Asterisk como gateway

**navaismo** · 02-20-2013, 07:12 PM

Si es posible. Solo asegúrate de que el viejo pbx igual use una E1 y listo.

**IOCAPP** · 02-22-2013, 11:14 AM

Hola buen dia ya crealize el trunk con el conmutador
y estos son algunos datos

El esquema es

E1 --- ASTERISK PBX ------ ==========3COM- --- PHONES 3COM
TRUNK SIP

sip_additional.conf

[pbx]
disallow=all
allow=g729
dtmfmode=rfc2833
context=incoming-pbx
host=10.10.200.253
type=peer

y mi diaplan es el siguiente

ivr

[incoming-pbx]

exten => 3000,1,Wait(1) ;espera un segunfo
exten => 3000,2,Set(TIMEOUT(digit)=7) ;
exten => 3000,3,Background(bienvenida)
exten => 3000,n,Background(ext-or-zero)

exten => 3000,n,WaitExten()
exten => 0,1,Dial(SIP/3061@10.10.200.253:5060,30)
exten => _3XXX,1,Dial(SIP/${EXTEN}@10.10.200.253:5060,30)
exten => _3XXX,2,Hangup

Salida

[sip-pbx]
exten => _9XXXXXXX,1,Dial(DAHDI/g0/${EXTEN:1},90,Tt)
exten => _9XXXXXXX,n,hangup()

El detalle ahora es que puedo tano recibir y realizar llamadas desde las extensiones conectadas al conmutador 3com

1.- Pero cuando se trasnfiere una llamada no se escuhcha mada del MOH musiconhold
2.- las llamadas que se reciben
si los telefonos 3com cuelgan en el tel del llamnte no se escucha que xcuelguen

es decir no hay hangup
pero si cuelga el llamnet no hay problema

Y si se realiza una llamada de los tels 3com hacia afuera no hay problema si hay hangu

este es un degug de una netrada de una llamada
entrante

---
[Feb 22 10:13:17] VERBOSE[30996] app_dial.c: -- Called SIP/3166@10.10.200.253:5060
[Feb 22 10:13:17] VERBOSE[25194] chan_sip.c:
<--- SIP read from UDP:10.10.200.253:5060 --->
SIP/2.0 100 Trying
From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b
To: <sip:3166@10.10.200.253:5060>
Call-Id: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0
Cseq: 102 INVITE
Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK33c3146d;rport=50 60
Content-Length: 0

<------------->
[Feb 22 10:13:17] VERBOSE[25194] chan_sip.c: --- (7 headers 0 lines) ---
[Feb 22 10:13:17] VERBOSE[25194] chan_sip.c:
<--- SIP read from UDP:10.10.200.253:5060 --->
SIP/2.0 180 Ringing

From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b
To: <sip:3166@10.10.200.253:5060>
Call-Id: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0
Cseq: 102 INVITE
Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK33c3146d;rport=50 60
Date: Fri, 22 Feb 2013 16:13:11 GMT
Contact: sip:10.10.200.253
User-Agent: 3Com-SIP-Phone/v2.6.0 (sipXphone) (VxWorks)
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------->
[Feb 22 10:13:17] VERBOSE[25194] chan_sip.c: --- (13 headers 0 lines) ---
[Feb 22 10:13:17] VERBOSE[25194] chan_sip.c: list_route: hop: <sip:10.10.200.253>
[Feb 22 10:13:17] VERBOSE[30996] app_dial.c: -- SIP/10.10.200.253:5060-0000020f is ringing
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c:
<--- SIP read from UDP:10.10.200.253:5060 --->
SIP/2.0 200 OK
From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b
To: <sip:3166@10.10.200.253:5060>;tag=1361549593
Call-Id: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0
Cseq: 102 INVITE
Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK33c3146d;rport=50 60
Content-Type: application/sdp
Content-Length: 225
P-Asserted-Identity: "SITE Central"<sip:3166@10.10.200.253>
Contact: "3ComCallProcessor"<sip:3ComCallProcessor@10.10.20 0.253:5060;transport=UDP>
Date: Fri, 22 Feb 2013 16:13:13 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE, NOTIFY
User-Agent: 3Com-SIP-Phone/v2.6.0 (sipXphone) (VxWorks)
Accept-Language: en
Supported: replaces

v=0
o=- 0 0 IN IP4 10.10.200.138
s=NIP Session
c=IN IP4 10.10.200.138
t=0 0
m=audio 8048 RTP/AVP 18 101
a=SilenceSuppff
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: --- (15 headers 11 lines) ---
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Found RTP audio format 18
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Found RTP audio format 101
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Found audio description format G729 for ID 18
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)

[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Peer audio RTP is at port 10.10.200.138:8048
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: list_route: hop: <sip:3ComCallProcessor@10.10.200.253:5060;transpor t=UDP>
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: set_destination: Parsing <sip:3ComCallProcessor@10.10.200.253:5060;transpor t=UDP> for address/port to send to
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: set_destination: set destination to 10.10.200.253:5060
[Feb 22 10:13:19] VERBOSE[25194] chan_sip.c: Transmitting (NAT) to 10.10.200.253:5060:
ACK sip:3ComCallProcessor@10.10.200.253:5060;transport =UDP SIP/2.0^M
Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK4ff04ab2;rport^M
Max-Forwards: 70^M
From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b^M
To: <sip:3166@10.10.200.253:5060>;tag=1361549593^M
Contact: <sip:PBX AST@10.10.200.210:5060>^M
Call-ID: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0^M
CSeq: 102 ACK^M
User-Agent: Asterisk PBX 1.8.11.0^M
Content-Length: 0^M
[Feb 22 10:13:19] VERBOSE[30996] app_dial.c: -- SIP/10.10.200.253:5060-0000020f answered DAHDI/1-1
[Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: bits changed in chan 1
[Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - Bits changed from 0x00 to 0x08
[Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - CAS Rx << [CLEAR FORWARD] 0x08
[Feb 22 10:13:36] VERBOSE[30996] chan_dahdi.c: Chan 1 - Far end disconnected. Reason: Normal Clearing
[Feb 22 10:13:36] VERBOSE[30996] chan_dahdi.c: MFC/R2 call disconnected on channel 1
[Feb 22 10:13:36] VERBOSE[30996] pbx.c: -- Executing [h@incoming-pbx:1] Hangup("DAHDI/1-1", "") in new stack
[Feb 22 10:13:36] VERBOSE[30996] features.c: == Spawn extension (incoming-pbx, h, 1) exited non-zero on 'DAHDI/1-1'
[Feb 22 10:13:36] VERBOSE[30996] chan_sip.c: Scheduling destruction of SIP dialog '1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:50 60' in 32000 ms (Method: INVITE)
[Feb 22 10:13:36] VERBOSE[30996] chan_sip.c: set_destination: Parsing <sip:3ComCallProcessor@10.10.200.253:5060;transpor t=UDP> for address/port to send to
[Feb 22 10:13:36] VERBOSE[30996] chan_sip.c: set_destination: set destination to 10.10.200.253:5060
[Feb 22 10:13:36] VERBOSE[30996] chan_sip.c: Reliably Transmitting (NAT) to 10.10.200.253:5060:
BYE sip:3ComCallProcessor@10.10.200.253:5060;transport =UDP SIP/2.0^M
Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK45c1d39f;rport^M
Max-Forwards: 70^M
From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b^M
To: <sip:3166@10.10.200.253:5060>;tag=1361549593^M
Call-ID: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0^M
CSeq: 103 BYE^M
User-Agent: Asterisk PBX 1.8.11.0^M

---
[Feb 22 10:13:36] VERBOSE[30996] pbx.c: == Spawn extension (incoming-pbx, 3166, 1) exited non-zero on 'DAHDI/1-1'
[Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: disconnecting MFC/R2 call on chan 1
[Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: ast cause 16 resulted in openr2 cause 6/Normal Clearing
[Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - Call ended
[Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - CAS Tx >> [IDLE] 0x08
[Feb 22 10:13:36] DEBUG[30996] chan_dahdi.c: Chan 1 - CAS Raw Tx >> 0x09
[Feb 22 10:13:36] VERBOSE[30996] chan_dahdi.c: MFC/R2 call end on channel 1
[Feb 22 10:13:36] VERBOSE[30996] chan_dahdi.c: -- Hungup 'DAHDI/1-1'
[Feb 22 10:13:36] VERBOSE[25194] chan_sip.c:
<--- SIP read from UDP:10.10.200.253:5060 --->
SIP/2.0 100 Trying
From: "PBX AST" <sip:PBX AST@10.10.200.210>;tag=as7f9a4d5b
To: <sip:3166@10.10.200.253:5060>;tag=1361549593
Call-Id: 1f1ef7857baf4a3a526fccf66358636c@10.10.200.210:506 0
Cseq: 103 BYE
Via: SIP/2.0/UDP 10.10.200.210:5060;branch=z9hG4bK45c1d39f;rport=50 60
Content-Length: 0

<------------->

Espero me puedan ayudara resolver el problema

Saludos

Asterisk como gateway

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