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SPA504G Subscriptions timing out

MARK CASEY
Level 1
Level 1

Hi

I have 14 SPA504G with the SPA500DS connected to a generic Asterisk system

Each phone is configured with the Unit Key Configurations below

If I restart all the phones and do a core show hints, I see each of the hints with 14 watches (after it all settles down).

Over  time some of these decrease randomly. If I do a sip show subscriptions I can see random subscriptions dropping off

I am assuming its a timeout issue so I have been looking into the sip settings

As examples

  <Reg_Max_Expires ua="na">7200</Reg_Max_Expires>
  <Sub_Max_Expires ua="na">7200</Sub_Max_Expires>

  <Subscription_Expires_1_ ua="na">3600</Subscription_Expires_1_>

   <Register_Expires_1_ ua="na">3600</Register_Expires_1_>

 

Is there documentation on what these settings are ??

Similarly, I would like to do more with the unit keys settings, I know there is the function below and fnc=xml

but apart from searching on the web, is there documentation defining what options are available for fnc= ??

Dont answer this here - I will start another stream

 

  <Unit_1_Key_1 ua="na">fnc=blf+sd+cp;sub=MOA-01@10.159.2.254;nme=Michael Osborne </Unit_1_Key_1>
  <Unit_1_Key_2 ua="na">fnc=blf+sd+cp;sub=MOA-P01@10.159.2.254;nme=Michael - Direct </Unit_1_Key_2>
  <Unit_1_Key_3 ua="na">fnc=blf+sd+cp;sub=MOA-02@10.159.2.254;nme=Wendy Gregory</Unit_1_Key_3>
  <Unit_1_Key_4 ua="na">fnc=blf+sd+cp;sub=MOA-P02@10.159.2.254;nme=Wendy - Direct</Unit_1_Key_4>
  <Unit_1_Key_5 ua="na">fnc=blf+sd+cp;sub=MOA-03@10.159.2.254;nme=Loretta Yabsley</Unit_1_Key_5>
  <Unit_1_Key_6 ua="na">fnc=blf+sd+cp;sub=MOA-P03@10.159.2.254;nme=Loretta - Direct</Unit_1_Key_6>
  <Unit_1_Key_7 ua="na">fnc=blf+sd+cp;sub=MOA-04@10.159.2.254;nme=John Valmas</Unit_1_Key_7>
  <Unit_1_Key_8 ua="na">fnc=blf+sd+cp;sub=MOA-P04@10.159.2.254;nme=John - Direct</Unit_1_Key_8>
  <Unit_1_Key_9 ua="na">fnc=blf+sd+cp;sub=MOA-05@10.159.2.254;nme=Naomi Meth</Unit_1_Key_9>
  <Unit_1_Key_10 ua="na">fnc=blf+sd+cp;sub=MOA-P05@10.159.2.254;nme=Naomi - Direct</Unit_1_Key_10>
  <Unit_1_Key_11 ua="na">fnc=blf+sd+cp;sub=MOA-06@10.159.2.254;nme=Alick Osborne</Unit_1_Key_11>
  <Unit_1_Key_12 ua="na">fnc=blf+sd+cp;sub=MOA-P06@10.159.2.254;nme=Alick - Direct</Unit_1_Key_12>
  <Unit_1_Key_13 ua="na">fnc=blf+sd+cp;sub=MOA-07@10.159.2.254;nme=Mary-Jane Turner</Unit_1_Key_13>
  <Unit_1_Key_14 ua="na">fnc=blf+sd+cp;sub=MOA-P07@10.159.2.254;nme=Mary-Jane - Direct</Unit_1_Key_14>
  <Unit_1_Key_15 ua="na">fnc=blf+sd+cp;sub=MOA-08@10.159.2.254;nme=Lucy Papastatikos</Unit_1_Key_15>
  <Unit_1_Key_16 ua="na">fnc=blf+sd+cp;sub=MOA-P08@10.159.2.254;nme=Lucy - Direct</Unit_1_Key_16>
  <Unit_1_Key_17 ua="na">fnc=blf+sd+cp;sub=MOA-09@10.159.2.254;nme=Board Room</Unit_1_Key_17>
  <Unit_1_Key_18 ua="na">fnc=blf+sd+cp;sub=MOA-10@10.159.2.254;nme=Small Board Room</Unit_1_Key_18>
  <Unit_1_Key_19 ua="na">fnc=blf+sd+cp;sub=MOA-11@10.159.2.254;nme=Phone 11</Unit_1_Key_19>
  <Unit_1_Key_20 ua="na">fnc=blf+sd+cp;sub=MOA-12@10.159.2.254;nme=Phone 12</Unit_1_Key_20>
  <Unit_1_Key_21 ua="na">fnc=blf+sd+cp;sub=MOA-13@10.159.2.254;nme=Phone 13</Unit_1_Key_21>
  <Unit_1_Key_22 ua="na">fnc=blf+sd+cp;sub=MOA-14@10.159.2.254;nme=Kitchen</Unit_1_Key_22>
  <Unit_1_Key_23 ua="na">fnc=blf+sd+cp;sub=MOA-15@10.159.2.254;nme=Reception</Unit_1_Key_23>
 

 

5 Replies 5

Dan Lukes
VIP Alumni
VIP Alumni

*_Expires - the proposed expiration time embedded into outgoing registration/subscription request.

Server will respond with either same or different number.

This number will be considered expiration time of particular registration/subscription as long as it's less or equal to *_Max_Expires. Otherwise *_Max_Expires will be considered expiration time.

 

Session should not expire, because client will renew in either time requested by server, or before it (*_Max_Expires is smaller than time requested by server).

Hi Dan

 

Thanks for the feedback

I agree that they are supposed to negotiate the timeout

However, I am seeing

[Mar 20 09:01:17] WARNING[27958]: chan_sip.c:4103 retrans_pkt: Retransmission timeout reached on transmission bcc3d3c-5998d647@10.159.2.17 for seqno 107 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response

The phone is still registered, it only seems to be the sip notify subscriptions that expire.

 

I have been through the associated url

- no nat. and qos seems to be working correctly

If it was a network issue, I would expect that the sip phone registrations would be failing

sip show peers shows an average of 9ms for the status

- I have outbound proxy defined and enabled back to the same asterisk server - should it be on these phones if its the same server?

 

I have syslog enabled

What do I need to have installed to pickup debug, or does the syslog  cover it ?

 

Any help would be appreciated.

 

Mark

 

 

 

 

 

[Mar 20 09:01:17] WARNING[27958]: chan_sip.c:4103 retrans_pkt: Retransmission timeout reached on transmission bcc3d3c-5998d647@10.159.2.17 for seqno 107 (Critical Request) ... timed out after 6400ms with no response

 

Register/Subscription is initiated by phone and confirmed by PBX. But the message in question seems to be related to something initiated by PBX but not responded by phone.  So I wish it is related neither to REGISTER nor SUBSCRIBE dialog.

At the first, you need to know what packet is it. So turn on packet debugging on Asterisk (sip set debug ip ...) or use a packed catcher like wireshark or so. You are interested to see the packet with sequential number mentioned in Asterisk's message. If you have syslog&debug (highest level possible) from the phone at the same time, it may be helpful as well. Also, a REGISTER and SUBSCRIBE packet may reveal something important to us.

Then we will continue ...

 

 

Hi Dan

Spent the whole weekend with wireshark (at both ends) and SIP debug.

Found the problem. It was the Netgear POE M4100-26 switch (newly purchased and now been replaced with a second hand Cisco 3750 POE).

It appeared to be trying to do something intelligent with SIP traffic.

When I had 14 Phones, 30 Sip Registration and about 300 Notify Subscriptions (24 per phone) and did anything more that a basic call it collapsed and failed to pass SIP traffic across it. All other traffic, including Voice traffic worked perfectly.

Go Figure.

Anyway - Its working perfectly now

 

Mark

 

 

Thanks for your help

Mark

So smart devices with insufficient documentation are rather evil than helpful ...

Glad to hear you found the solution.

 

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