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CUCM Inbound Call Configuration?

Signups
Level 1
Level 1

Hi, Ive been working on a lab setup for CCNA. I have a 2901 with the UC license applied and a SIP provider.

I have outbound calls working on the phones in the lab and im now trying to get inbound dialing working.

 

I have a DID setup at the SIP provider which is set to route calls via sip registration. I have debug ccsip all enabled on the router and when i place an incoming call from another system i can see via the debug logs that there are inbound sip packets first to the router and then to the CUCM which is where the trouble seems to be.

 

SIP/2.0 503 Service Unavailable

SIP/2.0/UDP 192.168.0.221:5060 (CUBE Address)

From: <sip:+18629544082@sip.flowroute.com>;tag=9CD2740-7AC

To: <sip:14434606486@192.168.0.211>;tag=810831095

Date: Sat, 13 Apr 2019 14:19:45 GMT

Call-ID: A4DFEE94-5D2811E9-9D0092FF-D30DD721@192.168.0.221

CSeq: 101 INVITE

Allow-Events: presence

Warning: 399 CallManager1 "Unable to find a device handler for the request received on port 56053 from 192.168.0.221"

Content-Length: 0

 

There is only one sip trunk configured in CUCM. I think the problem is that i have not told CUCM where to send the incoming calls to. Can anyone tell me how to do this ? I have a hunt pilot number i would like to send all incoming calls to

2 Accepted Solutions

Accepted Solutions

You can create a translation pattern and route call to a DN .

Say you want to send the call to 2001 and you are receiving a call on 5XXX number then just create translation pattern 5XXX and convert it to 2001 assign proper CSS and it should work.

*** Please rate helpful post; Mark "Accept as a Solution" if applicable

Thanks,
Haris

View solution in original post

It says
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.221:5060;branch=z9hG4bK634B
From: <sip:Anonymous@sip.flowroute.com>;tag=6A0D0-14B7
To: <sip:666@10.0.0.211>;tag=24565~2c72f3a0-14c7-41e0-9e45-3221c7b922f1-26509572
Date: Sun, 14 Apr 2019 14:34:50 GMT
Call-ID: 4B8E04B0-5DF911E9-8026D076-EFA2ED15@10.0.0.221
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Server: Cisco-CUCM12.0
Session-ID: 00000000000000000000000000000000;remote=20beed4e03fa509eaf6143706bb39eb5
Content-Length: 0


Most probably you have not applied proper CSS on the SIP Trunk in CUCM. Go to SIP Trunk look for Calling Search Space and apply appropriate.
*** Please rate helpful post; Mark "Accept as a Solution" if applicable

Thanks,
Haris

View solution in original post

8 Replies 8

HARIS_HUSSAIN
VIP Alumni
VIP Alumni
1) It seems that CUCM SIP Service is not identifying you gateway IP If you have the SIP Trunk Configured to Gateway from CUCM make sure to bind the SIP traffic to proper IP Address on Voice gateway.

*** Please rate helpful post; Mark "Accept as a Solution" if applicable

Thanks,
Haris

Yes! after adding 
voice-class sip bind control source-interface BVI10
voice-class sip bind media source-interface BVI10

to my inbound dial peer the error has disappeared.

 

The only thing im missing is how do I get CUCM to route all inbound calls from this dial peer to a directory number ? Normally with CME I would just do an earphone bit its not an option here.

You can create a translation pattern and route call to a DN .

Say you want to send the call to 2001 and you are receiving a call on 5XXX number then just create translation pattern 5XXX and convert it to 2001 assign proper CSS and it should work.

*** Please rate helpful post; Mark "Accept as a Solution" if applicable

Thanks,
Haris

I thought that would resolve the problem however when I dial the DID from another phone system I get the following 

 

Here is the revelent part of my config 

 

voice translation-rule 3
rule 1 /^.*/ /666/
!
!
voice translation-profile External_To_Hunt
translate called 3
!


dial-peer voice 2002 voip
description SIP Provider Inbound Dial Peer
translation-profile incoming External_To_Hunt
session protocol sipv2
session target ipv4:10.0.0.211
incoming called-number 14434206996
incoming uri via IN
voice-class sip bind control source-interface BVI10
voice-class sip bind media source-interface BVI10
dtmf-relay sip-kpml rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 2003 voip
description CUCM Porting Dial Peer
destination-pattern 666
session protocol sipv2
session target ipv4:10.0.0.211
incoming uri via IN
voice-class sip bind control source-interface BVI10
voice-class sip bind media source-interface BVI10
dtmf-relay sip-kpml rtp-nte
codec g711ulaw
no vad

 

https://pastebin.com/NtzMUXWz

 

It says
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.221:5060;branch=z9hG4bK634B
From: <sip:Anonymous@sip.flowroute.com>;tag=6A0D0-14B7
To: <sip:666@10.0.0.211>;tag=24565~2c72f3a0-14c7-41e0-9e45-3221c7b922f1-26509572
Date: Sun, 14 Apr 2019 14:34:50 GMT
Call-ID: 4B8E04B0-5DF911E9-8026D076-EFA2ED15@10.0.0.221
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Server: Cisco-CUCM12.0
Session-ID: 00000000000000000000000000000000;remote=20beed4e03fa509eaf6143706bb39eb5
Content-Length: 0


Most probably you have not applied proper CSS on the SIP Trunk in CUCM. Go to SIP Trunk look for Calling Search Space and apply appropriate.
*** Please rate helpful post; Mark "Accept as a Solution" if applicable

Thanks,
Haris

hi.....

I don't have SDI recording file

I did all procedures for call recording but not working ......i have CUCM 9.1 

kgibs42
Level 1
Level 1

The External Phone Number Mask MUST match the extension’s FULL 10 digit DID or calls made externally to the device will not go through. Even when all other functions of the phone work like internal inbound/outbound, internal outbound to external, etc.

The External Phone Number Mask (EPNM) do not have any correlation with any inbound calls, neither external nor internal. It is only used for outbound calls, if the applicable configuration is in place.



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