Hello and thanks for answering,
As I just answer to Yuan LIU, what I don't understand, is that I can
place an outbound call from asterisk to a gsm at the same time I
can't get asterisk thought a inbound call. But I'll try what you
advice me.
I'll tell you the result of it
Jean-Marc LE FEVRE
Le 19 avr. 07 à 20:08, Bala Neelakantan a écrit :
If your SIP server loses REGISTERs then it cant place an inbound
SIP call. Try changing the REGISTER frequency to lower value.
When you see incoming SIP call fail, you might want to check
whether the REGISTERs are working.
Thanks,
Neel
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:asterisk-
[EMAIL PROTECTED] On Behalf Of Jean Marc Le Fevre
Sent: Wednesday, April 18, 2007 11:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] incoming SIP call
Hello all,
I'm having a quite simple configuration like:
SIP provider <=> asterisk SIP <=> lan
Everythings works fine but sometime I can't get incoming call.
here are some of the logs from set debug 25 set verbosity 25 sip
show debug and sip.conf and a part of extension.conf
thanks in advance
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;branch= z9hG4bK67c2df66;rport
From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as01265eaf
To: <sip:freephonie.net>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 18 Apr 2007 13:57:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:5060;b ranch=z9hG4bK253c1a3d;rport
From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as372da2cb
To: <sip:freephonie.net>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 18 Apr 2007 13:57:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Zpro*CLI>
<-- SIP read from 212.27.52.5:5060:
SIP/2.0 403 not registered
Call-ID: 7263e88c20c9f3 [EMAIL PROTECTED]
CSeq: 102 OPTIONS
From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as01265eaf
To: <sip:freephonie.net>;tag=00-31057-001dc208-591e1ca81
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66
Content-Length: 0
--- (7 headers 0 lines) ---
Destroying call '[EMAIL PROTECTED]'
Zpro*CLI>
<-- SIP read from 212.27.52.5:5060:
SIP/2.0 403 not registered
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
From: "asteris k" <sip:[EMAIL PROTECTED]>;tag=as372da2cb
To: <sip:freephonie.net>;tag=00-32700-001dc209-6fc2b3303
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK253c1a3d
Content-Length: 0
--- (7 headers 0 lines) ---
Destroying call '[EMAIL PROTECTED]'
sip.conf
[general]
context=incoming
realm=etatcritik.dyndns.org
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
maxexpiry=3600
defaultexpiry=1800
videosupport=yes
disallow=all
all ow=ulaw
allow=ilbc
allow=alaw
allow=gsm
musicclass=default
language=fr
useragent=Asterisk PBX
dtmfmode = auto
register => 09XXXXXXXX:[EMAIL PROTECTED]
registertimeout=40
externip = 82.XXX.XXX.XXX
localnet=10.XXX.XXX.XXX/255.255.255.0
qualify=60000
nat = yes
[test]
type=friend
username=test
secret=test
host=dynamic
context=home
callerid =test <2222>
dmtfmode=rfc2833
authuser=test
fromuser=test
allow=all
[freephonie_outbound]
type=peer
allow=all
host=freephonie.net
secret=SECRET
fromuser=09XXXXXXX
username=09XXXXXXX
dtmfmode=inband
qualify=60000
fromdomain=freephonie.net
[freep honie_inbound]
type=peer
context=incoming
host=freephonie.net
qualify=60000
allow=all
deny=0.0.0.0/0..0.0.0
permit=212.27.52.5/255.255.255.255 ; ip de freephonie.net
etension.conf
...
[incoming]
exten => s,1,Ringing
exten => s,2,Noop(I receive a sip call);
exten => s,n,Goto(home,1000,1)
exten => s,n,Congestion
;
...
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