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})
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?
here is my Console log of asterisk server
[Feb 15 12:17:49] WARNING[3558][C-00000000]: res_rtp_asterisk.c:2141 dtlsetup: Could not set policies when setting up DTLS-SRTP on '0x7fd64400caa0
[Feb 15 12:17:49] WARNING[3558][C-00000000]: res_rtp_asterisk.c:4465 ast_d: RTP Read error: Unspecified. Hanging up.
Channel SIP/7005-00000000 left 'simple_bridge' basic-bridge <222810-4890-bedf-84d549cea2b0>
== Spawn extension (default, 7008, 1) exited non-zero on 'SIP/7005-0000
-- Channel SIP/7008-00000001 left 'simple_bridge' basic-bridge <222810-4890-bedf-84d549cea2b0>
IS Any changes needed in sip.conf ?
in sip.conf
[7005] ; This will be WebRTC client
type=peer ;
username=7005 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=Z-jj! ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=yes ; Asterisk will relay media for this peer
transport=udp,ws,wss,tcp ; Asterisk will allow this peer to register on UDP or WebSockets
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
disallow=all
disallow=all
allow=ulaw
allow=alaw
allow=speex
allow=gsm
dtlsverify=fingerprint
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
;nat=force_rport,comedia
force_avp=yes
Read error: Unspecified
Check your firewall and NAT settings.
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/${EXTEN},30)
In sip.conf please leave all what you have, just update lines what I pasted here:
[<myNumber123>] ; this number is at soft phone client
type=friend
context=nexmo-sip2
host=dynamic
secret=<myNumber123>
qualify=yes
[<mynumber123456>] ; this is my mobile number
type=friend
context=nexmo-sip2
host=dynamic
secret=<secretkey>
qualify=yes
As you can see we need to have 2 separate contexts for calls from your SIP extensions(nexmo-sip2) and for calls from your sip provider(nexmo-sip1).
I have installed Asterisk on Ubuntu
sip.conf
[10000001]
type=friend
host=dynamic
qualify=yes
secret=pw_random
context=demo
[10000002]
type=friend
host=dynamic
qualify=yes
secret=pw_random
context=demo
...
extensions.conf
[demo]
exten => _1XXXXXXX,1,Dial(SIP/${EXTEN})
exten => _1XXXXXXX,2,Set(CALLFILENAME=${EXTEN:1})
exten => _1XXXXXXX,3,Monitor(wav,${CALLFILENAME},m)
However, Asterisk runs Dial and gets stuck, the users can talk each other on call, but Asterisk doesn't record the audio
run asterisk -rvvv, I get
-- Executing [10000001#demo:1] Dial("SIP/10000002-00000045", "SIP/10000001") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/10000001
-- SIP/10000001-00000046 is ringing
-- SIP/10000001-00000046 answered SIP/10000002-00000045
-- Channel SIP/10000002-00000045 joined 'simple_bridge' basic-bridge <1b882cee-b0f0-473f-aafb-651169788159>
-- Channel SIP/10000001-00000046 joined 'simple_bridge' basic-bridge <1b882cee-b0f0-473f-aafb-651169788159>
Any idea? Thanks!!
Update:
If I modify extensions.conf to
exten => _1XXXXXXX,1,Set(CALLFILENAME=${EXTEN})
exten => _1XXXXXXX,2,Monitor(wav,${CALLFILENAME},m)
exten => _1XXXXXXX,3,Dial(SIP/${EXTEN})
log is
-- Executing [10000001#demo:1] Set("SIP/10000002-00000000", "CALLFILENAME=10000001") in new stack
[Apr 14 00:56:50] WARNING[8649][C-00000000]: pbx.c:4910 pbx_extension_helper: No application 'Monitor' for extension (demo, 10000001, 2)
== Spawn extension (demo, 10000001, 2) exited non-zero on 'SIP/10000002-00000000'
error pbx_extension_helper: No application 'Monitor' for extension is weird.
I fixed it.
It's due to pbx_extension_helper: No application 'Monitor' for extension
Application 'Monitor' is not found because asterisk doesn't load res_monitor module according to this link
So, I add load=res_monitor.so in /etc/asterisk/modules.conf
The file looks like
[modules]
autoload=yes
load=pbx_config.so
load=chan_sip.so
load=chan_iax2.so
load=res_rtp_asterisk.so
load=app_hangup.so
load=app_dial.so
load=app_stack.so
load=res_monitor.so
load=pbx_functions.so
load=codec_ulaw.so
load=codec_gsm.so
load=bridge_simple.so
Thank Shu Zhang!
Have you checked the directory /var/lib/asterisk/sounds/, /var/spool/asterisk/monitor, or you can find your ubuntu files by your file name. Since the verbose didn't show any warning I believe you are doing this right but the recorded file is hiding somewhere. You can fix the file name .
exten => _1XXXXXXX,3,Monitor(wav,fixedfilename)
And find this file name in linux.
Moreover you can take a look of this
[a link]http://www.voip-info.org/wiki/view/Asterisk+cmd+Record
this is an old but i had the same problem, the way you solved it its correct, but you are just loading ALL MODULES... instead of making an slim module load.
[modules]
autoload=yes <---- THIS
load=pbx_config.so <---- DISABLED THIS
load=chan_sip.so
load=chan_iax2.so
So in order to make it work in a slim module load configuracion you must load
[modules]
autoload=no <---- THIS to NO
...
load=func_periodic_hook.so <---- THIS IS NEEDED IN ORDER TO LOAD MONITOR
load=load=res_monitor.so.so
...
Obviouly you need codecs and formats, here is a good slim config.
https://www.voip-info.org/asterisk-slimming/