[asterisk-users] Cisco 7960 not always receiving incoming calls

[Stephen Reese]

On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese <rsreese at gmail.com> wrote:
> 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 [9045622082 at inbound:1] Goto("SIP/rsreese-082a8358",
> "default,101,1") in new stack
>    -- Goto (default,101,1)
>    -- Executing [101 at default:1] Dial("SIP/rsreese-082a8358",
> "SIP/101&SIP/9046260700 at vitel-outbound,30") in new stack
>  == Using SIP RTP CoS mark 5
>    -- Called 101
>  == Using SIP RTP CoS mark 5
>    -- Called 9046260700 at vitel-outbound
>    -- 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 [9045622082 at inbound:1] Goto("SIP/rsreese-082a8358",
> "default,101,1") in new stack
>    -- Goto (default,101,1)
>    -- Executing [101 at default:1] Dial("SIP/rsreese-082a8358",
> "SIP/101&SIP/9046260700 at vitel-outbound,30") in new stack
>  == Using SIP RTP CoS mark 5
>    -- Called 101
>  == Using SIP RTP CoS mark 5
>    -- Called 9046260700 at vitel-outbound
>    -- 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/9046260700 at vitel-outbound,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/9046260700 at vitel-outbond,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.
>

I'm still unable to wrap my head around this problem. I can recieve a
call after I first call out from the line/phone. I didn't think it's a
NAT issue since it kind of works.