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Elastix + Cisco phone 7940

Asterisk GUI 3 сообщений -
#1

Получил 2 аппарата Cisco ip phone 7940. теперь как бы их нужно приладить к астериску. на tftp залил конфигурацию прошивку итп, телефон нормально подцепился к астериску.
но возникли проблеммы с набором.
стоит карта астериск с 4мя портами.
2 аналоговые телефоны
2 входящие линии
аналоговые телефоны висят на chanel 3 chanel 4 присвоины внутринии номера 4003 4004.
на циско телефоны присвоил 101 102.
еще 4 телефона (2 софт фона + 2 на мобильных телефонах).
с циско фона на циско фон звонок проходит нормально и можно переключить званок на другой номер пример "##4003" и званок уходит по назначению, но позвонить на другой номер не выходит при наборе номера 4003 с номера 102 завнок уходит на номер 111 "установлен на моём"
пробывал авторизоватся под номером 101-102 на софто фоне всё работает нормально и в город и на остальные намера.
ну и самая главная проблема не уходят звонки с циско фонов в город.
asterisk -r пишет
Код:
asterisk*CLI>
asterisk*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [2943558@from-internal:1] Macro("SIP/102-0000003c", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/102-0000003c", "AMPUSER=102") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/102-0000003c", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/102-0000003c", "1?Set(REALCALLERIDNUM=102)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/102-0000003c", "AMPUSER=102") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/102-0000003c", "AMPUSERCIDNAME=102") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/102-0000003c", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/102-0000003c", "AMPUSERCID=102") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/102-0000003c", "CALLERID(all)="102" ") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/102-0000003c", "1?Set(CHANNEL(language)=ru)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/102-0000003c", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] Set("SIP/102-0000003c", "CALLERID(number)=102") in new stack
-- Executing [s@macro-user-callerid:20] Set("SIP/102-0000003c", "CALLERID(name)=102") in new stack
-- Executing [s@macro-user-callerid:21] NoOp("SIP/102-0000003c", "Using CallerID "102" ") in new stack
-- Executing [2943558@from-internal:2] NoOp("SIP/102-0000003c", "Calling Out Route: 9_outside") in new stack
-- Executing [2943558@from-internal:3] Set("SIP/102-0000003c", "MOHCLASS=default") in new stack
-- Executing [2943558@from-internal:4] Set("SIP/102-0000003c", "_NODEST=") in new stack
-- Executing [2943558@from-internal:5] Macro("SIP/102-0000003c", "record-enable,102,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/102-0000003c", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/102-0000003c", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/102-0000003c", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/102-0000003c", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/102-0000003c", "1?MacroExit()") in new stack
-- Executing [2943558@from-internal:6] Macro("SIP/102-0000003c", "dialout-trunk,1,2943558,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/102-0000003c", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/102-0000003c", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/102-0000003c", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/102-0000003c", "DIAL_NUMBER=2943558") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/102-0000003c", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/102-0000003c", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/102-0000003c", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/102-0000003c", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/102-0000003c", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/102-0000003c", "outbound-callerid,1") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/102-0000003c", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/102-0000003c", "0?Set(REALCALLERIDNUM=102)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/102-0000003c", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/102-0000003c", "USEROUTCID=102") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/102-0000003c", "EMERGENCYCID=102") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/102-0000003c", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/102-0000003c", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/102-0000003c", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/102-0000003c", "1?Set(CALLERID(all)=102)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/102-0000003c", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/102-0000003c", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/102-0000003c", "1?sub-flp-1,s,1") in new stack
-- Executing [s@sub-flp-1:1] ExecIf("SIP/102-0000003c", "1?Return()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/102-0000003c", "OUTNUM=2943558") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/102-0000003c", "custom=DAHDI/g0") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/102-0000003c", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/102-0000003c", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/102-0000003c", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/102-0000003c", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/102-0000003c", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/102-0000003c", "DAHDI/g0/2943558,300,") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/102-0000003c", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 58") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/102-0000003c", "s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/102-0000003c", "RC=58") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/102-0000003c", "58,1") in new stack
-- Goto (macro-dialout-trunk,58,1)
-- Executing [58@macro-dialout-trunk:1] Goto("SIP/102-0000003c", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/102-0000003c", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/102-0000003c", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 58 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/102-0000003c", "CALLERID(number)=102") in new stack
-- Executing [2943558@from-internal:7] Macro("SIP/102-0000003c", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/102-0000003c", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/102-0000003c", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/102-0000003c", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/102-0000003c", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- Executing [s@macro-outisbusy:5] Congestion("SIP/102-0000003c", "20") in new stack
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/102-0000003c' in macro 'outisbusy'
== Spawn extension (from-internal, 2943558, 7) exited non-zero on 'SIP/102-0000003c'
-- Executing [h@from-internal:1] Macro("SIP/102-0000003c", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/102-0000003c", "1?endmixmoncheck") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] NoOp("SIP/102-0000003c", "End of MIXMON check") in new stack
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/102-0000003c", "1?nomeetmemon") in new stack
-- Goto (macro-hangupcall,s,15)
-- Executing [s@macro-hangupcall:15] NoOp("SIP/102-0000003c", "MEETME_RECORDINGFILE=") in new stack
-- Executing [s@macro-hangupcall:16] GotoIf("SIP/102-0000003c", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,18)
-- Executing [s@macro-hangupcall:18] NoOp("SIP/102-0000003c", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:19] GotoIf("SIP/102-0000003c", "1?noautomon2") in new stack
-- Goto (macro-hangupcall,s,25)
-- Executing [s@macro-hangupcall:25] NoOp("SIP/102-0000003c", "MONITOR_FILENAME=") in new stack
-- Executing [s@macro-hangupcall:26] GotoIf("SIP/102-0000003c", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,29)
-- Executing [s@macro-hangupcall:29] GotoIf("SIP/102-0000003c", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,32)
-- Executing [s@macro-hangupcall:32] GotoIf("SIP/102-0000003c", "1?theend") in new stack
-- Goto (macro-hangupcall,s,34)
-- Executing [s@macro-hangupcall:34] Hangup("SIP/102-0000003c", "") in new stack
== Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/102-0000003c' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/102-0000003c'
-- Remote UNIX connection
-- Remote UNIX connection disconnected
asterisk*CLI>


дашельше конфиги циско фонов.
то что в SIPDefault.cnf
Код:
# SIP Default Generic Configuration File

#erase protflash
#reset
#Disable debug: tty mon 0

# Image Version
image_version: P0S3-8-12-00

# Proxy Server
proxy1_address: "192.168.1.118" ; Can be dotted IP or FQDN
proxy1_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: g729a

# 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: "pool.ntp.org" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: MSK ; Time Zone Phone is in
dst_offset: 01/00 ; Offset from Phone's time when DST is in effect
dst_start_month: March ; Month in which DST starts
dst_start_day: "" ; Day of month in which DST starts
dst_start_day_of_week: Sunday ; Day of week in which DST starts
dst_start_week_of_month: 8 ; Week of month in which DST starts
dst_start_time: 2 ; 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: 3 ; Time of day in which DST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
date_format: "D/M/Y"

# 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-disabled, 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: "sip.pbxware.ru" ; Dotted IP of Backup Proxy
#proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)

# Emergency Proxy Support
#proxy_emergency: "sip.pbxware.ru" ; 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: 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: 16384 ; Start RTP range for media (default - 16384)
end_media_port: 16484 ; End RTP range for media (default - 32766)
nat_received_processing: 1 ; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
#outbound_proxy: "sip.pbxware.ru" ; 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: 0 ; 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-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: 1 ; 0-Disabled (default), 1-Enabled

# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 1 ; 0-Disabled (default), 1-Enabled



в dialplan.xml
Код:



ну и конкретная конфигурация по циско фону SIP0018ba856766.cnf
Код:
# Cisco SIP Configuration

phone_label: "itcomm"
line1_name: "101"
line1_authname: "101"
line1_shortname: "101"
line1_displayname: "101"
line1_password: "тут password"
#line2_name: ""
#line2_authname: ""
#line2_shortname: ""
#line2_displayname: ""
#line2_password: ""

буду блогодарен услышать подсказку или напровление куда рыть.
#2

UP ребят всё борюсь нет мыслей?
#3

Как обойтись без Cisco Call Manager пробывал работает на ура
http://habrahabr.ru/post/148455/
http://habrahabr.ru/post/121140/