no TX audio == no fun

Submitted by findlay on Fri, 2006-10-13 09:14.

I have a softphone, kiax, which connects to asterisk via IAX which connects to my provider, arrivaltel.com, also via IAX. I have somehow managed to get to the point that I can call out and talk with people in the manner than one expects from an ordinary phone except that some numbers that I call there is exactly no audio heard at the other end. What did I do wrong? Why have the gods chosen to punish me in this manner? I can post my configs if that is helpful. Hopefully when I have some more time today `iax2 debug` will be informative, otherwise I am at the mercy of the relative reticence of the gurus to dispensate as to aid this matter with condescensions of their most august knowledge.

# cat /etc/asterisk/iax.conf | obfuscate_stuff
[general]
bandwidth=low
disallow=all
allow=ulaw
tos=lowdelay
autokill=yes
echocancel=yes
echotraining=yes
jitterbuffer=yes
forcejitterbuffer=yes

[kiax]
type=friend
context=internal
host=dynamic
username=uname
secret=secret

[arrivaltel-out]
type=peer
host=s1.arrivaltel.com
username=me
secret=secret
callerid=me
txgain=1.0

[arrivaltel-in]
type=user
context=incoming

# cat /etc/asterisk/extensions.conf
[incoming]
exten => 1,1,Dial(IAX2/kiax,30)
exten => 1,2,Hangup

[internal]
exten => _1NXXXXXX,1,Dial(IAX2/s1.arrivaltel.com/${EXTEN},60)
exten => _1NXXXXXX,2,Hangup

exten => _1NXXNXXXXXX,1,Dial(IAX2/s1.arrivaltel.com/${EXTEN},60)
exten => _1NXXNXXXXXX,2,Hangup

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hope this helps

#69 On Fri, 2006 10 13 15:18 admin said,

not all the gods are out to punish you for your hubris in attempting to install your own PBX. Prometheus is on your side but a buzzard keeps pecking out his liver and he is chained to a rock.

for inbound to work you will want to change [arrivaltel-in] to your username.

you will want to get rid of the txgain line (this is just for zaptel -- correct me if i am wrong)

what is your ping to s1 like? i would not force the jitterbuffer.

i would get rid of the autokill, echocancel and echotraining lines too. the last two i am pretty sure are not going to be used by IAX.

Your dialstrings should be

dial(IAX2/arrivaltel-out/${EXTEN},60)

not sure if any of these will fix your specific issues. surprised you can dial out at all -- impressive! maybe it is sheer will or even telekinesis. perhaps the gods are with you after all!

also, which version of asterisk are you using? release 1.2.11 has some regressions. 1.2.12.1 is stable and works well.

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