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.
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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
Now my incomming dial peer look like:
dial-peer voice 5000 voip
translation-profile incoming TP_IN_SIP
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
ip qos dscp ef signaling
Incomming calls still not working...
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
bind media source-interface fastEthernet0/1
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)#voice service voip
CME(conf-voi-serv)#allow-connections sip to sip
CME(conf-voi-serv)#signaling forward unconditional
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)# 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?
No, it's not normal.
Let's try one more thing.
Add this :
dial-peer voice 100 pots
incoming called-number .
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.
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Sorry for not being more helpful.
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
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
description To ISP
ip address a.a.a.a 255.255.255.a
ip nat outside
ip address 10.1.1.254 255.255.255.0
ip nat inside
This could be issue...maybe the signalling is PERFECT but media is NOT.
We could try the following
voice services voip
bind all source interface fa0/1
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