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

[Stephen Reese]

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.