[asterisk-users] UAC and UAS for timer refresher header

Wed May 4 15:22:49 CDT 2016

Hi all,

I have a intriguing issue that the RFC is not really clear about. 

Sometimes call hang-up on 45min mark because no-one refresh the call ( far-end hangup)

On both good and bad calls:

1)We initiate an invite
2)200OK is answered as refresher=UAS
3)Send ACK
4) 'follow-up' re-invite is received changing refresher to UAC
 5) my sip accept with 200OK
6) receive ACK

after 15 minutes on a good call:
1) we send reinvite continuing the CSeq of initial invite refresher=uac
2) receive 200OK refresher=uac
 and continues to do so every 15 min

After 15 min for bad call:
1) we send reinvite continuing the CSeq of initial invite refresher=uac
2) receive 200OK refresher=uac
3) receive a reinvite changing refresher=UAS
4) we 200ok refresher=uas

Call hangup 30 min later (timer is set to 1800s)

The snip of RFC 3261 states that UAS AND UAC changes on each sip transaction. But what about the timer refresher? Does it use the UAS and UAC based on who initiated the dialog???

Asterisk seems to think/use whatever was set when dialog was opened, otherwise it wouldn't send the first refresh and it wouldnt be kept from sending.

Whats confusing is that if it uses per transaction then why would it send the first reinvite if it was changed on the begging?

User Agent Client (UAC): A user agent client is a logical entity
         that creates a new request, and then uses the client
         transaction state machinery to send it.  The role of UAC lasts
         only for the duration of that transaction.  In other words, if
         a piece of software initiates a request, it acts as a UAC for
         the duration of that transaction.  If it receives a request
         later, it assumes the role of a user agent server for the
         processing of that transaction.

      UAC Core: The set of processing functions required of a UAC that
         reside above the transaction and transport layers.

      User Agent Server (UAS): A user agent server is a logical entity
         that generates a response to a SIP request.  The response
         accepts, rejects, or redirects the request.  This role lasts
         only for the duration of that transaction.  In other words, if
         a piece of software responds to a request, it acts as a UAS for
         the duration of that transaction.  If it generates a request
         later, it assumes the role of a user agent client for the
         processing of that transaction.

      UAS Core: The set of processing functions required at a UAS that
         resides above the transaction and transport layers.

      User Agent (UA): A logical entity that can act as both a user
         agent client and user agent server.

Right now they are set  the following on sip.conf 
I've tried setting to UAS and same result.
I've tried to refuse timer, far-end hang up after 30min

Session Timers:         Accept
 Session Refresher:      uac
 Session Expires:        1800 secs
 Session Min-SE:         90 secs

Marlon Araujo

> On May 4, 2016, at 13:00, asterisk-users-request at lists.digium.com wrote:
> 
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> Today's Topics:
> 
>   1. Re: Compatibilty between agi for asterisk 13.8.0    and php5.6
>      (A J Stiles)
>   2. Re: Compatibilty between agi for asterisk 13.8.0 and    php5.6
>      (Michael L. Young)
>   3. Anyone have problems with HPE 5130 EI Switch Series (Eric Klein)
>   4. Asterisk 1.8 secure SIP session only (Motty Cruz)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Wed, 4 May 2016 14:13:01 +0100
> From: A J Stiles <asterisk_list at earthshod.co.uk>
> To: Mamadou NGOM <ngom at numericap.com>, "Asterisk Users Mailing List -
>    Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Compatibilty between agi for asterisk
>    13.8.0    and php5.6
> Message-ID: <201605041413.01708.asterisk_list at earthshod.co.uk>
> Content-Type: Text/Plain;  charset="utf-8"
> 
>> On Wednesday 04 May 2016, Mamadou NGOM wrote:
>> Hello everybody,
>> When I call my extension  the  agi script don't work well. when I look at
>> the cli, that is what I have:
>> [stuff deleted] 
>> <SIP/myprovider-00000007>AGI Tx >> agi_arg_1: 56
>> <SIP/myprovider-00000007>AGI Tx >>
>> <SIP/myprovider-00000007>AGI Rx << SET VARIABLE ****** 2
>> <SIP/myprovider-00000007>AGI Tx >> 510 Invalid or unknown command
>> -- <SIP/myprovider-00000007>AGI Script *******.php completed, returning 0
>> I looked on the Internet but I saw a clear answer
>> it is sure that it is for the compatibility between php5.6 and agi. if
>> somebody can help me.
> 
> It looks as though something might be going wrong in the AGI script itself.  
> Did you use a "proper" AGI library, or a quick-and-dirty homebrew solution?  
> (There is little virtue in walking all the way to the tool shed to fetch a 
> chisel, if you know the screwdriver you already have in your drawer can be 
> used for the job.  On the other hand, breaking your screwdriver by 
> inappropriately using it as a chisel does not look too good either.  Knowing 
> the difference is one of the qualities of a great programmer.)
> 
> SET VARIABLE is a legitimate enough command, but ****** probably is not a 
> valid variable name.  Maybe there was a forbidden character in there before 
> you redacted it?
> 
> 
> I would try to isolate the problem, by writing an AGI script that *just* sets 
> a variable to some fixed value and exits; and having a corresponding extension 
> that *just* calls the AGI script, displays the variable's value in the CLI 
> with a NoOp() statement and then hangs up.  When you know you can do that, and 
> successfully read back the value from within your dialplan, *then* you can 
> make it decide what value to put in that variable.
> 
> -- 
> AJS
> 
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
> 
> 
> 
> ------------------------------
> 
> Message: 2
> Date: Wed, 4 May 2016 06:20:54 -0700 (PDT)
> From: "Michael L. Young" <myoung at acsacc.com>
> To: Asterisk Users Mailing List <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Compatibilty between agi for asterisk
>    13.8.0 and    php5.6
> Message-ID:
>    <2053955574.97281.1462368054977.JavaMail.zimbra at acsacc.com>
> Content-Type: text/plain; charset=utf-8
> 
> ----- On May 4, 2016, at 8:49 AM, Mamadou NGOM ngom at numericap.com wrote:
> 
>> Hello everybody,
> 
>> When I call my extension the agi script don't work well. when I look at the cli,
>> that is what I have:
> 
>> <SIP/myprovider-00000007>AGI Tx >> agi_request: ******.php
>> <SIP/myprovider-00000007>AGI Tx >> agi_channel: SIP/myprovider-00000007
>> <SIP/myprovider-00000007>AGI Tx >> agi_language: fr
>> <SIP/myprovider-00000007>AGI Tx >> agi_type: SIP
>> <SIP/myprovider-00000007>AGI Tx >> agi_uniqueid: ***************
>> <SIP/myprovider-00000007>AGI Tx >> agi_version: 13.8.0
>> <SIP/myprovider-00000007>AGI Tx >> agi_callerid:*********
>> <SIP/myprovider-00000007>AGI Tx >> agi_calleridname: unknown
>> <SIP/myprovider-00000007>AGI Tx >> agi_callingpres: 0
>> <SIP/myprovider-00000007>AGI Tx >> agi_callingani2: 0
>> <SIP/myprovider-00000007>AGI Tx >> agi_callington: 0
>> <SIP/myprovider-00000007>AGI Tx >> agi_callingtns: 0
>> <SIP/myprovider-00000007>AGI Tx >> agi_dnid: ********
>> <SIP/myprovider-00000007>AGI Tx >> agi_rdnis: unknown
>> <SIP/myprovider-00000007>AGI Tx >> agi_context: default
>> <SIP/myprovider-00000007>AGI Tx >> agi_extension: ********
>> <SIP/myprovider-00000007>AGI Tx >> agi_priority: 13
>> <SIP/myprovider-00000007>AGI Tx >> agi_enhanced: 0.0
>> <SIP/myprovider-00000007>AGI Tx >> agi_accountcode:
>> <SIP/myprovider-00000007>AGI Tx >> agi_threadid: *********************
>> <SIP/myprovider-00000007>AGI Tx >> agi_arg_1: 56
>> <SIP/myprovider-00000007>AGI Tx >>
>> <SIP/myprovider-00000007>AGI Rx << SET VARIABLE ****** 2
>> <SIP/myprovider-00000007>AGI Tx >> 510 Invalid or unknown command
>> -- <SIP/myprovider-00000007>AGI Script *******.php completed, returning 0
> 
>> I looked on the Internet but I saw a clear answer
> 
>> it is sure that it is for the compatibility between php5.6 and agi. if somebody
>> can help me.
> 
> Make sure there are no windows or dos line endings in that php script.  Try running it through dos2unix and see if that solves your issue.
> 
> Regards,
> Michael
> 
> (elguero)
> 
> 
> 
> ------------------------------
> 
> Message: 3
> Date: Wed, 4 May 2016 16:23:29 +0300
> From: Eric Klein <eric.klein at greenfieldtech.net>
> To: asterisk-users <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Anyone have problems with HPE 5130 EI Switch
>    Series
> Message-ID:
>    <CAD4dZ0kR2e=aSnWbZ32AmnnrrqcHi90rMMtqPpNZ2_c27ToLBQ at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
> 
> Have a strange issue at a customer, they went and replaced all of their old
> PoE switches with brand new HPE 5130 EI Switch Series.
> 
> Their PBX has been up and stable for several years with no recent changes,
> but since they change the switches they are having a problem with some
> their Yealink t-26 phones.
> 
> Seems that only the Yealinks are repeatedly rebooting  whenever the user
> clicks the speaker button, otherwise the phones work properly.
> 
> Their Cisco and Polycom phones work fine, and the Yealinks were fine prior
> to the upgrade.
> 
> Anyone see anything like this before?
> 
> -- 
> Eric Klein
> Sr. Consultant
> GreenfieldTech
> Mobile +972-54-666-0933
> Email Eric.Klein at greenfieldtech.net
> Skype: EricLKlein
> Web: http://www.greenfieldtech.net/
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> ------------------------------
> 
> Message: 4
> Date: Wed, 4 May 2016 09:43:02 -0700
> From: "Motty Cruz" <motty.cruz at gmail.com>
> To: <asterisk-users at lists.digium.com>
> Subject: [asterisk-users] Asterisk 1.8 secure SIP session only
> Message-ID: <572a2698.066d620a.6724b.1b01 at mx.google.com>
> Content-Type: text/plain; charset="us-ascii"
> 
> Hello, I am trying to secure SIP session with TLS on Asterisk Server 1.8. I
> keep getter an error, 
> 
>  == Problem setting up ssl connection: error:14094418:SSL
> routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
> [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection:
> FILE * open failed!
> 
> I tried both signed and self-signed cert to no avail. 
> 
> Here is my Configuration: 
> 
> Sip.conf
> 
> tlsenable=yes
> 
> tlsbindaddr=0.0.0.0
> 
> tlscertfile=/etc/asterisk/keys/box1.pem
> 
> tlscapath=/etc/asterisk/keys
> 
> tlscipher=ALL
> 
> tlsclientmethod=tlsv1
> 
> 
> 
> sip.conf ext.
> 
> [5006]
> 
> type=peer
> 
> context=sipext
> 
> call-limit=3
> 
> trustrpid=no
> 
> callerid="Rec" <5006>
> 
> disallow=all
> 
> allow=ulaw
> 
> allow=alaw
> 
> username=5006
> 
> secret=9fcbb025200881850526bc57d59885c3
> 
> dtmfmode=rfc2833
> 
> host=dynamic
> 
> mailbox=5006
> 
> nat=yes
> 
> canreinvite=no
> 
> transport=tls
> 
> 
> 
>  == Problem setting up ssl connection: error:14094418:SSL
> routines:SSL3_READ_BYTES:tlsv1 alert unknown ca
> [2016-05-04 09:31:17] WARNING[30032]: tcptls.c:254 handle_tcptls_connection:
> FILE * open failed!
> 
> Any ideas? 
> 
> 
> 
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