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marlonfca at me.com Guest
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Posted: Wed May 04, 2016 3:23 pm Post subject: [asterisk-users] UAC and UAS for timer refresher header |
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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
Quote: | On May 4, 2016, at 13:00, asterisk-users-request@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@earthshod.co.uk>
To: Mamadou NGOM <ngom@numericap.com>, "Asterisk Users Mailing List -
Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Compatibilty between agi for asterisk
13.8.0 and php5.6
Message-ID: <201605041413.01708.asterisk_list@earthshod.co.uk>
Content-Type: Text/Plain; charset="utf-8"
Quote: | 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@acsacc.com>
To: Asterisk Users Mailing List <asterisk-users@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@acsacc.com>
Content-Type: text/plain; charset=utf-8
----- On May 4, 2016, at 8:49 AM, Mamadou NGOM ngom@numericap.com wrote:
Quote: | When I call my extension the agi script don't work well. when I look at the cli,
that is what I have:
|
Quote: | <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
|
Quote: | I looked on the Internet but I saw a clear answer
|
Quote: | 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@greenfieldtech.net>
To: asterisk-users <asterisk-users@lists.digium.com>
Subject: [asterisk-users] Anyone have problems with HPE 5130 EI Switch
Series
Message-ID:
<CAD4dZ0kR2e=aSnWbZ32AmnnrrqcHi90rMMtqPpNZ2_c27ToLBQ@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@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@gmail.com>
To: <asterisk-users@lists.digium.com>
Subject: [asterisk-users] Asterisk 1.8 secure SIP session only
Message-ID: <572a2698.066d620a.6724b.1b01@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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