CME incomming calls failure

[dimitargrozev]

Hello!

I am trying to configure my CME with SIP trunk. My outgoing calls work fine, but i have problems with an incomming calls.

When i call from my mobile, the extention behind CME ring....but there is no voice.

Thanks in advanced.

[Amer rajai Sha'er · Accepted solution]

Hello,

I can see in the incoming dial-peer a session target ras , can you please explain this ?

Amer

[dimitargrozev]

Hello Amer!

With or without this command it's not working. I tried it. If you need some debug commands...

[Amer rajai Sha'er]

Hello,

Yes please , and please delete the command ,  the session target ras is for gatekeeper and i am suessing you don't  have a gatekeeper , can you please instert the command

session target ipv4:10.1.1.254 into the dial-peer voice 5000 voip

can you capture the debug voice ccapi inout

Amer

[dimitargrozev]

Hello!

Now my incomming dial peer look like:

dial-peer voice 5000 voip
translation-profile incoming TP_IN_SIP
huntstop
answer-address .T
destination-pattern 101562T
voice-class codec 1
session target ipv4:10.1.1.254
dtmf-relay rtp-nte h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 9600
fax nsf 000000
fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw
icpif 0
expect-factor 0
ip qos dscp ef signaling
no vad

Incomming calls still not working...

[dimitargrozev]

Is there anyone who can help me?

[Amer rajai Sha'er]

Hello Dimitar,

Can you please add the below lines and test.

voice service voip
allow-connections sip to sip
signaling forward unconditional
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
  bind media source-interface fastEthernet0/1
  rel1xx disable
  min-se 360
  header-passing
  midcall-signaling passthru
!

[dimitargrozev]

Hello!

Thanks for your sesponse. I made what you adviced, but without effect. Incomming calls still not working.

Interesting for me is the following:

CME(config)#no voice service voip
CME(config)#
CME(config)#
CME(config)#
CME(config)#
CME(config)#
CME(config)#
CME(config)#voice service voip
CME(conf-voi-serv)#allow-connections sip to sip
CME(conf-voi-serv)#signaling forward unconditional
CME(conf-voi-serv)#
CME(conf-voi-serv)#h323
CME(conf-serv-h323)#sip
CME(conf-serv-sip)#  bind media source-interface fastEthernet0/1
CME(conf-serv-sip)#  rel1xx disable
CME(conf-serv-sip)#  min-se 360
Warning: Setting the Min-SE value allowed to a small value may
degrade router performance due to frequent re-INVITES.

CME(conf-serv-sip)#  header-passing
CME(conf-serv-sip)#  midcall-signaling passthru
*Mar 19 14:14:12.871: //-1/xxxxxxxxxxxx/CCAPI/cc_get_call_active_next:
   NULL from dialMibActiveRBTree, setup_time=0, index=0
*Mar 19 14:14:12.871: //-1/xxxxxxxxxxxx/CCAPI/ccGetCallActive:
   Call Entry Is Not Found; Count=0

Is that normal? Could it be from that i installed CME basik on my router?

[Amer rajai Sha'er]

Hello,

No, it's not normal.

Let's try one more thing.

Add this :

dial-peer voice 100 pots
incoming called-number .
direct-inward-dial

Amer

[dimitargrozev]

Hi, Amer!

Is that neccessary? I am using a SIP trunk.

How can i fix this error i posted before?

Thanks

[Amer rajai Sha'er]

Hello,

I am trying to find that error , i have been scrolling around for this , i did it once before but it was only outgoing , i never did a incoming without port termination , every case i had to insert the pots dial-peer with the direct inward dial , and my guess also is that we need MTP.

Did you try to insert the pots dial-peer before.

Amer

[dimitargrozev]

Hello!

No i didn't. What i have is a SIP account. It should work.

Do you know cisco TAC number to call them?

[Amer rajai Sha'er]

Hello,

North America: 1-800-553-2447
   Europe: 32-2-704-5555
   Asia Pacific: +61-2-8446-7411
   Australia: 1-800-805-227
   USA Non-Domestic: 1-408-526-7209

Sorry for not being more helpful.

Amer

[saif musa]

Amer,

Really I have the same issue with outgoing calls. the incoming bound works prefect but when I want to call from FXS port. it just giving me fastbusy the moment i hit any digit. can you explain how did you fix that for outgoing calls.

thanks in advanced

[dijohn]

POTS dial-peer isn't necessary in this scenario!

IP Phone(SCCP) ----- CME ---- SIP TRUNK ----- ITSP

When you call your mobile from an internal EXTENSION does it work?

Can you please make a test call from Mobile(PSTN) to an internal extenstion

enable these debugs

debug voip ccapi inout

debug ccsip messages

debug ephone detail

I see you are using NAT too

interface FastEthernet0/0
description To ISP
ip address a.a.a.a 255.255.255.a
ip nat outside
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 10.1.1.254 255.255.255.0
ip nat inside
duplex auto
speed auto

This could be issue...maybe the signalling is PERFECT but media is NOT.

http://www.cisco.com/en/US/docs/ios/12_2/12_2x/12_2xb/feature/guide/ftbind.html#wp1048717

We could try the following

voice services voip

media flow-through

address hiding

sip

bind all source interface fa0/1

HTH

/divin

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