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?
Related
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.
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).
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.
I have a continental calling card and I'm not sure how to make it possible to dial out with my asterisk server.
It is a VOIP prepaid card. I can call out on a softphone using their server address and my username and password.
I can't figure out my sip.conf or my dial plan.
Here is what I have.
sip.conf:
[continentalcard]
host=continental.com
defaultuser=username ;; user on continental's server
secret=password
register => username:password#continental.com
context=global
[frank]
type=friend
defaultuser=frank ;; user on my local asterisk server
secret=password
host=dynamic
context=internal
extensions.conf:
[global]
CARD=SIP/continentalcard
[internal]
exten => 100,1,Dial(SIP/frank)
same => n,Hangup()
include => continentalcard
[continentalcard] ;; outgoing
exten => _1NXXNXXXXXX,1,Dial(${CARD}/${EXTEN})
I get the following message on the CLI as I try to dial out 1-222-333-4444 (not the real number):
== Using SIP RTP CoS mark 5
-- Executing [12223334444#internal:1] Dial("SIP/frank-00000151", "SIP/continentalcard:12223334444") in new stack
== Using SIP RTP CoS mark 5
[Oct 3 04:02:57] ERROR[22923]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("continentalcard", "12223334444", ...): Servname not supported for ai_socktype
[Oct 3 04:02:57] WARNING[22923]: chan_sip.c:5866 create_addr: No such host: continentalcard:12223334444
[Oct 3 04:02:57] WARNING[22923]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/frank-00000151' status is 'CHANUNAVAIL'
Update: Filled sip.conf with the global context. Also just noticed your missing a / in extensions.conf. Please look below
You have your sip.conf formatted incorrectly.
[global]
register => username:password#continental.com
context=continentalcard
[continentalcard]
host=continental.com
defaultuser=username
secret=password
context=continentalcard
Registration should be placed under the [global] context in sip.conf.
Context should be continentalcard not global. When the softphone dials 1NXXNXXXXXX it should start using the continentalcard context from extensions and perform the Dial(${CARD}/${EXTEN})
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