No Unity Connection access from PSTN

Paul Freiberg · ‎07-20-2015

Hi all,

I have a frustrating probleme.

UCv10.5.2 ---SIPS----CUCMv10.5----SIPS-----CUBEv15.4-----SIPS------ITSP------PSTN

And SRTP.

I can access the UC from an intern SIP-Client, encrypted. But I can not access it from PSTN.

The Phone rings but:

The difference between an intern call and an extern call to UC I mentioned was that CUCM do not send the ciphers to UC in the [8]: ACK an in the [9]: 200 OK. And debug ccsip error shows: Unexpected VoIPCodec Type :No Codec.

The consequence is the [12]: BYE with "Reason: Q.850; cause=65"

I hope someone can help me.

Paul Freiberg

Paul Freiberg · ‎07-22-2015

Hi,

now I got the error Code "Reason: Q.850; cause=47" on [10]: BYE

And still this debug ccsip error output:

Jul 22 15:06:51.277 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free:
Freeing NULL pointer!
Jul 22 15:06:51.317 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free:
Freeing NULL pointer!
SIP: (138580) Attribute mid, level 1 instance 1 not found.
SIP: (138580) setup attribute, level 1 instance 1 not found.
SIP: (138580) connection attribute, level 1 instance 1 not found.
SIP: (138580) Attribute label, level 1 instance 1 not found.
SIP: (138580) a=framerate attribute, level 1 instance 1 not found.
SIP: (138580) Attribute ptime, level 1 instance 1 not found.
Jul 22 15:06:51.325 LUEBECK: //138581/5A27608BB98C/SIP/Error/sipSPIGetCallServerGroupTargets:
No server group configured
Jul 22 15:06:51.325 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Error/voipCodec_to_rtpAvpCodec:
Unexpected VoIPCodec Type :No Codec
SIP: (138581) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: (138581) Group (a= group line) attribute, level 65535 instance 1 not found.
SIP: (138581) Group (a= group line) attribute, level 65535 instance 1 not found.
Jul 22 15:06:51.349 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_search_reg_number_table:
No entry found in reg Number Table for +49451500XXXX
isr2#
Jul 22 15:06:51.349 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_reg_search_reg_number_table:
No entry found in reg Number Table for +494513101XXXX
Jul 22 15:06:51.349 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr:
ReqLine IP addr does not match with host IP addr
SIP: Attribute mid, level 1 instance 1 not found.
SIP: setup attribute, level 1 instance 1 not found.
SIP: connection attribute, level 1 instance 1 not found.
SIP: Attribute label, level 1 instance 1 not found.
SIP: a=framerate attribute, level 1 instance 1 not found.
Jul 22 15:06:51.573 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Error/httpish_msg_free:
Freeing NULL pointer!

Paul Freiberg · ‎07-23-2015

I compared A Call from an intern SIP-Phone with UC with a call from extern.

During an intern Call, INVITE [1] and [3] and OK [7] and [9] transmits ciphers (a=crypto:XXXXXXXXXXXXX...)

During an call from extern to UC, OK [7] does not. I think thats why CUCM sends the [10]:BYE with Reason: Q.850; cause=47.

Is it a bug?

Paul Freiberg · ‎07-23-2015

I found it and I am not impressed.

I had to change srtp-auth back to sha1-32 on the cube. I just sha1-80. Now it works.

So it seems that UC just supports sha1-32 :(

Or is there a way to change it to sha1-80, because the cube and the itsp supports it?

Paul Freiberg · ‎07-27-2015

Hi,

here is my cube config:

version 15.4

voice-card 0
dspfarm
dsp services dspfarm
!
voice service voip
ip address trusted list
ipv4 <publisher ip>
ipv4 <subscriber1 ip>
ipv4 <subscriber2 ip>
ipv4 <itsp ip>
ipv4 <itsp ip>
ipv4 <itsp ip>
ipv4 <itsp ip>
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redundancy
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
srtp-auth sha1-80 sha1-32
early-offer forced
no silent-discard untrusted
midcall-signaling passthru
registration passthrough
reset timer expires 183
!
voice class uri uzl sip
host uni-luebeck.de
voice class codec 1
codec preference 1 g722-64
codec preference 2 g722-56
codec preference 3 g722-48
codec preference 4 g711ulaw
codec preference 5 g711alaw
!
voice class sip-profiles 100
request ANY sip-header Via modify "sips" "sip"
request ANY sip-header From modify "sips" "sip"
response ANY sip-header From modify "sips" "sip"
response ANY sip-header Via modify "sips" "sip"
response ANY sip-header To modify "sips" "sip"
request ANY sip-header Contact modify "sips" "sip"
request ANY sip-header From modify "sips" "sip"
request ANY sip-header Via modify "sips" "sip"
request ANY sip-header Requested-By modify "sips" "sip"
request ANY sip-header To modify "sips:49" "sip:+49"
request ANY sip-header SIP-Req-URI modify "sips:49" "sip:+49"
request ANY sip-header SIP-Req-URI modify "sips" "sip"
request ANY sdp-header Audio-Attribute modify "a=direction:passive" ""
request ANY sdp-header Attribute modify "a=direction:passive" ""
!
license udi pid CISCO2911/K9 sn FGL19231160
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
track 1 interface GigabitEthernet0/0 line-protocol
!
track 2 interface GigabitEthernet0/1 line-protocol
!
interface GigabitEthernet0/0
description Uplink VoIP-DFN
ip address *.*.*.6 255.255.255.240
standby delay minimum 30 reload 60
standby version 2
standby 0 ip *.*.*.4
standby 0 priority 60
standby 0 preempt delay minimum 10
standby 0 name DFN
standby 0 track 2 decrement 20
duplex auto
speed auto
!
interface GigabitEthernet0/1
description Uplink CUCM
ip address *.*.*.6 255.255.255.0
standby delay minimum 30 reload 60
standby version 2
standby 6 ip *.*.*.4
standby 6 priority 60
standby 6 preempt delay minimum 10
standby 6 track 1 decrement 20
duplex auto
speed auto
!
ip rtcp report interval 6000
ip route 0.0.0.0 0.0.0.0 *.*.*.1
ip route *.*.*.0 255.255.255.0 *.*.*.1
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/1
sccp ccm *.*.*.4 identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register TRANSCODER
!
dspfarm profile 1 transcode universal security
trustpoint DSP-SEC
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g722-64
maximum sessions 16
associate application SCCP
!
dial-peer voice 1 voip
preference 1
destination-pattern .T
session protocol sipv2
session target ipv4:*.*.*.8:5062
session transport tcp tls
voice-class codec 1
voice-class sip url sips
voice-class sip options-ping 60
voice-class sip profiles 100
no voice-class sip anat
srtp
no vad
!
dial-peer voice 2 voip
preference 2
destination-pattern .T
session protocol sipv2
session target ipv4:*.*.*.40:5062
session transport tcp tls
voice-class codec 1
voice-class sip url sips
voice-class sip options-ping 60
voice-class sip profiles 100
no voice-class sip anat
srtp
no vad
!
dial-peer voice 101 voip
preference 1
destination-pattern +494513101T
session protocol sipv2
session target ipv4:*.*.*.101:5061
session transport tcp tls
destination uri uzl
incoming uri from uzl
voice-class codec 1
voice-class sip options-ping 60
srtp
no vad
!
dial-peer voice 102 voip
preference 2
destination-pattern +494513101T
session protocol sipv2
session target ipv4:*.*.*.102:5061
session transport tcp tls
destination uri uzl
incoming uri from uzl
voice-class codec 1
voice-class sip options-ping 60
srtp
no vad
!

dial-peer voice 201 voip
session protocol sipv2
session target sip-server
session transport tcp tls
incoming called-number .
voice-class codec 1
srtp
no vad

!
gateway
media-inactivity-criteria all
timer receive-rtcp 30
timer receive-rtp 86400
!
sip-ua
no remote-party-id
timers notify 200
sip-server ipv4:*.*.*.8:5062
crypto signaling remote-addr *.*.*.0 255.255.255.0 trustpoint VOIP-TRUST strict-cipher
crypto signaling remote-addr *.*.*.0 255.255.255.0 trustpoint VOIP-TRUST strict-cipher
!
gatekeeper
shutdown
!
telephony-service
secure-signaling trustpoint DSP-SEC
tftp-server-credentials trustpoint DSP-SEC
sdspfarm units 1
sdspfarm transcode sessions 16
sdspfarm tag 1 TRANSCODER
max-ephones 16
max-dn 16
ip source-address *.*.*.4 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Jul 07 2015 13:35:36
!
end