Asterisk wont answer calls

General Help
1

Hello everyone. I’m very new to asterisk and i’m sure I overlooked something simple, but I have my FreePBX/Asterisk setup configured with an Openvox A400P, that should answer a call and send it to my IVR. However when I call the line it rings twice, then hangs up. I also cannot get an outside line from my SIP phones, but I can use all the internal features including voicemail. I am not quite sure where to begin looking and I don’t know what info you guys need to help me, but Ill be checking in frequently until I get this resolved. I appreciate all the help you can give me!

Thanks!

2

This is what the chan_dahdi.conf file looks like on my server…

; Copied from DAHDI Module of FreePBX

[general]

#include chan_dahdi_general.conf

[channels]

; include dahdi groups defined by DAHDI module of FreePBX
#include chan_dahdi_groups.conf

; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf

3

This afternoon I was checking some of my config files and I noticed the /tftpboot/ directory is not writable. I fixed that and my phones are now finding my server, which was another problem, but I didn’t have enough time into investigating. I also went into the Others page and opened up the FXO port config. The Context field was blank… Should there be something there? I added from-pstn to the context line and asterisk answered the call, but there was no audio. I am also still unable to call out from my SIP phones. I seem to remember somebody on this forum with a similar issue so I am going to do some digging. In the meantime, if you have any suggestions for me, please let me know. Thanks!

4

Can you post your DAHDI configuration files? Have a look in /etc/dahdi and also in /etc/asterisk for any chan_dahdi*.conf files.

5

This is what the dahdi-channels.conf file looks like on my server…

; Autogenerated by /usr/sbin/dahdi_genconf on Fri Nov 9 14:22:22 2012
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
;

; Span 1: WCTDM/4 “Wildcard TDM400P REV E/F Board 5” (MASTER)
;;; line="1 WCTDM/4/0 FXSKS (In use)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

6

I was doing some more playing, and I did some checking into my Outbound routes and my trunk setup. I had a few things wrong. I can now get an outside line from my SIP phones, however it will not dial. The only thing I ever get is a dial tone. Even when I dial over the dial tone. The server is still answering incoming calls, but still has no audio. If I run a simulated call from a SIP phone, everything works perfect, but when I call from outside the network there is silence. What really confuses me is if I dial an extension from the outside line after the server answers, it completes like it should - ringing the extensions as set up in the IVR. When the call goes to voice mail, you can hear the sound from the outside phone but it is very, very faint. I don’t know what to think now…

7

Can you also post the three files mentioned by “#included” statements in chan_dadhi.conf.

8

smoran22, seems like you have group defined twice… Do you have a dailplan setup?

; Span 1: WCTDM/4 “Wildcard TDM400P REV E/F Board 5” (MASTER)
;;; line="1 WCTDM/4/0 FXSKS (In use)"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 1
callerid=
group=
context=default

At the CLI what is the output of dahdi show channels?

9

Hello and thanks for the help, but I gave up on the DAHDI configs. I went to a SIP trunk through SIPSTATION. That has it’s own unique problems, but I have a lot of browsing yet to do on the forum. The entire dahdi config files changed and nothing that I had posted up above was accurate anymore, the card was not working… blah blah blah. Another case of you get what you pay for… (should have bought an actual digium card). Ill keep going on the SIP stuff, but for now thanks for all the help!