It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing calls to come
in until it expires (default on many devices is 60 seconds). You may
also receive inbound calls when the phone reregisters regularly. Try
'qualify=yes' in your phones section in sip.conf to send keepalives
(option packets in this case) every two seconds to the phone to keep it
from going idle. You can see the state of the phone from the console
with a 'sip show peers', if unreachable, your NAT device has killed the
NAT forward.
Should look like one of these:
xxx/xxx x.x.x.x D N 5060 OK (46 ms)
xxx/xxx x.x.x.x D N 5060 UNREACHABLE
As another troubleshooting step, you can telnet to the phone and have it
reregister with Asterisk manually ("register line 1 1") to see if that
brings it back to life.
If qualify doesn't do it, see if you can increase UDP timeouts in your
firewall/NAT device.
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese <[EMAIL PROTECTED]>
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 [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.
>
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.
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