Sip Trunk Registration Problem

[Anju Josua]

Dear All,

Currently i trying to register sip trunk on cube to itsp.

i already configure the sip-ua, here is my configuration:

credentials username 9477 password 7 1315051B060D557878707D realm 10.0.31.10
authentication username 9477 password 7 105E1B10081643595F507F realm 10.0.31.10
no remote-party-id
retry invite 5
retry register 5
retry options 10
timers connect 100
registrar ipv4:10.0.31.10 expires 3600
sip-server ipv4:10.0.31.10

but it still not register.

When i use "debug ccsip message", i can't see the register message. i think it still not trying to register to itsp.

Do i miss something?

Thanks

Anju

[Dennis Mink]

credentials username blah R password 7 xyz realm domainname.com
authentication usernameblah password 7 xyz realm domainname.com
retry invite 2
retry response 3
retry bye 2
retry prack 6
retry register 2
timers expires 300000
registrar dns:domainname.com expires 3600
sip-server dns:domainname.com
connection-reuse

under the sip configuration also add:

sip

outbound-proxy dns:<hostname or IP of provider proxy
asymmetric payload full
options-ping 90
early-offer forced
midcall-signaling passthru
privacy-policy passthru

do you get anything inbound?

also make sure you have dial peers pointing to your provider.

please rate if useful

[Anju Josua]

Hi Dennis,

Aprreciate your response.

Here is my sh run after add configuration from you:

sip
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
registrar server expires max 600 min 600
outbound-proxy ipv4:10.0.31.10
associate registered-number 9477
asymmetric payload full
options-ping 90
early-offer forced
midcall-signaling passthru
privacy-policy passthru

dial-peer voice 10 voip
description "TO TELKOM"
translation-profile incoming IncomingPSTN
translation-profile outgoing OutgoingPSTN
destination-pattern 9T
session protocol sipv2
session target ipv4:10.0.31.10
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
authentication username 9477 password 7 0103140D560A575D72181B realm 10.0.31.10
!

sip-ua
credentials username 9477 password 7 1315051B060D557878707D realm 10.0.31.10
authentication username 9477 password 7 105E1B10081643595F507F realm 10.0.31.10
no remote-party-id
retry invite 2
retry response 3
retry bye 2
retry prack 6
retry register 2
retry options 10
timers expires 300000
timers connect 100
registrar ipv4:10.0.31.10 expires 3600
sip-server ipv4:10.0.31.10

i already add dial-peer to itsp (dial-peer voice 10).i don't get inbound message from itsp. 

the weird thing is, when i turn debug ccsip message, i don't see register message, it seems my sip trunk doesn't try to register to itsp.

Thanks,

Anju

[Anju Josua]

Dear All,

i try to telnet their sip server using port 5060, but connection refused.

usually, does sip register using that port?

Thanks,

Anju

[Dennis Mink]

what actually happens when you make an outbound call?  do you get any debug ccsip message output?

Please rate if useful

[Anju Josua]

Hi Denis,

Sorry for late response.

i already my sip trunk in cube finally can register. i can make outgoing call.

but still have issue with incoming call from ITSP to callmanager. 

if i see the debug ccsip message, my cube already send invite message to cucm, but seems cucm doesn't response. finally the call canceled.

is there any idea, why cucm not response to invite message from cube?

Thanks,

Anju

[Dennis Mink]

Do you have an outbound dial peer from your CUBE, going to your CUCM.

and if you make an inbound call and do debug voip dialpeer, does this dial peer get hit?

start with that first

also, what did you do to make the registration to your provider work?

please rate if useful

[Anju Josua]

Yes, i already configure incoming and outgoing dialpeer. 

i haven't try debug voip dialpeer, but already try debug ccapi in out, and see my inbound call from itsp hit the incoming and outgoing dialpeer.

when i use debug ccsip message, i see cube already sent 'invite' to cucm but no response from cucm.

Any idea?

Thanks,

Anju

[Dennis Mink]

do you have a SIP trunk configured on your cucm to point to the cube? you probably do as you are making outbound calls through the same cube.

I would do a trace on the call manager side on an inbound call and see if the cucm actually sending a response to the invite, and add the trace to the post including timestamp and calling and called number

[Anju Josua]

Yes, i already can make outgoing call to itsp.

But, the incoming still problem.

Thanks,

Anju

[Anju Josua]

Already done,

i just need to delete some configuration in voice service voice to be:

sip
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1

and incoming call work perfectly.

[Igor Aliyev]

Could you please share the solution to the problem?

I have to register SIP-trunk as well yet my CUBE doesn't send register requests.

[HARIS_HUSSAIN]

Use Below working configuration  from my router

sip-ua
credentials number XXXXXXXX username 4XXXXXX.domain password 7 0000060B0942 realm etisalat.com
authentication username 4XXXXXX.domainpassword 7 14130706011D
no remote-party-id
retry invite 10
retry response 4
retry bye 1
retry register 10
timers trying 1000
timers expires 360000
timers connect 100
timers buffer-invite 500
registrar dns:4XXXXXX.domainexpires 3600
sip-server dns:4XXXXXXX.doamin

voice service voip
ip address trusted list
ipv4 192.168.1.6 255.255.255.255
ipv4 192.168.10.15 255.255.255.255
ipv4 192.168.10.0 255.255.255.0
dtmf-interworking rtp-nte
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
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
g729 annexb-all

ip host 4XXXXXX.domain 192.168.1.6
ip name-server 172.24.155.65
ip name-server 172.24.155.83

voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
voice class sip-profiles 101
request ANY sip-header Contact modify "7770000" "XXXXXXXX"
response ANY sip-header Contact modify "7770000" "04XXXXX"

dial-peer voice 103 voip
description ******outgoing call to ITSP trunk******
translation-profile outgoing PSTNOUT
shutdown
destination-pattern 9T
session protocol sipv2
session target ipv4:192.168.1.6
session transport udp
voice-class codec 1
voice-class sip localhost dns:4XXXXXX.domain
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 101
dtmf-relay rtp-nte
no vad

[hasnain.raza]

I see that you're not using 'mode border-element' command. Is it because you don't need to make the Voice router as CUBE? I've to configure the SIP trunk with Etisalat on a Cisco 4321-V/K9 router and have not ordered CUBE licenses. Am I safe without the licenses?