09-27-2012 05:07 AM - edited 03-16-2019 01:25 PM
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.
Solved! Go to Solution.
09-27-2012 07:25 AM
\dial-peer voice 101 pots
forward-digits all
09-27-2012 07:30 AM
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.
Sent from Cisco Technical Support iPhone App
09-27-2012 05:16 AM
Maybe you have something in config stripping out the called number.
09-27-2012 05:22 AM
Please post your config.
Please rate all useful posts.
Sent from Cisco Technical Support iPhone App
09-27-2012 07:06 AM
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.
09-27-2012 07:25 AM
\dial-peer voice 101 pots
forward-digits all
09-27-2012 07:29 AM
Thank you Paolo.
09-27-2012 07:31 AM
Thanks for the nice rating and good luck!
09-27-2012 07:30 AM
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.
Sent from Cisco Technical Support iPhone App
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