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

[Darryl Dunkin]

Sorry, I missed the Cisco router bit.

As a last resort (if qualify doesn't help), you could enter this
(global) to increase the timeout on UDP translations:
ip nat translation udp-timeout 300 (or greater if you prefer)

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Darryl
Dunkin
Sent: Friday, October 17, 2008 17:28
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users at 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 <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.

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