I'm a Cisco 7940 phone using SCCP. My setup is a private network with the * box acting as dhcp server and also tftp server. The phone loads and dials out fine. I can hear the other person, but there is no outgoing audio. I've read that this is an RTP problem and have tried making some changes in /etc/hosts to point to my * box IP but with no luck. When I do a tcpdump I see that the RTP packets are sent to 0.0.0.0. How do I get the phone to send to the * box? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050406/41cc4e00/attachment.htm
Check iptables with iptables -l You may be blocking the RTP streams going INTO your * box. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Bellows, Jared Sent: Wednesday, April 06, 2005 3:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 7940 Outgoing Audio I'm a Cisco 7940 phone using SCCP. My setup is a private network with the * box acting as dhcp server and also tftp server. The phone loads and dials out fine. I can hear the other person, but there is no outgoing audio. I've read that this is an RTP problem and have tried making some changes in /etc/hosts to point to my * box IP but with no luck. When I do a tcpdump I see that the RTP packets are sent to 0.0.0.0. How do I get the phone to send to the * box? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050406/a200019a/attachment.htm
Which skinny driver are you using? Also, what about incoming calls? Post your skinny.conf or sccp.conf as well as the config file for your 7940 from the TFTP server. It may help also if you include any other generic Cisco config files from your TFTP directory, for example an OS79XX.TXT. -Andy On Apr 6, 2005 3:27 PM, Bellows, Jared <jmbell@byu.edu> wrote:> > > > I'm a Cisco 7940 phone using SCCP. My setup is a private network with the * > box acting as dhcp server and also tftp server. The phone loads and dials > out fine. I can hear the other person, but there is no outgoing audio. > I've read that this is an RTP problem and have tried making some changes in > /etc/hosts to point to my * box IP but with no luck. When I do a tcpdump I > see that the RTP packets are sent to 0.0.0.0. How do I get the phone to > send to the * box? > > > > Thanks > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >
On Wed, Apr 06, 2005 at 02:27:56PM -0600, Bellows, Jared arranged a set of bits into the following:> I'm a Cisco 7940 phone using SCCP. My setup is a private network with theWow, first time I think we've ever had a *phone* post to the list ;-)> * box acting as dhcp server and also tftp server. The phone loads and > dials out fine. I can hear the other person, but there is no outgoing > audio. I've read that this is an RTP problem and have tried making some > changes in /etc/hosts to point to my * box IP but with no luck. When I do > a tcpdump I see that the RTP packets are sent to 0.0.0.0. How do I get > the phone to send to the * box?If you're using chan_sccp update it to the latest CVS HEAD, that fixes the RTP issues. Thanks, Julien chan_sccp project lead -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 232 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050407/70e2aa36/attachment.pgp