Cucme direct to SIP gateway connection doesnt work

[PETER MARTLAND]

                   Can anyone assist, I have a call manager express version 8.6 IOS 15.1.4T and i am trying to connect directly to a Gamma provided SIP gateway in the UK. The config looks ok, debugs indicate calls hitting our router but dont get any ringing??

My questions, firstly is this a supported setup I am using config guides from version 4 CME??

Secondly, any assistance with the debugs.

thanks in advance.

[chrysostomos1980]

Hi

debug ccsip messages

debug ccsip error

debug ccsip events

[PETER MARTLAND]

i have attached those debugs, following making a call into the system

[chrysostomos1980]

Hi

did you allow the subnet of the destination sip server?

voice service voip

ip address trusted list

  ipv4 0.0.0.0 0.0.0.0

dtmf-interworking rtp-nte

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

[PETER MARTLAND]

yes

voice service voip

ip address trusted list

  ipv4 0.0.0.0

  ipv4 87.238.219.194

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip refer

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

  registrar server expires max 3600 min 3600

I added the redirect ip2ip as an after thought as well as the bind commands but this did not help at all

[chrysostomos1980]

Ok

Good

I see that you are getting  SIP/2.0 403 Forbidden

Can you show us the

show sip reg status

[PETER MARTLAND]

DL-VG#sh sip-ua registration statu
DL-VG#sh sip-ua registration status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED

SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED  172.28.255.253
SIP User Agent bind status(media): ENABLED  172.28.255.253
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
protocol mode is ipv4

SDP application configuration:

[chrysostomos1980]

DD_UC#show  sip-ua register status

[PETER MARTLAND]

I have ringing outside to inside cant do any other tests at the moment?

i noticed in your config that ip trusted list was 0.0.0.0 0.0.0.0 mine was 0.0.0.0.

I changed this and it rings through to the phone?

I will do some more testing but thanks at the moment

[chrysostomos1980]

Cheers mate

i didnt see that

If you are unable to recieve incoming calls then most probably you dont have the correcttranslation rules applied to the incoming dial peers

Another one command that may is usefful for you

Under sip-ua config add the below command if you have issues to dial any pstn number

calling-info pstn-to-sip  from number set xxx

Pls rate usefull posts

[PETER MARTLAND]

hello,

unfortunately my eureka moment was short lived. We have ringing but not on any phones? its a strange setup with the internet provision on 1 interface of the CUCME and the otherside LAN connecting the phones?

Question can I natively configure my phones as SCCP or do they require to be SIP endpoints?

The reason i ask is my understading is we are not a SIP proxy server like CUBE but an endpoint connection?

Hope this makes sense, thanks for your help so far?

[PETER MARTLAND]

tearing my hair out.

If i take off G729R8 codec support we dont ring!!! with it on we ring call does not complete onto the phone

[chrysostomos1980]

Hi

I dont believe that you will have any issue if you use sip or sccp phones

where did you try to call.What are the numbers that you tried.PSTN numbers?

If yes then you will use ONLY g711 codec..

send your debugs

[PETER MARTLAND]

debugs of ccsip

[PETER MARTLAND]

Sorry, the call seems to come in as a g729 call. I have added a transcoder as far as I know but no changes. Just checing that config now.

voice-card 0

dspfarm

dsp services dspfarm

!

sccp local GigabitEthernet0/0

sccp ccm 172.28.6.200 priority 1 version 7.0

sccp

!

dspfarm profile 1 transcode 

description ***** Transcoder *****

codec g729r8

codec g729abr8

codec g729ar8

codec g711alaw

codec g711ulaw

maximum sessions 5

associate application SCCP

!

Dspfarm Profile Configuration

Profile ID = 1, Service = TRANSCODING, Resource ID = 1 
Profile Description : ***** Transcoder *****
Profile Service Mode : Non Secure
Profile Admin State : UP
Profile Operation State : RESOURCE ALLOCATED
Application : SCCP   Status : ASSOCIATION IN PROGRESS
Resource Provider : FLEX_DSPRM   Status : UP
Number of Resource Configured : 5
Number of Resource Available : 5
Codec Configuration: num_of_codecs:5
Codec : g729r8, Maximum Packetization Period : 60
Codec : g729abr8, Maximum Packetization Period : 60
Codec : g729ar8, Maximum Packetization Period : 60
Codec : g711alaw, Maximum Packetization Period : 30
Codec : g711ulaw, Maximum Packetization Period : 30

SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0    1   28.3.5   UP     N/A  FREE  xcode   1      -         -         -       
0    1   28.3.5   UP     N/A  FREE  xcode   1      -         -         -       
0    1   28.3.5   UP     N/A  FREE  xcode   1      -         -         -       
0    1   28.3.5   UP     N/A  FREE  xcode   1      -         -         -       
0    1   28.3.5   UP     N/A  FREE  xcode   1      -         -         -       
         
Total number of DSPFARM DSP channel(s) 5