I can't make calls from SIP to DAHDI [FIXED]

Asterisk Asterisk Support
1

I can make calls between extensions DAHDI and DAHDI for SIP, however, SIP for DAHDI I can not.
Does anyone have any idea?
Thank you so much.

2

[quote=“lcvleo”]I can make calls between extensions DAHDI and DAHDI for SIP, however, SIP for DAHDI I can not.
Does anyone have any idea?
Thank you so much.[/quote]

I’m afraid you will need to give us more info.
And re-formulate your question please because I don’t really understand it.

3

Firstly, thank you for your attention!
Well, In my Asterisk box there is a Digium Wildcard AEX410P (2 FXS and 2 FXO).
In one FXO port is connected to PSTN. The other port is disconnected. And in the other two ports FXS analog I have two devices connected.
There are two SIP accounts, an IP phone connected to one and another connected to X-Lite.
When I make a call from a phone connected to an FXS port to another device connected to another port FXS, everything ok. The same happens when I make another call to a SIP to another SIP account, all ok. The problem is when I try to make a SIP call from one account to one of the devices connected to one of the FXS ports. The message I get is that the person is not available, go straight to voicemail. When I make a call from a device connected to one of the FXS ports for SIP account, everything is ok. The only problem is the calls from SIP to DAHDI.

4

How does your extensions.conf look like ?
Can you post the part that handles the call to the device that is connected to the FXS-port ?

Also, how does the configuration of your FXS-port look like in /etc/asterisk/chan_dahdi.conf ?

5

Again, thank you for your attention. :smiley:
I am new to the Asterisk world. Today I just use packages of AsteriskNOW and the FreePBX.
There is a way for me to send you these two files? They are rather large.
chan_dahdi.conf is bellow:

[channels]
;General options
#include chan_dahdi_additional.conf
#include dahdi-channels.conf

usecallerid = yes
hidecallerid = no
callwaiting = yes
threewaycalling = yes
transfer = yes
echocancel = yes
language = pt-BR
usedistinctiveringdetection = yes
callwaitingcallerid = yes
echocancelwhenbridged = yes
rxgain = 0.0
txgain = 0.0
callerid = asreceived
;cidsignalling = dtmf <---- not working
;cidstart = polarity <---- not working
useincomingcalleridondahditransfer = yes
busydetect = yes
busycount = 3
progzone = br
callprogress = yes
tonezone = 20

;FXS Modules
group = 1
signalling = fxo_ls
context = from-internal
channel = 3-4

;FXO Modules
group = 2
echocancel = yes
signalling = fxs_ks
context = from-zaptel
channel = 1-2

6

Is it possible to post your extensions.conf ? Not the whole file (if it is large), just the part where your created an extension to call your phones connected to the FXS-module.

With the following line :exten => 100,1,Dial(DAHDI/3,60)You should be able to call the phone connected to FXS-module number 3, as defined in your chan_dahdi.conf :;FXS Modules group = 1 signalling = fxo_ls context = from-internal channel = 3-4 if you want this phone to have the extension 100.

Make sure you define this extension (100) in the context where your SIP-phone enters the dialplan.

If you have defined a SIP-client with context [SIP-phones], then you need to put [SIP-phones] exten => 100,1,Dial(DAHDI/3,60) in this context in order to be able to reach the phone from your SIP-client.

You could define your extension 100 in another context, but then you need to include this context :[code][FXO-phones]
exten => 100,1,Dial(DAHDI/3,60) ; FXS-phone 1
exten => 200,1,Dial(DAHDI/4,60) ; FXS-phone 2

[SIP-phones]
include => FXO-phones
exten => 50,1,Dial(SIP/50,60) ; SIP-phone 1
exten => 60,1,Dial(SIP/60,60) ; SIP-phone 2[/code](if you have [50] and [60] defined in sip.conf)

7

Still not working.
This is mi first DAHDI extension:

;;;;;;[7000]
signalling=fxo_ks
pickupgroup=
mailbox=7000@device
immediate=no
echotraining=800
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=device <7000>
busydetect=no
busycount=7
accountcode=
channel=>3

This is my first SIP Extension:

[6000]
deny=0.0.0.0/0.0.0.0
type=friend
secret=xxxx
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=6000@default
host=dynamic
dtmfmode=rfc2833
dial=SIP/6000
context=from-internal
canreinvite=no
callgroup=
callerid=device <6000>
accountcode=
call-limit=50

This is the context from-internal:

[from-internal]
include => from-internal-xfer
include => bad-number
include => FXS-phones

[FXS-phones]
exten => 7000,1,Dial(DAHDI/3,60)
exten => 8000,2,Dial(DAHDI/4,60)

8

You don’t have a step (“priority”) 1 for extension 8000.

9

Thank you my friends, but I managed to discover and solve the problem.
Thank you so much! :smiley: