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
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
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!
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
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.
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