Hello Terry,
thank you for your answer.
2011/10/19 Terry Wilson twil...@digium.com
If I had to guess, I'd say that you don't have canreinvite/directmedia=no
in sip.conf and there is possibly a NAT between the phones and Asterisk.
When they have the same codec and directmedia is enabled, the
Hello,
I have a strange audio delay behaviour when placing a call between two SIP
devices using the same codec.
In my example, I have two devices forced to use GSM codec.
When placing a call, the first ~9sec have no audio, then the audio starts
trasmitting.
If I force one phone to use GSM and
DNID: 330
ADSICPE : 0
CALLTOKEN : Present
FW BLOCK DATA : 16 bytes
any hint?
bye,
stefano
--
Stefano Sasso
http://stefano.dscnet.org/
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Hello folks,
for a customer of us we are trying to make asterisk and windows RTC
library work together, but without success.
We *need* to use gsm codec, so in the peer section we have
disallow=all
allow=gsm
the sip signaling is ok, and when sniffing we got this session description:
INVITE from
);
} else if (!strcasecmp(v-name, notifymimetype)) {
--
Stefano Sasso
http://stefano.dscnet.org/
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New to Asterisk? Join us for a live
for this part, or for the global setup?
Thanks in advance for the help!
Stefano
PS: sorry for my weak english
PS2: sorry for cross posting but, if the general setup is more
asterisk related, the OpenSIPS part is, obviously, OpenSIP specific.
--
Stefano Sasso
http://stefano.dscnet.org
