Asterisk in a cloud - Asterisk 1.6.0.5 optimized for Amazon EC2

eric · #1 (**permalink**) February 17th, 2009, 07:57 PM

We've received a number of requests for an Amazon EC2 AMI built based on the Asterisk in a cloud tutorial, so we built one.

Open source Asterisk PBX running on your corner of the Amazon cloud. Asterisk 1.6.0.5 with a VoIP optimized kernel timer.

AMI ID: ami-0bfa1d62
AMI Manifest: voxilla/asterisk-1.6.0.5-i386.manifest.xml
License: Public
Operating System: Linux/Unix

About this AMI
Published by Voxilla (http://voxilla.com).

This AMI is a build of Asterisk 1.6.0.5 on Fedora 8 with the kernel timer optimized for VoIP (1000Hz timer).

Asterisk was buit from the Asterisk in a cloud tutorial.

This AMI contains:

Asterisk 1.6.0.5
DAHDI (dahdi_dummy - used by MeetMe conferencing and IAX2)
VoIP optimized 1000Hz kernel timer

Release notes

Asterisk runs under the asterisk user and group, instead of root.
For help identifying network ports used by Asterisk or configuring an Asterisk EC2 Security Group, see the Asterisk in a cloud tutorial.
The Asterisk configuration was made by running make samples.
sip.conf was changed to treat the 10.0.0.0/8 subnet as a localnetwork, SIP TCP is enabled, and STUN is used to determine the public IP address of the server.

The following modules are enabled by the sample configuration:

Code:

Module                      Description
app_adsiprog                Asterisk ADSI Programming Application
app_alarmreceiver           Alarm Receiver for Asterisk
app_amd                     Answering Machine Detection Application
app_authenticate            Authentication Application
app_cdr                     Tell Asterisk to not maintain a CDR for
app_chanisavail             Check channel availability
app_channelredirect         Redirects a given channel to a dialplan
app_chanspy                 Listen to the audio of an active channel
app_controlplayback         Control Playback Application
app_dahdibarge              Barge in on DAHDI channel application
app_dahdiras                DAHDI ISDN Remote Access Server
app_dahdiscan               Scan DAHDI channels application
app_db                      Database Access Functions
app_dial                    Dialing Application
app_dictate                 Virtual Dictation Machine
app_directed_pickup         Directed Call Pickup Application
app_directory               Extension Directory
app_disa                    DISA (Direct Inward System Access) Appli
app_dumpchan                Dump Info About The Calling Channel
app_echo                    Simple Echo Application
app_exec                    Executes dialplan applications
app_externalivr             External IVR Interface Application
app_festival                Simple Festival Interface
app_flash                   Flash channel application
app_followme                Find-Me/Follow-Me Application
app_forkcdr                 Fork The CDR into 2 separate entities
app_getcpeid                Get ADSI CPE ID
app_ices                    Encode and Stream via icecast and ices
app_image                   Image Transmission Application
app_jack                    JACK Interface
app_macro                   Extension Macros
app_meetme                  MeetMe conference bridge
app_milliwatt               Digital Milliwatt (mu-law) Test Applicat
app_minivm                  Mini VoiceMail (A minimal Voicemail e-ma
app_mixmonitor              Mixed Audio Monitoring Application
app_morsecode               Morse code
app_mp3                     Silly MP3 Application
app_nbscat                  Silly NBS Stream Application
app_page                    Page Multiple Phones
app_parkandannounce         Call Parking and Announce Application
app_pickupchan              Channel Pickup Application
app_playback                Sound File Playback Application
app_privacy                 Require phone number to be entered, if n
app_queue                   True Call Queueing
app_read                    Read Variable Application
app_readexten               Read and evaluate extension validity
app_readfile                Stores output of file into a variable
app_record                  Trivial Record Application
app_sayunixtime             Say time
app_senddtmf                Send DTMF digits Application
app_sendtext                Send Text Applications
app_setcallerid             Set CallerID Presentation Application
app_sms                     SMS/PSTN handler
app_softhangup              Hangs up the requested channel
app_speech_utils            Dialplan Speech Applications
app_stack                   Dialplan subroutines (Gosub, Return, etc
app_system                  Generic System() application
app_talkdetect              Playback with Talk Detection
app_test                    Interface Test Application
app_transfer                Transfers a caller to another extension
app_url                     Send URL Applications
app_userevent               Custom User Event Application
app_verbose                 Send verbose output
app_voicemail               Comedian Mail (Voicemail System)
app_waitforring             Waits until first ring after time
app_waitforsilence          Wait For Silence
app_waituntil               Wait until specified time
app_while                   While Loops and Conditional Execution
app_zapateller              Block Telemarketers with Special Informa
cdr_adaptive_odbc           Adaptive ODBC CDR backend
cdr_csv                     Comma Separated Values CDR Backend
cdr_custom                  Customizable Comma Separated Values CDR
cdr_manager                 Asterisk Manager Interface CDR Backend
cdr_odbc                    ODBC CDR Backend
cdr_pgsql                   PostgreSQL CDR Backend
cdr_radius                  RADIUS CDR Backend
cdr_sqlite3_custom          SQLite3 Custom CDR Module
chan_agent                  Agent Proxy Channel
chan_dahdi                  DAHDI Telephony
chan_gtalk                  Gtalk Channel Driver
chan_iax2                   Inter Asterisk eXchange (Ver 2)
chan_jingle                 Jingle Channel Driver
chan_local                  Local Proxy Channel (Note: used internal
chan_mgcp                   Media Gateway Control Protocol (MGCP)
chan_oss                    OSS Console Channel Driver
chan_phone                  Linux Telephony API Support
chan_sip                    Session Initiation Protocol (SIP)
chan_skinny                 Skinny Client Control Protocol (Skinny)
chan_unistim                UNISTIM Protocol (USTM)
codec_a_mu                  A-law and Mulaw direct Coder/Decoder
codec_adpcm                 Adaptive Differential PCM Coder/Decoder
codec_alaw                  A-law Coder/Decoder
codec_dahdi                 Generic DAHDI Transcoder Codec Translato
codec_g722                  ITU G.722-64kbps G722 Transcoder
codec_g726                  ITU G.726-32kbps G726 Transcoder
codec_gsm                   GSM Coder/Decoder
codec_ilbc                  iLBC Coder/Decoder
codec_lpc10                 LPC10 2.4kbps Coder/Decoder
codec_resample              SLIN Resampling Codec
codec_speex                 Speex Coder/Decoder
codec_ulaw                  mu-Law Coder/Decoder
format_g723                 G.723.1 Simple Timestamp File Format
format_g726                 Raw G.726 (16/24/32/40kbps) data
format_g729                 Raw G729 data
format_gsm                  Raw GSM data
format_h263                 Raw H.263 data
format_h264                 Raw H.264 data
format_ilbc                 Raw iLBC data
format_jpeg                 JPEG (Joint Picture Experts Group) Image
format_ogg_vorbis           OGG/Vorbis audio
format_pcm                  Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.
format_sln                  Raw Signed Linear Audio support (SLN)
format_sln16                Raw Signed Linear 16KHz Audio support (S
format_vox                  Dialogic VOX (ADPCM) File Format
format_wav                  Microsoft WAV format (8000Hz Signed Line
format_wav_gsm              Microsoft WAV format (Proprietary GSM)
func_base64                 base64 encode/decode dialplan functions
func_blacklist              Look up Caller*ID name/number from black
func_callerid               Caller ID related dialplan functions
func_cdr                    Call Detail Record (CDR) dialplan functi
func_channel                Channel information dialplan function
func_curl                   Load external URL
func_cut                    Cut out information from a string
func_db                     Database (astdb) related dialplan functi
func_devstate               Gets or sets a device state in the dialp
func_dialgroup              Dialgroup dialplan function
func_dialplan               Dialplan Context/Extension/Priority Chec
func_enum                   ENUM related dialplan functions
func_env                    Environment/filesystem dialplan function
func_extstate               Gets an extension's state in the dialpla
func_global                 Variable dialplan functions
func_groupcount             Channel group dialplan functions
func_iconv                  Charset conversions
func_lock                   Dialplan mutexes
func_logic                  Logical dialplan functions
func_math                   Mathematical dialplan function
func_md5                    MD5 digest dialplan functions
func_module                 Checks if Asterisk module is loaded in m
func_odbc                   ODBC lookups
func_rand                   Random number dialplan function
func_realtime               Read/Write/Store/Destroy values from a R
func_sha1                   SHA-1 computation dialplan function
func_shell                  Returns the output of a shell command
func_strings                String handling dialplan functions
func_sysinfo                System information related functions
func_timeout                Channel timeout dialplan functions
func_uri                    URI encode/decode dialplan functions
func_version                Get Asterisk Version/Build Info
func_vmcount                Indicator for whether a voice mailbox ha
func_volume                 Technology independent volume control
pbx_ael                     Asterisk Extension Language Compiler
pbx_config                  Text Extension Configuration
pbx_dundi                   Distributed Universal Number Discovery (
pbx_loopback                Loopback Switch
pbx_realtime                Realtime Switch
pbx_spool                   Outgoing Spool Support
res_adsi                    ADSI Resource
res_ael_share               share-able code for AEL
res_agi                     Asterisk Gateway Interface (AGI)
res_clioriginate            Call origination from the CLI
res_config_curl             Realtime Curl configuration
res_config_ldap             LDAP realtime interface
res_config_odbc             Realtime ODBC configuration
res_config_pgsql            PostgreSQL RealTime Configuration Driver
res_convert                 File format conversion CLI command
res_crypto                  Cryptographic Digital Signatures
res_indications             Region-specific tones
res_jabber                  AJI - Asterisk Jabber Interface
res_limit                   Resource limits
res_monitor                 Call Monitoring Resource
res_musiconhold             Music On Hold Resource
res_odbc                    ODBC resource
res_phoneprov               HTTP Phone Provisioning
res_realtime                Realtime Data Lookup/Rewrite
res_smdi                    Simplified Message Desk Interface (SMDI)
res_snmp                    SNMP [Sub]Agent for Asterisk
res_speech                  Generic Speech Recognition API

dgiulianelli · #2 (**permalink**) February 25th, 2009, 03:38 PM

Eric,

I tried your tutorial and everything went fine except that I can't find a way to register my SIP phones with my instance of Asterisk in the cloud. I double checked the Asterisk group permissions, but when I send an invite to the amazon server, I get not response.
In my test I added the following lines to sip.conf:

[2001]
type=friend
username=2001
host=dynamic
context=default

I restarted Asterisk and then configured and tried to register first with my CISCO 7960 IP phone, and then with Zoiper softphone, but the request doesn't seem to go through (no activity on the Asterisk console at all).
Any ideas on what I'm doing wrong?
Thanks in advance.
Danilo Giulianelli

eric · #3 (**permalink**) February 25th, 2009, 04:50 PM

Danilo,

Can you ping the box?

Then make sure asterisk is listening on the correct ports, from the command line type:

Code:

netstat -nap | grep asterisk

The output will show you what processes are listening and on which port. You should see something like:

Code:

tcp        0      0 0.0.0.0:5060                0.0.0.0:*                   LISTEN      21431/asterisk      
tcp        0      0 0.0.0.0:5061                0.0.0.0:*                   LISTEN      21431/asterisk      
udp        0      0 0.0.0.0:5000                0.0.0.0:*                               21431/asterisk      
udp        0      0 0.0.0.0:2727                0.0.0.0:*                               21431/asterisk      
udp        0      0 0.0.0.0:4520                0.0.0.0:*                               21431/asterisk      
udp        0      0 0.0.0.0:5060                0.0.0.0:*                               21431/asterisk      
udp        0      0 0.0.0.0:4569                0.0.0.0:*                               21431/asterisk      
unix  2      [ ACC ]     STREAM     LISTENING     4808258 21431/asterisk      /var/run/asterisk/asterisk.ctl
unix  3      [ ]         STREAM     CONNECTED     6458571 21431/asterisk      
unix  3      [ ]         STREAM     CONNECTED     4808274 21431/asterisk

The important line is to make sure asterisk is listening on UDP port 5060.

Have you checked the SIP settings and tried enabling SIP debug to confirm that the traffic is getting to the asterisk box?

From the command line:

Code:

asterisk -rvvvvvvvvvv

then type:

Code:

sip show settings

Make sure the externip address is your EC2 public IP address.

then type:

Code:

sip set debug on

Then restart your phone and see if you see any traffic registration traffic.

dgiulianelli · #4 (**permalink**) February 25th, 2009, 05:09 PM

I can ping the box without problems. The output of my netstat is the following:

tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN 810/asterisk
tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 810/asterisk
udp 0 0 0.0.0.0:5000 0.0.0.0:* 810/asterisk
udp 0 0 0.0.0.0:2727 0.0.0.0:* 810/asterisk
udp 0 0 0.0.0.0:4520 0.0.0.0:* 810/asterisk
udp 0 0 0.0.0.0:5060 0.0.0.0:* 810/asterisk
udp 0 0 0.0.0.0:4569 0.0.0.0:* 810/asterisk
unix 2 [ ACC ] STREAM LISTENING 49077 810/asterisk /var/run/asterisk/asterisk.ctl

I don't see port 5061 but hopefully that should be ok.
Also my Externip matches the public elastic IP I assigned to my instance. But when I enable sip debug I don't see any trace coming out. Also I used wireshark on my machine and verified that no response is coming back from the instance box.
BTW I tried enabling httpd and authorize port 80 and everything works fine, but at this point I believe it is still something related to my Asterisk group permissions:

$ ec2-describe-group Asterisk
GROUP 761954543573 Asterisk Security Group for Asterisk instances
PERMISSION 761954543573 Asterisk ALLOWS icmp -1 -1 FROM CIDR 0.0.0.0/0
PERMISSION 761954543573 Asterisk ALLOWS tcp 80 80 FROM CIDR 0.0.0.0/0
PERMISSION 761954543573 Asterisk ALLOWS tcp 5060 5061 FROM CIDR 0.0.0.0/0
PERMISSION 761954543573 Asterisk ALLOWS tcp 22 22 FROM CIDR 135.207.168.62/32
PERMISSION 761954543573 Asterisk ALLOWS udp 4569 4569 FROM CIDR 0.0.0.0/0
PERMISSION 761954543573 Asterisk ALLOWS udp 5060 5060 FROM CIDR 0.0.0.0/0
PERMISSION 761954543573 Asterisk ALLOWS udp 10000 20000 FROM CIDR 0.0.0.0/0

Do you see anything strange? Should the "Source User:group" field populated?
Thanks for the help.
Danilo

dgiulianelli · #5 (**permalink**) February 26th, 2009, 05:58 PM

Eric,

I found the problem: my company firewall is blocking all UDP traffic in and out of our internal network. I tried to register with the machine from my home internet connection and everything works just fine.
Thanks again for the support.

kudroz · #6 (**permalink**) April 28th, 2009, 09:41 PM

Great AMI, it saved me a lot of time.

I'm getting choppy audio with the MOH and IVR, and also in VM menus. Any ideas?