I will try disallow=all, thanks, Nat is off. Sip.conf below.
If I call from my mobile to the isdn number, the Cisco phone rings, I pick up, they can hear me, I can not hear them, if I transfer it to the Sipura I get AUDIO!! ....
It is also happening over IAX with the Cisco phones.
I followed a lot of the examples on loligo.com, which were a great help, but this is so hard to troubleshoot as I cannot see any errors in debug, asterisk thinks a good call is in progress.
Anything internal is perfect. The CAPI works fine. Its just the audio from the other end.
Every now and then I can hear a quick bit of sound. One in 20 calls may work.
[general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind to allow=ulaw allow=alaw tos=lowdelay
[20] type=friend username=20 secret=20 canreinvite=no host=dynamic mailbox=20 callerid="Cisco Phone" <20> accountcode=20 qualify=yes context=sip
Thanks.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Cuthie Sent: 16 April 2004 18:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 no audio
Craig Waddington wrote:
When we receive or make a call to the outside - they can hear us, but we cant hear them.Make sure you don't have the Cisco phone set to do NAT.
It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem.
Current setup:
Isdn Eicon Diva card / Capi -> Asterisk � network.
I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck.
Anyone had this problem, and has a fix?
Thanks.
-brian _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Just to be clear, you need at least the following (or at least I did):
sip.conf:
nat=yes reinvite=no
SIPDefault.conf (in your tftp directory)
nat_enable="0"
-brian _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
