The Lenght Of The Uri Affects On Dialplan?

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Asterisk Users 5 Comments

Hi Gurus.. I use Asterisk for just for ivr. My issue is that when the switch changes it’s host name from MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call is rejected with “No matching peer” and the “handle_request_invite: Sending fake auth rejection for device x”. It doesn’t match it’s own default context.

Also, it has somethig to do with the numbers of digits of the dialed number. Few digits works ok, 14 to more works wrong. Do you know what am i missing?
Thanks in advance.

Debug with long hostname (B is considered as an ‘*’)
================================
<--- SIP read from TCP:10.146.9.70:6240 --->
INVITE sip:B56510123456789012345@SISIVR03.MYDOMAIN.COM.PY;user=phone SIP/2.0
From: ;tag=3016589695
To:
Max-Forwards: 70
Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK00000035391821780096
Call-ID: 9caX8060616182201-AAAABOPA-@MSSASU1.MYDOMAIN.COM.PY
CSeq: 7313 INVITE
P-Asserted-Identity:
Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
P-Charging-Vector: icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY
Supported: 100rel Content-Type: application/sdp Contact:
Content-Length: 414

v=0
o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY
s=-
t=0 0
a=sendrecv m=audio 13802 RTP/AVP 8 96 18 97
c=IN IP4 10.143.1.67
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
— (15 headers 17 lines) –

5 thoughts on - The Lenght Of The Uri Affects On Dialplan?

  • mention the complete scnario and your sip.conf.

    Regards,

    Faisal
    (sent from phone)

    Rafael Visser wrote:

  • Ok…

    sip.conf
    [general]
    context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
    allowoverlap=no ; Disable overlap dialing support. (Default is yes)
    udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
    tcpenable=yes ; Enable server for incoming TCP connections (default is no)
    tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
    srvlookup=yes ; Enable DNS SRV lookups on outbound calls relaxdtmf=yes dtmfmode=inband
    ;rfc2833compensate=yes

    users.conf
    [general]
    fullname = New User userbase = 6000
    hasvoicemail = yes vmsecret = 1234
    hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1
    pickupgroup = 1
    allowguest=no ; Allow or reject guest calls -sin password- (default is yes)

    [sip.ericsson]
    ;cambios allowguest hosts
    ;allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
    type=friend calllimit=200
    fromuser=ivr1
    dtmfmode=inband username=administrador context=incoming-sip-ericsson host=10.146.9.70
    host=ericsson host=MSSASU1.MYDOMAIN.COM.PY
    port=5060
    disallow=all allow=alaw allow=gsm allow=ulaw qualify=yes insecure=no

  • Sorry, the last config was not clear. I replaced for the following sip.conf

    [general]
    context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
    allowoverlap=no ; Disable overlap dialing support. (Default is yes)
    udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
    tcpenable=yes ; Enable server for incoming TCP connections (default is no)
    tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
    srvlookup=yes ; Enable DNS SRV lookups on outbound calls relaxdtmf=yes dtmfmode=inband
    ;rfc2833compensate=yes

    [sip.ericsson]
    ;cambios allowguest hosts allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
    type=friend calllimit=200
    fromuser=ivr1
    dtmfmode=inband username=administrador context=incoming-sip-ericsson host=10.146.9.70
    host=ericsson host=MSSASU1.MYDOMAIN.COM.PY
    port=5060
    disallow=all allow=alaw allow=gsm allow=ulaw qualify=yes insecure=no

    From: rafael_visser@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 26 Aug 2012 19:52:43 -0400
    Subject: Re: [asterisk-users] the lenght of the uri affects on dialplan?

    Ok…

    sip.conf
    [general]
    context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
    allowoverlap=no ; Disable overlap dialing support. (Default is yes)
    udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
    tcpenable=yes ; Enable server for incoming TCP connections (default is no)
    tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
    srvlookup=yes ; Enable DNS SRV lookups on outbound calls relaxdtmf=yes dtmfmode=inband
    ;rfc2833compensate=yes

    users.conf
    [general]
    fullname = New User userbase = 6000
    hasvoicemail = yes vmsecret = 1234
    hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1
    pickupgroup = 1
    allowguest=no ; Allow or reject guest calls -sin password- (default is yes)

    [sip.ericsson]
    ;cambios allowguest hosts
    ;allowguest=no ; Allow or reject guest calls -sin password- (default is yes)
    type=friend calllimit=200
    fromuser=ivr1
    dtmfmode=inband username=administrador context=incoming-sip-ericsson host=10.146.9.70
    host=ericsson host=MSSASU1.MYDOMAIN.COM.PY
    port=5060
    disallow=all allow=alaw allow=gsm allow=ulaw qualify=yes insecure=no

  • Rafael Visser wrote:

    The device sending SIP is using a different protocol (TCP vs. UDP) and port (6240 vs. 5060) in your two examples. You have Asterisk configured to listen for UDP and TCP connections, so the protocol isn’t the problem.

    However, the first example fails to find a matching peer because the
    “sip.ericsson” SIP entity is defined with “portP60” and
    “insecure=no”. Try changing the insecure option to “insecure=port”. This should resolve your problem by allowing the peer to be matched by IP address regardless of the port number.

    Regards,

    Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer