Hi Mark, Yes we had exactly the same issue with a 7940. It is with the .cnf xml file. You need to make sure that the line name is populated; line1_name: "<<USERNAME>>"line1_authname: "<<USERNAME>>"line1_password: "supersecret" See an example file attached, hopefully will give you an idea. Thanks Jon
From: m...@darkorigins.com Date: Thu, 19 Jun 2014 11:03:25 +0100 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Cisco 79XX Series Phones and TLS integration with kamailio Hi Jon I’m working with a variety of routers here at the moment (testing / lab environment :-) ranging from Mikrotik, Draytek to Firebricks. So far; - Phone on test is a 7970 - Variety of routers here at the moment (testing / lab environment :-) ranging from Mikrotik, Draytek to Firebricks. All configured to NAT, no firewall filtering - SIP v9.3(1)SR4 firmware on the phone - Provisioning side ‘appears’ to be working. In that it requests the files, says ‘registering’ - Doesn’t appear to generate any SIP traffic during registration (tcpdump on router/kamailio server) Wondering if I’ve got something wrong in the .cnf xml … Cheers Mark On 19 Jun 2014, at 10:48, Jonathan Hunter <hunter...@hotmail.com> wrote:Hi Mark, Sure of course, they are some what painful to get working due to their asymmetric NAT behaviour. What handset models are you working with, and what firewall devices do you have on site, as I have them working with a Cisco ASA on the network edge. Then I can give you some more details on at least them trying to register to kamailio. Thanks Jon From: m...@darkorigins.com Date: Thu, 19 Jun 2014 10:37:42 +0100 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Cisco 79XX Series Phones and TLS integration with kamailio Hi Jon It sounds like you’re a few steps in front of me. I’m just starting to try and get a working set of config files etc for provisioning to the phones. Which at the moment is going through the motions of registering but not actually sending anything. Would you mind sharing your experiences so far? Thanks Mark On 19 Jun 2014, at 09:50, Jonathan Hunter <hunter...@hotmail.com> wrote:Hi All, As you guys might remember I have been doing alot of work with these legacy handsets recently, in particular the Cisco IP phones 7945G and 7965G. They work well now with kamailio, using UDP or TCP as the transport protocol. I am now looking to implement SIP over TLS with them, and wondered if anyone had completed the same, as it appears they were designed very much to support SIP over TLS and SRTP but with Cisco call manager and not other SIP devices. I am trying to understand if its possible to integrate with kamailio, as from Cisco documentation it appears a CTL file (client trust list) is required which seems to be generated by cisco software, and the handset needs it before it will try to initiate a SIP connection over TLS. I have implemented TLS /SRTP with Cisco SPA's, Bria etc on kamailio fine, so its more a question around the handset and if anyone has achieved this. Thanks Jon_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
# Image Version image_version: P0S3-8-12-00 # Proxy Server proxy1_address: "proxyIP" # Proxy Server Port (default - 5060) proxy1_port:"5060" # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 ####### New Parameters added in Release 2.0 ####### # All user_parameters have been removed # Phone Label (Text desired to be displayed in upper right corner) phone_label: "test1" # Outbound Proxy Support outbound_proxy: "proxyIP" outbound_proxy_port: 5060 ; default is 5060 # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "IPaddress" ; SNTP Server IP Address sntp_mode: unicast ; unicast, multicast, anycast, or directedbroadcast (default) # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 3600 # Codec for media stream (g711ulaw (default), g711alaw, g729a) preferred_codec: g729a # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 0 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt_always # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control) dnd_control: 0 ; Default 0 (Do Not Disturb feature is off) # Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous) # Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: 101 ; Default 101 # Sync value of the phone used for remote reset sync: 1 ; Default 1 ####### New Parameters added in Release 2.1 ####### # Backup Proxy Support #proxy_backup: "proxyIP" #proxy_backup: "proxyIP" # Emergency Proxy Support #proxy_emergency: "proxyIP" #proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable ####### New Parameters added in Release 2.2 ###### # NAT/Firewall Traversal nat_enable: 1 ; 0-Disabled (default), 1-Enabled #nat_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: 10000 ; Start RTP range for media (default - 16384) end_media_port: 20000 ; End RTP range for media (default - 32766) #nat_received_processing: 0 ; 0-Disabled (default), 1-Enabled ####### New Parameter added in Release 3.0 ####### # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable : 1 ; 0-Disabled, 1-Enabled (default) ####### New Parameters added in Release 3.1 ####### # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: 1 ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the ability to telnet into the phone) telnet_level: 2 ; 0-Disabled (default), 1-Enabled, 2-Privileged ####### New Parameters added in Release 4.0 ####### # XML URLs services_url: "http://xxx"; directory_url: "http://xxx"; //logo_url: "http://xxx.bmp"; logo_url: "http://xxx.bmp"; # HTTP Proxy Support http_proxy_addr: "" ; Address of HTTP Proxy server http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default) # Dynamic DNS/TFTP Support dyn_dns_addr_1: "" ; restricted to dotted IP dyn_dns_addr_2: "" ; restricted to dotted IP dyn_tftp_addr: "" ; restricted to dotted IP # Remote Party ID remote_party_id: 1 ; 0-Disabled (default), 1-Enabled ####### New Parameters added in Release 4.4 ####### # Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control) call_hold_ringback: 0 ; Default 0 (Call Hold Ringback feature is off) ####### New Parameters added in Release 6.0 ####### # Dialtone Stutter for MWI stutter_msg_waiting: 0 ; 0-Disabled (default), 1-Enabled # RTP Call Statistics (SIP BYE/200 OK message exchange) call_stats: 0 ; 0-Disabled (default), 1-Enabled ################################ # SIP Configuration Generic File # Line 2 appearance line2_name: "UNPROVISIONED" # Line 2 Registration Authentication line2_authname: "UNPROVISIONED" # Line 2 Registration Password line2_password: "UNPROVISIONED" ####### New Parameters added in Release 2.0 ####### # All user_parameters have been removed # Phone Label (Text desired to be displayed in upper right corner) phone_label: "test1" # SIP Configuration Generic File # Line 1 appearance line1_name: "2023-tenant" line1_authname: "2023-tenant" line1_shortname: "2023 Larry Davis line1_password: "supersecret" # Setting for Message speeddial to UOne box line1_displayname: "Larry Davis" messages_uri: "*71" mwi_status : "1" dial_template: "dialplan_test1_"
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