Hi, i am a little bit frustrated.
I am connecting to my newly installed asterisk with a softphone.
The extensions.conf is simple:
exten => 102,1,Answer()
exten => 102,2,Playback(beep)
exten => 102,3,Hangup()
I can connect to this rule, but the log hangs with
*CLI> – Executing Answer(“SIP/mathias-0ef3”, “”) in new stack
– Executing Playback(“SIP/mathias-0ef3”, “beep”) in new stack
– Playing ‘beep’ (language ‘en’)
There it stays until i hang up. The sip debug looks reasonable (for me), they
agree on A-Law or GSM, depending which codecs i am allowing.
I did test this with linphone, twinkle, kphone and PhonerLite, under Linux and Windows. I used the hints setting up kphone and linphone.
Snooping the network i see RTP traffic from the phone (host) to the server, with stagging sequence numbers, but only one RTP packet from the server to the phone, with a high sequence number. Etherial says that the payload of these packages fits to the codecs.
The network is a straight LAN, no router, FWs or NAT. Asterisk is a fresh install “1.2.1 built by mathias @ ngiri on a x86_64”. But the same situation also with a SuSE 10.0 Asterisk.
A phone to phone communication did work before,using Asterisk for dialing.
Any hints where to look further? I have got no clue