Hi Salvatore The firmware is PS03-08-2-00.
Unfortunately I can only packet capture on the asterisk server itself, but I am seeing: P 172.16.8.22 > 172.16.8.1: ICMP 172.16.8.22 udp port 2224 unreachable, length 36 IP 172.16.8.20 > 172.16.8.1: ICMP 172.16.8.20 udp port 17099 unreachable, length 36 IP 172.16.8.20 > 172.16.8.1: ICMP 172.16.8.20 udp port 17099 unreachable, length 36 IP 172.16.8.1 > 172.16.8.20: ICMP 172.16.8.1 udp port 17228 unreachable, length 208 Where x.1 is asterisk and x.20 is the cisco and x.22 is a polycom test phone. We also see these errors on our working network (asterisk 1.0.10) so they are possibly a red herring. I suspect RTP issues but am unsure how to proceed as the Cisco phones do not seem to allow rtp debugging via their console. For reference our rtp.conf is: [general] rtpstart=10000 rtpend=20000 rtpchecksums=yes (have tried with no and made no difference) Regards Simon Salvatore Giudice wrote: > You should get a packet capture of both cisco-cisco and > grandstream/polycom-cisco. Compare the SDP's. The cisco phone may not be > able to understand the other vendor's devices. BTW, what version of firmware > are you running on the cisco phones? > > -------------------------------------------------- > Salvatore Giudice > [EMAIL PROTECTED] > > VoIP Security Training, LLC > http://VoIPSecurityTraining.com > > 848 N. Rainbow Blvd. #1676 > Las Vegas, NV 89107 > Phone: (617) 959-7625 > Fax: (214) 279-2906 > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Simon Alman > Sent: Tuesday, May 01, 2007 11:27 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Cisco 7940 no outgoing audio > > Hi All > > We have a private network setup (no nat) with three types of phones > connecting to asterisk via SIP. We have Polycom, Grandstream and Cisco > 7940 IP phones. > > When we ring polycom to grandstream or grandstream to polycom then both > phones can send and receive voice fine and all is well. > > When we dial any combination of Cisco and either Polycom, or Granstream > the Cisco, no voice is being sent but the Cisco can receive voice from > the remote phone fine. > > When we dial Cisco to Cisco it all works fine. > > I am at a loss to figure this out and any help pointing me in the right > direction would be appreciated. We are running an old Asterisk server > with version 1.0.10 (yeah we know) and the same mix of hardware and > configs works fine. > > On the new (problem) setup we are running Asterisk 1.4.2 and our Cisco > firmware is 08-2-00. > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users