No extension found ?

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Asterisk Users 5 Comments

Hi

I have a small problems with incoming call.

I have a peer actually configured for outcall :

sip.conf:

[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=port,invite
context=incoming

This SIP account work for outgoing call. when i want receive call from
this sipplier, i have a “extension not found”.

In extensions.conf for incoming:

[incoming]
exten => _X.,1,Dial(IAX2/VoIP/${EXTEN},180,rt)

in dialplan show incoming, no problems i see the dialplan.

when i call, i have:

< --- SIP read from UDP://84.xx.xx.72:5060 —>
INVITE sip:331NUMNOFOUND@78.IPOFMYSERVER:5060 SIP/2.0
Record-Route:
Record-Route:
Record-Route:

Record-Route:

Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: “+331MYCLID”
;tag=2RUVP51HBW30000E1D00001u0K4NFQC0QNAN31
To:

Call-ID:
60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Contact:
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Max-Forwards: 67
P-Asserted-Identity:

Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value=”4f924d2c1e20abe1d@172.16.20.119″
X-PSN-Trunk: ME

v=0
o=- 18406958643964291255 1 IN IP4 172.16.21.11
s=session
c=IN IP4 84.xx.xx.34
t=0 0
m=audio 64296 RTP/AVP 8 18 4 0 105 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=fmtp:4 bitrate=6.3
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes

< ------------->

5 thoughts on - No extension found ?

  • Hi,

    No matching peer for ‘+331MYCLID’ from ’84.xx.xx.72:5060′

    This line is telling you everything. The peer you’ve declared isn’t being
    matched for the incoming call and hence it tries to look in “default”
    context (I assume allowguest=yes in your sip.conf)

    Make sure that your peer is matched, since you’ve qualify=yes defined
    execute the command “sip show peer Trunk-Telco” in asterisl CLI and see the
    status of the peer.

    What I’m guessing is that the telco has multiple IPs to send you calls and
    the incoming call isn’t coming from the IP you’ve declared in your sip
    telco-trunk section. I don’t think we can set a subnet in
    host=87.XX.XX.XX/28 parameter.!!

    Regards,
    Sammy.

  • Thats not gonna work TOOTAi,
    that’s just ACL thing you wrote. The peer IP is only going to be matched
    against the host= field.
    correct me if I’m wrong on this.

  • Le 24/04/2012 12:37, SamyGo a écrit :

    Well from the permit-deny-mask link it say that it works on type=user or
    type=peer. As type=peer can also be used for users, I would expect that
    something like (without being registred):

    [MyTelco]
    type=peer
    host=sip.mytelco.com ;to place call to
    deny=0.0.0.0/0.0.0.0 ;deny all incoming IPs
    permit=1.2.3.4/255.255.255.255 ;but not this one
    permit=5.6.7.8/255.255.255.240 ;but not for this subnet

    If it only match on host field, why are multiple permit field allowed?
    And for what they are usable then?

    Anyway, from other threads I saw it seems that your right … my
    questions stays open 😉

  • Peers are matched against IPs in “host” field, however the permit/deny
    fields restricts the peers in case host=dynamic. That’s what I’ve learned
    so far.

    But for OP I think he definitely needs to define multiple peers for all
    incoming IPs inorder to goto the correct context and match the desired
    extension.