[asterisk-users] incoming SIP call

Thu Apr 19 15:58:20 MST 2007

Well, for outbound calls, the SIP Server challenges the INVITE with 401/407.
Then Re-INVITE is sent which explains why outgoing call works.  It is
possible that the SIP Server doesn’t check to see whether the caller is
Registered.

For inbound call, the SIP server needs to know the gateway contact
information, and it is obtained only through REGISTER (if not statically
configured in the SIP server, which is very unlikely).

Thanks,

Neel

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jean Marc Le
Fevre
Sent: Thursday, April 19, 2007 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] incoming SIP call

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 REGI STER 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jean Marc Le
Fevre
Sent: Wednesday, April 18, 2007 11:15 AM
To: asterisk-users at 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:asterisk at 82.XXX.XXX.XXX>;tag=as01265eaf

To: & lt;sip:freephonie.net>

Contact: <sip:asterisk at 82.XXX.XXX.XXX>

Call-ID: 7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX

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:asterisk at 82.XXX.XXX.XXX>;tag=as372da2cb

To: <sip:freephonie.net>

Contact: <sip:asterisk at 82.XXX.XXX.XXX>

Call-ID: 793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX

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> 

<-- S IP read from 212.27.52.5:5060: 

SIP/2.0 403 not registered

Call-ID: 7263e88c20c9f3
<mailto:7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX>
8c34963cef6704cf07 at 82.XXX.XXX.XXX

CSeq: 102 OPTIONS

From: "asterisk" <sip:asterisk at 82.XXX.XXX.XXX>;tag=as01265eaf

To: <sip:freephonie.net>;tag=00-31057-001dc 208-591e1ca81

Via: SIP/2.0/UDP
82.XXX.XXX.XXX:5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df6
6

Content-Length: 0

--- (7 headers 0 lines) ---

Destroying call '7263e88c20c9f38c34963cef6704cf07 at 82.XXX.XXX.XXX
<mailto:7263e88c20c9f38c34963cef670%0d%0a%204cf07 at 82.XXX.XXX.XXX> '

Zpro*CLI> 

<-- SIP read from 212.27.52.5:5060: 

SIP/2.0 403 not registered

Call-ID: 793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX

CSeq: 102 OPTIONS

From: "asteris k" <sip:asterisk at 82.XXX.XXX.XXX>;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=z9hG4bK253c1a3
d

Content-Length: 0

--- (7 headers 0 lines) ---

Destroying call '793bf24e5290d562787c8d9451baedd7 at 82.XXX.XXX.XXX'

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:SECRET at freephonie.net

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

quali fy=60000

fromdomain=freephonie.net

[freep honie_inbound]

type=peer

context=incoming

host=freephonie.net

qualify=60000

allow=all< /P> 

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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