Sip to ISDN, called not present

[david.sua]

Hi all,

I have a CUBE connected to an asterisk.

In the CUBE I have a VIC-2BRI connected to a PBX.

In the PBX I have a user with its own phone number, 941010714, that another than main number.

Asterisk -------IP NETWORK----------- CUBE ---------ISDN--------- PBX

Ok, come on with my problem.

When I place a call from Asterisk to PBX 941010714 I can see calling and called number in SIP packets into CUBE but I can't see called number into ISDN.

Router#debug ccsip messages

INVITE sip:941010714@X.X.X.X:5060 SIP/2.0

Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bK05920d4e;rport

Max-Forwards: 70

From: "Knet Comunicaciones" <sip:941519151@Y.Y.Y.Y>;tag=as4c18f8d4

To: <sip:941010714@X.X.X.X:5060>

Router#debug isdn q931

        Bearer Capability i = 0x8090A2

                Standard = CCITT

                Transfer Capability = Speech

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0x89

                Exclusive, B1

        Display i = 'Knet Comunicaciones'

        Calling Party Number i = 0x0080, '941519151'

                Plan:Unknown, Type:Unknown

        Called Party Number i = 0x80

                Plan:Unknown, Type:Unknown

Somebody can help me?

Regards.

[paolo bevilacqua]

Maybe you have something in config stripping out the called number.

[Roger Kallberg]

Please post your config.

Please rate all useful posts.

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[david.sua]

Here is my config.

version 12.4

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol cisco

h323

sip

  registrar server

  no update-callerid

  sip-profiles 1000

!

voice class sip-profiles 1000

request ANY sdp-header Connection-Info remove

response ANY sdp-header Connection-Info remove

!

interface BRI0/0/0

no ip address

isdn switch-type basic-net3

isdn protocol-emulate network

isdn layer1-emulate network

isdn incoming-voice voice

isdn map address transparent

isdn skipsend-idverify

line-power

!

voice-port 0/0/0

cptone ES

!

dial-peer voice 3006 voip

description Llamadas entrantes SIP

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

incoming called-number 941010714

dtmf-relay rtp-nte

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

!

dial-peer voice 101 pots

destination-pattern 941010714

direct-inward-dial

port 0/0/0

no sip-register

!

sip-ua

credentials username 941010714 password 7 xxx realm asterisk

authentication username 941010714 password 7 xxx realm asterisk

no remote-party-id

retry invite 2

retry register 10

timers connect 100

registrar ipv4:Y.Y.Y.Y:5060 expires 3600

sip-server ipv4:Y.Y.Y.Y:5060

host-registrar

!

Thank you.

[paolo bevilacqua · Accepted solution]

\dial-peer voice 101 pots
 forward-digits all

[david.sua]

Thank you Paolo.

[paolo bevilacqua]

Thanks for the nice rating and good luck!

[Roger Kallberg · Accepted solution]

The problem is on your dial peer that points to ISDN.

dial-peer voice 101 pots

destination-pattern 941010714

direct-inward-dial

port 0/0/0

no sip-register

By default a directly match digit in the destination pattern is dropped. To stop this you need to add the command no digit-strip or forward-digits all.

Please rate all useful posts.

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