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Beginner

WIP310 Fails To Re-Register.

Customer has several WIP310 connected wirelessly to an accesspoint and then to a internal PBX Tribox V2.6.22. Using Teliax ITSP.

 
The phones work great for a while but randomly fail to re-register. The only way for it to work again is to power cycle the device.   Customer has tried three diferent access point in order to elimate the possibility of bad wireless connectivity. Has latest firmware 5.0.11
 
Please advice in any way possible thanks in advance.
233 REPLIES 233
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The WIP310 don't loose the connection since one week. Just adjusting the config files like i described in my last posting. Also the "Peer '7227-wlan-cisco1' is now Lagged" is never coming up again. I never update the firmware in the phone (5.0.11(10301355)) and for me, it's not the problem from the phone, it's a problem between trixbox or asterisk and the phone.

Please check the config on the trixbox or asterisk (on the cli: "sip show settings" then it is under "Reg. max duration") and adjust the settings in the sip config or on the phone.

cu ivo

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Thank you, I understand that you are suggesting

"Change the value in the sip.conf to 4800 (and the phone 3600)"

Got it!

Problem is, on my version of Trixbox (2.6.2.3) I cannot change the sip.conf.

The file says:

;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;

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Hi Mark

That's interesting, can you post the output from the cli command "sip show settings"?

cu ivo

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Hey Guys,

Environment at my office:

1xWIP 310

Asterisk

WRT54G

Environment at my customer:

3x WIP 310

Asterisk

4xCisco 1250 AP's managed by a Cisco 2106 Controller

I've been having some issues with the WIP 310, too. It happens pretty much the same thing as in another post.. Conversations occur normally (or don't), wireless signal is entirely lost, and the phone shows "Acquiring Network" shortly afterwards.

This issue with the "Acquiring Network" happens in sync with the "registry timeout setting". If it's three minutes, it'll "Acquiring Network" in three minutes and so on.

The longer I put this timeout to happen, the more it takes for the problem to get back. But if it'll have to re-register let's say, in two minutes and my call will take five minutes, the call will drop.

My WIP 310 can always register after the call drops.

Should I update it to the alpha version?

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Thand you.

From the Trixbox Menu I went to

Pbx -> PBX Settings -> Tools - Asterisk CLI and type Sip Show Setting. I got the information below.

Is this what I need to Change?

  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs

If so, I can look in the Trixbox forum to find out how.

Note that, on my SNOM phones, I was told to change the Sip Session Timer from the Default of 3600 to 60 -- which solved a similar problem of remote SNOM phones disconnecting frequently. This setting, from the SNOM website is described as:

"If SIP Session Timer Support is enabled, this option specifies the SIP session timer in seconds. For instance, a Re-INVITE will be sent after 50% of its value has elapsed."

Help would be appreciated!

Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           Yes
  AutoCreatePeer:         No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          asterisk
  Realm. auth:            No
  Always auth rejects:    No
  Call limit peers only:  Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              Unknown
  From: Domain:          
  Record SIP history:     Off
  Call Events:            Off
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  T38 fax pt UDPTL:       No
  RFC2833 Compensation:   No
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 0x28000c (ulaw|alaw|h263|h264)
  Codec Order:            ulaw:20,alaw:20
  T1 minimum:             100
  Relax DTMF:             No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
  Notify hold state:      Yes
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No

Default Settings:
-----------------
  Context:                from-sip-external
  Nat:                    Always
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               (Defaults to English)
  MOH Interpret:          default
  MOH Suggest:           

  Voice Mail Extension:   *97
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Hi ivo,

Great input. Thanks so much for sharing your findings.

Regards,

Patrick

----------

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Beginner

I changed my SIP T1 to 2 and Reg Max Expires to 3600.

This is with asterisk 1.4 and FreePBX 2.6

My WIP310 used to lock up and never reregister until I made these changes now I have not had any problems with it. I made the changes 3 days ago.

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Hi cspiess24,

Thanks for sharing.

Can you confirm please:

1. what your *CLI> sip show conf reports

2. that you made the the  SIP T1: 2 and the Reg Max Expires: 3600 changes on the the WIP310, not on your Asterisk server

Regards,

Patrick

----------

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I made the changes to the WIP310 and not the asterisk server.

freepbx*CLI> sip show settings

freepbx*CLI>

Global Settings:

----------------

  SIP Port:               5060

  Bindaddress:            0.0.0.0

  Videosupport:           No

  AutoCreatePeer:         No

  Allow unknown access:   Yes

  Allow subscriptions:    Yes

  Allow overlap dialing:  Yes

  Promsic. redir:         No

  SIP domain support:     No

  Call to non-local dom.: Yes

  URI user is phone no:   No

  Our auth realm          asterisk

  Realm. auth:            No

  Always auth rejects:    Yes

  Call limit peers only:  Yes

  Direct RTP setup:       No

  User Agent:             Asterisk PBX

  MWI checking interval:  10 secs

  Reg. context:           (not set)

  Caller ID:              Unknown

  From: Domain:

  Record SIP history:     Off

  Call Events:            Off

  IP ToS SIP:             CS3

  IP ToS RTP audio:       EF

  IP ToS RTP video:       AF41

  T38 fax pt UDPTL:       No

  RFC2833 Compensation:   No

  SIP realtime:           Disabled

Global Signalling Settings:

---------------------------

  Codecs:                 0xe (gsm|ulaw|alaw)

  Codec Order:            ulaw:20,alaw:20,gsm:20

  T1 minimum:             100

  No premature media:     No

  Relax DTMF:             No

  Compact SIP headers:    No

  RTP Keepalive:          0 (Disabled)

  RTP Timeout:            0 (Disabled)

  RTP Hold Timeout:       0 (Disabled)

  MWI NOTIFY mime type:   application/simple-message-summary

  DNS SRV lookup:         Yes

  Pedantic SIP support:   No

  Reg. min duration       60 secs

  Reg. max duration:      3600 secs

  Reg. default duration:  120 secs

  Outbound reg. timeout:  20 secs

  Outbound reg. attempts: 0

  Notify ringing state:   Yes

  Notify hold state:      Yes

  SIP Transfer mode:      open

  Max Call Bitrate:       384 kbps

  Auto-Framing:           No

Default Settings:

-----------------

  Context:                from-sip-external

  Nat:                    RFC3581

  DTMF:                   rfc2833

  Qualify:                0

  Use ClientCode:         No

  Progress inband:        Never

  Language:               (Defaults to English)

  MOH Interpret:          default

  MOH Suggest:

  Voice Mail Extension:   *97

----

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cspless24,

Which firmware version are you running? The alpha one or 5.0.11?

Thanks!

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I am running the 5.0.11 firmware.

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Patrick,

any news on the new FW 5.0.13? thanks

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Beginner

It seems that I am not alone... I have installed several WIP310 for SPA9000 on customer sites and faced the same problem - WIP310 was running unstable; sometimes shown as "Acquring Network", sometimes seems running normal but no response actually (i.e. no incoming/outgoing call could be made).

I tried to assign fixed ip to WIP310 but no improvement, it seems to be lost network connection after 20-30 seconds in idle state; even ping test was no response!!

I have upgrade firmware to 5.0.11 and change wifi router but the problem still could not be fixed :(

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Hi,

In response to Nelson-Wong's post - we were experience very similar issues with the handsets showing "aquiring network" or locking up, which we have seem to have got rid of by upgrading the firmware to the beta release posted earlier in this thread. It puts the firmware to version 5.0.12.

To everyone else - However we are still getting issues with the handsets dropping calls. We have traced this issue to the fact that when we roam with the handset between WiFi cells the call gets dropped as the WIP310 re-associates to the nearer Access Point. First there is a drop in the Voice stream (RTP) and then the call gets dropped. We have carried out the same test and walked the same route with a Cisco 7921G WiFi handset and it 'hands-off' seamlessly between WiFi cells. We have even walked the entire building (which has 9 access points) with the 7921, and the 'hand-offs' were seamless, with the call staying active.

We are now contemplating replacing all the WIP310s at the customers premise at a big cost to our company :(

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Ok, se we now have 6 months or so of debate and 140 odd emails all confirming that the device in question is defective. Can someone from CISCO advise when we are all going to get a replacement device that works so I supply my postal address for shipment please? I hope CISCO is not waiting for the warranty period to expire.

Thank you

Dimitrios:(