[asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

Fri Nov 12 09:37:51 CST 2010

On Fri, Nov 12, 2010 at 9:36 AM, Sherwood McGowan
<sherwood.mcgowan at gmail.com> wrote:
> On Fri, Nov 12, 2010 at 7:52 AM, Brett Woollum <brett at woollum.com> wrote:
>> More information:  When I have "rtcachefriends = yes" in sip.conf,
>> everything seems fine. With "rtcachefriends = no" I see this behavior.
>>
>> I'd rather not cache. I'm aiming for as near real-time as possible.
>>
>> Any thoughts?
>>
>> Brett Woollum
>> Brett at Woollum.com
>>
>>
>> ----- Original Message -----
>> From: "Brett Woollum" <brett at woollum.com>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at lists.digium.com>
>> Sent: Friday, November 12, 2010 5:34:03 AM GMT -08:00 US/Canada Pacific
>> Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6
>> Realtime ODBC Tables
>>
>> Hi Brad,
>>
>> I did notice that bug in the bug tracker. That's different from the behavior
>> I am seeing. I don't get multiple values in the "Mailbox". I just upgraded
>> to 1.6.2.14 and it's still there.
>>
>> By the way, the quantity of SIP NOTIFY's generated is significant. It
>> appears to be way more that the number of peers I have (3) times a handful
>> of duplicates per peer. I've been doing a Wireshark capture, and it appears
>> as though any time there is a new message in the ODBC voicemail store for a
>> mailbox that has been subscribed to, Asterisk continually generates as many
>> of the messages as possible. At one point I noticed my CPU jump from 0% to
>> ~50% just by moving one message from an mailbox that hadn't been subscribed
>> to to a mailbox that was subscribed to by the 3 peers. It only came back to
>> ~0-1% by moving the message back to an unsubscribed user.
>>
>> When I set rtcachefriends = yes in sip.conf, I get the following for each
>> peer:
>>
>> ast01*CLI> sip show peer 412
>>
>>
>>   * Name       : 412
>>   Realtime peer: Yes, cached
>>   Secret       : <Set>
>>   MD5Secret    : <Not set>
>>   Remote Secret: <Not set>
>>   Context      : sipphones
>>   Subscr.Cont. : blf_subscriptions
>>   Language     : en
>>   AMA flags    : Unknown
>>   Transfer mode: open
>>   CallingPres  : Presentation Allowed, Not Screened
>>   Callgroup    :
>>   Pickupgroup  :
>>   Mailbox      : vm_bob at default
>>   VM Extension : asterisk
>>   LastMsgsSent : 32767/65535
>>   Call limit   : 0
>>   Dynamic      : Yes
>>   Callerid     : "" <>
>>   MaxCallBR    : 384 kbps
>>   Expire       : 69
>>   Insecure     : no
>>   Nat          : RFC3581
>>   ACL          : No
>>   T.38 support : No
>>   T.38 EC mode : Unknown
>>   T.38 MaxDtgrm: -1
>>   DirectMedia  : Yes
>>   PromiscRedir : No
>>   User=Phone   : No
>>   Video Support: No
>>   Text Support : No
>>   Ign SDP ver  : No
>>   Trust RPID   : No
>>   Send RPID    : No
>>   Subscriptions: Yes
>>   Overlap dial : Yes
>>   Forward Loop : Yes
>>   DTMFmode     : rfc2833
>>   Timer T1     : 500
>>   Timer B      : 32000
>>   ToHost       :
>>   Addr->IP     : 10.20.1.225 Port 5064
>>   Defaddr->IP  : 0.0.0.0 Port 5060
>>   Prim.Transp. : UDP
>>   Allowed.Trsp : UDP
>>   Def. Username: 412
>>   SIP Options  : (none)
>>   Codecs       : 0x1004 (ulaw|g722)
>>   Codec Order  : (g722:20,ulaw:20)
>>   Auto-Framing :  No
>>   100 on REG   : Yes
>>   Status       : Unmonitored
>>   Useragent    : Yealink SIP-T28P 2.50.0.52
>>   Reg. Contact : sip:412 at 10.20.1.225:5064
>>   Qualify Freq : 120000 ms
>>   Sess-Timers  : Accept
>>   Sess-Refresh : uas
>>   Sess-Expires : 1800 secs
>>   Min-Sess     : 90 secs
>>   Parkinglot   :
>>
>> This is Asterisk 1.6.2.14 using the ODBC store for voicemail and ODBC for
>> sip_peers.
>>
>> Brett Woollum
>> Brett at Woollum.com
>>
>>
>> ----- Original Message -----
>> From: "Bradley Watkins" <Bradley.Watkins at compuware.com>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at lists.digium.com>
>> Sent: Friday, November 12, 2010 5:14:49 AM GMT -08:00 US/Canada Pacific
>> Subject: Re: [asterisk-users] Official Documentation for Asterisk 1.6
>> Realtime ODBC Tables
>>
>>
>>
>>>-----Original Message-----
>>>From: asterisk-users-bounces at lists.digium.com
>>>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
>>>Paul Belanger
>>>Sent: Friday, November 12, 2010 7:58 AM
>>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>Subject: Re: [asterisk-users] Official Documentation for
>>>Asterisk 1.6 Realtime ODBC Tables
>>>
>>>On Fri, Nov 12, 2010 at 6:07 AM, Brett Woollum
>>><brett at woollum.com> wrote:
>>>> I'm having an issue where Asterisk continuously sends out a
>>>GAZILLION
>>>> "SIP NOTIFY" messages when a user has a voice message in
>>>their INBOX.
>>>> This issue is only present when my SIP users and peers are
>>>configured
>>>> from my ODBC backend (MySQL). A static configuration of users in
>>>> sip.conf resolves this and everything works fine.
>>>>
>>>What version of 1.6?  I _think_ this may have been a bug, that
>>>was fixed.
>>>
>>>Don't hold me to that.
>>
>> I agree with Paul, this sounds like a bugs that's been fixed.
>>
>> What does the 'Mailbox :' line look like when you do a 'sip show peers'?
>>
>> My guess is that there will be multiple entries of the same mailbox, and
>> that's why you're receiving a bunch of NOTIFY messages.
>>
>> - Brad
>>
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>
> That's the problem, you've got rtcache friends turned off. If full
> realtime is that important, modify whatever scripts you have that make
> updates to your sip accounts to run "asterisk -rx 'sip prune realtime
> peer PEERNAME' " and then "asterisk -rx 'sip show peer PEERNAME load'
> " after it makes the update to the sip table. That clears Asterisk's
> cache for the modified sip peer and then loads the information from
> the database. Technically, I believe you might be able to get away
> with not clearing the cached info, but I've always played it safe.
>
> Cheers,
> Sherwood McGowan
> A LOOOOONG Time user of all things Asterisk Realtime
>

oooh, also, I saw a blf subscription in there. Have you performed a
capture of the sip signaling between asterisk and the peer(s) in
question? If you haven't, you might want to do that, and investigate
the contents of the notify message. Just a troubleshooting thought :D

Cheers