03-18-2011 05:12 AM - edited 03-16-2019 04:01 AM
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.
Solved! Go to Solution.
03-18-2011 05:23 AM
Hello,
I can see in the incoming dial-peer a session target ras , can you please explain this ?
Amer
03-18-2011 05:23 AM
Hello,
I can see in the incoming dial-peer a session target ras , can you please explain this ?
Amer
03-18-2011 05:51 AM
Hello Amer!
With or without this command it's not working. I tried it. If you need some debug commands...
03-18-2011 05:57 AM
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
03-18-2011 06:26 AM
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...
03-19-2011 01:51 AM
Is there anyone who can help me?
03-19-2011 03:07 AM
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
!
03-19-2011 05:20 AM
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?
03-19-2011 05:36 AM
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
03-19-2011 05:45 AM
Hi, Amer!
Is that neccessary? I am using a SIP trunk.
How can i fix this error i posted before?
Thanks
03-19-2011 05:50 AM
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
03-19-2011 06:39 AM
Hello!
No i didn't. What i have is a SIP account. It should work.
Do you know cisco TAC number to call them?
03-19-2011 06:43 AM
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
08-24-2015 03:59 AM
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
03-19-2011 06:55 AM
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
PS: Rate only useful posts!
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide