New in Asterisk v1.2.0: The naming convention for IAX channels has changed in a minor way such that the call number follows a "-" rather than a "/" character.
As of january 2004, IAX is now the same as IAX2. The old version is now labeled IAX1. For backwards
compability, the source for IAX1 is still in the source tree and compilation can be enabled in the Makefile.
For more information on the protocol, see IAX2.
IAXtel from Dec. 2003 only supports IAX2.
IAX2 can be used to
- Connect Asterisk servers with each other
- Connect to IAXtel
- Connect IAX clients
Channels are configured in iax.conf and used in extensions.conf
Channel Naming
The format of an IAX channel name used for an outgoing connection is:IAX/[<user>[:<secret>]@]<peer>[:<portno>][/<exten>[@<context>][/<options>]]
- <user>: UserID on remote peer, or name of client configured in iax.conf (optional)
- <secret>: Password (optional). Alternatively it can be the filename of an RSA key, without the trailing extension (.key or .pub), and enclosed in [square brackets], like this: [thefilename]
- <peer>: Name of server to connect to
- <portno>: Port number for connection on server (optional)
- <exten>: Extension in the remote Asterisk server (optional)
- <context>: Context to use in the remote Asteriskserver (optional)
- <options>: The only option available is 'a' meaning 'request autoanswer'
Example outgoing channel names:
- IAX/mark:asdf@myserver/6275@default – Call to "myserver" using "mark" as username and "asdf" as password, and requesting extension 6275 in default context
- IAX/iaxphone/s/a – Call to "iaxphone" requesting immediate answer.
- IAX/guest@misery.digium.com – Call Digium
- IAX/john:[johnrsa]@somewhere.com — Call to somewhere.com, using "john" as a username and an RSA key for authentication.
The format of an IAX channel name used for an incoming connection is simply:
IAX[[<username>@]<host>]/<callno>
- <username>: the username, if known
- <host>: the apparent host connecting
- <callno>: the local call number
Example incoming channel names:
- IAX[mark@192.168.0.1]/14 – Call number 14 from user "mark" at 192.168.0.1
- IAX[192.168.10.1]/13 – Call 13 from 192.168.10.1
Trunking
IAX Trunking needs support of a hardware timer. See Asterisk timer for more information.IAX Trunking allows multiple voice streams to share a single "trunk" to another server, reducing overhead created by IP packets. Basically after 4 concurrent calls you start seeing a per packet savings when trunking is enabled.
Use experience (early 2005): "A long time ago i tried to make one big iax2 trunk for one of my customers, i soon changed this to several small trunks. (bandwith doesnt rise all that much if you use 2 trunks instead of 1.) Asterisk didnt seem to like my big trunk very much (i don't remember how big it was, but probably over 100 calls). I think, but am not sure, that with a lot of calls inside the trunk, some calls seemed to go suddenly go outside of the trunk in one or more directions, bursts of error messages appeared on the cli etc." The user settled for ~60 calls/trunk and had that working fine.
i didnt investigate it a lot more, my problems went away with splitting them up in smaller trunks.
Jitter buffer
There is a new jitter buffer implementation in Asterisk 1.2. This new jitter buffer has a feature that enables trunk time stamps to be sent within the trunked stream. This enables the use of the jitter buffer with a trunked IAX2 connection (previously impossible). However, enabling trunk time stamps in a trunk connection to an older version of asterisk causes problems including one-way audio. See The New Jitter Buffer in Asterisk by Steve Kann for more info.
DTMF issues: The IAX jitterbuffer is known to cause DTMF recognition troubles (1.2, 1.4), so set jitterbuffer=no if you experience trouble.
Jitter buffer debugging
In order to obtain information on the current state of the jitter buffer, enter on the console:
iax2 set debug jb
or
iax2 set debug jb off
to disable jb debugging. Whenever frames are leaving the jitter buffer, a letter is printed (letter: event, action):
G: growing jitter buffer, interpolating last frame
o: not a voice frame, forwarding it to the decoder
v: early or late voice frame, forwarding it to the decoder
V: voice frame, forwarding it to the decoder
l: late voice frame, too late to play so it's dropped
L: lost frame, interpolating last frame
s: shrinking jb, frame lost
S: shrinking jb, no frame lost
See also
- Intl. success stories of benj with IAX beating SIP in problematic infrastructures
- Asterisk readme iax: Readme.iax from the standard distribution
- Asterisk iax rsa auth: About RSA authentication for IAX2 connections
- Asterisk config iax.conf: Configuration file for IAX connectivity
- Asterisk handbook
- Asterisk security
- Asterisk channels: Overview of Asterisk channels
- Asterisk IAX media path: Native bridging of IAX calls
- IAX: The IAX protocol
- Asterisk RealTime IAX
- IAX2 trunking statistics
- IAX load balancer by Stanisław Pitucha
- Important thing to look at if you get one way audio problem with Asterisk 1.4.10 and FreePBX 2.3.0
- Asterisk | FAQ | Tips & Tricks


Comments
333!Our iax2 phone and ata can support your voip system
We manufacture ipphone and ata which support sip and iax2 running sychronously. It do support asterisk core
-
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Fax: +86-755-26505858
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Email: v15914106575@gmail.com
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MSN: ludanmigai@hotmail.com
Skype: abc33345905
Address: R506,Jianda Bldg., 10North Keyuan Rd, High-Tech Industrial Park, Shenzhen518057China
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333VPN and VGCP - for VoIP Blocking issue
In today’s market, VoIP for business has become more and more popular and necessary than ever before.
Dubai has become a big market, many big companies need to open branch offices in the UAE allowing more profit and larger market access. Technology Issues become apparent during this process that can cripple communications for that company. The primary communications issues are with VOIP blocking policies implemented in Dubai.
Now, here is the good news, A Canada based company SpeedVoIP with their integral R&D team have work out a new way to solve this VoIP blocking issue. This new system VGCP (VoiceGuard@ Control Protocol) has now laid the path to streamline low cost telephony solutions removing country limitations.
VGCP is a proprietary layer 2 link protocol working at between IP stack and NIC driver for VoIP anti-blocking. The core patent-pending VGCP is industry's most state-of-art voice service provider class security protocol whose scalability and flexibility results in not to compromise voice quality and overhead. VGCP controls and monitors full voice signalling and media flow intelligently, meanwhile disguises sip and RTP packets into normal allowed data packets such as DNS and TFTP, and makes two-way encryption and decryption driven by user-customized policy. VGCP is fully transparent to upper SIP proxy or UA which means VoiceGuard@ can work with any 3rd party soft phone/ATA/Gateway/IP Phone/IADs and SIP Proxy or Server not like some competitors which take effect on their own device and soft switch.
Korea Telecom has implemented this solution successfully for more than one year. And it has been operational within a group of Dubai companies. The trials and implementations proves that, The VGCP solution is the best solution to solve the VoIP Blocking issue and provides stable communications platforms providing an indispensable part of the business network.
Andy Wong ~ ~
MSN: andywong-01@hotmail.com
Email: xd.wong@speed-voip.com
www.speed-voip.com
333CAllback
to realize a callback with the french provider free.fr.
So far i have two problems:
first dtmf tones are not catched in the disa section of asterisk
second the legb part of the callback isnt possible, because call-limit is eaual 1?
koc
333CAllback
to realize a callback with the french provider free.fr.
So far i have two problems:
first dtmf tones are not catched in the disa section of asterisk
second the legb part of the callback isnt possible, because call-limit is eaual 1?
koc
333CAllback
to realize a callback with the french provider free.fr.
So far i have two problems:
first dtmf tones are not catched in the disa section of asterisk
second the legb part of the callback isnt possible, because call-limit is eaual 1?
koc
333SIP Phone <->Asterisk --- IAX --- Asterisk SIP Provider
Outgoing Call flow: SIP-Phone--->(LAN)-->LocalAsterisk---->(VPN)---IAX---(VPN)-->remoteAsterisk->(Internet)---SIP provider.
Incomming flow: SIP-provider--->Internet--->RemoteAsterisk-->(VPN)--IAX2--(VPN)-->LocalAsterisk--->SIP-phone
IAX.CONF
- Fridu-Follijen
- type=friend
- host=10.10.95.1 (for local) and 10.10.95.14(for remote)
- secret=MySecret
- context=FollijenCtx (for local) FriduCtx(for remote)
- qualify=60000
- transfert=no
SIP.conf
Freephonie-Provider ; **** Remote Asterisk ****
- type=peer
- username=095xxxxx
- fromuser=095xxxxx
- secret=MyProviderSecret
- host=freephonie.net
- qualify=60000 ; implement supervision conserve la session NAT
- dtmfmode=inband ; free does not handle SIP/DTMF
- call-limit=1 ; Free does not allow multiple call
- canreinvite=yes
Follijen-PA-Z1 ; *** local asterisk ****
- type=friend
- secret=MyPhoneSecret
- callerid="Fulup Meeting" <1121>
- host=dynamic ; This device needs to register
- canreinvite=yes ; Typically set to NO if behind NAT
- insecure=port ; Follijen has a fix IP adress
- qualify=60000 ; force monitoring
- context=FollijenCtx
My extensions.conf (I use *124 for my test that translate in a real PSTN french number)
FollijenCtx ; local asterisk
- exten => *124,1,Dial(IAX2/Fridu-Follijen/${EXTEN}@FriduCtx)
FriduCtx ; remote asterisk
- exten => *124,1,Dial(SIP/Freephonie-Provider/0297xxxxxx,,r)
Warning: When calling out, receiving stream from localasterisk to your localphone is an unsolicited UDP stream and should be allowed by your firewall. If you do not get audio when calling out, then check your firewall on your LAN from your local asterisk to your local sip phone.
333
Any help will be greatly appreciated.
Leo
333IAX2/ not IAX/
Asterisk 1.2.7.1 under Debian.