Does anyone know why Asterisk 15.3 uses its Public IP-Address in the Contact field in a '200 OK' from an INVITE, but Asterisk 14.6 uses its Private IP-Address?
Can this be manipulated in Asterisk?
Thanks All.
In Asterisk 15, ICE support is enabled by default. In 14, it's disabled by default. See the icesupport option in /etc/asterisk/rtp.conf
Related
How can we achieve forking/multi-registration of same endpoint in asterisk 11.2 cert1 or greater version ?
Multiple registrations is supported only by PJSIP stack(starting with Asterisk 12).
To support 5 simultaneous devices you need to make sure that PJSIP user have following line "max_contacts=5".
To dial all devices of extension 1000 you can use following code:
Dial(PJSIP/1000)
Customer calls into my Asterisk Box.
Customer key presses that he wants to talk to an Agent.
Asterisk dials him to Agent.
Agent phone tells Asterisk to "Ring Back".
Agent picks up the call.
Call Completes.
I need to disable the "Ring Back". I do not want it to ring at all.
When Agent is using free version of X-Lite, there is a ring.
I have a SDK license from X-Lite, and there is a code line where I can disable the phone from sending the Ring Back code.
Is there a setting in Asterisk where I can tell it to just ignore the Ring Back regardless if it's sent?
Thanks.
If you not need ringing from phone, you can setup asterisk play musiconhold instead using m(holdclass) param in dial command.
If you want silence, you always can put silence message as moh.
I was running a asterisk 11.5 with no error. After installed Opus codec with patch in my asterisk
btw i am using browser based sip softphone jssip
Now whenever I am making audio call between 2 peer, no call is getting establish and throwing below errors:
btw i am using browser based sip softphone jssip
[Sep 29 09:27:36] WARNING[17065][C-00000003]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 29 09:27:36] WARNING[17065][C-00000003]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 29 09:27:36] WARNING[17065][C-00000003]: chan_sip.c:11100 process_sdp_a_audio: Got Opus minptime=10
[Sep 29 09:27:36] WARNING[17065][C-00000003]: chan_sip.c:10437 process_sdp: Rejecting secure audio stream without encryption details: audio 17002 RTP/SAVPF 111 103 104 0 8 106 105 13 126
[1000]
secret=test123
context=local
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,wss,ws
encryption=yes
dial=SIP/8000
callerid=testuser <8000>
callcounter=yes
avpf=yes
icesupport=yes
directmedia=no
Can any one cooperate?
Use tcpdump or wireshark to look at the packets, and make sure that they're actually encrypted packets. I looks like it may be rejecting them because it's not SRTP. However, that is only an educated guess. Additionally at the asterisk command line you might want to look at rtp set debug on.
You may look to Asterisk 12 for Opus support, where it will be officially supported, or so it seems from the development activity for Asterisk 12.
Here's the reference to the issue and discussion of the fix. And you can see that it's been approved by the review board here.
I, too, am interested in using Opus. However, without knowing what patch you are using, and the stability of said patch -- it may be difficult (and rather time consuming) to give specific instruction here. I would recommend reporting the issue to the person/group who released the patch as it may help the community project (if it is such).
I have recently installed Asterisk 11.4 on my VM. I wanted to know if Asterisk 11 can act as a ICE server ? In Asterisk 11's wiki page , there is a mention of ICE but those configurations are for Asterisk to act as a end device and not as a server. There is a mention of stunserver and turnserver which the Asterisk device must use to send STUN/TURN requests.... Is it not possible for Asterisk itself to act as STUN/TURN server?
It can not be ICE and STUN and TURN, but it supports both after version 11.
see bellow:
https://wiki.asterisk.org/wiki/display/AST/Interactive+Connectivity+Establishment+(ICE)+in+Asterisk
https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support
No, asterisk not support of ice/stun server inside asterisk.
Asterisk is VOIP !!!! server.
Very bad idea put anything in it.
So it not support ice, stun,jabber etc.
Read this about ice servers availibe:
http://www.voip-info.org/wiki/view/ICE
http://www.voip-info.org/wiki/view/STUN
https://code.google.com/p/rfc5766-turn-server/
I'm a veteran of Asterisk 1.4 and am looking to build a new application on Asterisk 11 (which is currently beta, but is planned to be LTS release some time before I need it.)
I can't get Asterisk Manager Interface on 11 to send me any events. (Now, obviously, in production, I need to cut down these AMI rights drastically, but as I'm exploring I've opened the firehose, if you will.)
manager.conf looks like this:
[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1
[manager]
secret = squirrel
deny = 0.0.0.0/0.0.0.0
permit = 127.0.0.1/255.0.0.0
read = all
write = all
I then use telnet to try to get in and explore the event stream:
$ telnet localhost 5038
Trying ::1...
telnet: connect to address ::1: Connection refused
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.3
Action: Login
Username: manager
Secret: squirrel
Events: on
Response: Success
Message: Authentication accepted
Event: FullyBooted
Privilege: system,all
Status: Fully Booted
...and there it sits, not moving, no matter what I actually do with the system. I've also tried using the Event manager action with EventMask: on to try to get something out of it; the command is accepted, but nothing changes. It will happily respond to any other actions I send it, though.
Any leads? This sort of thing worked fine under 1.4, and I'm not finding anything in any documentation suggesting I'm doing something wrong. I suppose the next thing to try is 1.8...
(There is little else in /etc/asterisk; I'm using example configs only for reference. This is as minimal as we get...)
It's may be bug in Asteriks / FreePBX. I had same situation, and my API php script didn't receive any events from AMI.
For fix this bug, you must install "Conferences" module and restart Asterisk from SSH: service asterisk restart
I just tested this with the latest 11 from subversion using your configs. I see events being generated. For example, executing this from the CLI:
*CLI> channel originate Local/Foo application Bar
While invalid, will cause some events to be spit out to the manager interface.