search for: proxy_register

...ip.conf: [55] type=friend defaultuser=55 secret=12345655 context=home_castle callerid="Lukasz Cisco 7960" <55> canreinvite=no host=dynamic dtmfmode=rfc2833 qualify=200 mailbox=55 SIPDefault.cnf: image_version: P0S3-8-12-00 proxy1_address: 192.168.1.109 ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: 80 phone_password: cisco ; Limited to 31 characters (Default - cisco) sntp_mode: directedbroadcast ;"unicast" sntp_server: 192.168.1.77 time_zone: GMT+01/00 ; assuming you're in GMT time_format_24hr: 1 ; to show the time in 24hour format date_format: D/M/Y ; format...

...on Password # Emergency Proxy info proxy_emergency: "" proxy_emergency_port: "5060" # Backup Proxy info proxy_backup: "10.6.0.223" proxy_backup_port: "5060" # Outbound Proxy info outbound_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...

...om' failed for '64.169.xx.yyy' - Username/auth name mismatch What's the right way to do this. Shown below are my configuration files: </edg> -----SIPDefault and SIPMacAddr ----- image_version: P0S3-07-4-00 tftp_cfg_dir: "" ; Example: ./sip_phone/ proxy_register: 1 timer_register_expires: 3600 dial_template: dialplan messages_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....

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

Hello all, I've got a Cisco 7960 running version 7.4 firmware (heard there were problems with 7.5) and I can't get it to register with Asterisk. I've stripped down my configs on the phone to a bare minimum, and posted them below. Basically, the Cisco phone sends absolutely no packets to the proxy when it gets booted. If I make an outgoing call I see traffic getting to Asterisk, but

...n => 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.neo...

...t: "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" # TOS bits in media stream [0-5] (Default - 5) tos_media: "5" #...

I have several (various brand) sip devices with static IP's. I understand that asterisk will not accept a registration from these devices if the host= parameter is not set to 'dynamic' in sip.conf. I want calls to these extensions to be routable even before the device registers. I understand that is what defaultip= is supposed to do, but it doesn't work. I get a busy tone when

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

Hi, We've got a customer site with Cisco 7940G Phones connecting to our Asterisk Server via SIP. However, upon running a "sip show peers" it states that the phone is : user/user (Unspecified) D 255.255.255.255 0 Unmonitored All phones can call out via our Asterisk Server fine regardless. The problem is when they try and place a call to one another, where it

...rmediate SIP image and then in SIPDefault.cnf put the image you wish to ultimately upgrade to. Here's my current files: [me@here tftpboot]# more OS79XX.TXT P0S30202 [me@here tftpboot]# more SIPDefault.cnf image_version:P0S3-04-4-00 proxy1_address: xx.xx.xx.xx tftp_cfg_dir: /configs/cisco7900/ proxy_register : 1 Phone starts off at Skinny 2.x, then upgrades to SIP 2.2 then to SIP 4.4. 4.4 could be assumably replaced by 5.3.... HTH, --- Gavin

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

...dress to bind to (all addresses on machine) context = from-sip-external ; Send unknown SIP callers to this context ;context = from-budgetphone ; Send unknown SIP callers to this context disallow=all allow=ulaw allow=alaw allow=g723.1 allow=g726 allow=g729 callerid = Unknown srvlookup=yes proxy_register = 1 dtmfmode=inband register => 31437110323:mypassword@budgetphone.nl/31437110323 #include sip_nat.conf #include sip_additional.conf [31437110323] type=friend context=from-sip-t2y ;context=from-talkin2ya host=budgetphone.nl fromuser=31437110323 fromdomain=budgetphone.nl...

...Password # Emergency Proxy info proxy_emergency: "" proxy_emergency_port: "5060" # Backup Proxy info proxy_backup: "192.168.2.1" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "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...

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

...rt: "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" # TOS bits in media stream [0-5] (Default - 5) tos_media: "5" # En...

...assword: "lounge" ; Line 1 Registration Password ------------------------- sipdefault.cnf # Image Version image_version: P0S3-06-3-00 # Proxy Server proxy1_address: ipaddress of A30P ; Can be dotted IP or FQDN proxy1_port: 5060 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 0 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (...

...an be dotted IP or FQDN proxy6_address: "192.168.17.13" ; Can be dotted IP or FQDN # Proxy Server Port (default - 5060) proxy1_port: 5060 proxy2_port: 5060 proxy3_port: 5060 proxy4_port: 5060 proxy5_port: 5060 proxy6_port: 5060 # 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, g729a) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)...

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have

Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47