integracion de elastix y callmanager cisco v7

Joined
Feb 4, 2009
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hola qu etal tengo un call manager de cisco el cual maneja extensiones con SCCP

la ip es 192.168.211.2

del otro lado tengo un pbx elastix version 1.6 con la ip 192.168.4.201
en este pbx tengo ext. sip


necesito crear una troncal sip entre elastix y callmanager


mis datos del peer son



host=192.168.211.2
secret=digestpbx
type=friend
canreivite=yes
qualify=yes
disallow=all
allow=ulaw
context=from-internal
nat=yes


los datos de user details

type=friend
canreivite=yes
qualify=yes
disallow=all
allow=ulaw
context=from-internal
nat=yes



en mi pbx elastix hago una ruta saliente 9|. saliendo por mi truk sip

een el call manager tengo una ext 5001 y en elastix tengo un 7004


este es el sip set debug

-- Executing [s@macro-dialout-trunk:19] Dial("SIP/7004-08ebbe80", "SIP/callman01/5001|300|") in new stack
Audio is at 192.168.40.201 port 10530
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (NAT) to 192.168.211.2:5060:
INVITE sip:5001@192.168.211.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.201:5060;branch=z9hG4bK393c4e17;rport
From: "7004" <sip:7004@192.168.40.201>;tag=as05005cb1
To: <sip:5001@192.168.211.2>
Contact: <sip:7004@192.168.40.201>
Call-ID: 4bdb491e3da7ed3874ec721702c55ddd@192.168.40.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 18 May 2010 19:08:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 186

v=0
o=root 2782 2782 IN IP4 192.168.40.201
s=session
c=IN IP4 192.168.40.201
t=0 0
m=audio 10530 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:eek:ff - - - -
a=ptime:20
a=sendrecv

---
-- Called callman01/5001
elastix*CLI>
<--- SIP read from 192.168.211.2:5060 --->
SIP/2.0 100 Trying
Date: Tue, 18 May 2010 19:27:46 GMT
From: "7004" <sip:7004@192.168.40.201>;tag=as05005cb1
Allow-Events: presence
Content-Length: 0
To: <sip:5001@192.168.211.2>
Call-ID: 4bdb491e3da7ed3874ec721702c55ddd@192.168.40.201
Via: SIP/2.0/UDP 192.168.40.201:5060;branch=z9hG4bK393c4e17;rport
CSeq: 102 INVITE


<------------->
--- (9 headers 0 lines) ---
elastix*CLI>
<--- SIP read from 192.168.211.2:5060 --->
SIP/2.0 401 Unauthorized
Date: Tue, 18 May 2010 19:27:46 GMT
From: "7004" <sip:7004@192.168.40.201>;tag=as05005cb1
Allow-Events: presence
WWW-Authenticate: Digest realm="StandAloneCluster", nonce="ZdxPuvBXxOpByslCQevOuvo6cLF+lGaO", algorithm=MD5
Content-Length: 0
To: <sip:5001@192.168.211.2>;tag=1359530374
Call-ID: 4bdb491e3da7ed3874ec721702c55ddd@192.168.40.201
Via: SIP/2.0/UDP 192.168.40.201:5060;branch=z9hG4bK393c4e17;rport
CSeq: 102 INVITE


<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.211.2:5060:
ACK sip:5001@192.168.211.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.201:5060;branch=z9hG4bK393c4e17;rport
From: "7004" <sip:7004@192.168.40.201>;tag=as05005cb1
To: <sip:5001@192.168.211.2>;tag=1359530374
Contact: <sip:7004@192.168.40.201>
Call-ID: 4bdb491e3da7ed3874ec721702c55ddd@192.168.40.201
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/callman01-08ebfea0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/7004-08ebbe80", "s-CONGESTION|1") in new st


me dice que todas las lineas estan ocupadas

necesito ayuda


saludos
 
Joined
Sep 24, 2008
Messages
134
Points
0
Hola,

Ya pudiste realizar la integracion?

He tratado de todo y nada, llamo desde mi softphone registrado en asterisk y me dice "todas las líneas están ocupadas"

Cualquier ayuda se agradece
 
Joined
Sep 24, 2008
Messages
134
Points
0
Hola,

Ya pudiste realizar la integracion?

He tratado de todo y nada, llamo desde mi softphone registrado en asterisk y me dice "todas las líneas están ocupadas"

Cualquier ayuda se agradece
 

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