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Phibee Network Operation Center

2009-Oct-28 05:09 UTC

[asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)

Hi

Now, my Cisco AS5300 sent call to my asterisk, but two problems:

When i call the phone number, i have:

[Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: 
Call from '' to extension '0426000000' rejected because
extension not found.
[Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: 
Call from '' to extension '0426000000' rejected because
extension not found.

(0426000000 = my phone number)



First problems:
   
    Why he don't see the extension ?

sip.conf:

[AS5300]
host=192.168.50.125
context=as5300-incoming
type=peer
dtmf=rfc2833
nat=no
canreinvite=yes
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw


extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp

[as5300-incoming]
        exten => 0426000000,1,Ringing
        exten => 0426000000,2,Answer
        exten => 0426000000,3,Dial(SIP/Jpc,25,m)
        exten => 0426000000,4,Hangup



And second problems:

    "Call from '' to", AS5300 don't sent the number of the
caller ?


Thanks for your help
Jerome

Phibee Network Operation Center

2009-Oct-28 06:16 UTC

head link

[asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)

Phibee Network Operation Center a ?crit :> Hi
>
> Now, my Cisco AS5300 sent call to my asterisk, but two problems:
>
> When i call the phone number, i have:
>
> [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: 
> Call from '' to extension '0426000000' rejected because
extension not found.
> [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: 
> Call from '' to extension '0426000000' rejected because
extension not found.
>
> (0426000000 = my phone number)
> <..>
>   
I have put a debug:

[Kvoip*CLI>
<--- SIP read from UDP://192.168.50.125:59124 --->
INVITE sip:0426000000 at 192.168.50.130:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.50.125:5060
From: <sip:477000000 at 192.168.50.125>;tag=6950F0-25C7
To: <sip:0426000000 at 192.168.50.130>
Date: Wed, 28 Oct 2009 05:16:26 GMT
Call-ID: E02F04A1-C2B711DE-82B09FC7-B045D36F at 192.168.50.125
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 3761097657-3266777566-2192416711-2957366127
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: 
<sip:477000000 at 192.168.50.125>;party=calling;screen=yes;privacy=off
Timestamp: 1256706986
Contact: <sip:477000000 at 192.168.50.125:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 8642 2741 IN IP4 192.168.50.125
s=SIP Call
c=IN IP4 192.168.50.125
t=0 0
m=audio 18726 RTP/AVP 8 101
c=IN IP4 192.168.50.125
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
[Kvoip*CLI> --- (20 headers 11 lines) ---
[Kvoip*CLI> Sending to 192.168.50.125 : 5060 (no NAT)
[Kvoip*CLI> Using INVITE request as basis request - 
E02F04A1-C2B711DE-82B09FC7-B045D36F at 192.168.50.125
[Kvoip*CLI> No matching peer for '477000000' from
'192.168.50.125:59124'
[Kvoip*CLI> Found RTP audio format 8
[Kvoip*CLI> Found RTP audio format 101
[Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726
[Kvoip*CLI> Found audio description format PCMA for ID 8
[Kvoip*CLI> Found audio description format telephone-event for ID 101
[Kvoip*CLI> Got unsupported a:fmtp in SDP offer
[Kvoip*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - 
audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 
(alaw)
[Kvoip*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), 
peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Kvoip*CLI> Peer audio RTP is at port 192.168.50.125:18726
[Kvoip*CLI> Looking for 0426000000 in default (domain 192.168.50.130)
[Kvoip*CLI> <--- Reliably Transmitting (no NAT) to 192.168.50.125:5060
--->
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP  192.168.50.125:5060;received=192.168.50.125
From: <sip:477000000 at 192.168.50.125>;tag=6950F0-25C7
To: <sip:0426000000 at 192.168.50.130>;tag=as25696e60
Call-ID: E02F04A1-C2B711DE-82B09FC7-B045D36F at 192.168.50.125
CSeq: 101 INVITE

Server: Asterisk PBX 1.6.1.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

Ok, i see that:

    1- Cisco sent the phone number of the caller (477000000)
    2- I have a "To: <sip:0426000000 at 192.168.50.130>"
       192.168.50.130 = My Asterisk Server
       192.168.50.125 = My Cisco AS5300
    3- i have a "No matching peer for '477000000' from 
'192.168.50.125:59124'"
       why he search a peer with "477000000" ??

bye
Jerome

Alex Balashov

2009-Oct-28 10:41 UTC

head link

[asterisk-users] Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)

Double-check the IP and port associated with the AS5300 peer.  The 
messages below indicate that calls coming in from it are not being 
matched to the right peer, and as a result, not routed to the correct 
dial plan context.

Phibee Network Operation Center wrote:
> Hi
> 
> Now, my Cisco AS5300 sent call to my asterisk, but two problems:
> 
> When i call the phone number, i have:
> 
> [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: 
> Call from '' to extension '0426000000' rejected because
extension not found.
> [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: 
> Call from '' to extension '0426000000' rejected because
extension not found.
> 
> (0426000000 = my phone number)
> 
> 
> 
> First problems:
>    
>     Why he don't see the extension ?
> 
> sip.conf:
> 
> [AS5300]
> host=192.168.50.125
> context=as5300-incoming
> type=peer
> dtmf=rfc2833
> nat=no
> canreinvite=yes
> dtmfmode=rfc2833
> disallow=all
> allow=g729
> allow=alaw
> 
> 
> extensions.conf:
> [general]
> static=yes
> writeprotect=no
> autofallthrough=yes
> clearglobalvars=no
> priorityjumping=no
> 
> [globals]
> CONSOLE=Console/dsp
> 
> [as5300-incoming]
>         exten => 0426000000,1,Ringing
>         exten => 0426000000,2,Answer
>         exten => 0426000000,3,Dial(SIP/Jpc,25,m)
>         exten => 0426000000,4,Hangup
> 
> 
> 
> And second problems:
> 
>     "Call from '' to", AS5300 don't sent the number
of the caller ?
> 
> 
> Thanks for your help
> Jerome
> 
> 
> 
> 
> 
> 
> _______________________________________________
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-- 
Alex Balashov - Principal
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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