Hi,
I have following setup: Asterisk 1.4 (IP 10.1.1.250) connect to Asterisk
13 (IP 10.1.1.251) with PJSIP, this one connected to the provider also
with PJSIP. Both LAN Asteriks are also connected via IAX.
Everything is working fine except SIP call from 1.4 to external number:
there is no audio. SIP call to eg demo@Asterisk13 is OK. If I replace
the SIP link between 1.4 and 13 with the IAX trunk it's OK. Other way is
OK in full SIP.
I tried few parameters on both Asteriks, no luck. The RTP port range is
the same on both instance.
If someone had a clue on this, welcome ;) and thanks in advance.
Daniel
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