dreaded "user is busy" calling sip

AU Troll

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#1
I have an Elastix 1.3-2 server in place that is holding 5 SIP trunks from MyNetFone here in Australia.

I also have an Elastix 0.9 server at home with three VoIP trunks, two with MyNetFone.

Issue is - no one can call in on the VoIP lines onthe 1.3-2 server. Logs show the calls hitting the server but the calls are rejected (like sip anonymous is no) kind of thing. I have checked the mynetfone accounts by placing them on my own server and they work with no problems, incoming calls on the channels are picked up and routed appropriately.

I have compared the two servers for configuration differences and can't find any other than the way the 1.3-2 handles the sip.conf files. there i can see differences but can't find where the place the info i feel the server needs. I have placed the required info in the sip-general-custom.conf (see below for what i added).

Can anyone point me in the right direction so the 1.3-2 server will accept incoming sip calls?

Mark

sip-general-custom.conf

allow=g729
externip=xxx.xxx.xxx.xxx (marked out for security)
localnet=192.168.0.0/255.255.255.0
relaxdtmf=yes
bind=5060
bindaddr=0.0.0.0
 

wiseoldowl

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#2
Why are you putting this in sip-general-custom.conf? That's NOT the correct way to do things in Elastix. You need to create a trunk, and an inbound route. Use Google to search for "Elastix Without Tears", download a free copy and read it (especially the parts on Trunks and Incoming Routes) - then you will know the correct way to do this sort of thing.

Also this page on the FreePBX site about MyNetFone trunk configuration might help you.

If you are going to build custom dial plans to do the work of trunks then you are on your own!

Also, you REALLY should think about upgrading that 0.9 box.
 

AU Troll

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#3
I made the changes as per the website you referred to, removed all the other settings from the sip_general_custom.conf and still no joy. Still get the user is busy.

The reason I put those settings in the sip_general_custom.conf is those settings make my old version of Elastix at home work properly (externip, internet, etc.) and there was no other place to put them.

Before you ask, yes i do have all the appropriate ports forwarded from the router to the box, etc...and all out going calls over the VoIP lines do work. Only incoming is the problem.

Mark
 

wiseoldowl

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#4
Re:dreaded

AU Troll said:
I made the changes as per the website you referred to, removed all the other settings from the sip_general_custom.conf and still no joy. Still get the user is busy.

The reason I put those settings in the sip_general_custom.conf is those settings make my old version of Elastix at home work properly (externip, internet, etc.) and there was no other place to put them.
(Emphasis added) Really? No other place?? How about the place they are supposed to go - in a trunk? Sorry if I sound a bit irritable about this but I always feel as though if you are going to try and defy the way Elastix/FreePBX does things (on never look in any documentation to learn the correct way) then why you you expect others to bail you out when things stop working?

Just because you get some hack to work doesn't mean you've done it right. In FreePBX especially, newer versions can sometimes break your hacks. It's the risk you take when you try to do things your way, rather than the way FreePBX has designed for you to do it. I have no quarrel with people who really want to do their own thing, but then why are you using a FreePBX-based distribution at all?

Anyway, I'll make another suggestion. Try omitting the following:

User Details

User Context: 091XXXXX

canreinvite=no
context=from-trunk
fromuser=091xxxxx
insecure=very
qualify=no
secret=<password>
type=friend
username=091xxxxx

Blank all those out, but move the context=from-trunk to the PEER details - also in the PEER details you may (probably will) have to change the type= to type=peer. Then make sure that the number after the forward slash (/) in your registration string is in fact the DID assigned by your provider, and that you use that same number as the DID in your inbound route (you DID create an inbound route, right?)
 

jgutierrez

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#5
And I suppose that you have already installed on your server the appropiate g729 (depends on hardware) codec on your server?

When someone calls you, do they get a fastbusy tone, or do they get a message that the line is out of service?

is your sip trunk registered?
asterisk -rx "sip show peers"

paste the configuration for your sip trunk.
 

AU Troll

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#6
Yes, we have the g729 codecs installed and 10 licenses all of which does register in properly per the Asterisk log files and the fact that we do get the music on hold with no problems. further, the system is able to make outgoing calls via the sip trunks (that is the only way the company makes outgoing calls) and there has never been an issue with this.

a sip show registry gives the following:



Host Username Refresh State Reg.Time
sip.myfone.com.au:5060 09157662@sip 105 Registered Thu, 26 Feb 2009 16:10:18
sip.myfone.com.au:5060 09214603 105 Registered Thu, 26 Feb 2009 16:10:18
sip.myfone.com.au:5060 09214602 105 Registered Thu, 26 Feb 2009 16:10:18
sip.myfone.com.au:5060 09214601 105 Registered Thu, 26 Feb 2009 16:10:18
sip.myfone.com.au:5060 09214600 105 Registered Thu, 26 Feb 2009 16:10:18

sip show peers gives (less the extensions for brevity):



Name/username Host Dyn Nat ACL Port Status
Out 4/09214603 125.213.160.82 N 5060 OK (60 ms)
Out 3/09214602 125.213.160.82 N 5060 OK (75 ms)
Out 2/09214601 125.213.160.81 N 5060 OK (46 ms)
Out 1/09214600 125.213.160.81 N 5060 OK (79 ms)
moffs-sip/09157662 125.213.160.81 N 5060 OK (73 ms)

the sip configurations below are first from the non-working Elastix 1.3 and the second from my own working 0.9. You will note the information the working 0.9 sip file that I can not incorporate into the non-working 1.3. I did try to put this information into the sip_general_additional.conf but that did not fix this issue either.

Configs from Elastix 1.3 that is not accepting Sip calls

Sip.conf

; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui

[general]
;
; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes

; These will all be included in the [general] context
;
#include sip_general_additional.conf
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
#include sip_custom_post.conf

Sip_additional.conf

; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui


[09157662]
canreinvite=no
context=from-trunk
fromuser=09157662
insecure=port,invite
qualify=no
secret=xxxxxx
type=user
username=09157662

[09214600]
context=from-trunk

[09214601]
canreinvite=no
context=from-trunk
fromuser=09214601
insecure=port,invite
qualify=no
secret=xxxxxx
type=user
username=09214601

[09214602]
canreinvite=no
context=from-trunk
fromuser=09214602
insecure=port,invite
qualify=no
secret=xxxxxx
type=user
username=09214602

[09214603]
canreinvite=no
context=from-trunk
fromuser=09214603
insecure=port,invite
qualify=no
secret=xxxxxx
type=user
username=09214603



[moffs-sip]
disallow=all
allow=g729
allow=alaw
allow=ulaw
authname=09157662
canreinvite=no
dtmfmode=rfc2833
fromuser=09157662
host=sip.myfone.com.au
insecure=port,invite
nat=yes
pedantic=no
qualify=yes
secret=xxxxxx
type=peer
username=09157662

[Out 1]
disallow=all
allow=g729
allow=ulaw
allow=alaw
authname=09214600
canreinvite=no
dtmfmode=rfc2833
fromuser=09214600
host=sip.myfone.com.au
insecure=port,invite
nat=yes
pedantic=no
qualify=yes
secret=xxxxxx
type=peer
username=09214600

[Out 2]
allow=g729
disallow=all
canreinvite=no
authname=09214601
dtmfmode=rfc2833
fromuser=09214601
host=sip.myfone.com.au
insecure=port,invite
nat=yes
pedantic=no
qualify=yes
secret=xxxxxx
type=peer
username=09214601

[Out 3]
disallow=all
allow=g729
authname=09214602
canreinvite=no
dtmfmode=rfc2833
fromuser=09214602
host=sip10.mynetfone.com.au
insecure=port,invite
nat=yes
pedantic=no
qualify=yes
secret=xxxxxx
type=peer
username=09214602

[Out 4]
disallow=all
allow=g729
authname=09414603
canreinvite=no
dtmfmode=rfc2833
fromuser=09214603
host=sip10.mynetfone.com.au
insecure=port,invite
nat=yes
pedantic=no
qualify=yes
secret=xxxxxx
type=peer
username=09214603


sip_general_additional.conf

; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui


vmexten=*97
disallow=all
allow=ulaw
allow=alaw
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41

sip_registrations.conf

; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui


register=09214600:{password}@sip.myfone.com.au/09214600
register=09214601:{password}@sip.myfone.com.au/09214601
register=09214602:{password}@sip.myfone.com.au/09214602
register=09214603:{password}@sip.myfone.com.au/09214603
register=09157662@sip.myfone.com.au:{password}:09167662@sip.myfone.com.au/09157662

Same files but from working Elastix 0.9

sip.conf

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.


[general]
#include sip_general_additional.conf


bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-trunk ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68
relaxdtmf=yes
localnet=192.168.1.0/255.255.255.0
externhost=(my external yi org host name)
t38pt_udptl=yes



; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes


; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes


; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf

sip_additional.conf

; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui

[09150372]
username=0915xxxx
type=user
secret=xxxxxx
qualify=no
insecure=very
fromuser=0915xxxx
coontext=from-trunk
canreinvite=no

[0917xxxx]
username=0917xxxx
type=user
secret=xxxxxx
qualify=no
insecure=very
fromuser=0917xxxx
coontext=from-trunk
canreinvite=no

[200]
type=friend
secret=legend51
record_out=Always
record_in=Always
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=200@default
host=dynamic
dtmfmode=rfc2833
disallow=all
dial=SIP/200
context=from-internal
canreinvite=no
callgroup=
callerid=device <200>
allow=g729
allow=ulaw
allow=alaw
accountcode=
call-limit=50


[Exetel-in]
username=0242730172
type=user
secret=xxxxxx
qualify=no
insecure=very
fromuser=0242XXXXXX
coontext=from-trunk
canreinvite=no

[Exetel-out]
username=0242XXXXXX
type=peer
secret=xxxxxx
qualify=yes
pedantic=no
nat=yes
insecure=very
host=58.96.1.2
fromuser=0242XXXXXX
dtmfmode=rfc2833
canreinvite=no
authname=0242XXXXXX
allow=g729

[Myfone Line 1]
username=0915xxxx
type=peer
secret=xxxxxx
qualify=yes
pedantic=no
nat=yes
insecure=very
host=sip.myfone.com.au
fromuser=0915xxxx
dtmfmode=rfc2833
disallow=all
canreinvite=no
authname=0915xxxx
allow=g729
allow=ulaw
allow=alaw

[Myfone Line 2]
username=0917xxxx
type=peer
secret=xxxxxx
qualify=yes
pedantic=no
nat=yes
insecure=very
host=sip00.myfone.com.au
fromuser=0917xxxx
dtmfmode=rfc2833
disallow=all
canreinvite=no
authname=0917xxxx
allow=g729
allow=ulaw
allow=alaw

sip_registrations.conf

; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui

register=0915xxxx:{password}@sip.myfone.com.au/0915xxxx
register=0917xxxx:{password}@sip01.myfone.com.au/0917xxxx
register=0242XXXXXX:{password}@58.96.1.2/0242XXXXXX

Thanks for taking a look.

Mark
 

ramoncio

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#7
Try to debug the problem.
In the asterisk console:

core set debug 99
core set verbose 99

Then try to receive a call.
The console will be too fast to read, so watch whe asterisk logs at /var/log/asterisk/full

There you should be able to see what is wrong.


wiseodowl: you should calm down and control your anger.
This is not a good way to treat new visitors. If you get pissed of, just forget about the thread and keep going.
Your help has always been very valuable, but you must understand people can get offended by your comments.

Au Troll: you were the first shouting here and he just was trying to help you. You should read Elastix without tears, it can be very helpful. If you use freePBX the way of doing most things is at the web interface, manually editing files can cause more trouble if you don't know exactly what you are doing. Keep trying and post here your debugs, but also have a look at google to see if you find the solution there.
 

AU Troll

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#8
i did the debug, but after a lot of searching through all the log files, i finally found that the calls are being rejected because " no compatible codec".

I have the g729 codecs installed and the log shows them loading in. Calls out via the sip channels are no problem. Music on hold is no problem.

I have replicated all of this issue again here at home, my installing the 1.3 Elastix on a second system, and sure nuff the calls are rejected because of no compatible codecs. therefore something in the FreePBX stuff has severlly changed someplace. I again have placed the following inthe sip_general_custom.conf where FreePBX says it should be placed:
allow=g729
externip=XXX.XXX.XXX.XXX
localnet=192.168.0.0/255.255.255.0

So this problem has now boiled down to getting the codecs through properly. A core show translation gives me:

Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)

g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 2 2 2 2 1 3 7 11 - 2 -
ulaw - 3 - 1 2 2 1 3 7 11 - 2 -
alaw - 3 1 - 2 2 1 3 7 11 - 2 -
g726aal2 - 3 2 2 - 2 1 3 7 11 - 1 -
adpcm - 3 2 2 2 - 1 3 7 11 - 2 -
slin - 2 1 1 1 1 - 2 6 10 - 1 -
lpc10 - 3 2 2 2 2 1 - 7 11 - 2 -
g729 - 4 3 3 3 3 2 4 - 12 - 3 -
speex - 3 2 2 2 2 1 3 7 - - 2 -
ilbc - - - - - - - - - - - - -
g726 - 3 2 2 1 2 1 3 7 11 - - -
g722 - - - - - - - - - - - - -

and the g729 is there but is not being brought into play on an incoming call....

ideas???

Ramonco - send me a private email please...

Mark
 

rafael

Joined
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#9
ramoncio said
wiseodowl: you should calm down and control your anger.
This is not a good way to treat new visitors. If you get pissed of, just forget about the thread and keep going.
Your help has always been very valuable, but you must understand people can get offended by your comments.

Au Troll: you were the first shouting here and he just was trying to help you. You should read Elastix without tears, it can be very helpful. If you use freePBX the way of doing most things is at the web interface, manually editing files can cause more trouble if you don't know exactly what you are doing. Keep trying and post here your debugs, but also have a look at google to see if you find the solution there.
Thank you ramoncio you have all the reason. There is no reason to take things personally in this forums. They are made to help each other to have successful implementations of Elastix. For this reason I deleted two posts that where not helpfull to anyone.

So please be focus on the subject and friendly with each other.

Best regards,

Rafael
 

dicko

Joined
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#10
Quote:
from your posts you have variously
.
.

[Out 4]
disallow=all
allow=g729
.
.


sip_general_additional.conf

; do not edit this file, this is an auto-generated file by freepbx
; all modifications must be done from the web gui


vmexten=*97
disallow=all
allow=ulaw
allow=alaw
.
.


if the context from which the call is placed only allows some g729 then the carrier must support it and you must have a "working" codec, if it is licensable (and the digium one is) and you don't have that license, then . . ..

I suggest while trouble shooting you allow g711 until your are sure that all parties can use the g729 you are restricting the calls to

good luck

(p.s. he codec order is significant, if you have disallow=all after aany allowed codecs, you have in fact redisallowed them)
 

ramoncio

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#11
Dicko is fully right.
I also had the same problem, and dicko also helped me.
In older asterisk versions you could have

allow=g729
disallow=all

But this doesn't work in latest versions. You must first disallow and then allow.

disallow=all
allow=g729
 

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