Can call to SIP but not from SIP phones?

Asterisk Asterisk Support
1

First of all I’m new to all of this but thanks for the hard work guys!

Now here is my setup. I have a computer running Debian with Zaptel 1.4.* and Asterisk 1.4.* installed. There is a TDM800 card installed with 4 FXS and 4 FXO ports. Now I basically followed the example configuration that is given in the TDM800 installation instructions. Like add this to the end of your zapata.conf/extensions.conf file…

From here I added two SIP phones with new voice mail and extensions. One is a Cisco ATA188 and the other is a UTStarCom F1000G phone. Now I call a SIP extension from any phone on the TDM800 card but if I pick up a SIP phone and try to call an extension on the TDM800 card I get a busy signal. The only things that works from the SIP is ext 1000 with gets me the “Congrats you successfully configured Asterisk dialog.” And extension 8500 which is set up to access voice mail. Otherwise busy signal.

So is there anything I could obviously be doing wrong?

I’ll be sure to post my config files once I’m at the computer tomorrow.

A.J.

2

I added these to the end of the default config files.

zapata.conf:
;;General options
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
;;FXS Moduels
Group=1
signalling=fxo_ks
context=Internal
channel=5-8
;;FXO Modules
Group=2
echocancel=yes
signalling=fxs_ks
context=Incoming
channel=1-4

sip.conf:
[3000]
type=friend
username=3000
secret=test
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=3000
nat=0

[6000]
type=friend
username=6000
secret=test
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=6000
nat=0

extensions.conf:
[Internal]
exten => 1000,1,Dial(zap/8,20,rt)
exten => 1000,2,Voicemail(u1000)
exten => 1000,102,Voicemail(b1000)

exten => 2000,1,Dial(zap/7,20,rt)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)

exten => 3000,1,Dial(sip/3000,45,o)
exten => 3000,2,Voicemail(u3000)
exten => 3000,102,Voicemail(b3000)

exten => 4000,1,Dial(zap/6,20,rt)
exten => 4000,2,Voicemail(u4000)
exten => 4000,102,Voicemail(b4000)

exten => 5000,1,Dial(zap/5,20,rt)
exten => 5000,2,Voicemail(u5000)
exten => 5000,102,Voicemail(b5000)

exten => 6000,1,Dial(sip/6000,45,o)
exten => 6000,2,Voicemail(u6000)
exten => 6000,102,Voicemail(b6000)

exten => 8500,1,VoiceMailMain
exten => 8501,1,MusicOnHold

exten => _9.,1,Dial(zap/g2/www${EXTEN:1})
exten => _9.,2,Congestion

[Incoming]
exten => s,1,Answer
exten => s,2,Dial(zap/g1,20,rt)
exten => s,3,Voicemail(u1000)
exten => s,103,Voicemail(b1000)

A.J.

3

Your sip.conf doesn’t specify which context those friends should use. (Asterisk will fall back to the name [default], but your [default] is not defined.) You can add

either to [general] section or to friends who need to use this context.

4

So my sip should become:

[general]
context=Internal
srvlookup=yes

[3000]
type=friend
username=3000
secret=test
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=3000
nat=0

[6000]
type=friend
username=6000
secret=test
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=6000
nat=0

I’ll give it a try on Monday. THANKS!

A.J.

5

I think your users are not pointing to internal context.

Thanks,
Suresh