I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date. I am also using a Cisco router at the gateway. Is there anything special I should to to make this work? Note my soft phone does not have any issues using the same dialing rules and extension information. Here is some of my config stuff: ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116 5060 Unmonitored 101/101 68.156.63.118 D N 1038 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline] Inbound call in progress when the SIP Cisco phone doesn't ring........ Verbosity is at least 5 == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Goto("SIP/rsreese-082a8358", "default,101,1") in new stack -- Goto (default,101,1) -- Executing [EMAIL PROTECTED]:1] Dial("SIP/rsreese-082a8358", "SIP/101&SIP/[EMAIL PROTECTED],30") in new stack == Using SIP RTP CoS mark 5 -- Called 101 == Using SIP RTP CoS mark 5 -- Called [EMAIL PROTECTED] -- SIP/vitel-outbound-08270130 is making progress passing it to SIP/rsreese-082a8358 -- SIP/vitel-outbound-08270130 is ringing == Spawn extension (default, 101, 1) exited non-zero on 'SIP/rsreese-082a8358' Inbound call in progress when the SIP Cisco does ring after I first make an outbound call........ == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Goto("SIP/rsreese-082a8358", "default,101,1") in new stack -- Goto (default,101,1) -- Executing [EMAIL PROTECTED]:1] Dial("SIP/rsreese-082a8358", "SIP/101&SIP/[EMAIL PROTECTED],30") in new stack == Using SIP RTP CoS mark 5 -- Called 101 == Using SIP RTP CoS mark 5 -- Called [EMAIL PROTECTED] -- SIP/101-0825cab8 is ringing -- SIP/vitel-outbound-08270130 is making progress passing it to SIP/rsreese-082a8358 -- SIP/vitel-outbound-08270130 is ringing == Spawn extension (default, 101, 1) exited non-zero on 'SIP/rsreese-082a8358' Extensions.conf, which I don't think is relevent, I've changed it to just a simple dial the sip phone and it still fails. exten => 101,1,Dial(SIP/101&SIP/[EMAIL PROTECTED],30) exten => 101,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?lbl_default_1:) exten => 101,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?lbl_default_1:) exten => 101,n(lbl_default_0),Hangup() exten => 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30) exten => 101,n,Goto(lbl_default_0) Cisco phone stuff from a Cisco 7960: SIPDefault.cnf image_version: P0S3-08-9-00 proxy1_address: neocipher.net ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: "100" phone_password: "cisco" ; Limited to 31 characters (Default - cisco) sntp_server: 10.10.10.1 time_zone: EST dial_template: DIALPLAN nat_enable: 1 nat_address: 172.16.2.1 nat_received_processing: 1 outbound_proxy_port: 5060 outbond_proxy: ns1.neocipher.net SIP0112B9EAFF72.cnf image_version: P0S3-08-9-00 # Line 1 Setup line1_name: 101 line1_authname: 101 line1_shortname: "Line 101" line1_password: "test" line1_displayname: "Stephen Reese"; # Line 1 Display Name (Display name to use for SIP messaging) # Line 2 Setup #line2_name: "scott" #line2_authname: "scott" #line2_shortname: "201" #line2_password: "tiger" #line2_displayname: "Larry Ellison"; # Line 2 Display Name (Display name to use for SIP messaging) # Phone Label (Text desired to be displayed in upper right corner) phone_label: "Stephen Reese" ; Has no effect on SIP messaging # Phone Password (Password to be used for console or telnet login) phone_password: "goaway" ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none telnet_level: 2 Any ideas or help would be great, thanks. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users