[asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!

[Eric "ManxPower" Wieling]

You have too many codecs allowed.

disallow=all and allow=ulaw in [general] and in each of the device 
sections of iax.conf.  If that works, then you can start from there and 
try to get the codec you really want.

BSumrall wrote:
> after 18 hours, over 200 pages of reading, a complete reinstall of asterisk
> I am down to this. 
> 
> extensions.conf 
> 
> [globals] 
> CONSOLE=Console/dsp 
> IAXINFO=guest 
> TRUNK=Zap/g2 
> TRUNKMSD=1 
> 
> [default] 
> exten => 8005181896,1,Dial,(IAX2/UXMC) 
> exten => s,1,Answer() 
> 
> (I tried) 
> exten => _1XXXXXXXXXX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr) 
> (as well) 
> 
> iax.conf 
> 
> [general] 
> port=4569 
> bandwidth=low 
> disallow=lpc10 
> jitterbuffer=no 
> forcejitterbuffer=no 
> tos=lowdelay 
> autokill=yes 
> 
> register => xxxx:xxxxxxxxxx at voip-co3.teliax.com 
> 
> [teliax] 
> context=default 
> type=friend 
> host=voip-co3.teliax.com 
> auth=md5 
> user=xxxx 
> secret=xxxxxxxxx 
> disallow=all 
> allow=ulaw 
> allow=alaw 
> allow=gsm 
> 
> sip.conf 
> 
> [UXMC] 
> user=xxxxxxx 
> context=internal 
> type=friend 
> qualify=yes 
> nat=no 
> secret=xxxxxxxx 
> canreinvite=no 
> host=dynamic 
> nat=no 
> 
> If I put back previous config, I can call into the 1800 number and here that
> silly chick heckle me from my server!
> 
> 
> 
> 
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