Hi All.

I’m running some tests with the latest Asterisk SVN-branch-12-r410493M
compiled with fresh github pjsip and srtp 1.4.2 on an i386 CentOS
machine (2.6.32-358.18.1.el6.i686). As a client I’m using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m.

I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call from the softphone, Asterisk answers with a 488 not acceptable here.

I’m probably missing something but I’m not able to find what and where. Is there someone able to point me to the right direction?
Below is my configuration. The sofpthone is registered as 1060.

Thanks in advance. Marco Signorini.

pjsip.conf:

[transport-tls]
type=transport protocol=tls bind=0.0.0.0
cert_file=/etc/asterisk/sslcert.pem method=tlsv1

[1060]
type=endpoint transport=transport-tls context=from-internal use_avpf=yes media_encryption=sdes disallow=all allow=alaw allow=ulaw aors60
auth60

[1060]
type=auth auth_type=userpass password60
username60

[1060]
type=aor max_contacts

[204]
….

http.conf:

enabled=yes bindaddr.10.5.49
bindport