search for: nat_received_processing

Hi List, I've got a 7960 that's behind NAT (nat_enabled: 1 and nat_received_processing: 1) and for whatever reason doesn't seem to register, or at least hold a registration. If both the phone and the router (netgear) are rebooted, the phone will register, take a few incoming/outgoing calls no problems, then a few hours later, it drops the registration and never re-registers. If t...

...proxy: "10.6.0.223" outbound_proxy_port: "5060" proxy_register: 1 timer_register_expires : 120 # NAT/Firewall Traversal nat_enable: "1" nat_address: "" voip_control_port: "5060" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "1" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "Garrett - " ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: PST sntp_server: "136.159.2.254" ; SNTP Server IP Address sntp_m...

...e and has a public IP address. I've set up port-forwarding on the firewall for both phones to tunnel the SIP messages initiated by the Asterisk box. It works like a charm with the Cisco phone by using the following config info: voip_control_port: 5077 nat_enable: 1 nat_address: "" nat_received_processing: 0 Every time the Cisco phone registers with Asterisk, it does so using port 5077 and with the corresponding port-forwarding rule added to the firewall, it works great. However, for the life of me, I can't get the Aastra to do the same thing. I thought that should be able to do it by using &q...

...sages_uri: 3688 telnet_level: 2 phone_label: "Nat One" line1_name: 3115552368 line1_shortname: 3115552368 line1_authname: 3115552368 line1_password: 3115552368 line1_displayname: "Nat One" logo_url: "http://192.168.1.45/pts.bmp" nat_address: 64.169.xx.xxx nat_enable: 1 nat_received_processing: 1 proxy1_address: myserver.outthere.com My NatTwo phone is similar. The only difference is the name/password using 8115552368. ----sip.conf------------- [3115552368] type=friend host=dynamic username=3115552368 secret=3115552368 nat=1 context=wholesale9 disallow=all allow=ulaw accountcode=testi...

...ake the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] Thank you. -- James (Please CC me on all replies)

..._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 N...

Hi, I'm experimenting with the dev kit lite and now past the USB unpleasantness it's working great with standard phones and lines. The priority right now is getting soft phones (under Windows XP) working well. So far, I've only been able to get the XTEN Lite phone working and I really don't understand how I set it up. I used "xten" for every option everywhere (display

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

...port: "5060" # Outbound Proxy info outbound_proxy: "" outbound_proxy_port: "5060" # NAT/Firewall Traversal nat_enable: "0" nat_address: "" voip_control_port: "5061" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "0" # Proxy Registration (0-disable (default), 1-enable) proxy_register: "1" # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: "3600" # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: "none...

I have a really simple question about a fairly complex problem: I have a Cisco 7960 behind a NAT. I have an Asterisk server behind a different NAT. I have a SER server (with rtpproxy installed) on a public IP adress. I've opened ports with static NAT to * and the Cisco. Without using SER, I can register the phone to *, I can complete calls, I just can't move audio. Reading the

...nother location. This one is on a private LAN, and uses NAT to get out on to the Internet. I have been successful in registering the 7960 to the local * server. There's no NAT here, so it's easy. I have registered the phone to the remote * server (using nat=yes in *, and nat_enable plus nat_received_processing on the 7960). This works fine too. BUT, I want a line button on the local * box, plus a line button on the remote * box. This works too, for a while. After a short while, usually once I've completed a call to/from the remote * box, the phone starts dishing out its public address to the local...

We have a strange issue here - we have the following setup: Asterisk CVS-HEAD-12/15/04-07:42:16 40 SIP Cisco 7940 phones, linking to PSTN via EuroiSDN 30 channels. Often, when someone tries to dial any internal extension or external number, they get the "Reorder" message. If they try again, they get another "Reorder" message. If they try a third time, the call gets

...quot;5060" # Outbound Proxy info outbound_proxy: "" outbound_proxy_port: "" # NAT/Firewall Traversal nat_enable: "1" nat_address: "64.123.190.68" voip_control_port: "5060" start_media_port: "16000" end_media_port: "32768" nat_received_processing: "1" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "Home " ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: CST # Telnet Level (enable or disable the ability to telnet into this phone telnet_level: &quo...

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

...roxy: "192.168.2.1" outbound_proxy_port: "5060" proxy_register: 1 timer_register_expires : 120 # NAT/Firewall Traversal nat_enable: "1" nat_address: "" voip_control_port: "5060" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "1" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "Cisco - " ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: EST sntp_server: "136.159.2.254" ; SNTP Server IP Address sntp_mode...

...5060" # Outbound Proxy info outbound_proxy: "192.168.1.17" outbound_proxy_port: "5060" # NAT/Firewall Traversal nat_enable: "0" nat_address: "" voip_control_port: "5061" start_media_port: "16384" end_media_port: "32766" nat_received_processing: "0" # Proxy Registration (0-disable (default), 1-enable) proxy_register: "1" # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: "120" # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: "none&q...

Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing "55 <phone icon with x>" so it looks like the phone is not registered. The phone and the asterisk are in the same local

...at_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled outbound_proxy: "" ; restricted to dotted IP or DNS A record only outbound_proxy_port: 5060 ; default is 5060 # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default) # Allow...

...; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP range for media (default - 16384) end_media_port: 32766 ; End RTP range for media (default - 32766) nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled # Outbound Proxy Support outbound_proxy: "" ; restricted to dotted IP or DNS A record only outbound_proxy_port: 5060 ; default is 5060 ####### New Parameter added in Release 3.0 ####### # Allow for the bridge on a 3way call to join...