TRY THIS Gigaset C610IP - handsets registration failed

grzmrc

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Hi,
I'm on piaf 2.0.6.4.5 + incrediblepbx 11 + incrediblefax11.
I bought a c610IP base and 3 handsets, but I'm not able to register more than 1 handset at a time .
Every time I try to register more than 1 handset I get "registration failed" on every tree handset.
I thought that asterisk identifies an extension by IP address. As the gigaset base is the IP address holder, not the individual handpieces, perhaps the concept of extension for the individual handset will not work.
Anybody have the C610IP under asterisk? How may I get all the handsets to work?
Thanks in advance for every clarification
 
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I looked into that a bit but didn't get one.
If it works like the other SIP DECT phones, the base would be configured with all the accounts and the handsets should just be registered to the base like any other DECT phone and then you control which handset(s) are allowed to use which accounts from the base.
That one may be a little different due to the POTS as well, but it should be pretty much the same.
 

leemason

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For more than the first handset you will probably have to use a different SIP port number from the standard. Try going up from the standard 5060 and use 5061,5062, etc for the subsequent extensions.
 
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I guess the first question should be, is it the SIP registrations failing or is it the handset to base registrations failing?
 
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For more than the first handset you will probably have to use a different SIP port number from the standard. Try going up from the standard 5060 and use 5061,5062, etc for the subsequent extensions.

I just took a look and my Yealink and Panasonic DECT stations automatically used different ports for each account registration (It doesn't say so in the interface, but when I looked at Peers in Asterisk Info I saw them all on different ports).
 

leemason

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Yes most will do that but for some reason certain handsets won't do it if you are registering multiple extensions against the same PBX.
 

grzmrc

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Thanks
Do I have to change the port in the extension account on asterisk or the listen ports for voip connections on the handset base?
I tried to set a different port for every gigaset extension on asterisk (ext 640 port 5060, ext 641 port 5061, ext 642 port 5062).
Accordingly I thought to change the "Proxy server port:" and "Registration server port:" on the gigaset base, but when I configure the second handset, I get "registration failed" on every handset.
 

rossiv

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All ports should be 5060 IIRC - that's what I used with my Snom M3s. I've been wrong before though.
 

leemason

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You have "Use Random ports" set so this should take care of any port conflicts. You should find that the port used by each extension is different. However the RTP port range doesn't look correct to me (it's too narrow a range really). But I don't think this will cause extensions not to register. Check you Asterisk config (/etc/asterisk/rtp.conf) and set the RTP port range on the Gigaset to match that.
 

grzmrc

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peers.png
Actually it seems that gigaset uses the same port for every sip account despite the above configuration.
After a while the phone status become unreachable/unknown.
 

gpuser

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OK.........here is what you need to do.

Use Firefox/Chrome browser not IE!

PIAF extension settings for 627, 642, 687 (strange way to assign extensions :))

type = friend
port = 5070

Gigaset settings:

settings>telephony>connections>1>edit

connection name or number = 627
auth name = 627
auth passwd = extension password
username = 627
display name = 627
click 'show advanced settings'
domain = piaf ip
proxy server address = piaf ip
proxy server port = 5060
registrar server = piaf ip
reg server port = 5060
reg refresh time = 600 or anything from 120 to 3600
STUN enabled = no
outbound proxy = piaf ip (not really necessary)
outbound proxy port = 5060

click 'set'

Do the same for other two extensions by selecting 2 & 3 from the connections page.

settings>telephony>number asssignment
Under 'handset' you should have all 3 handsets listed as INT1, INT2 and INT3

For INT1:

for outgoing calls select radio button against 627
for incoming calls tick all checkboxes except 642 and 687

For INT2:

for outgoing calls select radio button against 642
for incoming calls tick all checkboxes except 627 and 687

For INT3:

for outgoing calls select radio button against 687
for incoming calls tick all checkboxes except 627 and 642

Click 'set'

settings>telephony>advanced settings

Under 'Listen ports for voip connections'

Use random ports = no
SIP port = 5070
RTP port = 16000 - 16100

Click 'set'

settings>telephony>network mailbox

enter *97 in 'call number' against all 3 extensions and tick 'active' checkbox.
Click 'set'

Make sure you have no dial plans defined settings>telephony>dialing plans
Even better do a factory reset and start over.

Good luck!
 

grzmrc

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Thanks for help.

I tryed your configuration, but actually the problem is the same.

Use random ports = no

I thought those ports had to be different.
 

gpuser

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Thanks for help.

I tryed your configuration, but actually the problem is the same.

I posted a working configuration from my set up. So please reset the device and start over, it will work. Have you registered 3 separate providers (extension in your case) in the base? Do they show up as registered in the web gui?

I thought those ports had to be different.

There is no requirement like that. In fact you can have everything on 5060 (for SIP signalling), just need to adjust settings to suit.

Please post a screenshot of the extension page from piaf

Oh.........just noticed something from the screenshots you posted earlier. You have 10.0.8.111 as the pbx ip in your gaigaset base but the asterisk info screenshot has been taken from 10.0.8.117.
 

grzmrc

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Thanks for your help, but I give up, I'm not able to make work this phone. I remember that in past I had a c610 with only one handset that worked with asterisk 1.8, but on asterisk 11.7.0, perhaps, I need to do some other tweaks that I'm not able to do.

The screenshot is taken from a non production asterisk box. I use this one to makbe "experiments".
Thanks
 

matthew

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From my experience, asterisk is looking at the Contact header (which may contain a different IP than the one shown on the SIP Info report). We have multiple handsets working on C610 without issue. We didn't do anything particularly special. It just works. gpuser's instructions would appear to be on the money. I can only imagine you have something else going on. As suggested, default the unit, make a couple of new extensions, and start again with gpuser's instructions.
 

grzmrc

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Only for feedback: the issue was related to the cisco switch pord. I forgot to enable "FastPort". Now it is working great. Thanks all for you patient and your suggestion.
 

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