Stephen Reese
2008-Oct-15 23:57 UTC
[asterisk-users] Cisco 7960 not always receiving incoming calls
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.
Stephen Reese
2008-Oct-18 00:03 UTC
[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.