Topic: Siemens S675 IP Dect and SAIL - no sound coming in.

[jester]

I've recently bought a Siemens S675IP DECT phone (SIP) for home use and am trying to connect it up with SAIL 2.1.14-507. It connects nicely to SAIL but when calling audio is going out, but nothing is coming in. I've tried about everything i can think of, and searching the internet did not provide me with any answers.

Anyone suggestions?!
Thanx!
jester.

[SARK devs]

Hi Jester,

We have lots of these units running at customer sites.  They are very popular.  However, there is no S675 in ther UK but it looks, from the photo I googled, like a device we call the S450 (we also sell a lot of C460 units).

Either way, we've never had any problems with them.. They just work.  So, I am guessing that you have set something up not quite correctly on the phone or in the SARK extension.

Usually, we just use the "general sip" phone type for these units.  Fill out the details in the phone browser (usually, just point everything at SARK and make sure you have the user name and password and auth user all set to the same value and off you go.

If it is still giving problems then you will need to run a packet trace to see where the voice packets are going.  do this...

yum install wireshark --enablerepo=base

Then do

tethereal -R rtp -i eth1 -f "host your.phone.ip.address"

Make a call and watch the packets fly.  It is usually obvious where the problem lies when you do a trace. 

use Ctrl+C to exit ethereal when you've fnished.

Good luck

S

[jester]

Hi Selintra,

I'm also using the general sip account and all addresses are pointing to SAIL so i went on to you packet trace suggestion... but i have a hard time interpreting the lines (see below). It looks like from the moment i pick up the line my RTP packets are lost ?!


 17.605879   10.0.0.197 -> 42.97.116.124 RTP Payload type=ITU-T G.711 PCMU, SSRC=2814289406, Seq=644, Time=2642560
 17.623355     10.0.0.1 -> 10.0.0.197   RTP Payload type=ITU-T G.711 PCMU, SSRC=1997782555, Seq=35254, Time=92320
 17.625743   10.0.0.197 -> 42.97.116.124 RTP Payload type=ITU-T G.711 PCMU, SSRC=2814289406, Seq=645, Time=2642720
 17.643355     10.0.0.1 -> 10.0.0.197   RTP Payload type=ITU-T G.711 PCMU, SSRC=1997782555, Seq=35255, Time=92480
 17.645606   10.0.0.197 -> 42.97.116.124 RTP Payload type=ITU-T G.711 PCMU, SSRC=2814289406, Seq=646, Time=2642880
 17.663355     10.0.0.1 -> 10.0.0.197   RTP Payload type=ITU-T G.711 PCMU, SSRC=1997782555, Seq=35256, Time=92640
 17.665439   10.0.0.197 -> 42.97.116.124 RTP Payload type=ITU-T G.711 PCMU, SSRC=2814289406, Seq=647, Time=2643040
 17.684972   10.0.0.197 -> 42.97.116.124 RTP Payload type=ITU-T G.711 PCMU, SSRC=2814289406, Seq=648, Time=2643200
 17.704851   10.0.0.197 -> 42.97.116.124 RTP Payload type=ITU-T G.711 PCMU, SSRC=2814289406, Seq=649, Time=2643360
 17.724854   10.0.0.197 -> 42.97.116.124 RTP Payload type=ITU-T G.711 PCMU, SSRC=2814289406, Seq=650, Time=2643520

I could post the screenshots of my S675IP configuration if this would reveal anything interesting relevant to the above, if the phone is similar to the S450/S460 misconfiguration wouldn't be hard to recognize.

I wish i bought a known to work set from you, i normally like to test/mess around with stuff but for once i don't really have the time for it and wished it just worked... BTW congrats on your new online-shop, looking good!

regards,
jester.

[SARK devs]

Hi There

The trace does look a little odd.  SARK is reliably sending RTP to the phone but it isn't clear that it is returning correctly.  I'm guessing (from the gateway IP) that this SME server is running server-only.  Make sure that you have the RTP ports on the router open and forwarded to the SARK box (10000-20000 UDP). 

Check that you have defined the phone as "EXTERNAL" in extensions and make sure that the External IP address field is correctly set in Globals.

The phone is registering OK so I think you have filled everything out OK.

Kind Regards

S

[jester]

Just for anyone running into the same problem:
This issue solved itself when i installed Asterisk 1.4 (with SAIL 2.2.1). Apparently it had something to do with Asterisk 1.2

Thanks Selintra/Jeff for this SUPER contrib and your tireless patience & support!!



-- edit: i think a was a bit to quick with my conclusion... had the fixed line still stuck in <duh!>

[SARK devs]

One other thing to look at (which I forgot -  )...

Check the localnet setting in Headers->sip.conf.

The default looks like this...

Code: [Select]

localnet=192.168.1.0/255.255.255.0

Yours will need to look like this...

Code: [Select]

localnet=10.0.0.0/255.255.255.0

Kind Regards and thankyou for your kind comments.

S

[jester]

Hi Selintra,

Well that DID solve my problem!!
...and thanks again! Next time i'm in the UK remind me to buy you a beer 

jester.