Working configuration for incoming Asterisk IAX2

If you have not signed up for a number under My Numbers, do so now, then edit that number, edit Destination One click on IAX2 and enter...

08444846041:PLAINTEXTPASS@67.172.238.000

Where 08444846041 is your UK number, PLAINTEXTPASS matches your iax.conf, and 67.172.238.000 is your asterisk server.

replace the 08444846041 number with your UK number, the settings below are just what I use on my machine, feel free to tweak them once you get it working. Username may not be needed.


In IAX.conf

[general]
authdebug=yes
bindport=4569

disallow=all
allow=ulaw ;(g711)64 Kbps, sample-based
allow=ilbc ;15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
allow=gsm ;13 Kbps (full rate), 20ms frame size
allow=alaw ;(g711)64 Kbps, sample-based

jitterbuffer=yes
dropcount=1
tos=lowdelay ;IAX can optionally set the TOS (Type of Service) bits to
;specified values to help improve performance in routing.
;The recommended value is "lowdelay", which many routers
;(including any Linux routers with 2.4 kernels that have not
;been altered with ip tables) will give priority to these
;packets, improving voice quality.

[08444846041]
username=SIPUSERNAME
secret=PLAINTEXTPASS
type=user
context=inbound-voipuser






In extensions.conf - replace the numbers after the 44 with your number without the 0, and replace the Sip/21 with the extension you want to ring.
This will make my phone ring for 20 seconds and in case of busy/unavailable it will go to voicemail.

[inbound-voipuser]
exten => 448444846041,1,Dial(SIP/21,20)
exten => 448444846041,2,Voicemail2(u12)
exten => 448444846041,3,HangUp



If you are behind a NAT/firewall, you'll need port 4569 forwarding in.

The following codec's are supported and known to work, g729 works on outbound sip, but not incoming IAX.
g711u (ulaw)
g711a (alaw)
gsm
ilbc


Under My Numbers you can also specify a PSTN number to ring if your Asterisk server is not available.

Big thanks to ichilton for his original post that I based my configuration on, and to #asterisk on irc.freenode.net for their help. I will update this as my experience grows.

Thanks!

I've made this a sticky.

--ian

Hi

Just a couple of things

Personally i have the codecs set up as so

disallow=all ; same as bandwidth=high
allow=g729 ; Only if ou have the licence
allow=ilbc ; The prefered codec of IAX
allow=gsm
allow=ulaw
allow=alaw

This way I get the best use of bandwidth. I do notice that calls from voipuser support ilbc but not G729 so unless you have the G729 licence you can miss that one out. Also make sure both alaw and ulaw are in the conf as well.

Also the tos should be

tos=lowdelay -- Minimize delay

Including the trailing remarks. It may work without but I had strange errors when I didnt have the trailing remarks including failing calls ..

Cheers Ian

ianplain :

disallow=all allow=g729 allow=ilbc allow=gsm allow=ulaw allow=alaw

I have the same, except I took gsm out because it's not very good.

Quote:

I do notice that calls from voipuser support ilbc but not G729 so unless you have the G729 licence you can miss that one out.

voipuser works with g729 on the outbound (over SIP) but not on the incoming over IAX. I believe this is because they dont have licences for it on IAX.

Quote:

Including the trailing remarks. It may work without but I had strange errors when I didnt have the trailing remarks including failing calls ..

What do you mean by trailing remarks?

--ian

Hi

I just leave GSM in as my PDA supports it, but 90% of the calls are ilbc anyway.

The trailing bit is the "-- Minimize delay "

tos=lowdelay -- Minimize delay

Now whether its still causing errors I dont know ,and Im not going to see if it does either .

Ian

Ahh, I see now

--ian

ianplain :

Hi The trailing bit is the "-- Minimize delay " tos=lowdelay -- Minimize delay Ian

I can't find a single example of a config file or documentation with that -- Minimize delay, I feel you are mistaken.

Here are some examples clipped from numerous iax.conf files found on the asterisk mailing list, and voip wiki.

from voip wiki: tos = [lowdelay|throughput|reliability|mincost|none]
tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;tos=lowdelay

I upated my sticky with some comments.

Hi.

The format used was what was spec'd in June last year as My iax.conf was from the suplied samples then.

here is the full section.

; Finally, you can set values for your TOS bits to help improve
; performance. Valid values are:
; tos=lowdelay -- Minimize delay
; tos=throughput -- Maximize throughput
; tos=reliability -- Maximize reliability
; tos=mincost -- Minimize cost
; tos=none -- No flags
;
tos=reliability -- Maximize reliability
;

No I agree having read the wiki it is now different , Thats opensource development for you, Maybe I will try it without the end bits, But in August when I was cleaning files I removed the -- Maximize reliability bit i had failing calls and strange errors.

I suppose this is the thing with having a reliable working system, I havent had any need to change the IAX.conf other than to add peers since I built the first system in June last year.

Ian

ianplain :

; Finally, you can set values for your TOS bits to help improve ; performance. Valid values are: ; tos=lowdelay -- Minimize delay ; tos=throughput -- Maximize throughput ; tos=reliability -- Maximize reliability ; tos=mincost -- Minimize cost ; tos=none -- No flags tos=reliability -- Maximize reliability Ian

The example files now look like this...
; Finally, you can set values for your TOS bits to help improve
; performance. Valid values are:
; lowdelay -- Minimize delay
; throughput -- Maximize throughput
; reliability -- Maximize reliability
; mincost -- Minimize cost
; none -- No flags
;
tos=lowdelay

They are more clear now in that the -- Minimize delay is just a description of that value, because they omit the tos=reliability which was most likely confusing people, because they would uncomment it, rather then add one of those values to the tos=

At first, I couldn't get this to work -- it kept rejecting the incoming connection. However, I have found the problem.

In all config files, and in the IAX2 string in destination1, you must set the number with the full 44844 international prefix. Putting the 44844 version only in extensions.conf breaks it, as does having 44844 in local configs and 0844 in the VoIP management page.

Hope that's helpful.

Hey guys,

Can I configure IAX Destination One as:

voipuser:secret@ciavox-pegasus.homedns.org/08700688205

and declare in iax.conf:

[voipuser]
type=user
context=inbound-voipuser
username=fgomes ; my username @ voipuser
secret=secret
nat=yes

... and expect voipuser is able to call my * box?

CLI> iax2 show users
seems to be ok.

I'm trying to call 44 870 0688205 from outside UK but without success. I didnt see an inbound call in * console.

Any ideas?

Thank you all !

Fernando Gomes
sip:200200@voip.ciavox.com

Hi

welcome to voipuser

Read the first post and set it up as that.

Ian

Hi all,

I'm trying to get 0844 pstn calls into my Asterisk server but am failing

If I set the destination to be my ekiga.net sip account it works fine. I have another IAX link to my colleagues' asterisk server working just fine.

I have configured my web account as follows:

Code:

08444849XXX:verybigsecret@my.ip.addr.165(or)DomainName

My asterisk server's iax.conf has

Code:

[general] bindport=4569 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=ilbc mailboxdetail=yes jitterbuffer=yes tos=ef ;low-delay is deprecated in asterisk 1.4... [08444849XXX] ;username=voipusername ;with and without comment secret=verybigsecret type=user context=inbound-voipuser

I have tried the port tester and all I get is:

Code:

UDP Port Tester Linux V1.0 by Dean (c) www.voipuser.org Testing Port : 4569      [FAILED] Hit enter to exit...

I have turned my firewall OFF! for this test.

When I set a higher debug level on asterisk I do not see any incoming traffic when I call my 0844 PSTN number.

I am behind a "natting" router/dsl modem so for the other IAX trunk I register to the other Asterisk server. I can't see any notes here about doing this so I guess it is not necessary.

Can anyone suggest anything?

Thanks

Alan

Hi Alan

Welcome to voipuser.

Have you forwarded port 4569 to your server ?

Ian

ianplain :

Hi Alan Welcome to voipuser. Have you forwarded port 4569 to your server ? Ian

Thanks IAN - I hadn't, a good idea. I have now! But I still don't get any joy

I'm really puzzled by this:

Code:

UDP Port Tester Linux V1.0 by Dean (c) www.voipuser.org Testing Port : 4569      [FAILED] Hit enter to exit...

And I see nothing incoming on my asterisk server...[/quote]