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SIP trunk not registering on voice gateway

Sean McQuade
Level 5
Level 5

‎05-29-2014 06:10 AM - edited ‎03-16-2019 10:55 PM

Hi guys, a bit stuck on this one. I have a voice gateway which is SIP to CUCM and has both an E1 and SIP trunk.

CUCM ----- SIP Trunk ---  Gig0/0  --> Voice Gateway ---- Serial 0/0/0:15 ----> E1 PRI

                                                                                                                  |___   Gig0/2  __> SIP

 

I have configured the calling on the PRI and all works correctly, but can't seem to get the SIP to work. The provider is from South Africa and provided a username and password for me to use, as well as specifying a network range. I have one end of the range set on the router and can ping across the interface, but I am getting the following from debug sip-ua register status:

 

#sho sip-ua register status    
Line                             peer       expires(sec) reg survival P-Associ-URI
================================ ========== ============ === ======== ============
27XXXXXXXXXX                     -1         153          no  normal

 

My config is as follows:

 

!
isdn switch-type primary-net5
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
no voice hunt unassigned-number
no voice hunt invalid-number
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 10.60.13.30
  ipv4 10.82.13.31
  ipv4 172.18.0.1
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 modem passthrough nse codec g711ulaw
 sip
  bind control source-interface GigabitEthernet0/0 <---- I'm not sure if this is correct. Should this be GI0/2?
  bind media source-interface GigabitEthernet0/0
  registrar server expires max 600 min 60
  early-offer forced
  midcall-signaling passthru
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8
!
!
voice register global
 mode srst
 system message CM Fallback Service Operating
 max-dn 300
 max-pool 58
!
voice register pool  58
 id network 10.0.0.0 mask 255.0.0.0
 preference 2
 incoming called-number
 no digit collect kpml
 dtmf-relay rtp-nte cisco-rtp sip-notify
 voice-class codec 1
 no vad
!
!
!
voice translation-rule 1
 rule 1 /^\(.*\)/ /9\1/ type subscriber subscriber
 rule 2 /^\(.*\)/ /90\1/ type national national
 rule 3 /^\(.*\)/ /900\1/ type international international
!
voice translation-rule 2
 rule 1 /^6...$/ /0XX\0/
 rule 2 /^7...$/ /0XX\0/
!
voice translation-rule 3
 rule 1 /^6...$/ /+27XX\0/
 rule 2 /^7...$/ /+27XX\0/
 rule 3 /^9\(.*\)/ /\1/
!
!
voice translation-profile INBOUND
 translate calling 1
 translate called 2
!
voice translation-profile OUTBOUND
 translate called 3
!
!
redundancy
!
!
controller E1 0/0/0
 pri-group timeslots 1-31
!
interface GigabitEthernet0/0
 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
 ip address 10.38.20.251 255.255.255.0
 duplex auto
 speed auto
!
interface GigabitEthernet0/2
 description SIP PSTN
 ip address 172.18.0.5 255.255.255.0
 duplex auto
 speed auto
!
interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-net5
 isdn overlap-receiving
 isdn incoming-voice voice
 isdn send-alerting
 isdn bchan-number-order ascending
 isdn sending-complete
 no cdp enable
!
!
ip route 0.0.0.0 0.0.0.0 10.38.20.201
ip route 172.18.0.0 255.255.0.0 172.18.0.1
!
!

!
voice-port 0/0/0:15
 translation-profile incoming INBOUND
 cptone ZA
!
voice-port 0/1/0
!
voice-port 0/1/1
 !
!
!
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 GigabitEthernet0/0
sccp ccm 10.82.13.31 identifier 2 version 7.0
sccp ccm 10.60.13.30 identifier 1 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate profile 1 register CFB
 associate profile 2 register XCODE
 switchback method graceful
!
dspfarm profile 2 transcode  
 codec g729r8
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 2
 associate application SCCP
!
dspfarm profile 1 conference  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 codec g729br8
 maximum sessions 2
 associate application SCCP
!
!
dial-peer voice 1001 voip
 incoming called-number 27XXXXXXXXX
!
dial-peer voice 1000 voip
 description TO SA SIP PSTN
 translation-profile outgoing SIP_DDI_TRANSLATE_OUTBOUND
 preference 1
 destination-pattern 0T
 session protocol sipv2
 session target sip-server
 voice-class codec 1  
 dtmf-relay rtp-nte
 fax-relay ecm disable
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711ulaw
 no vad
!
dial-peer voice 3001 voip
 destination-pattern 0XXXX....
 session protocol sipv2
 session target ipv4:10.60.13.30
 voice-class codec 1  
 dtmf-relay rtp-nte
 fax-relay ecm disable
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711ulaw
 no vad
!
dial-peer voice 3002 voip
 preference 1
 destination-pattern 0XXXX....
 session protocol sipv2
 session target ipv4:10.82.13.31
 voice-class codec 1  
 dtmf-relay rtp-nte
 fax-relay ecm disable
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711ulaw
 no vad
!
dial-peer voice 100 pots
 incoming called-number .
 direct-inward-dial
 no sip-register
!
dial-peer voice 101 pots
 description Emergency 10111
 destination-pattern 10111
 port 0/0/0:15
 forward-digits 5
 no sip-register
!
dial-peer voice 102 pots
 description Emergency 910111
 destination-pattern 910111
 port 0/0/0:15
 forward-digits 5
 no sip-register
!
dial-peer voice 103 pots
 description Emergency 9112
 destination-pattern 9112
 port 0/0/0:15
 forward-digits 3
 no sip-register
!
dial-peer voice 104 pots
 description Emergency 112
 destination-pattern 112
 port 0/0/0:15
 forward-digits 3
 no sip-register
!
dial-peer voice 105 pots
 description Outgoing SRST 9T
 translation-profile outgoing OUTBOUND
 destination-pattern 9T
 port 0/0/0:15
 no sip-register
!
dial-peer voice 106 pots
 description Outgoing
 destination-pattern 0T
 port 0/0/0:15
 forward-digits all
 no sip-register
!
dial-peer voice 107 pots
 description Fax 011 996 6610
 destination-pattern 0119966610
 port 0/1/0
 no sip-register
!
sip-ua
 credentials username 27XXXXXXXXX password 7 14270A5A0F0078 realm sip-x01.trc.sadv.co.za
 authentication username 27XXXXXXXXX password 7 02361C0A08025D realm sip-x01.trc.sadv.co.za
 retry invite 2
 retry bye 2
 retry cancel 2
 registrar dns:sip-x01.trc.sadv.co.za expires 3600
 sip-server dns:sip-x01.trc.sadv.co.za
!
!
!
gatekeeper
 shutdown
!
!
call-manager-fallback
 secondary-dialtone 0
 max-conferences 20 gain -6
 transfer-system full-consult
 timeouts interdigit 4
 timeouts ringing 20
 ip source-address 10.38.20.251 port 2000
 max-ephones 110
 max-dn 400
 system message primary CM Fallback Service Operating
 transfer-pattern .T
 keepalive 20
 no huntstop
 mwi relay
 moh "music-on-hold.au"
 multicast moh 239.1.1.1 port 16384
 time-zone 29
 time-format 24
 date-format dd-mm-yy
!

 

When I make incoming calls to the SIP, I get a busy tone and don't see anything coming into the gateway on the ccsip messages debug.

 

Thanks

Sean

Labels:
0 Helpful
4 Replies 4

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

‎05-29-2014 07:14 AM

Its not all providers that require registrations, so I wont worry about the registration status just yet. If you cant see the call hitting your gateway at all, then that's a different problem. You need to involve your ITSP on why they are sending calls to you

Please rate all useful posts
0 Helpful

mmoulson1
Level 4
Level 4
In response to Ayodeji Okanlawon

‎05-30-2014 03:53 AM

One other thing you are pointing to a SIP server via DNS:
dns:sip-x01.trc.sadv.co.za

But I don't see a DNS server in your configuration, add:
ip name-server 8.8.8.8
(Google DNS)

Also as other have suggested are you sure you have connectivity to your ITSP?

Run a 'debug ccsip message' and try and call from the PSTN, if you don't see a load of SIP message then you need to check your connectivity.

Hope that helps!

0 Helpful

Sreekanth Narayanan
Cisco Employee
Cisco Employee

‎05-29-2014 08:25 AM

Hi Sean,

Your sip registrar configuration seems ok. Like Ayodeji said in his post, if you can't see sip messages coming to your gateway, then you need to involve the provider.

A packet capture on the gateway's gig0/2 interface should reveal more whether any messages are coming to the gateway on that interface.

You have bound sip to the gig0/0 interface. Sometimes providers, want the source address of messages to them to contain the IP address that they recognize, which means you would need to bind gig0/2 for sip and not gig0/0. This can lead to call issues.

 

Also do you see the router sending any register messages to the ITSP on the debug ccsip messages?

 

Thanks

Sreekanth

0 Helpful

Brian Meade
Level 7
Level 7

‎05-29-2014 09:54 AM

It looks like you only have private IP addresses on this box.  That may be a problem with getting the registration to work.  What's the full topology heading from this box to the internet?  You may need to enable/disable SIP inspection at the edge.

0 Helpful
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