[HELP] x86_64 problem?


#1

Hi, i am a little bit frustrated.

I am connecting to my newly installed asterisk with a softphone.
The extensions.conf is simple:
[general]
static=yes
writeprotect=no
[localsip]
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 :frowning:

Thanks, Mathias


#2

Addon: i did a fresh reinstall of SuSE 10.0 Asterisk on a 32bit host and on a 64bit host.
With exactly the same config files. On 32bit it works, on 64bit it doesn’t :question:

On the 64 bit i also get

Can this affect the Playback function :question: Loading ALSA module removes message, but Asterisk doesn’t work :frowning:

Any ideas? Mathias

See also http://forums.digium.com/viewtopic.php?t=3715