With changes in Kamailio.cfg Richard Good on Dec 17, 2014
Updated P-CSCF example with additional Rx_AAR params, incorporated in kamailio.cfg, I am able to REGISTER 200 ok with Rx interface.
But I want to send AAR message only after receiving 200 ok at pcscf. Currently message sequences are
CER ------->
<-------CEA
AAR --------->
<------AAA
REGISTER--->
<-------401
REGISTER---->
<----200ok
Screen shot is for reference.
But I want to Send AAR/AAA only after REGISTER 200ok.
Below is the Sequence:
CER ------->
<-------CEA
REGISTER--->
<-------401
REGISTER---->
<----200 ok
AAR --------->
<------AAA
I using PCSCF as Diameter client and Seagull as a Diameter server.
I am able to REGISTER but not able to send AAR/AAA after 200 ok.(Refer picture)
Kindly let me know any other changes required in kamailio.cfg or seagull scenario.xml or some where else.
and What changes.
Under ims_qos module, you need to edit mod.c. Here Rx_AAR_Register will be enabled by default for Register requests. If you want to enable it for Register-replies, you need to add ON_REPLY_ROUTE in that line.
By default AAR will go only for 200 OK of register requests.
Thanks, Senthil.
Related
I am having issues with mail bouncing when sending from my own server to my own active yahoo account using JavaMail. The mails are passing SPF, DKIM and DMARC according to google mail that receives the same messages being bounced by yahoo. I can send messages from other accounts to my yahoo account without issue.
The messages send fine from my server to ZMail, GMail, Microsoft mail. Looking at the emails, the only thing that I have noticed is the message header for the Message-Id. My messages have the following header:
Message-ID: <923936395.17.1634776639078#[internally visible hostname]>
I am wondering if this header could be the problem and whether there is a way in JavaMail or in the Apache James to set the hostname or IP address that gets used in this message so that rather than using the "internally visible hostname", I can get the hostname that is externally visible. I have been searching the available documentation for Apache James and JavaMail but have not found any parameters to try in order to resolve this.
According to the Decompiled SRC of sun mail it should be possible by setting some properties for your session.
props.setProperty("mail.from", user);
props.setProperty("mail.host", host);
//props.setProperty("mail.user", user);
The Id will be updated by the save method (saveChanges()) and will trigger an new ID generation (updateHeaders() -> updateMessageID()). (Looked up in the decompiled MimeMessage.class)
Leading to the HostPart called in javax.mail.internet.InternetAddress.
The relevant method is _getLocalAddress.
Here you can see that the values get extracted from the Properties or will fallback to your local machine.
Used Fields:
user.name
mail.from
mail.user
mail.host
The user.name property can also be looked up from the system props.
Here is their outging email.
---------- Forwarded message ----------
From: tomnora
To: GcxwtJgnCTWXlhaq
Cc:
Bcc:
Date: Tue, 17 Dec 2019 10:18:39 +0000
Subject: Confirm your subscription to TravlGusto
Hello!
Hurray! You've subscribed to our site.
We need you to activate your subscription to the list(s): My first list by clicking the link below:
Click here to confirm your subscription.
Thank you,
The team!
They get bounced, but why is this happening? How do I stop it?
You cannot. Sending email means saying not only what you want to send and who you want to send it to, but also who you are. And you can say that you're anybody you want:
https://en.wikipedia.org/wiki/Email_spoofing
I am build Linphone from official site for Debian:
$ ./linphone --version
linphone 4.1.1-655-g95245907
$ ./linphonec --version
version: 3.12.0
I am try voice call in linphonec to test phone, but get error:
> call 891********8
2019-03-06 17:13:20:391 liblinphone-error-LinphoneCore has video disabled for both capture and display, but video policy is to start the call with video. This is a possible mis-use of the API. In this case, video is disabled in default LinphoneCallParams
Error from linphone_core_invite.
Okay, may be enable set only voice? Yes, it is possible:
> help call
'call <sip-url or number> [options]' : initiate a call to the specified destination.
Options can be:
--audio-only : initiate the call without video.
--early-media : sends audio and video stream immediately when remote proposes early media.
Good option --audio-only. Try it:
> call 891********8 --audio-only
2019-03-06 17:14:01:951 liblinphone-error-LinphoneCore has video disabled for both capture and display, but video policy is to start the call with video. This is a possible mis-use of the API. In this case, video is disabled in default LinphoneCallParams
Error from linphone_core_invite.
This option is not work.
Q: How to disable video in default LinphoneCallParams? How to do it?
I ran into a similar problem and the solution I came up with may help.
My problem:
linphonec> call 5201
2019-04-13 02:47:38:771 liblinphone-error-LinphoneCore has video disabled for both capture and display, but video policy is to start the call with video. This is a possible mis-use of the API. In this case, video is disabled in default LinphoneCallParams
Error from linphone_core_invite. <----- THE REAL PROBLEM
The solution was to include the domain. For me this was:
linphonec> call sip:5201#172.31.0.1
2019-04-13 02:53:00:103 liblinphone-error-LinphoneCore has video disabled for both capture and display, but video policy is to start the call with video. This is a possible mis-use of the API. In this case, video is disabled in default LinphoneCallParams
Establishing call id to sip:5201#172.31.0.1, assigned id 2
Call 2 to sip:5201#172.31.0.1 in progress.
Media streams established with sip:5201#172.31.0.1 for call 2 (audio).
I just tried to get into Astersik and FreePBX because I'd like to set up a office phone network with different extensions for different people.
So I took a Raspberry Pi 3, downloaded the image here and followed this documentation. I thought for testing it should be fine.
It was so. Everything worked, no errors or something else while installation. So I opened the FreePBX site in my browser, logged in as admin, created a new user, a new Chain_SIP extension and linked both. Submitted this and applied the config (the red button in the right upper corner). I logged out, logged in with the credentials of the new user in FreePBX admin interface and the UCP interface. Everything was still fine.
Now I wanted to connect via a client to it and chose Empathy and came back to reality. I wasn't able to connect. Astersik full log tells me:
[2016-11-24 19:07:08] NOTICE[2004] chan_sip.c: Registration from
'' failed for '192.168.0.54:34061' - Wrong
password
I used that password which was shown during the process to create the extension. I tried the user (not the extension) credentials too, which didn't work and returned the same error, only 1003 is replaced by the user name.
Somewhere I found the command
asterisk -rx "sip show users"
which returns:
Username Secret Accountcode Def.Context ACL Forcerport
1003 12345678 from-internal Yes Yes
I'm sure, I missed something, but I have no ides what.
Could someone help me?
In FreePBX terminology user <> extension look at http://wiki.freepbx.org/pages/viewpage.action?pageId=5242941 for details. I believe, in your case you need to
asterisk -rx "sip show peers"
not
asterisk -rx "sip show users"
I am trying to configure an example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support.
I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout
I copied sample code into web root directory and example loaded successfully and also able to register 2 extensions.
I have tried both browser Google Chrome and Firefox with their latest versions.
For asterisk, I made some configuration like below.
Please check : http://pastebin.com/7KCvtcNf
For Outbound calls :
when i am dialling 8002 -> 8001 every time Chrome Browser asking for allow microphone. Is there any way to disable asking permission and allowing it by default ?
when i allow microphone then SIpml5 phone showing like "Not Allow".
Here is the asterisk logs : http://pastebin.com/JZeDjyay
For Incoming calls :
When call come to browser,And allow microphone then Call rejected and asterisk showing like "Got SIP response 603 "Failed to get local SDP" in asterisk CLI.
But After some google i found new link https://code.google.com/p/sipml5/wiki/Downloads for "SIPml-api.js" and after replacing that JS File Calls are comming in browser even i am able to answer that calls,Also in browser it says "In call" but in asterisk CLI it keep showing ringing and other end showing like "remote ringing" .
Here is the asterisk logs : http://pastebin.com/e8Ap3bhq
Can anyone please let me know what am i doing wrong?
If you are going to test webRTC, i successfully tested the Flashphoner Web Call Server with Asterisk 1.8.x versions using different call scenarios. Regarding sipML5, i would suggest you to please try using this tutorial by Sanjay Willy : http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html
I hope it will be helpful for you.
Regards,