I had the same problem recently, and it didn't have anything to do with NAT. Try looking at the port range in the rtp.conf file. Make sure it matches the port range configured on your 7960s.
- Darren -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter J VERNON Sent: Wednesday, March 30, 2005 10:54 PM To: [email protected] Subject: [Asterisk-Users] Cisco 7960 and Asterisk,I think I have a curly one here Guys...... I have Asterisk CVS-NHEAD-03/19/05-21:56:28 running on a box here and have a couple of Cisco 7960s and a Grandstream phone. I can make calls from the 7960. When I get a call placed to the 7960 the call is setup but there is no audio in either direction. This is for a call placed on the local subnet between extensions so I doubt that it is a NAT issue though I have tried a number of combinations of this to no avail. I have tried firmware versions 6 & 7 on the Cisco phones, same result. I have tried the phones on two other Asterisk installs and they work fine. I have compared sip.conf with these and can see no differences. If I configure the grandstream up to replace the 7960 it works fine. I have noticed that the src port (TCP port on the phone) increments during the session which seems to be the issue. Anyone seen this before? Any assistance would be appreciated. Regards Peter Here are some of the settings. Sip.conf for the extension: [9001] type=friend ; either "friend" (peer+user), "peer" or "user" context=extensions secret=9001 fromuser=Cisco ; overrides the callerid, e.g. required by FWD callerid=9001 host=dynamic ; we have a static but private IP address nat=never ; there is not NAT between phone and Asterisk dtmfmode=rfc2833 canreinvite=no ; allow RTP voice traffic to bypass Asterisk incominglimit=5 ; permit only 1 outgoing call at a time outgoinglimit=5 progressinband=yes disallow=all allow=ulaw mailbox=9001 Phone config: Loadid: SW: P0S3-05-3-00 ARM: PAS3ARM1 Boot: PC030300 DSP: PS03AS30 SIP Phone> show config ------ Current *FLASH* Configuration ------ Platform : Cisco IP Phone 7960 Elasped Time: 15:15:24 dhcp_server : 192.168.10.254 my_ip_addr : 192.168.10.17 subnet_mask : 255.255.255.0 defaultgw : 192.168.10.254 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 203.194.26.236 dns_backup_1: 203.194.26.235 tftp_addr : 192.168.11.2 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0003:e348:e5cb domain_name : home.jbo.com.au my_name : SIP0003E348E5CB Status Flags : 12310000 image_version : "P0S3-07-3-00" FirmLoadID : "PC030300" network_media_type : Auto network_port2_type : Hub/Switch tos_media : 5 phone_label : "UNPROVISIONED" tftp_cfg_dir : "" phone_password : ********** phone_prompt : "SIP Phone" language : english sntp_mode : DirectedBroadcast sntp_server : 0.0.0.0 time_zone : EAST dst_offset : 01/00 dst_start_month : April dst_start_day : 0 dst_start_day_of_week : Sunday dst_start_week_of_month : 1 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 0 dst_stop_time : 02/00 dst_auto_adjust : 1 time_format_24hr : 1 date_format : D/M/Y nat_enable : 0 nat_address : UNPROVISIONED voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : "1" xml_card_dir : "" xml_card_file : "CARD.XML" telnet_level : 2 services_url : "http://192.168.11.2/cgi-bin/rss2cisco.pl"; directory_url : "http://192.168.11.2/directory.html"; logo_url : "http://192.168.11.2/asterisk-tux.bmp"; http_proxy_addr : UNPROVISIONED http_proxy_port : 80 enable_vad : 0 dial_template : "" callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 1 messages_uri : "199" dnd_control : 0 preferred_codec : g711ulaw dtmf_outofband : avt dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 1 line1_name : "9001" line2_name : "UNPROVISIONED" line3_name : "UNPROVISIONED" line4_name : "UNPROVISIONED" line5_name : "UNPROVISIONED" line6_name : "UNPROVISIONED" line1_authname : "9001" line2_authname : "UNPROVISIONED" line3_authname : "UNPROVISIONED" line4_authname : "UNPROVISIONED" line5_authname : "UNPROVISIONED" line6_authname : "UNPROVISIONED" line1_password : ********** line2_password : ********** line3_password : ********** line4_password : ********** line5_password : ********** line6_password : ********** line1_shortname : "9001" line2_shortname : "UNPROVISIONED" line3_shortname : "UNPROVISIONED" line4_shortname : "UNPROVISIONED" line5_shortname : "UNPROVISIONED" line6_shortname : "UNPROVISIONED" line1_displayname : "9001" line2_displayname : "UNPROVISIONED" line3_displayname : "UNPROVISIONED" line4_displayname : "UNPROVISIONED" line5_displayname : "UNPROVISIONED" line6_displayname : "UNPROVISIONED" proxy1_address : "192.168.10.106" proxy2_address : "UNPROVISIONED" proxy3_address : "UNPROVISIONED" proxy4_address : "UNPROVISIONED" proxy5_address : "UNPROVISIONED" proxy6_address : "UNPROVISIONED" proxy1_port : 5060 proxy2_port : 5060 proxy3_port : 5060 proxy4_port : 5060 proxy5_port : 5060 proxy6_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : "UNPROVISIONED" proxy_emergency : "UNPROVISIONED" proxy_backup_port : 5060 proxy_emergency_port : 5060 outbound_proxy : 192.168.10.106 outbound_proxy_port : 5060 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : none cnf_join_enable : 1 remote_party_id : 0 semi_attended_transfer : 1 call_hold_ringback : 0 SIP Phone> _______________________________________________ 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
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