I am trying to setup caller ID spoofing using asterisk running on Ubuntu 18.04.4 LTS. I am following this tutorial: https://www.youtube.com/watch?v=DZ0czppbamo and I am currently stuck at 29:20. The problem is when I attempt to call the test number 12120001234 given to me by GoTrunk, X-Lite throws the error "Failed to establish call - Forbidden number". My sip.conf file is as follows:
[general]
allowguest=no
context=default
bindport=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723
allow=g726
allow=speex
; replace INBOUND_SIP_USERNAME and INBOUND_SIP_PASSWORD
; with your Inbound SIP Registration credentials
register => ####:#####
[201]
type=friend
host=dynamic
context=from-internal
username=201
password=####
qualify=yes
nat=force_rport,comedia
[202]
type=friend
host=dynamic
context=from-internal
username=202
password=######
qualify=yes
nat=force_rport,comedia
[trunk]
type=peer
host=eu.st.ssl7.net ; Europe POP
;host=amn.st.ssl7.net ; North America POP
context=from-trunk
qualify=yes
defaultuser=#####
remotesecret=#####
I would also like to add that I have never setup caller ID spoofing before.
That just means your provider not allow you do spoofing. Not all providers allow that. Considering latest changes in USA law most of providers should not allow.
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?
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).
This is my sip.conf
; inbound configuration
[nexmo-sip]
fromdomain=sip.nexmo.com
type=friend
context=nexmo
insecure=port,invite
nat=no
;Add your codec list here.
; Note: Use "ulaw" for US only, "alaw" for the rest of the world.
allow=ulaw
allow=alaw
allow=G729
dtmfmode=rfc2833
[nexmo-sip-01](nexmo-sip)
host=173.193.199.24
[nexmo-sip-02](nexmo-sip)
host=174.37.245.34
[nexmo-sip-03](nexmo-sip)
host=5.10.112.121
[nexmo-sip-04](nexmo-sip)
host=5.10.112.122
[nexmo-sip-05](nexmo-sip)
host=119.81.44.6
[nexmo-sip-06](nexmo-sip)
host=119.81.44.7
;outbound configuration
[general]
register => <api-key>:<api-secret>#sip.nexmo.com
registerattempts=0
srvlookup=yes
context=nexmo-sip1
[nexmo]
username=<api-key>
host=sip.nexmo.com
defaultuser=<api-key>
fromuser=<myNumber123>
fromdomain=sip.nexmo.com
secret=<api-secret>
type=friend
context=nexmo-sip1
insecure=very
qualify=yes
nat=no
;Add your codec list here.
; Note: Use "ulaw" for US only, "alaw" for the rest of the world.
allow=ulaw
allow=alaw
allow=G729
dtmfmode=rfc2833
[<myNumber123>] ; this number is at soft phone client
type=friend
context=nexmo-sip1
host=dynamic
secret=<myNumber123>
qualify=yes
[<mynumber123456>] ; this is my mobile number
type=friend
context=nexmo-sip1
host=dynamic
secret=<secretkey>
qualify=yes
This is extensions.conf
[general]
live_dangerously=yes
[globals]
[nexmo-sip1]
exten => _X.,1,Dial(SIP/${EXTEN}#nexmo)
[default]
exten => s,1,gosub(nexmo-sip1,${EXTEN},1)
Setting 1: If above is the setting of extensions.conf, I am able to make outbound calls from my soft client, but not able to get inbound calls to that soft client
Setting 2: If I change the settings of extensions.conf as follows, I am able to get incoming calls at softclient, but not able to make outbound calls.
[general]
live_dangerously=yes
[globals]
[nexmo-sip1]
exten => _X.,1,Dial(SIP/${EXTEN},30)
[default]
exten => s,1,gosub(nexmo-sip1,${EXTEN},1)
Question 1) What should I change so that I get both outbound and inbound calls?
Question 2: When I set extensions.conf as in Setting 1, I don't hear the other side, but I hear both the side conversation when extensions.conf is set as in Setting 2. How to fix that? And this is the log I see when I don't hear
[Jul 1 22:50:38] WARNING[11299]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission tvK9cRGNN- for seqno 21 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 8383ms with no response
[Jul 1 22:50:38] WARNING[11299]: chan_sip.c:4204 retrans_pkt: Hanging up call tvK9cRGNN- - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
I understand that there are lot of wrong configurations like insecure=very etc. But right now I want to make this prototype to work successfully
To make inbound and outbound calls work, you need to have 2 separate contexts inbound and for outbound.
Try to change your configs in following way, extensions.conf:
[general]
[globals]
[nexmo-sip2]
exten => _X.,1,Dial(SIP/${EXTEN}#nexmo)
[nexmo-sip1]
exten => _X.,1,Dial(SIP/${EXTEN},30)
In sip.conf please leave all what you have, just update lines what I pasted here:
[<myNumber123>] ; this number is at soft phone client
type=friend
context=nexmo-sip2
host=dynamic
secret=<myNumber123>
qualify=yes
[<mynumber123456>] ; this is my mobile number
type=friend
context=nexmo-sip2
host=dynamic
secret=<secretkey>
qualify=yes
As you can see we need to have 2 separate contexts for calls from your SIP extensions(nexmo-sip2) and for calls from your sip provider(nexmo-sip1).
Asterisk unable to receive DTMF tone from sip client.
Im using the (d) flag in dial application to perfume one digit exit during ringing state. But unfortunately doesn't work.
Here is my sip configuration :-
[100]
type=friend
username=100
host=dynamic
nat=yes
canreinvite=no
allow=all
secret=xxxxx
context=sipphones
relaxdtmf=yes
dtmfmode=auto
rfc2833compensate=yes
[200]
type=friend
username=200
host=dynamic
nat=yes
canreinvite=no
allow=all
qualify=yes
secret=xxxxx
context=sipphones
relaxdtmf=yes
dtmfmode=auto
rfc2833compensate=yes
here is my extensions.conf:-
exten => 100,1,Set(EXITCONTEXT=exitContext)
exten => 100,n,Dial(SIP/100,30,dTt)
exten => 200,1,Set(EXITCONTEXT=exitContext)
exten => 200,n,Dial(SIP/200,30,dTt)
[exitContext]
exten =>9,1,Goto(sipphones,1,1)
Regards
-Hadi.Salem
In logger.conf add to console line
console=>dtmf,verbose,debug
After that see debug output.
You may want also change dtmfmode param in your trunk config.
http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode
If your peers are in the same network, set nat to "no", that may help.
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