Cisco_Twilio_Incoming Call failing

[anto.sathish0213]

Hi,

I've setup a Elastic SIP trunk using Public IP NAT'ting towards Cisco Voice Gateway.

Twilio(Elastic SIP Trunk)-->Public IP NAtting--->Cisco Voice Gateway--->CUCM

Public IP-115.111.182.150

We are trying a establish a incoming call using the above call flow.

Am able to see calls in Voice gateway. But I See few errors like SIP/2.0 100 Trying or SIP/2.0 404 Not Found.

The way I setup the call routing is when I call the main number (2177714466) the call is supposed to get routed to Cisco CUCM.

Translation Rule created is,

voice translation-rule 3
rule 1 /^\+/ //

voice translation-profile NEW
translate called 3

Dial-Peer:

dial-peer voice 7011111 voip
translation-profile incoming NEW
destination-pattern 721..
progress_ind alert enable 8
session protocol sipv2
session target ipv4:10.131.62.146----CUCM IP
session transport udp
incoming called-number .
voice-class sip rel1xx supported "100rel"
voice-class sip profiles 1
no voice-class sip reset timer expires 183
dtmf-relay sip-notify
codec g711ulaw
no vad

Attaching the VG Configs and debugs. Can you guys help to sort out this issue

Regards,

Anto

[Vitthal Thutte]

Define your incoming DNIS in "incoming called-number" and test once.

dial-peer voice 7011111 voip
translation-profile incoming NEW
destination-pattern 721..
progress_ind alert enable 8
session protocol sipv2
session target ipv4:10.131.62.146----CUCM IP
session transport udp
incoming called-number 90046812400521
voice-class sip rel1xx supported "100rel"
voice-class sip profiles 1
no voice-class sip reset timer expires 183
dtmf-relay sip-notify
codec g711ulaw
no vad

INVITE sip:90046812400521@115.111.182.150 SIP/2.0

Via: SIP/2.0/UDP 193.46.255.79:63702;branch=z9hG4bK1001664554

Max-Forwards: 70

From: <sip:620@115.111.182.150>;tag=347059791

To: <sip:90046812400521@115.111.182.150>

Call-ID: 111393798-1473263611-2106785414

[AntoSathish8587]

Hi,

Have made few changes in the Translation rule and in the dial-peer,

voice translation-rule 3
rule 1 /.*/ /72100/

dial-peer voice 7011111 voip
translation-profile incoming NEW
destination-pattern 72100
session protocol sipv2
session target ipv4:10.131.62.146
incoming called-number .
no voice-class sip reset timer expires 183
dtmf-relay sip-notify
codec g711ulaw

Now calls are getting connected.

But the issue here is once call is getting connected and if I accept the call in Cisco phone immediately calls are getting disconnected 

Also I see "delay media to slow start case, codec negotation is not done" in the debugs.

Attaching the debugs and configs.

Could you guys help on this issue.

Regards,

Anto