $  $Date: 2004/12/21 18:28:05 $ -->
<directory>
        <item_list>
                <item>
                        <ln>Doe</ln>
                        <fn>John</fn>
                        <ct>1001</ct>
                        <sd>1</sd>
                        <rt>3</rt>
                        <dc/>
                        <ad>0</ad>
                        <ar>0</ar>
                        <bw>0</bw>
                        <bb>0</bb>
                </item>

        </item_list>
</directory>
] 

It provisions {$mac}.cfg, phone.{$ext} and {$mac}-directory.xml, all in one go.  Now, this is pretty smart as provisioning systems go, but it does mean that these files get rebuilt at each commit.  So...  You have a whole series of choices; you can disable local directory provisioning at the phone and just do everything centrally in the provisioning window of each extension, safe in the knowledge that the directory will get built correctly after each commit; or, you can remove the directory xml from the provisoning window (either at the individual phone level or globally by modifying the ip320/330 device entry in IP Devices).  In this way, each user will manage her own directory entries using the phone touchpad. 

The way the provisioning parser works is that it will start a new file between [ ] brackets and it expects the substitional files name (e.g. $mac-directory.xml) in double quotes, immediately after the [.  So, to delete/suppress a file simply remove the entry from [ to ].

Up to you really, but it's nice to know we can do these things without needing to change any of the SAIL code so maybe the time Sam put into designing it was well spent after all.

Kind Regards

S

 

[d6hq]

Thanks Gents - that solves the Polycom directoy issue (its good to know that we were on the right track). In this case there are only going to be 2 users so individual self managed directory files make sense.

C450 - we will need to test but I thought the left soft key [INT] was used to transfer calls to other C450 handsets on the same base station and not to other phones on the PBX. In this case we only have one C450 intended to allow the customer to walk around his warehouse while talking - the Polycom's are the main phones - but we do need to be able to transfer from the c450 to a Polycom.

Will report back tomorrow when we have had a chance to test.

Thanks again.

[d6hq]

On the Siemens C450 - the left hand soft key [INT] does not give any transfer option except to other C450's registered to the same base station. Can anyone recommend a suitable alternative roaming handset that is known to work with SARK - the key feature missing on the C450 is the ability to transfer calls to other extensions

[SARK devs]

Its me who gave you duff gen, not the phone....


Here is the UK Siemens Disti's FAQ on C450 transfer...

http://www.provu.co.uk/siemens_faq.html

See the last FAQ on transfer

Best

S




 

[d6hq]

Top man !

I have trawled the web for the last seven days looking for that.

[SARK devs]

Note the last bit about turning on transfer-on-hangup in the browser...  Most of our customers set this ON because it makes transfer a fair bit slicker at the phone.

Best

S

[d6hq]

We might be at cross purposes here

If you have multiple C450's connected to one base station then call transfers are a breeze between those handsets.

What I am talking about is transferring a call from a C450 to another type of phone (Polycom in this case)

The left soft key will put a caller on hold but that seems to be all it will do. If you dial a number while they are on hold nothing happens.

There is no option in the web configurator to "hang up" on transfer if there is only one attached handset.

[SARK devs]

We might be at cross purposes here

Nope - don't think so.  We must have sold 50 or 60 of these units.  However, most of our customers only have one or two each and they just use 'em like regular SIP extensions.  They happily transfer to Snom, Polycom, Aastra, Cisco...  whatever they're running in their shops.


Best

S

[d6hq]

I am sure I am not being thick.

If I recieve a call on the C450 and press INT the caller goes on hold and gets music on hold. Thereafter the screen shows:

BACK : OK

where the colon is actually an up and down arrow. Pressing up or down does nothing - the phone beeps at you in complaint.
If you dial a number and press the green key you cut the caller off but nothing else happens

The instructions on Nimans site that you referred to say

Answer call.
Left soft key- ext call (caller hears hold music).
Phone number- dial number, press send.
Can select between the two calls using the up and down keys.
Right soft key- press to conference all 3 users.
Left soft key- press option, select down to call transfer.

I cannot get this to work in this single handset scenario.

I have another customer with multiple handsets and they can transfer to and fro to each other but they do not have any other type of phone in play.

[SARK devs]

The instructions on Nimans site that you referred to say
...

and

I cannot get this to work in this single handset scenario.

Then I think you need to talk to the handset supplier and take their advice.  All we can do is assure you that we have lots pf these phones deployed in the field and they work pretty well.  I suspect you may have a parameter set wrongly, either in the phone or in the asterisk definitions and you'll need to track it down.  Check things like call-limit and dtmfmode at both ends.

Best

S

[d6hq]

Thanks for all your help.

There seems to be contradictory info around about this handset. As you know it supports one PSTN line and a SIP connection at the same time. The instructions that I quoted above assume that the incoming call that you want to transfer is on the PSTN line.

In this instance there is no PSTN line - there is but it isn't plugged in at this stage - the call that we wish to transfer is on the SIP connection. I know (because another customer has this set up) that when the PSTN line is used and there a multiple handsets on the base station you can transfer incoming calls to whichever handset you want.

So the question is does this handset support transfer of SIP calls.

Siemens Technical Support say (quoted)
"Thank you for your email. We are glad to be at your assistance.
Unfortunately our support does not extend to covering Asterisk functions. We advise you to search for the information on the support section of the Asterisk website. "

Voip Info.org have some information here which suggests that the R key is useless
http://www.voip-info.org/wiki/view/Siemens+Gigaset+C450IP

If anyone knows of a working installation where calls can be transferred from a C450 or C460 to another SIP phone please will they let me know. Any further pointers will be gratefully appreciated.


 

[SARK devs]

who did you buy the phone from?