Bom dia,
Estamos participando da integração de um asterisk 1.6.2.11 com uma
Alcatel via SIP. Quando ligo do asterisk para a Alcatel recebo um
forwarding da chamada de volta para o asterisk. Segue maiores
informações:
sip.conf:
[alcatel]
host=172.16.1.3
context=from-Alcatel
type=friend
nat=no
disallow=all
allow=alaw
extensions.conf
exten => _6X.,1,Dial(SIP/${EXTEN:1}@alcatel)
same => n,HangUp
no console do asterisk durante a chamada
-- Executing [69202@saida:1] Dial("SIP/jefferson-00001a43",
"SIP/9202@alcatel") in new stack
== Using SIP RTP CoS mark 5
-- Called 9202@alcatel
-- Now forwarding SIP/jefferson-00001a43 to
'Local/9202@from-Alcatel' (thanks to SIP/alcatel-00001a44)
[Feb 22 09:36:24] NOTICE[19305]: chan_local.c:538 local_call: No such
extension/context 9202@from-Alcatel while calling Local channel
[Feb 22 09:36:24] NOTICE[19305]: app_dial.c:789 do_forward: Failed to
dial on local channel for call forward to '9202@from-Alcatel'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [69202@saida:2] Hangup("SIP/jefferson-00001a43", "")
in new stack
== Spawn extension (TI, 69202, 2) exited non-zero on
'SIP/jefferson-00001a43'
Segue sip debug deste peer
asterisk*CLI> sip set debug peer alcatel
SIP Debugging Enabled for IP: 172.16.1.3:5060
== Using SIP RTP CoS mark 5
-- Executing [69202@TI:1] Dial("SIP/jefferson-00001a46",
"SIP/9202@alcatel") in new stack
== Using SIP RTP CoS mark 5
Audio is at 172.16.200.92 port 5404
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.1.3:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport
Max-Forwards: 70
From: "jefferson" <sip:[email protected]>;tag=as1b46e101
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Fri, 22 Feb 2013 12:38:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 303517300 303517300 IN IP4 172.16.200.92
s=Asterisk PBX 1.6.2.11
c=IN IP4 172.16.200.92
t=0 0
m=audio 5404 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called 9202@alcatel
<--- SIP read from UDP:172.16.1.3:5060 --->
SIP/2.0 100 Trying
To: <sip:[email protected]>
From: "jefferson" <sip:[email protected]>;tag=as1b46e101
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:172.16.1.3:5060 --->
SIP/2.0 100 Trying
To: <sip:[email protected]>
From: "jefferson" <sip:[email protected]>;tag=as1b46e101
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:172.16.1.3:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:172.16.1.3;lr;transport=UDP>
Via: SIP/2.0/UDP
172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d
Via: SIP/2.0/UDP
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Max-Forwards: 69
From: "jefferson" <sip:[email protected]>;tag=as1b46e101
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.11
Date: Fri, 22 Feb 2013 12:38:28
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO
Supported: replaces,timer
Content-Type: application/sdp
Content-Length: 236
Session-Expires: 1800
v=0
o=root 303517300 303517300 IN IP4 172.16.1.3
s=Asterisk PBX 1.6.2.11
c=IN IP4 172.16.1.3
t=0 0
m=audio 5404 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (17 headers 11 lines) ---
<--- Transmitting (no NAT) to 172.16.1.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.16.1.3:5060;branch=z9hG4bK467b772a9f9ef266e7d514eede2bf348cc48719d6f59fafe0690167af90cbb1d;received=172.16.1.3
Via: SIP/2.0/UDP
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Record-Route: <sip:172.16.1.3;lr;transport=UDP>
From: "jefferson" <sip:[email protected]>;tag=as1b46e101
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
-- Now forwarding SIP/jefferson-00001a46 to
'Local/9202@from-Alcatel' (thanks to SIP/alcatel-00001a47)
[Feb 22 09:38:28] NOTICE[19309]: chan_local.c:538 local_call: No such
extension/context 9202@from-Alcatel while calling Local channel
[Feb 22 09:38:28] NOTICE[19309]: app_dial.c:789 do_forward: Failed to
dial on local channel for call forward to '9202@from-Alcatel'
Scheduling destruction of SIP dialog
'[email protected]' in 32000 ms (Method:
INVITE)
Reliably Transmitting (no NAT) to 172.16.1.3:5060:
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.16.200.92:5060;branch=z9hG4bK7b79bc0e;rport
Max-Forwards: 70
From: "jefferson" <sip:[email protected]>;tag=as1b46e101
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0
---
Scheduling destruction of SIP dialog
'[email protected]' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [69202@TI:2] Hangup("SIP/jefferson-00001a46", "") in
new stack
== Spawn extension (TI, 69202, 2) exited non-zero on
'SIP/jefferson-00001a46'
<--- SIP read from UDP:172.16.1.3:5060 --->
SIP/2.0 200 OK
To: <sip:[email protected]>
From: "jefferson" <sip:[email protected]>;tag=as1b46e101
Call-ID: [email protected]
CSeq: 102 CANCEL
Via: SIP/2.0/UDP
172.16.200.92:5060;received=172.16.200.92;branch=z9hG4bK7b79bc0e;rport=5060
Content-Length: 0
<------------->
--
[]'s Jefferson B. Limeira
[email protected]
(41) 9928-8628
_______________________________________________
EBS MODULAR: 3 slots para combinação entre E1, GSM, FXS ou FXO;
Linha de PORTEIROS IP, abrem até 2 dispositivos com acesso IP remoto;
Conheça esses e outros LANÇAMENTOS KHOMP em www.Khomp.com
_______________________________________________
DIGIVOICE Fabricante de Placas de Voz e Channel Bank
20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
Centro Treinamento - Curso de PABX IP - Asterisk - Site www.digivoice.com.br
_______________________________________________
ALIGERA Fabricante nacional de Gateways SIP-E1 para R2, ISDN e SS7.
Placas de 1E1, 2E1, 4E1 e 8E1 para PCI ou PCI Express.
Channel Bank Appliance Asterisk - Acesse www.aligera.com.br.
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