Scott A. Henderson
2004-Oct-04 22:42 UTC
[Asterisk-Users] Cisco 7960G w/ SIP not working properly
I have Asterisk version 1.0-RC1 running on Debian Woody. I have 1 analog phone working, 2 inbound lines working, X-Lite is working. The problem that I am having is with Cisco 7960 with SIP version 7.2 software. I can make outbound calls and they work fine, I even get a notice that I have voice mail on the phone and it seems to register properly but I can seem to dial to the phone. Configuration information is: =====================================================================argon*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status marc/marc (Unspecified) D 255.255.255.255 0 Unmonitored jarad/jarad (Unspecified) D 255.255.255.255 0 Unmonitored johnsip/johnsip (Unspecified) D 255.255.255.255 0 Unmonitored kevinsip/kevins (Unspecified) D 255.255.255.255 0 Unmonitored scott/scott 192.168.17.114 D 255.255.255.255 5060 Unmonitored <------- This is the phone in question =======================================================================sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson =========================================================================extensions.conf ; Scott Henderson exten => 6101,1,Dial(SIP/scott/s,20,Ttr) exten => 6101,2,Dial(Zap/R1/3372860) exten => 6101,3,Voicemail(u6101) ==========================================================================sip debug info from the CLI when I dial from one asterisk extension to the 7960 SIP phone in question argon*CLI> Destroying call '06d39dda6546c5a0148ec12a59af7d0a@127.0.0.1' We're at 192.168.17.13 port 19478 Answering/Requesting with root capability 4 Answering with capability 0x2(GSM) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) 12 headers, 12 lines Reliably Transmitting: INVITE sip:s@192.168.17.114 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK4d228686 From: "asterisk" <sip:asterisk@192.168.17.13>;tag=as73fe6d09 To: <sip:s@192.168.17.114> Contact: <sip:asterisk@192.168.17.13> Call-ID: 74ee54ba09a3c0d31bc3d6853229181f@192.168.17.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 05 Oct 2004 05:37:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 263 v=0 o=root 2816 2816 IN IP4 192.168.17.13 s=session c=IN IP4 192.168.17.13 t=0 0 m=audio 19478 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.168.17.114:5060 argon*CLI> Sip read: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK4d228686 From: "asterisk" <sip:asterisk@192.168.17.13>;tag=as73fe6d09 To: <sip:s@192.168.17.114>;tag=00115c407fa3000948b81d9b-144916e7 Call-ID: 74ee54ba09a3c0d31bc3d6853229181f@192.168.17.13 Date: Tue, 05 Oct 2004 05:37:05 GMT CSeq: 102 INVITE Server: CSCO/7 Contact: <sip:scott@192.168.17.114:5060> Content-Length: 0 10 headers, 0 lines Transmitting: ACK sip:scott@192.168.17.114:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.17.13:5060;branch=z9hG4bK4d228686 From: "asterisk" <sip:asterisk@192.168.17.13>;tag=as73fe6d09 To: <sip:s@192.168.17.114>;tag=00115c407fa3000948b81d9b-144916e7 Contact: <sip:asterisk@192.168.17.13> Call-ID: 74ee54ba09a3c0d31bc3d6853229181f@192.168.17.13 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.17.114:5060 Destroying call '74ee54ba09a3c0d31bc3d6853229181f@192.168.17.13' argon*CLI> ==================================================================SIPDefault.cnf # SIP Default Generic Configuration File # Image Version image_version: P0S3-07-2-00 ; # Proxy Server proxy1_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy2_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy3_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy4_address: "192.168.17.13" ; Can be dotted IP or FQDN proxy5_address: "192.168.17.13" ; Can 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)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # 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 ####### New Parameters added in Release 2.0 ####### # Dialplan template (.xml format file relative to the TFTP root directory) dial_template: dialplan # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "" ; Example: ./sip_phone/ # Time Server (There are multiple values and configurations refer to Admin Guide for Specifics) sntp_server: "192.168.17.11" ; SNTP Server IP Address sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default) time_zone: YST ; Time Zone Phone is in dst_offset: 1 ; Offset from Phone's time when DST is in effect dst_start_month: April ; Month in which DST starts dst_start_day: "" ; Day of month in which DST starts dst_start_day_of_week: Sun ; Day of week in which DST starts dst_start_week_of_month: 1 ; Week of month in which DST starts dst_start_time: 02 ; Time of day in which DST starts dst_stop_month: Oct ; Month in which DST stops dst_stop_day: "" ; Day of month in which DST stops dst_stop_day_of_week: Sunday ; Day of week in which DST stops dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month dst_stop_time: 2 ; Time of day in which DST stops dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment time_format_24hr: 0 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr) # 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: "" ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) # Emergency Proxy Support proxy_emergency: "" ; Dotted IP of Emergency Proxy 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: 0 ; 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: 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 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: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged ####### New Parameters added in Release 4.0 ####### # XML URLs services_url: "" ; URL for external Phone Services directory_url: "" ; URL for external Directory location logo_url: "" ; URL for branding logo to be used on phone display # 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: 0 ; 0-Disabled (default), 1-Enabled ####### New Parameters added in Release 4.4 ####### # Call Hold Ringback (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) call_hold_ringback: 0 ; Default 0 (Disable ringback of held call) ====================================================================SIP00115C407FA3.conf # SIP Configuration Generic File # Line 1 line1_name: scott line1_authname: "scott" line1_password: "scott" # Line 2 line2_name: Line 2 line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 3 line2_name: "Line 2" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 4 line2_name: "Line 4" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 5 line2_name: "Line 5" line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 6 line2_name: "Line 6" line2_authname: "UNPROVISIONED" 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: "" ; Has no effect on SIP messaging # Line 1 Display Name (Display name to use for SIP messaging) line1_displayname: "User ID" # Line 2 Display Name (Display name to use for SIP messaging) line2_displayname: "" ####### New Parameters added in Release 3.0 ###### # Phone Prompt (The prompt that will be displayed on console and telnet) phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP Phone) # Phone Password (Password to be used for console or telnet login) phone_password: "cisco" ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none ======================================================================= The result of this process is that the 6101 fails to dial so the system then dial 337-2860 and I can complete the call. Any help would be appreciated on this -- Scott Henderson =========================================Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.337.2860, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com ==========================================
Bryan Vyhmeister
2004-Oct-04 22:52 UTC
[Asterisk-Users] Cisco 7960G w/ SIP not working properly
Scott A. Henderson wrote:> I have Asterisk version 1.0-RC1 running on Debian Woody. > > I have 1 analog phone working, 2 inbound lines working, X-Lite is working. > > The problem that I am having is with Cisco 7960 with SIP version 7.2 > software. I can make outbound calls and they work fine, I even get a > notice that I have voice mail on the phone and it seems to register > properly but I can seem to dial to the phone. >Looking at your extensions.conf, you have Dial(SIP/scott/s,20,Ttr). Shouldn't it be Dial(SIP/scott,20,Ttr)? That /s after Scott is incorrect as far as I know. Maybe I'm wrong but that looks like the culprit to me.> ; Scott Henderson > exten => 6101,1,Dial(SIP/scott/s,20,Ttr)Hope that helps. Bryan