Jim Lynch
2006-Oct-02 06:46 UTC
[asterisk-users] Dial and connect to sip provider works, but no audio.
This is strange. I upgraded from an older Asterisk@home that was working to the latest Tribox. I also added a A204 board, but for some reason neither the Grandstream phone or a phone connected to the Linksys ATA has any audio either way via the Telasip connection. Audio works OK between the phones, so I'm pretty sure the extension configuration is OK.. Here's my sip configs. I added the [from-pstn] to this file because I didn't see it defined anywhere else. I realize it will go away when I change the extensions but it wasn't working so I thought I'd try it. I don't see much difference in configuration from when it worked and now, other than the missing [from-pstn] block. Thanks for any help. Jim. sip_additional.conf register=xxx.yyy@xxxxx.telasip.com [101] username=101 type=friend secret=xxx record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never mailbox=101@device host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=101 <101> [102] username=102 type=friend secret=xxx record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never mailbox=102@device host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=102 <102> [from-pstn] type=user qualify=yes insecure=very host=xxx.telasip.com [telasip] username=xxx type=friend secret=xxx.yyy qualify=yes insecure=very host=xxx.telasip.com fromuser=xxx fromdomain=xxx.telasip.com dtmfmode=rfc2833 context=from-pstn ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding "nat=1" to each peer definition to ; solve translation problems. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw ; If you need to answer unauthenticated calls, you should change this ; next line to 'from-trunk', rather than 'from-sip-external'. ; You'll know this is happening if when you call in you get a message ; saying "The number you have dialed is not in service. Please check the ; number and try again." context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown tos=0x68 ; #, in this configuration file, is NOT A COMMENT. This is exactly ; how it should be. #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf ~