Fran ois Marier: Setting up a JMP SIP account on Asterisk
JMP offers VoIP calling via XMPP, but it's also
possibly to use the VoIP using SIP.
The underlying VoIP calling functionality in JMP is provided by
Bandwidth, but their old Asterisk
instructions
didn't quite work for me. Here's how I set it up in my Asterisk server.
Get your SIP credentials
After signing up for JMP and setting it up in your favourite XMPP client,
send the following message to the
Get your SIP credentials
After signing up for JMP and setting it up in your favourite XMPP client,
send the following message to the cheogram.com
gateway contact:
reset sip account
In response, you will receive a message containing:
- a numerical username
- a password (e.g. three lowercase words separated by spaces)
Add SIP account to your Asterisk config
First of all, I added the following near the top of my
/etc/asterisk/sip.conf
:
[general]
register => username:three secret words@jmp.cbcbc7.auth.bandwidth.com:5008
The other non-default options I have set in [general]
are:
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
tlsenable=yes
transport=udp
srvlookup=no
vmexten=voicemail
relaxdtmf=yes
useragent=Asterisk PBX
tlscertfile=/etc/asterisk/asterisk.cert
tlsprivatekey=/etc/asterisk/asterisk.key
tlscapath=/etc/ssl/certs/
externhost=machinename.dyndns.org
localnet=192.168.0.0/255.255.0.0
Note that you can have more than one register
line in your config if you
use more than one SIP provider, but you must register with the server
whether you want to receive incoming calls or not.
Then I added a new blurb to the bottom of the same file:
[jmp]
type=peer
host=mp.cbcbc7.auth.bandwidth.com
port=5008
secret=three secret words
defaultuser=username
context=from-jmp
disallow=all
allow=ulaw
allow=g729
insecure=port,invite
canreinvite=no
dtmfmode=rfc2833
and for reference, here's the blurb for my Snom 300 SIP phone:
[1001]
; Snom 300
type=friend
qualify=yes
secret=password
encryption=no
context=full
host=dynamic
nat=no
directmedia=no
mailbox=1000@internal
vmexten=707
dtmfmode=rfc2833
call-limit=2
disallow=all
allow=g722
allow=ulaw
I checked that the registration was successful by running asterisk -r
and
then typing:
sip set debug on
before reloading the configuration using:
reload
Create Asterisk extensions to send and receive calls
Once I got registration to work, I hooked this up with my other extensions
so that I could send and receive calls using my JMP number.
In /etc/asterisk/extensions.conf
, I added the following:
[from-jmp]
include => home
exten => s,1,Goto(1000,1)
where home
is the context which includes my local SIP devices and 1000
is the extension I want to ring.
Then I added the following to enable calls to any destination within the
North American Numbering Plan:
[pstn-jmp]
exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231434>)
exten => _1NXXNXXXXXX,n,Dial(SIP/jmp/$ EXTEN )
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231234>)
exten => _NXXNXXXXXX,n,Dial(SIP/jmp/1$ EXTEN )
exten => _NXXNXXXXXX,n,Hangup()
Here 5551231234
is my JMP phone number, not my bwsip numerical username.
For reference, here's the rest of my dialplan in /etc/asterisk/extensions.conf
:
[general]
static=yes
writeprotect=no
clearglobalvars=no
[public]
exten => _X.,1,Hangup(3)
[sipdefault]
exten => _X.,1,Hangup(3)
[default]
exten => _X.,1,Hangup(3)
[internal]
include => home
[full]
include => internal
include => pstn-jmp
exten => 707,1,VoiceMailMain(1000@internal)
[home]
; Internal extensions
exten => 1000,1,Dial(SIP/1001,20)
exten => 1000,n,Goto(in1000-$ DIALSTATUS ,1)
exten => 1000,n,Hangup
exten => in1000-BUSY,1,Hangup(17)
exten => in1000-CONGESTION,1,Hangup(3)
exten => in1000-CHANUNAVAIL,1,VoiceMail(1000@internal,su)
exten => in1000-CHANUNAVAIL,n,Hangup(3)
exten => in1000-NOANSWER,1,VoiceMail(1000@internal,su)
exten => in1000-NOANSWER,n,Hangup(16)
exten => _in1000-.,1,Hangup(16)
Firewall
Finally, I opened a few ports in my firewall by putting the following in
/etc/network/iptables.up.rules
:
# SIP and RTP on UDP (jmp.cbcbc7.auth.bandwidth.com)
-A INPUT -s 67.231.2.13/32 -p udp --dport 5008 -j ACCEPT
-A INPUT -s 216.82.238.135/32 -p udp --dport 5008 -j ACCEPT
-A INPUT -s 67.231.2.13/32 -p udp --sport 5004:5005 --dport 10001:20000 -j ACCEPT
-A INPUT -s 216.82.238.135/32 -p udp --sport 5004:5005 --dport 10001:20000 -j ACCEPT
Outbound calls not working
While the above setup works for me for inbound calls, it doesn't currently
work for outbound calls.
The hostname currently resolves to one of two IP addresses:
$ dig +short jmp.cbcbc7.auth.bandwidth.com
67.231.2.13
216.82.238.135
If I pin it to the first one by putting the following in my /etc/hosts
file:
67.231.2.13 jmp.cbcbc7.auth.bandwidth.com
then I get a 486
error back from the server when I dial 1-555-456-4567
:
<--- SIP read from UDP:67.231.2.13:5008 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK03210a30
From: "Francois Marier" <sip:5551231234@127.0.0.1>
To: <sip:15554564567@jmp.cbcbc7.auth.bandwidth.com:5008>
Call-ID: 575f21f36f57951638c1a8062f3a5201@127.0.0.1:5060
CSeq: 103 INVITE
Content-Length: 0
On the other hand, if I pin it to 216.82.238.135
, then I get a 600
error:
<--- SIP read from UDP:216.82.238.135:5008 --->
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7b7f7ed9
From: "Francois Marier" <sip:5551231234@127.0.0.1>
To: <sip:15554564567@jmp.cbcbc7.auth.bandwidth.com:5008>
Call-ID: 5bebb8d05902c1732c6b9f4776844c66@127.0.0.1:5060
CSeq: 103 INVITE
Content-Length: 0
If you have any idea what might be wrong here, or if you got outbound calls to
work on Bandwidth.com, please leave a comment!
reset sip account
/etc/asterisk/sip.conf
:
[general]
register => username:three secret words@jmp.cbcbc7.auth.bandwidth.com:5008
The other non-default options I have set in [general]
are:
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
tlsenable=yes
transport=udp
srvlookup=no
vmexten=voicemail
relaxdtmf=yes
useragent=Asterisk PBX
tlscertfile=/etc/asterisk/asterisk.cert
tlsprivatekey=/etc/asterisk/asterisk.key
tlscapath=/etc/ssl/certs/
externhost=machinename.dyndns.org
localnet=192.168.0.0/255.255.0.0
Note that you can have more than one register
line in your config if you
use more than one SIP provider, but you must register with the server
whether you want to receive incoming calls or not.
Then I added a new blurb to the bottom of the same file:
[jmp]
type=peer
host=mp.cbcbc7.auth.bandwidth.com
port=5008
secret=three secret words
defaultuser=username
context=from-jmp
disallow=all
allow=ulaw
allow=g729
insecure=port,invite
canreinvite=no
dtmfmode=rfc2833
and for reference, here's the blurb for my Snom 300 SIP phone:
[1001]
; Snom 300
type=friend
qualify=yes
secret=password
encryption=no
context=full
host=dynamic
nat=no
directmedia=no
mailbox=1000@internal
vmexten=707
dtmfmode=rfc2833
call-limit=2
disallow=all
allow=g722
allow=ulaw
I checked that the registration was successful by running asterisk -r
and
then typing:
sip set debug on
before reloading the configuration using:
reload
Create Asterisk extensions to send and receive calls
Once I got registration to work, I hooked this up with my other extensions
so that I could send and receive calls using my JMP number.
In /etc/asterisk/extensions.conf
, I added the following:
[from-jmp]
include => home
exten => s,1,Goto(1000,1)
where home
is the context which includes my local SIP devices and 1000
is the extension I want to ring.
Then I added the following to enable calls to any destination within the
North American Numbering Plan:
[pstn-jmp]
exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231434>)
exten => _1NXXNXXXXXX,n,Dial(SIP/jmp/$ EXTEN )
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231234>)
exten => _NXXNXXXXXX,n,Dial(SIP/jmp/1$ EXTEN )
exten => _NXXNXXXXXX,n,Hangup()
Here 5551231234
is my JMP phone number, not my bwsip numerical username.
For reference, here's the rest of my dialplan in /etc/asterisk/extensions.conf
:
[general]
static=yes
writeprotect=no
clearglobalvars=no
[public]
exten => _X.,1,Hangup(3)
[sipdefault]
exten => _X.,1,Hangup(3)
[default]
exten => _X.,1,Hangup(3)
[internal]
include => home
[full]
include => internal
include => pstn-jmp
exten => 707,1,VoiceMailMain(1000@internal)
[home]
; Internal extensions
exten => 1000,1,Dial(SIP/1001,20)
exten => 1000,n,Goto(in1000-$ DIALSTATUS ,1)
exten => 1000,n,Hangup
exten => in1000-BUSY,1,Hangup(17)
exten => in1000-CONGESTION,1,Hangup(3)
exten => in1000-CHANUNAVAIL,1,VoiceMail(1000@internal,su)
exten => in1000-CHANUNAVAIL,n,Hangup(3)
exten => in1000-NOANSWER,1,VoiceMail(1000@internal,su)
exten => in1000-NOANSWER,n,Hangup(16)
exten => _in1000-.,1,Hangup(16)
Firewall
Finally, I opened a few ports in my firewall by putting the following in
/etc/network/iptables.up.rules
:
# SIP and RTP on UDP (jmp.cbcbc7.auth.bandwidth.com)
-A INPUT -s 67.231.2.13/32 -p udp --dport 5008 -j ACCEPT
-A INPUT -s 216.82.238.135/32 -p udp --dport 5008 -j ACCEPT
-A INPUT -s 67.231.2.13/32 -p udp --sport 5004:5005 --dport 10001:20000 -j ACCEPT
-A INPUT -s 216.82.238.135/32 -p udp --sport 5004:5005 --dport 10001:20000 -j ACCEPT
Outbound calls not working
While the above setup works for me for inbound calls, it doesn't currently
work for outbound calls.
The hostname currently resolves to one of two IP addresses:
$ dig +short jmp.cbcbc7.auth.bandwidth.com
67.231.2.13
216.82.238.135
If I pin it to the first one by putting the following in my /etc/hosts
file:
67.231.2.13 jmp.cbcbc7.auth.bandwidth.com
then I get a 486
error back from the server when I dial 1-555-456-4567
:
<--- SIP read from UDP:67.231.2.13:5008 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK03210a30
From: "Francois Marier" <sip:5551231234@127.0.0.1>
To: <sip:15554564567@jmp.cbcbc7.auth.bandwidth.com:5008>
Call-ID: 575f21f36f57951638c1a8062f3a5201@127.0.0.1:5060
CSeq: 103 INVITE
Content-Length: 0
On the other hand, if I pin it to 216.82.238.135
, then I get a 600
error:
<--- SIP read from UDP:216.82.238.135:5008 --->
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7b7f7ed9
From: "Francois Marier" <sip:5551231234@127.0.0.1>
To: <sip:15554564567@jmp.cbcbc7.auth.bandwidth.com:5008>
Call-ID: 5bebb8d05902c1732c6b9f4776844c66@127.0.0.1:5060
CSeq: 103 INVITE
Content-Length: 0
If you have any idea what might be wrong here, or if you got outbound calls to
work on Bandwidth.com, please leave a comment!
[from-jmp]
include => home
exten => s,1,Goto(1000,1)
[pstn-jmp]
exten => _1NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231434>)
exten => _1NXXNXXXXXX,n,Dial(SIP/jmp/$ EXTEN )
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Set(CALLERID(all)=Francois Marier <5551231234>)
exten => _NXXNXXXXXX,n,Dial(SIP/jmp/1$ EXTEN )
exten => _NXXNXXXXXX,n,Hangup()
[general]
static=yes
writeprotect=no
clearglobalvars=no
[public]
exten => _X.,1,Hangup(3)
[sipdefault]
exten => _X.,1,Hangup(3)
[default]
exten => _X.,1,Hangup(3)
[internal]
include => home
[full]
include => internal
include => pstn-jmp
exten => 707,1,VoiceMailMain(1000@internal)
[home]
; Internal extensions
exten => 1000,1,Dial(SIP/1001,20)
exten => 1000,n,Goto(in1000-$ DIALSTATUS ,1)
exten => 1000,n,Hangup
exten => in1000-BUSY,1,Hangup(17)
exten => in1000-CONGESTION,1,Hangup(3)
exten => in1000-CHANUNAVAIL,1,VoiceMail(1000@internal,su)
exten => in1000-CHANUNAVAIL,n,Hangup(3)
exten => in1000-NOANSWER,1,VoiceMail(1000@internal,su)
exten => in1000-NOANSWER,n,Hangup(16)
exten => _in1000-.,1,Hangup(16)
/etc/network/iptables.up.rules
:
# SIP and RTP on UDP (jmp.cbcbc7.auth.bandwidth.com)
-A INPUT -s 67.231.2.13/32 -p udp --dport 5008 -j ACCEPT
-A INPUT -s 216.82.238.135/32 -p udp --dport 5008 -j ACCEPT
-A INPUT -s 67.231.2.13/32 -p udp --sport 5004:5005 --dport 10001:20000 -j ACCEPT
-A INPUT -s 216.82.238.135/32 -p udp --sport 5004:5005 --dport 10001:20000 -j ACCEPT
Outbound calls not working
While the above setup works for me for inbound calls, it doesn't currently
work for outbound calls.
The hostname currently resolves to one of two IP addresses:
$ dig +short jmp.cbcbc7.auth.bandwidth.com
67.231.2.13
216.82.238.135
If I pin it to the first one by putting the following in my /etc/hosts
file:
67.231.2.13 jmp.cbcbc7.auth.bandwidth.com
then I get a 486
error back from the server when I dial 1-555-456-4567
:
<--- SIP read from UDP:67.231.2.13:5008 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK03210a30
From: "Francois Marier" <sip:5551231234@127.0.0.1>
To: <sip:15554564567@jmp.cbcbc7.auth.bandwidth.com:5008>
Call-ID: 575f21f36f57951638c1a8062f3a5201@127.0.0.1:5060
CSeq: 103 INVITE
Content-Length: 0
On the other hand, if I pin it to 216.82.238.135
, then I get a 600
error:
<--- SIP read from UDP:216.82.238.135:5008 --->
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7b7f7ed9
From: "Francois Marier" <sip:5551231234@127.0.0.1>
To: <sip:15554564567@jmp.cbcbc7.auth.bandwidth.com:5008>
Call-ID: 5bebb8d05902c1732c6b9f4776844c66@127.0.0.1:5060
CSeq: 103 INVITE
Content-Length: 0
If you have any idea what might be wrong here, or if you got outbound calls to
work on Bandwidth.com, please leave a comment!
$ dig +short jmp.cbcbc7.auth.bandwidth.com
67.231.2.13
216.82.238.135
67.231.2.13 jmp.cbcbc7.auth.bandwidth.com
<--- SIP read from UDP:67.231.2.13:5008 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK03210a30
From: "Francois Marier" <sip:5551231234@127.0.0.1>
To: <sip:15554564567@jmp.cbcbc7.auth.bandwidth.com:5008>
Call-ID: 575f21f36f57951638c1a8062f3a5201@127.0.0.1:5060
CSeq: 103 INVITE
Content-Length: 0
<--- SIP read from UDP:216.82.238.135:5008 --->
SIP/2.0 600 Busy Everywhere
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7b7f7ed9
From: "Francois Marier" <sip:5551231234@127.0.0.1>
To: <sip:15554564567@jmp.cbcbc7.auth.bandwidth.com:5008>
Call-ID: 5bebb8d05902c1732c6b9f4776844c66@127.0.0.1:5060
CSeq: 103 INVITE
Content-Length: 0