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Beginner

CME incomming calls failure

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.

1 ACCEPTED SOLUTION

Accepted Solutions
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Rising star

Hello,

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

Amer

View solution in original post

17 REPLIES 17
Highlighted
Rising star

Hello,

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

Amer

View solution in original post

Highlighted

Hello Amer!

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

Highlighted

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

Highlighted

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...

Highlighted

Is there anyone who can help me?

Highlighted

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
!

Highlighted

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?

Highlighted

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

Highlighted

Hi, Amer!

Is that neccessary? I am using a SIP trunk.

How can i fix this error i posted before?

Thanks

Highlighted

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

Highlighted

Hello!

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

Do you know cisco TAC number to call them?

Highlighted

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

Highlighted

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

 

Highlighted

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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