Hi,
I have a problem with a SIP trunk between Asterisk and central OXE Alcatel,
especially sometimes are not received inbound calls with following messages:
-- Executing [...@test:1] AGI("SIP/800-084250f8",
"agi://127.0.0.1/test.agi") in new stack
-- AGI Script agi://127.0.0.1/test.agi completed, returning 0
== Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN'
I configured the sip.conf file:
[800]
type=peer
host=172.XX.XX.XX
username=test
secret=XXXXXXX
insecure=very
context=test
disallow=all
allow=alaw
allow=ulaw
and the extensions.conf file:
exten => 375,1,AGI(agi://127.0.0.1/test.agi)
I attach to this email the sip messages receveid by Asterisk when the
problem occurs.
Thanks for your help.
Best regards,
GP
<--- SIP read from 172.25.51.1:10011 --->
INVITE sip:3...@172.24.10.188;user=phone SIP/2.0
Supported: replaces,100rel
User-Agent: ABS GW v5.1.0
P-Asserted-Identity: "ISDN_T2" <sip:+521776...@mercury>
Content-Type: application/sdp
To: <sip:3...@172.24.10.188;user=phone>
From: "ISDN_T2" <sip:+521776...@mercury>;tag=cc01ff37a60521d35da001f98edda0ac
Contact: sip:172.25.51.1
Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
CSeq: 684819861 INVITE
Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c
Max-Forwards: 70
Content-Length: 314
v=0
o=OXE 1279704517 1279704517 IN IP4 172.25.51.1
s=abs
c=IN IP4 172.25.51.4
t=0 0
m=audio 32712 RTP/AVP 8 0 4 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:30
a=rtpmap:4 G723/8000
a=ptime:30
a=maxptime:30
a=rtpmap:97 telephone-event/8000
<------------->
--- (13 headers 17 lines) ---
Sending to 172.25.51.1 : 5060 (no NAT)
Using INVITE request as basis request -
82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
Found peer '800'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 97
Peer audio RTP is at port 172.25.51.4:32712
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 97
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xd
(g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.25.51.4:32712
Looking for 375 in sedoc (domain 172.24.10.188)
list_route: hop: <sip:172.25.51.1>
<--- Transmitting (no NAT) to 172.25.51.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1
From: "ISDN_T2" <sip:+521776...@mercury>;tag=cc01ff37a60521d35da001f98edda0ac
To: <sip:3...@172.24.10.188;user=phone>
Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
CSeq: 684819861 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:3...@172.24.10.188>
Content-Length: 0
<------------>
-- Executing [...@sedoc:1] AGI("SIP/800-084250f8",
"agi://127.0.0.1/mercury.agi") in new stack
-- AGI Script agi://127.0.0.1/mercury.agi completed, returning 0
== Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN'
Scheduling destruction of SIP dialog
'82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 172.25.51.1:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1
From: "ISDN_T2" <sip:+521776...@mercury>;tag=cc01ff37a60521d35da001f98edda0ac
To: <sip:3...@172.24.10.188;user=phone>;tag=as3455cb36
Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
CSeq: 684819861 INVITE
ser-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
ccsedoc*CLI>
<--- SIP read from 172.25.51.1:10011 --->
ACK sip:3...@172.24.10.188;user=phone SIP/2.0
Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1
From: "ISDN_T2" <sip:+521776...@mercury>;tag=cc01ff37a60521d35da001f98edda0ac
To: <sip:3...@172.24.10.188;user=phone>;tag=as3455cb36
Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c
CSeq: 684819861 ACK
Content-Length: 0
<------------->
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
