Hi
I’ve just started to configure an Asterisk system for the first time. And I can get my iPhone make a call to a softphon on my pc via Asterisk on another Ubuntu system. Nice!
But theres no voice between the iPhone (runngin iSip) and the softphone (Zoiper).
I can call both of them (the iPhone or the softphone) from the CLI and can here me speaking on ubuntu. I’ve also a “Hello World” test, if I call it from on of the clients I can hear the “Hello World” message. But I cannot get the two clients to hear something from each other.
Any idea about what could be wrong?
Thanks for your help!
Ernst
This is my sip.conf:
[general]
port=5060
bindaddr=0.0.0.0
[2000]
type=friend
secret=1234
host=dynamic
[2001]
type=friend
secret=1234
host=dynamic
And my extensions.conf
[default]
; Die Extension 1001 wird als Funktionstest
; für alle Telefone benutzt.
;
exten => 1001,1,Answer() ; abgehoben
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup() ; aufgelegt
; Mit der Extension 2000 wird zum SIP-Telefon
; 2000 verbunden.
;
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
hi,
if the client rings, but not able to talk to each one, then it might be due to unmatched codec.
regards,
Thanks for the quick answer.
Sorry for the newbee question: How can I fix this?
Best Regards
Ernst
hi,
you might consider puting these in for each sip accounts in sip.conf
nat=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
context=
it tells asterisk that you are using codec g711 which in most cases supported by your sip clients.
Thanks for the tip. Unfortunately I’m not sure what I have to enter for context in
That is what I tried in sip.conf:
[general]
port=5060
bindaddr=0.0.0.0
[2000]
type=friend
secret=1234
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
context=<general>
[2001]
type=friend
secret=1234
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
context=<general>
But now I cannot make phone calls at all. CLI says:
[May 14 15:49:16] NOTICE[1915]: chan_sip.c:20039 handle_request_invite: Call from '2001' to extension '2000' rejected because extension not found.
I also tried
context=<2000>
and
context=<2001>
with the same result.
Any ideas?
Best Regards
Ernst
context would be the context in your dialplan (extensions.conf) that handles the incoming/outgoing calls. Perhaps you should (re)read the section on Basic Dialplans in the book, as this is pretty elementary stuff.
In your case since you only have the [default] context, you’d enter
under each extension in sip.conf.
Hurray! Internal basecal is working.
Thanks for all the support!
Ernst