Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
Very cool, I believe that did the trick. Thank you for your time. On Sat, Oct 18, 2008 at 7:42 PM, Darryl Dunkin <[EMAIL PROTECTED]> wrote: > Oh, you are using ip inspect as well. > > I have this setup on a few routers when using the FW feature set: > ip inspect udp idle-time 900 > > -Original Message- > From: Stephen Reese [mailto:[EMAIL PROTECTED] > Sent: Saturday, October 18, 2008 14:41 > To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl > Dunkin > Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming > calls > > I tried increasing the value and even set it to never and added the > qualify line but that did not help. Do I need to poke any holes in the > firewall on the nat device for the udp traffic to stay persistent? I > have included my routers configuration in case someone notices > something I may need to make the connection work correctly. Also when > I call the phone within the "OK" reachable time after the call > disconnects the status immediately become "UNREACHABLE". > > ns1*CLI>sip show peers > Name/username HostDyn Nat ACL Port > Status > vitel-outbound/rsreese 64.2.142.22 5060 > Unmonitored > vitel-inbound/rsreese 64.2.142.1165060 > Unmonitored > 101/10168.156.63.118D N 1038 > UNREACHABLE > 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 > offline] > > > [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 > handle_response_peerpoke: Peer '101' is now Reachable. (217ms / > 2000ms) > > ns1*CLI> sip show peers > Name/username HostDyn Nat ACL Port Status > vitel-outbound/rsreese 64.2.142.22 5060 > Unmonitored > vitel-inbound/rsreese 64.2.142.1165060 > Unmonitored > 101/10168.156.63.118D N 1038 OK (217 > ms) > 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 > offline] > > [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p > oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 > > CISCO CONF FOLLOWS: > > > ! > version 12.4 > service timestamps debug datetime msec > service timestamps log datetime > service password-encryption > ! > hostname 3725router > ! > boot-start-marker > boot system flash:/c3725-adventerprisek9-mz.124-21.bin > boot-end-marker > ! > logging buffered 8192 debugging > logging console informational > enable secret 5 > ! > aaa new-model > ! > ! > aaa authentication login default local > aaa authentication ppp default local > aaa authorization exec default local > aaa authorization network default local > ! > aaa session-id common > clock timezone EST -5 > clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 > network-clock-participate slot 1 > network-clock-participate slot 2 > no ip source-route > ! > ip traffic-export profile IDS-SNORT > interface FastEthernet0/0 > bidirectional > mac-address 000c.2989.f93a > ip cef > ! > ! > no ip dhcp use vrf connected > ip dhcp excluded-address 172.16.2.1 > ip dhcp excluded-address 172.16.3.1 > ! > ip dhcp pool VLAN2clients > network 172.16.2.0 255.255.255.0 > default-router 172.16.2.1 > dns-server 205.152.144.23 205.152.132.23 > option 66 ip 172.16.2.10 > option 150 ip 172.16.2.10 > ! > ip dhcp pool VLAN3clients > network 172.16.3.0 255.255.255.0 > default-router 172.16.3.1 > dns-server 205.152.144.23 205.152.132.23 > ! > ! > ip domain name neocipher.net > ip name-server 205.152.144.23 > ip name-server 205.152.132.23 > ip inspect name SDM_LOW cuseeme > ip inspect name SDM_LOW dns > ip inspect name SDM_LOW ftp > ip inspect name SDM_LOW h323 > ip inspect name SDM_LOW https > ip inspect name SDM_LOW icmp > ip inspect name SDM_LOW netshow > ip inspect name SDM_LOW rcmd > ip inspect name SDM_LOW realaudio > ip inspect name SDM_LOW rtsp > ip inspect name SDM_LOW sqlnet > ip inspect name SDM_LOW streamworks > ip inspect name SDM_LOW tftp > ip inspect name SDM_LOW tcp > ip inspect name SDM_LOW udp > ip inspect name SDM_LOW vdolive > ip inspect name SDM_LOW imap > ip inspect name SDM_LOW pop3 > ip inspect name SDM_LOW esmtp > ip auth-proxy max-nodata-conns 3 > ip admission max-nodata-conns 3 > ip ips sdf location flash://256MB.sdf > ip ips notify SDEE > ip ips name sdm_ips_rule > vpdn enable > ! > ! > ! > ! > ! > ! > ! > ! > ! > ! > ! > ! > ! > ! > ! > ! > ! > crypto pki trustpoint TP-self-signed-
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
Oh, you are using ip inspect as well. I have this setup on a few routers when using the FW feature set: ip inspect udp idle-time 900 -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Saturday, October 18, 2008 14:41 To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl Dunkin Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming calls I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the "OK" reachable time after the call disconnects the status immediately become "UNREACHABLE". ns1*CLI>sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI> sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki certificate chain TP-self-signed-995375956 certificate self-signed 01 quit username user privilege 15 secret 5 ! ! ip ssh authentication-retries 2 ! ! crypto isakmp policy 3 encr 3des authentication pre-share group 2 ! crypto isakmp policy 10 hash md5 authentication pre-share crypto isakmp key cisco address 10.0.0.2 no-xauth ! crypto isakmp client configuration group VPN-Users key dns 2 domain neocipher.net pool VPN_POOL acl 115 include-local-lan netmask 255.255.255.0 crypto isakmp profile IKE-PROFILE match identity group VPN-Users client authentication list default isakmp authorization list default client configuration address initiate client configuration address respond virtual-template 1 ! ! crypto
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
> 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)
>
> 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.
I tried increasing the value and even set it to never and added the
qualify line but that did not help. Do I need to poke any holes in the
firewall on the nat device for the udp traffic to stay persistent? I
have included my routers configuration in case someone notices
something I may need to make the connection work correctly. Also when
I call the phone within the "OK" reachable time after the call
disconnects the status immediately become "UNREACHABLE".
ns1*CLI>sip show peers
Name/username HostDyn Nat ACL Port
Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.1165060 Unmonitored
101/10168.156.63.118D N 1038 UNREACHABLE
3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline]
[Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231
handle_response_peerpoke: Peer '101' is now Reachable. (217ms /
2000ms)
ns1*CLI> sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.1165060 Unmonitored
101/10168.156.63.118D N 1038 OK (217 ms)
3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]
[Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p
oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134
CISCO CONF FOLLOWS:
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime
service password-encryption
!
hostname 3725router
!
boot-start-marker
boot system flash:/c3725-adventerprisek9-mz.124-21.bin
boot-end-marker
!
logging buffered 8192 debugging
logging console informational
enable secret 5
!
aaa new-model
!
!
aaa authentication login default local
aaa authentication ppp default local
aaa authorization exec default local
aaa authorization network default local
!
aaa session-id common
clock timezone EST -5
clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00
network-clock-participate slot 1
network-clock-participate slot 2
no ip source-route
!
ip traffic-export profile IDS-SNORT
interface FastEthernet0/0
bidirectional
mac-address 000c.2989.f93a
ip cef
!
!
no ip dhcp use vrf connected
ip dhcp excluded-address 172.16.2.1
ip dhcp excluded-address 172.16.3.1
!
ip dhcp pool VLAN2clients
network 172.16.2.0 255.255.255.0
default-router 172.16.2.1
dns-server 205.152.144.23 205.152.132.23
option 66 ip 172.16.2.10
option 150 ip 172.16.2.10
!
ip dhcp pool VLAN3clients
network 172.16.3.0 255.255.255.0
default-router 172.16.3.1
dns-server 205.152.144.23 205.152.132.23
!
!
ip domain name neocipher.net
ip name-server 205.152.144.23
ip name-server 205.152.132.23
ip inspect name SDM_LOW cuseeme
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp
ip inspect name SDM_LOW udp
ip inspect name SDM_LOW vdolive
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW esmtp
ip auth-proxy max-nodata-conns 3
ip admission max-nodata-conns 3
ip ips sdf location flash://256MB.sdf
ip ips notify SDEE
ip ips name sdm_ips_rule
vpdn enable
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
crypto pki trustpoint TP-self-signed-995375956
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-995375956
revocation-check none
rsakeypair TP-self-sig
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl
Dunkin
Sent: Friday, October 17, 2008 17:28
To: asterisk-users@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: [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 HostDyn Nat ACL Port Status
> vitel-outbound/rsreese 64.2.142.22 5060
Unmonitored
> vitel-inbound/rsreese 64.2.142.1165060
Unmonitored
> 101/10168.156.63.118D 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}" =
"C
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: [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 HostDyn Nat ACL Port Status
> vitel-outbound/rsreese 64.2.142.22 5060
Unmonitored
> vitel-inbound/rsreese 64.2.142.1165060
Unmonitored
> 101/10168.156.63.118D 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
> p
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 HostDyn Nat ACL Port Status
> vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
> vitel-inbound/rsreese 64.2.142.1165060 Unmonitored
> 101/10168.156.63.118D 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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[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 HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.1165060 Unmonitored
101/10168.156.63.118D 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.
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