Newbie trying to setup Cisco 7940 phones
I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin dsi> sh conf ------ Current *FLASH* Configuration ------ Platform : Cisco Systems, Inc. IP Phone CP-7940G Elapsed Time: 01:01:06 dhcp_server : Disabled my_ip_addr : 10.86.11.50 subnet_mask : 255.255.0.0 defaultgw : 10.86.0.1 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 10.85.0.11 dns_backup_1: 10.85.0.10 primary_tftp_addr : 10.86.10.58 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0012:7f98:eaa9 domain_name : dsi-corp.net my_name : SIP00127F98EAA9 Status Flags : 12300001 image_version : "P003-8-12-00" FirmLoadID : "PC030301" DSPLoadID : "PS03AT38" network_media_type : Auto network_port2_type : Hub/Switch dscpForAudio : 184 phone_label : "Matthew Kitchin" tftp_cfg_dir : "" phone_password : ********** phone_prompt : "dsi" language : english sntp_mode : Unicast sntp_server : 10.85.0.10 time_zone : CST dst_offset : 01/00 dst_start_month : March dst_start_day : 0 dst_start_day_of_week : Sunday dst_start_week_of_month : 8 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 02/00 dst_auto_adjust : 1 time_format_24hr : 1 date_format : M/D/Y nat_enable : 0 nat_address : UNPROVISIONED voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : "1" xml_card_dir : "" xml_card_file : "CARD.XML" telnet_level : 1 services_url : "" directory_url : "" logo_url : "" http_proxy_addr : UNPROVISIONED http_proxy_port : 80 garp_enable : 0 enable_vad : 0 dial_template : "dialplan" callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 0 messages_uri : "" dnd_control : 2 preferred_codec : g711ulaw dtmf_outofband : avt_always dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 0 call_manager1_addr : "UNPROVISIONED" call_manager2_addr : "UNPROVISIONED" call_manager3_addr : "UNPROVISIONED" call_manager1_sip_port : 5060 call_manager2_sip_port : 5060 call_manager3_sip_port : 5060 call_manager5_addr : "UNPROVISIONED" call_manager5_sip_port : 5060 call_manager4_addr : "UNPROVISIONED" call_manager4_sip_port : 0 line1_name : "1008" line2_name : "1001" line1_authname : "1008" line2_authname : "1001" line1_password : ********** line2_password : ********** line1_shortname : "1008" line2_shortname : "1001" line1_displayname : "1008" line2_displayname : "UNPROVISIONED" line1_contact : "UNPROVISIONED" line2_contact : "UNPROVISIONED" proxy1_address : "nshplpbx1.unix" proxy2_address : "nshplpbx1.unix" proxy1_port : 5060 proxy2_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : "" proxy_emergency : "" proxy_backup_port : 5060 proxy_emergency_port : 5060 outbound_proxy : UNPROVISIONED outbound_proxy_port : 5060 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : phone cnf_join_enable : 0 remote_party_id : 1 semi_attended_transfer : 1 transfer_onhook_enabled : 0 call_hold_ringback : 3 stutter_msg_waiting : 0 cfwd_url : "" call_stats : 0 auto_answer : 0 local_cfwd_enable : 1 timer_register_delta : 5 sip_max_forwards : 70 rfc_2543_hold : 0 version_stamp : "" timer_keepalive_expires : 120 connection_monitor_duration : 120 encrypt_key : ********** 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5067 0 acls to check for proxy 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5085 network ip is a proxy [0] 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl "domains". Falling back to Digest auth. 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5067 0 acls to check for proxy 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5085 network ip is a proxy [0] 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl "domains". Falling back to Digest auth. 2009-11-03 15:39:50.797901 [NOTICE] switch_channel.c:613 New Channel sofia/internal/1000@nshplpbx1.unix [8c133ad4-67cf-4ffa-8655-56ffa0e3933d] 2009-11-03 15:39:50.799061 [DEBUG] sofia.c:5812 Setting NAT mode based on nat.auto 2009-11-03 15:39:50.799061 [DEBUG] sofia.c:3588 Channel sofia/internal/1000@nshplpbx1.unix entering state [received][100] 2009-11-03 15:39:50.799061 [DEBUG] sofia.c:3599 Remote SDP: v=0 o=- 8 2 IN IP4 10.86.10.58 s=CounterPath X-Lite 3.0 c=IN IP4 10.86.10.58 t=0 0 m=audio 37250 RTP/AVP 107 0 8 101 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : cwKnUvoi NsQWtT4P 10.86.10.58 37250 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[G7221:115:32000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[G7221:107:16000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[G722:9:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[PCMU:0:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[PCMA:8:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [BV32:107:16000:0]/[GSM:3:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3160 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:2102 Set Codec sofia/internal/1000@nshplpbx1.unix PCMU/8000 20 ms 160 samples 2009-11-03 15:39:50.799061 [DEBUG] sofia_glue.c:3120 Set 2833 dtmf payload to 101 2009-11-03 15:39:50.799061 [DEBUG] sofia.c:3744 (sofia/internal/1000@nshplpbx1.unix) State Change CS_NEW -> CS_INIT 2009-11-03 15:39:50.799061 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000@nshplpbx1.unix [BREAK] 2009-11-03 15:39:50.799061 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000@nshplpbx1.unix) Running State Change CS_INIT 2009-11-03 15:39:50.799061 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/1000@nshplpbx1.unix) State INIT 2009-11-03 15:39:50.799061 [DEBUG] mod_sofia.c:83 sofia/internal/1000@nshplpbx1.unix SOFIA INIT 2009-11-03 15:39:50.799061 [DEBUG] mod_sofia.c:111 (sofia/internal/1000@nshplpbx1.unix) State Change CS_INIT -> CS_ROUTING 2009-11-03 15:39:50.799061 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000@nshplpbx1.unix [BREAK] 2009-11-03 15:39:50.800189 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/1000@nshplpbx1.unix) State INIT going to sleep 2009-11-03 15:39:50.800189 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000@nshplpbx1.unix) Running State Change CS_ROUTING 2009-11-03 15:39:50.800189 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1000@nshplpbx1.unix) State ROUTING 2009-11-03 15:39:50.800189 [DEBUG] mod_sofia.c:130 sofia/internal/1000@nshplpbx1.unix SOFIA ROUTING 2009-11-03 15:39:50.800189 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1000@nshplpbx1.unix Standard ROUTING 2009-11-03 15:39:50.800189 [INFO] mod_dialplan_xml.c:397 Processing 1000->1008 in context default Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->unloop] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->tod_example] continue=true Dialplan: day of week[3] =~ 2-6 (PASS) Dialplan: hour[15] =~ 9-18 (PASS) Dialplan: sofia/internal/1000@nshplpbx1.unix Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix Action set(open=true) Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [global-intercept] destination_number(1008) =~ /^886$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [group-intercept] destination_number(1008) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [intercept-ext] destination_number(1008) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->redial] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [redial] destination_number(1008) =~ /^870$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->global] continue=true Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/1000@nshplpbx1.unix Absolute Condition [global] Dialplan: sofia/internal/1000@nshplpbx1.unix Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1000@nshplpbx1.unix Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1000@nshplpbx1.unix Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [snom-demo-2] destination_number(1008) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [snom-demo-1] destination_number(1008) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [eavesdrop] destination_number(1008) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [eavesdrop] destination_number(1008) =~ /^779$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->call_return] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [call_return] destination_number(1008) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->del-group] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [del-group] destination_number(1008) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->add-group] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [add-group] destination_number(1008) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [call-group-simo] destination_number(1008) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [call-group-order] destination_number(1008) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (FAIL) [extension-intercom] destination_number(1008) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1000@nshplpbx1.unix Regex (PASS) [Local_Extension] destination_number(1008) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000@nshplpbx1.unix Action set(dialed_extension=1008) Dialplan: sofia/internal/1000@nshplpbx1.unix Action export(dialed_extension=1008) Dialplan: sofia/internal/1000@nshplpbx1.unix Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1000@nshplpbx1.unix Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1000@nshplpbx1.unix Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1000@nshplpbx1.unix Action set(ringback=${us-ring}) Dialplan: sofia/internal/1000@nshplpbx1.unix Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1000@nshplpbx1.unix Action set(call_timeout=30) Dialplan: sofia/internal/1000@nshplpbx1.unix Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1000@nshplpbx1.unix Action set(continue_on_fail=true) Dialplan: sofia/internal/1000@nshplpbx1.unix Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1000@nshplpbx1.unix Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1000@nshplpbx1.unix Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1000@nshplpbx1.unix Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1000@nshplpbx1.unix Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1000@nshplpbx1.unix Action answer() Dialplan: sofia/internal/1000@nshplpbx1.unix Action sleep(1000) Dialplan: sofia/internal/1000@nshplpbx1.unix Action voicemail(default ${domain_name} ${dialed_extension}) 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/1000@nshplpbx1.unix) State Change CS_ROUTING -> CS_EXECUTE 2009-11-03 15:39:50.801914 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000@nshplpbx1.unix [BREAK] 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/1000@nshplpbx1.unix) State ROUTING going to sleep 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000@nshplpbx1.unix) Running State Change CS_EXECUTE 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:340 (sofia/internal/1000@nshplpbx1.unix) State EXECUTE 2009-11-03 15:39:50.801914 [DEBUG] mod_sofia.c:173 sofia/internal/1000@nshplpbx1.unix SOFIA EXECUTE 2009-11-03 15:39:50.801914 [DEBUG] switch_core_state_machine.c:151 sofia/internal/1000@nshplpbx1.unix Standard EXECUTE EXECUTE sofia/internal/1000@nshplpbx1.unix set(open=true) 2009-11-03 15:39:50.803724 [DEBUG] mod_dptools.c:763 sofia/internal/1000@nshplpbx1.unix SET [open]=[true] EXECUTE sofia/internal/1000@nshplpbx1.unix hash(insert/10.85.0.53-spymap/1000/8c133ad4-67cf-4ffa-8655-56ffa0e3933d) EXECUTE sofia/internal/1000@nshplpbx1.unix hash(insert/10.85.0.53-last_dial/1000/1008) EXECUTE sofia/internal/1000@nshplpbx1.unix hash(insert/10.85.0.53-last_dial/global/8c133ad4-67cf-4ffa-8655-56ffa0e3933d) EXECUTE sofia/internal/1000@nshplpbx1.unix set(dialed_extension=1008) 2009-11-03 15:39:50.803724 [DEBUG] mod_dptools.c:763 sofia/internal/1000@nshplpbx1.unix SET [dialed_extension]=[1008] EXECUTE sofia/internal/1000@nshplpbx1.unix export(dialed_extension=1008) 2009-11-03 15:39:50.803724 [DEBUG] mod_dptools.c:846 EXPORT [dialed_extension]=[1008] EXECUTE sofia/internal/1000@nshplpbx1.unix bind_meta_app(1 b s execute_extension::dx XML features) 2009-11-03 15:39:50.804911 [INFO] switch_ivr_async.c:2151 Bound B-Leg: 1 execute_extension::dx XML features EXECUTE sofia/internal/1000@nshplpbx1.unix bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1000.2009-11-03-15-39-50.wav) 2009-11-03 15:39:50.804911 [INFO] switch_ivr_async.c:2151 Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/1000.2009-11-03-15-39-50.wav EXECUTE sofia/internal/1000@nshplpbx1.unix bind_meta_app(3 b s execute_extension::cf XML features) 2009-11-03 15:39:50.804911 [INFO] switch_ivr_async.c:2151 Bound B-Leg: 3 execute_extension::cf XML features EXECUTE sofia/internal/1000@nshplpbx1.unix set(ringback=%(2000,4000,440.0,480.0)) 2009-11-03 15:39:50.804911 [DEBUG] mod_dptools.c:763 sofia/internal/1000@nshplpbx1.unix SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/1000@nshplpbx1.unix set(transfer_ringback=local_stream://moh) 2009-11-03 15:39:50.804911 [DEBUG] mod_dptools.c:763 sofia/internal/1000@nshplpbx1.unix SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1000@nshplpbx1.unix set(call_timeout=30) 2009-11-03 15:39:50.804911 [DEBUG] mod_dptools.c:763 sofia/internal/1000@nshplpbx1.unix SET [call_timeout]=[30] EXECUTE sofia/internal/1000@nshplpbx1.unix set(hangup_after_bridge=true) 2009-11-03 15:39:50.806131 [DEBUG] mod_dptools.c:763 sofia/internal/1000@nshplpbx1.unix SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1000@nshplpbx1.unix set(continue_on_fail=true) 2009-11-03 15:39:50.806131 [DEBUG] mod_dptools.c:763 sofia/internal/1000@nshplpbx1.unix SET [continue_on_fail]=[true] EXECUTE sofia/internal/1000@nshplpbx1.unix hash(insert/10.85.0.53-call_return/1008/1000) EXECUTE sofia/internal/1000@nshplpbx1.unix hash(insert/10.85.0.53-last_dial_ext/1008/8c133ad4-67cf-4ffa-8655-56ffa0e3933d) EXECUTE sofia/internal/1000@nshplpbx1.unix set(called_party_callgroup=techsupport) 2009-11-03 15:39:50.806131 [DEBUG] mod_dptools.c:763 sofia/internal/1000@nshplpbx1.unix SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1000@nshplpbx1.unix hash(insert/10.85.0.53-last_dial/techsupport/8c133ad4-67cf-4ffa-8655-56ffa0e3933d) EXECUTE sofia/internal/1000@nshplpbx1.unix bridge(user/1008@10.85.0.53) 2009-11-03 15:39:50.812162 [DEBUG] switch_ivr_originate.c:1333 variable string 0 = [presence_id=1008@10.85.0.53] 2009-11-03 15:39:50.812162 [NOTICE] switch_channel.c:613 New Channel sofia/internal/sip:1008@10.86.11.50:5060 [32661f42-b0a9-4179-867d-156b46ae71ce] 2009-11-03 15:39:50.812162 [DEBUG] mod_sofia.c:3061 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_NEW -> CS_INIT 2009-11-03 15:39:50.812162 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.812162 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_INIT 2009-11-03 15:39:50.812162 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/sip:1008@10.86.11.50:5060) State INIT 2009-11-03 15:39:50.812162 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1008@10.86.11.50:5060 SOFIA INIT 2009-11-03 15:39:50.812162 [DEBUG] sofia_glue.c:1761 sip:1008@10.86.11.50:51061;user=phone;transport=udp Setting proxy route to sofia/internal/sip:1008@10.86.11.50:5060 2009-11-03 15:39:50.814226 [DEBUG] mod_sofia.c:111 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_INIT -> CS_ROUTING 2009-11-03 15:39:50.814226 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.814226 [DEBUG] sofia.c:3588 Channel sofia/internal/sip:1008@10.86.11.50:5060 entering state [calling][0] 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:330 (sofia/internal/sip:1008@10.86.11.50:5060) State INIT going to sleep 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_ROUTING 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:1008@10.86.11.50:5060) State ROUTING 2009-11-03 15:39:50.814226 [DEBUG] mod_sofia.c:130 sofia/internal/sip:1008@10.86.11.50:5060 SOFIA ROUTING 2009-11-03 15:39:50.814226 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-11-03 15:39:50.814226 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:333 (sofia/internal/sip:1008@10.86.11.50:5060) State ROUTING going to sleep 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_CONSUME_MEDIA 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/sip:1008@10.86.11.50:5060) State CONSUME_MEDIA 2009-11-03 15:39:50.814226 [DEBUG] switch_core_state_machine.c:352 (sofia/internal/sip:1008@10.86.11.50:5060) State CONSUME_MEDIA going to sleep 2009-11-03 15:39:50.817484 [DEBUG] sofia.c:3588 Channel sofia/internal/sip:1008@10.86.11.50:5060 entering state [terminated][503] 2009-11-03 15:39:50.817484 [NOTICE] sofia.c:4187 Hangup sofia/internal/sip:1008@10.86.11.50:5060 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-11-03 15:39:50.817484 [DEBUG] switch_channel.c:1896 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [KILL] 2009-11-03 15:39:50.817484 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.817484 [DEBUG] switch_core_state_machine.c:451 thread mismatch skipping state handler. 2009-11-03 15:39:50.817484 [DEBUG] switch_ivr_originate.c:2543 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2009-11-03 15:39:50.817484 [ERR] switch_ivr_originate.c:1829 Cannot create outgoing channel of type [user] cause: [NORMAL_TEMPORARY_FAILURE] 2009-11-03 15:39:50.817484 [DEBUG] switch_ivr_originate.c:2543 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2009-11-03 15:39:50.817484 [INFO] mod_dptools.c:2290 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2009-11-03 15:39:50.817484 [DEBUG] mod_dptools.c:2312 Continue on fail [true]: Cause: NORMAL_TEMPORARY_FAILURE EXECUTE sofia/internal/1000@nshplpbx1.unix answer() 2009-11-03 15:39:50.817484 [DEBUG] mod_dptools.c:653 sofia/internal/1000@nshplpbx1.unix receive message [ANSWER] 2009-11-03 15:39:50.817484 [DEBUG] sofia_glue.c:2336 AUDIO RTP [sofia/internal/1000@nshplpbx1.unix] 10.85.0.53 port 24712 -> 10.86.10.58 port 37250 codec: 0 ms: 20 2009-11-03 15:39:50.817484 [DEBUG] switch_rtp.c:1155 Starting timer [soft] 160 bytes per 20ms 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_HANGUP 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/sip:1008@10.86.11.50:5060) State HANGUP 2009-11-03 15:39:50.818924 [DEBUG] mod_sofia.c:329 sofia/internal/sip:1008@10.86.11.50:5060 Overriding SIP cause 503 with 503 from the other leg 2009-11-03 15:39:50.818924 [DEBUG] mod_sofia.c:361 Channel sofia/internal/sip:1008@10.86.11.50:5060 hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:1008@10.86.11.50:5060 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/sip:1008@10.86.11.50:5060) State HANGUP going to sleep 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_HANGUP -> CS_REPORTING 2009-11-03 15:39:50.818924 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_REPORTING 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:1008@10.86.11.50:5060) State REPORTING 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:1008@10.86.11.50:5060 Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:1008@10.86.11.50:5060) State REPORTING going to sleep 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/sip:1008@10.86.11.50:5060) State Change CS_REPORTING -> CS_DESTROY 2009-11-03 15:39:50.818924 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/sip:1008@10.86.11.50:5060 [BREAK] 2009-11-03 15:39:50.818924 [DEBUG] switch_core_session.c:1140 Session 32 (sofia/internal/sip:1008@10.86.11.50:5060) Locked, Waiting on external entities 2009-11-03 15:39:50.818924 [NOTICE] switch_core_session.c:1158 Session 32 (sofia/internal/sip:1008@10.86.11.50:5060) Ended 2009-11-03 15:39:50.818924 [NOTICE] switch_core_session.c:1160 Close Channel sofia/internal/sip:1008@10.86.11.50:5060 [CS_DESTROY] 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:1008@10.86.11.50:5060) Running State Change CS_DESTROY 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/sip:1008@10.86.11.50:5060) State DESTROY 2009-11-03 15:39:50.818924 [DEBUG] mod_sofia.c:278 sofia/internal/sip:1008@10.86.11.50:5060 SOFIA DESTROY 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:1008@10.86.11.50:5060 Standard DESTROY 2009-11-03 15:39:50.818924 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/sip:1008@10.86.11.50:5060) State DESTROY going to sleep 2009-11-03 15:39:50.820304 [DEBUG] mod_sofia.c:569 Local SDP sofia/internal/1000@nshplpbx1.unix: v=0 o=FreeSWITCH 1257259678 1257259679 IN IP4 10.85.0.53 s=FreeSWITCH c=IN IP4 10.85.0.53 t=0 0 m=audio 24712 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-11-03 15:39:50.821960 [DEBUG] sofia.c:3588 Channel sofia/internal/1000@nshplpbx1.unix entering state [completed][200] 2009-11-03 15:39:50.821960 [DEBUG] switch_core_session.c:664 Send signal sofia/internal/1000@nshplpbx1.unix [BREAK] 2009-11-03 15:39:50.821960 [NOTICE] mod_dptools.c:653 Channel [sofia/internal/1000@nshplpbx1.unix] has been answered 2009-11-03 15:39:50.821960 [DEBUG] switch_channel.c:182 sofia/internal/1000@nshplpbx1.unix receive message [AUDIO_SYNC] EXECUTE sofia/internal/1000@nshplpbx1.unix sleep(1000) 2009-11-03 15:39:50.821960 [DEBUG] switch_channel.c:182 sofia/internal/1000@nshplpbx1.unix receive message [AUDIO_SYNC] 2009-11-03 15:39:50.867612 [DEBUG] switch_rtp.c:1917 Correct ip/port confirmed. 2009-11-03 15:39:50.926503 [DEBUG] sofia.c:3588 Channel sofia/internal/1000@nshplpbx1.unix entering state [ready][200] EXECUTE sofia/internal/1000@nshplpbx1.unix voicemail(default 10.85.0.53 1008) 2009-11-03 15:39:51.840356 [DEBUG] mod_voicemail.c:799 [default] rwlock 2009-11-03 15:39:51.840356 [DEBUG] switch_channel.c:182 sofia/internal/1000@nshplpbx1.unix receive message [AUDIO_SYNC] 2009-11-03 15:39:51.959567 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-03 15:39:51.961607 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-person.wav] (en:en) 2009-11-03 15:39:51.961607 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-03 15:39:51.961607 [DEBUG] switch_core_io.c:660 sofia/internal/1000@nshplpbx1.unix receive message [TRANSCODING_NECESSARY] 2009-11-03 15:39:53.319706 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-03 15:39:53.439693 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1008] (en:en) 2009-11-03 15:39:53.439693 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-03 15:39:53.439693 [DEBUG] switch_core_io.c:660 sofia/internal/1000@nshplpbx1.unix receive message [TRANSCODING_NECESSARY] 2009-11-03 15:39:53.899622 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-03 15:39:53.899622 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-03 15:39:53.899622 [DEBUG] switch_core_io.c:660 sofia/internal/1000@nshplpbx1.unix receive message [TRANSCODING_NECESSARY] 2009-11-03 15:39:54.127652 [NOTICE] sofia.c:328 Hangup sofia/internal/1000@nshplpbx1.unix [CS_EXECUTE] [NORMAL_CLEARING] 2009-11-03 15:39:54.127652 [DEBUG] switch_channel.c:1896 Send signal sofia/internal/1000@nshplpbx1.unix [KILL] 2009-11-03 15:39:54.127652 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000@nshplpbx1.unix [BREAK] 2009-11-03 15:39:54.127652 [DEBUG] switch_core_state_machine.c:451 thread mismatch skipping state handler. 2009-11-03 15:39:54.139706 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:340 (sofia/internal/1000@nshplpbx1.unix) State EXECUTE going to sleep 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000@nshplpbx1.unix) Running State Change CS_HANGUP 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/1000@nshplpbx1.unix) State HANGUP 2009-11-03 15:39:54.239929 [DEBUG] mod_sofia.c:329 sofia/internal/1000@nshplpbx1.unix Overriding SIP cause 480 with 503 from the other leg 2009-11-03 15:39:54.239929 [DEBUG] mod_sofia.c:361 Channel sofia/internal/1000@nshplpbx1.unix hanging up, cause: NORMAL_CLEARING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1000@nshplpbx1.unix Standard HANGUP, cause: NORMAL_CLEARING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/1000@nshplpbx1.unix) State HANGUP going to sleep 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1000@nshplpbx1.unix) State Change CS_HANGUP -> CS_REPORTING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000@nshplpbx1.unix [BREAK] 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:306 (sofia/internal/1000@nshplpbx1.unix) Running State Change CS_REPORTING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000@nshplpbx1.unix) State REPORTING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1000@nshplpbx1.unix Standard REPORTING, cause: NORMAL_CLEARING 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/1000@nshplpbx1.unix) State REPORTING going to sleep 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:319 (sofia/internal/1000@nshplpbx1.unix) State Change CS_REPORTING -> CS_DESTROY 2009-11-03 15:39:54.239929 [DEBUG] switch_core_session.c:1003 Send signal sofia/internal/1000@nshplpbx1.unix [BREAK] 2009-11-03 15:39:54.239929 [DEBUG] switch_core_session.c:1140 Session 31 (sofia/internal/1000@nshplpbx1.unix) Locked, Waiting on external entities 2009-11-03 15:39:54.239929 [NOTICE] switch_core_session.c:1158 Session 31 (sofia/internal/1000@nshplpbx1.unix) Ended 2009-11-03 15:39:54.239929 [NOTICE] switch_core_session.c:1160 Close Channel sofia/internal/1000@nshplpbx1.unix [CS_DESTROY] 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1000@nshplpbx1.unix) Running State Change CS_DESTROY 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/1000@nshplpbx1.unix) State DESTROY 2009-11-03 15:39:54.239929 [DEBUG] mod_sofia.c:278 sofia/internal/1000@nshplpbx1.unix SOFIA DESTROY 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1000@nshplpbx1.unix Standard DESTROY 2009-11-03 15:39:54.239929 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/1000@nshplpbx1.unix) State DESTROY going to sleep ]0;root@NSHPLPBX1:/usr/local/freeswitch/log[root@NSHPLPBX1 log]#
On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public@gmail.com < mkitchin.public@gmail.com> wrote:
I'm working on an alternative to a $120,000 Cisco phone system that my
company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3.
Thanks, Matthew Kitchin
Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now. Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd say first thing to do is capture the SIP traffic to see if there are any clues. A "normal temporary failure" doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org) You can also join the IRC channel #freeswitch on irc.freenode.net and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers. -MC
Michael Collins wrote:
On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public@gmail.com <mailto:mkitchin.public@gmail.com> <mkitchin.public@gmail.com <mailto:mkitchin.public@gmail.com>> wrote:
I'm working on an alternative to a $120,000 Cisco phone system that my
company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53 <http://10.85.0.53>. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3.
Thanks, Matthew Kitchin
Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now.
Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs
I'd say first thing to do is capture the SIP traffic to see if there are any clues. A "normal temporary failure" doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org)
You can also join the IRC channel #freeswitch on irc.freenode.net <http://irc.freenode.net> and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers.
-MC
Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc.
Matthew, I'm about in the same boat as you are, just on a smaller scale. We have a ton of Nortel telephony gear, but it's time to move out of the 90's and enter this millennium. My Cisco quote was in the same ballpark as yours. The Cisco stuff is mature, rock solid, meshes very well with their network gear and is actually relatively easy to set up and maintain if you know your way around IOS. I just refuse to pay that kind of money for yet another semi-proprietary solution. After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The developers and the community are great and available, but just starting out with SIP and voip in general, this may not be the best platform. So let the blasphemy begin :) SipX was a breeze to install (insert CD, boot, next next next...) and looks pretty solid. I believe they actually use FreeSWITCH for their voicemail and conferencing, internally. I just couldn't get my head around their GUI, ACD was too basic and had all kinds of issues getting stuff to "just work". 3CX (Windows Only) was completely painless. It just worked. But I'm still not convinced that I want to run all my voice on a single windows box. Plus it's not free/open/etc and I don't want to lock myself in again. Although it's an asterisk based solution, I found trixbox to be very easy. Setup is automatic and everything "just worked". The GUI is simple and logical enough that I can let somebody else handle the day-to-day phone setup and basic admin. I have my doubts about it scaling to 250 users, though. This may be a completely flawed strategy and I may very well be shooting myself in the foot by doing this, but I plan on piloting a trixbox install with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH box next to it for the more advanced stuff. Once I get more comfortable with the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, I have a feeling that that trixbox is going to get phased out... Peter -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of mkitchin.public@gmail.com Sent: Tuesday, November 03, 2009 11:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Michael Collins wrote:
On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public@gmail.com <mailto:mkitchin.public@gmail.com> <mkitchin.public@gmail.com <mailto:mkitchin.public@gmail.com>> wrote:
I'm working on an alternative to a $120,000 Cisco phone system that my
company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53 <http://10.85.0.53>. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3.
Thanks, Matthew Kitchin
Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now.
Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs
I'd say first thing to do is capture the SIP traffic to see if there are any clues. A "normal temporary failure" doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org)
You can also join the IRC channel #freeswitch on irc.freenode.net <http://irc.freenode.net> and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers.
-MC
Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Peter J. Zandvoort <peter@cindyandpeter.com> wrote:
After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall.
The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki.
Absolutely agreed. To use Matthew's original car metaphor: When you just got your learner's permit, the old Chevy may be a better choice than the Ferrari. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, November 04, 2009 1:42 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Peter J. Zandvoort <peter@cindyandpeter.com> wrote:
After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall.
The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I had the exact same problem with the Cisco phones not being able to receive calls. I fixed it by messing around with the NAT settings in the internal sofia profile. From what I remember, I just removed the <param name="apply-nat-acl" value="nat.auto"/> line and everything worked fine. -- mm_202. On Wed, Nov 4, 2009 at 11:33 AM, Peter J. Zandvoort <peter@cindyandpeter.com> wrote:
Absolutely agreed. To use Matthew's original car metaphor: When you just got your learner's permit, the old Chevy may be a better choice than the Ferrari.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, November 04, 2009 1:42 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Peter J. Zandvoort <peter@cindyandpeter.com> wrote:
After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall.
The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I hate to say it, but I had to give in and try sipx. the ease of provisioning phones and the ability for helpdesk staff to reset passwords and such through a gui looks like it is too good for me to pass on. mm_202 wrote:
I had the exact same problem with the Cisco phones not being able to receive calls.
I fixed it by messing around with the NAT settings in the internal sofia profile. From what I remember, I just removed the <param name="apply-nat-acl" value="nat.auto"/> line and everything worked fine.
-- mm_202.
On Wed, Nov 4, 2009 at 11:33 AM, Peter J. Zandvoort <peter@cindyandpeter.com> wrote:
Absolutely agreed. To use Matthew's original car metaphor: When you just got your learner's permit, the old Chevy may be a better choice than the Ferrari.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, November 04, 2009 1:42 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Peter J. Zandvoort <peter@cindyandpeter.com> wrote:
After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall.
The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Peter, Did you look at http://www.cudatel.com? Probably just what you are looking for. GUI goodness based on FS. SDR Peter J. Zandvoort wrote:
Matthew,
I'm about in the same boat as you are, just on a smaller scale. We have a ton of Nortel telephony gear, but it's time to move out of the 90's and enter this millennium. My Cisco quote was in the same ballpark as yours.
The Cisco stuff is mature, rock solid, meshes very well with their network gear and is actually relatively easy to set up and maintain if you know your way around IOS. I just refuse to pay that kind of money for yet another semi-proprietary solution.
After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The developers and the community are great and available, but just starting out with SIP and voip in general, this may not be the best platform. So let the blasphemy begin :)
SipX was a breeze to install (insert CD, boot, next next next...) and looks pretty solid. I believe they actually use FreeSWITCH for their voicemail and conferencing, internally. I just couldn't get my head around their GUI, ACD was too basic and had all kinds of issues getting stuff to "just work".
3CX (Windows Only) was completely painless. It just worked. But I'm still not convinced that I want to run all my voice on a single windows box. Plus it's not free/open/etc and I don't want to lock myself in again.
Although it's an asterisk based solution, I found trixbox to be very easy. Setup is automatic and everything "just worked". The GUI is simple and logical enough that I can let somebody else handle the day-to-day phone setup and basic admin. I have my doubts about it scaling to 250 users, though.
This may be a completely flawed strategy and I may very well be shooting myself in the foot by doing this, but I plan on piloting a trixbox install with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH box next to it for the more advanced stuff. Once I get more comfortable with the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, I have a feeling that that trixbox is going to get phased out...
Peter
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of mkitchin.public@gmail.com Sent: Tuesday, November 03, 2009 11:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Michael Collins wrote:
On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.public@gmail.com <mailto:mkitchin.public@gmail.com> <mkitchin.public@gmail.com <mailto:mkitchin.public@gmail.com>> wrote:
I'm working on an alternative to a $120,000 Cisco phone system that my
company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53 <http://10.85.0.53>. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3.
Thanks, Matthew Kitchin
Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now.
Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs
I'd say first thing to do is capture the SIP traffic to see if there are any clues. A "normal temporary failure" doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org)
You can also join the IRC channel #freeswitch on irc.freenode.net <http://irc.freenode.net> and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers.
-MC
Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (6)
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Jason White
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Michael Collins
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mkitchin.public＠gmail.com
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mm_202
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Peter J. Zandvoort
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Shelby Ramsey