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h323 trunk CCM и Trixbox

0

Добрый день!

Помогите настроить h323 транк между Trixbox 2.0(Asterisk 1.2.13) и CCM(7.0.2)

Настраивал по этой инструкции http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration

скопировал файл ooh323.conf, находящийся здесь /etc/asterisk-1.2.12.1_samples в папку /etc/asterisk. Дальше редактировал уже файл /etc/asterisk/ooh323.conf.

IP CCM - 192.168.1.222 extencions CCM - 1XXXX IP Trixbox - 192.168.1.224 extencions Trixbon 5XXX

Никаких обновлений, описанных в этой части я не делал

!!!!!!!!!!!!!!!!!!!!!

Brothers i recommend you update your trixbox to version 1.2.3 as described in update trixbox section in that way your addon versions will be change and you will have more stable h323 trunk in long term.(you will just change addon versions in the upper part.)

FOR SER & TRIXBOX INTEGRATION FOLLOW THE LINK BELOW: http://www.trixbox.org/modules/newbb/viewtopic.php?topicid=7786&postid=30904&order=0&viewmode=thread&pid=30735&forum=2#forumpost30904

Brothers, The addons used in trixbox 2.0 and trixbox 1.2.3 are causing the asterisk crash... saying core dumped...

here is the solution; on trixbox 1.2.3 (NOT TRIXBOX 2.0 !!!);

STEP 1:DELETE CURRENT ADDONS RPM ; rpm -qa | grep asterisk-addons

rpm -e asterisk-addons-1.2.4_1.2.12.1-1.294

STEP2:LOAD ADDONS VERSION 1.2.3

rpm -i asterisk-addons-1.2.3-1.219.i386.rpm

(i put in the link->http://n.domaindlx.com/ergenay/rpms/asterisk-addons-1.2.3-1.219.i386.rpm)(sometimes page is under load so try your chance.) (also same link on "http://www.ergenay.com" alüminyum korkuluk )

amportal stop

amportal start

then it works perfect.

on trixbox 2.0 it is NOTworking whatever you do... so STAY on 1.2.3 and DO rpm change if you are using "ooh323".

!!!!!!!!!!!!!!!!!!!!!

В результате мой файл ooh323.conf выглядит так:

; Objective System's H323 Configuration example for Asterisk ; ooh323c driver configuration ; ; general section defines global parameters ; ; This is followed by profiles which can be of three types - user/peer/friend ; Name of the user profile should match with the h323id of the user device. ; For peer/friend profiles, host ip address must be provided as "dynamic" is ; not supported as of now. ; ; Syntax for specifying a H323 device in extensions.conf is

For Registered peers/friends profiles

; OOH323/name where name is the name of the peer/friend profile. ;

For unregistered H.323 phones

OOH323/ip:port OR if gk is used OOH323/alias where alias can be any H323

; alias ; ; For dialing into another asterisk peer at a specific exten ; OOH323/exten/peer OR OOH323/exten@ip ; ; Domain name resolution is not yet supported. ; ; When a H.323 user calls into asterisk, his H323ID is matched with the profile ; name and context is determined to route the call ; ; The channel driver will register all global aliases and aliases defined in ; peer profiles with the gatekeeper, if one exists. So, that when someone ; outside our pbx (non-user) calls an extension, gatekeeper will route that ; call to our asterisk box, from where it will be routed as per dial plan.

general ;Define the asetrisk server h323 endpoint

;The port asterisk should listen for incoming H323 connections. ;Default - 1720 port=1720

;The dotted IP address asterisk should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own bindaddr=192.168.1.224

;This parameter indicates whether channel driver should register with ;gatekeeper as a gateway or an endpoint. ;Default - no gateway=no

;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes ;faststart=no ;h245tunneling=no

;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=ObjSysAsterisk e164=100

;CallerID to use for calls ;Default - Same as h323id callerid=asterisk

;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER ;gatekeeper = a.b.c.d gatekeeper = DISABLE

;Location for H323 log file ;Default - /var/log/asterisk/h323log ;logfile=/var/log/asterisk/h323log

;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition

;Sets default context all clients will be placed in. ;Default - default context=default

;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold rtptimeout=3 ;do not drop this below 3 nor increase much...other wise ;you will not able to call same number again for some time because ;it hangs.Now 3 seconds waiting is needed and it is acceptable.

;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) ;tos=lowdelay

;amaflags = default

;The account code used by default for all clients. ;accountcode=h3230101

;The codecs to be used for all clients.Only ulaw and gsm supported as of now. ;Default - ulaw ; ONLY ulaw, gsm, g729 and g7231 supported as of now disallow=all ;Note order of disallow/allow is important. allow=gsm allow=ulaw allow=g729 allow=g723

; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad ; h245alphanumeric, h245signal. ;Default - rfc 2833 dtmfmode=rfc2833

User/peer/friend definitions

; User config options Peer config options ; ------------------ ------------------- ; context ; disallow disallow ; allow allow ; accountcode accountcode ; amaflags amaflags ; dtmfmode dtmfmode ; rtptimeout ip ; port ; h323id ; email ; url ; e164 ; rtptimeout

;

;Define users here ;Section header is extension myuser1 type=user context=context1 disallow=all allow=gsm allow=ulaw

mypeer1 type=peer context=context2 ip=a.b.c.d ; UPDATE with appropriate ip address port=1720 ; UPDATE with appropriate port e164=101

myfriend1 type=friend context=default ip=10.0.0.82 ; UPDATE with appropriate ip address port=1820 ; UPDATE with appropriate port disallow=all allow=ulaw e164=12345 rtptimeout=60 dtmfmode=rfc2833

Dial rules:1XXXX custom dial string:OOH323/$OUTNUM$@192.168.1.222:1720

Route name:h323trunk Dial rules:1XXXX OOH323/$OUTNUM$@192.168.1.222:1720

удалить закрыть спам изменить тег редактировать

спросил 2011-03-03 16:11:09 +0400

DADwq21 Gravatar DADwq21
1 1 1

3 Ответа

1

Я бы постарался переубедить в пользу работы по sip. Это позволит избежать кучи глюков.

ссылка удалить спам редактировать

ответил 2011-03-05 13:30:28 +0400

Tron Gravatar Tron
21 2 1 2
0

Мне нужно настроить H323 транк, не sip транк. И здесь описан порядок настройки H323, SIP транк описан ваше, читайте внимательней.

ссылка удалить спам редактировать

ответил 2011-03-05 11:02:51 +0400

DADwq21 Gravatar DADwq21
1 1 1
0

Причем здесь H323 ? Здесь описан SIP-транк. 1. Настроить sip-транк на CallManager. 2. Настроить SIP-пир на Астериске. Дальше все заработает.

ссылка удалить спам редактировать

ответил 2011-03-04 19:10:48 +0400

Tron Gravatar Tron
21 2 1 2

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Задан: 2011-03-03 16:11:09 +0400

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Обновлен: Mar 05 '11

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.