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Jean Marc Le Fevre

2007-Apr-18 09:14 UTC

[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@82.XXX.XXX.XXX>;tag=as01265eaf
To: <sip:freephonie.net>
Contact: <sip:asterisk@82.XXX.XXX.XXX>
Call-ID: 7263e88c20c9f38c34963cef6704cf07@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;branch=z9hG4bK253c1a3d;rport
From: "asterisk" <sip:asterisk@82.XXX.XXX.XXX>;tag=as372da2cb
To: <sip:freephonie.net>
Contact: <sip:asterisk@82.XXX.XXX.XXX>
Call-ID: 793bf24e5290d562787c8d9451baedd7@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>
<-- SIP read from 212.27.52.5:5060:
SIP/2.0 403 not registered
Call-ID: 7263e88c20c9f38c34963cef6704cf07@82.XXX.XXX.XXX
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk@82.XXX.XXX.XXX>;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 '7263e88c20c9f38c34963cef6704cf07@82.XXX.XXX.XXX'
Zpro*CLI>
<-- SIP read from 212.27.52.5:5060:
SIP/2.0 403 not registered
Call-ID: 793bf24e5290d562787c8d9451baedd7@82.XXX.XXX.XXX
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk@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=z9hG4bK253c1a3d
Content-Length: 0

--- (7 headers 0 lines) ---
Destroying call '793bf24e5290d562787c8d9451baedd7@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
allow=ulaw
allow=ilbc
allow=alaw
allow=gsm
musicclass=default
language=fr
useragent=Asterisk PBX
dtmfmode = auto
register => 09XXXXXXXX:SECRET@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
qualify=60000
fromdomain=freephonie.net
[freephonie_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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Yuan LIU

2007-Apr-18 15:39 UTC

head link

[asterisk-users] incoming SIP call

>From: Jean Marc Le Fevre <jm.lefevre@etatcritik.dyndns.org>
>Date: Wed, 18 Apr 2007 18:14:41 +0200
>
>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.
Define "sometimes" and from where the income call you can't get?
>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
>[good stuff sniffed]
Where do you suspect the error message is?
>---
>Zpro*CLI>
><-- SIP read from 212.27.52.5:5060:
>SIP/2.0 403 not registered
Does this message make sense, "not registered"?

Yuan Liu
>Call-ID: 7263e88c20c9f38c34963cef6704cf07@82.XXX.XXX.XXX
>CSeq: 102 OPTIONS
>From: "asterisk"
<sip:asterisk@82.XXX.XXX.XXX>;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 '7263e88c20c9f38c34963cef6704cf07@82.XXX.XXX.XXX'
>Zpro*CLI>
><-- SIP read from 212.27.52.5:5060:
>SIP/2.0 403 not registered
>Call-ID: 793bf24e5290d562787c8d9451baedd7@82.XXX.XXX.XXX
>CSeq: 102 OPTIONS
>From: "asterisk"
<sip:asterisk@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=z9hG4bK253c1a3d
>Content-Length: 0
>
>--- (7 headers 0 lines) ---
>Destroying call '793bf24e5290d562787c8d9451baedd7@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
>allow=ulaw
>allow=ilbc
>allow=alaw
>allow=gsm
>musicclass=default
>language=fr
>useragent=Asterisk PBX
>dtmfmode = auto
>register => 09XXXXXXXX:SECRET@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
>qualify=60000
>fromdomain=freephonie.net
>[freephonie_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
>;
>...
>
>
>
>
>
>
>
>
>!DSPAM:462643f450705772331342!
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Bala Neelakantan

2007-Apr-19 11:08 UTC

head link

[asterisk-users] incoming SIP call

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: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] 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:asterisk@82.XXX.XXX.XXX>;tag=as01265eaf

To: <sip:freephonie.net>

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

Call-ID: 7263e88c20c9f38c34963cef6704cf07@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@82.XXX.XXX.XXX>;tag=as372da2cb

To: <sip:freephonie.net>

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

Call-ID: 793bf24e5290d562787c8d9451baedd7@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> 

<-- SIP read from 212.27.52.5:5060: 

SIP/2.0 403 not registered

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

CSeq: 102 OPTIONS

From: "asterisk" <sip:asterisk@82.XXX.XXX.XXX>;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=z9hG4bK67c2df6
6

Content-Length: 0

 

 

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

Destroying call '7263e88c20c9f38c34963cef6704cf07@82.XXX.XXX.XXX'

Zpro*CLI> 

<-- SIP read from 212.27.52.5:5060: 

SIP/2.0 403 not registered

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

CSeq: 102 OPTIONS

From: "asteris k" <sip:asterisk@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@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@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

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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