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Cisco SIP Trunk configuration


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#1 tasneem

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Posted 12 December 2013 - 04:34 PM

Hi Chaps,

 

I need help and your suggestion in configuration SIP trunk configuration on C3945-CME-SRST/K9 router. We are running CUCM version 8.0.3 and we currently have two PRI's on our voice gateways to communicate over to PSTN but we want to add one more SIP trunk with 6MB bandwidth available to us for an additional DID range. I need to know what should be the configuration required on Call manager side and what configuration would be requried on C3945-CME router.  If you direct me to any link where information is available and also what should be my starting point on this project. My telco have provide me following configuration to be used

 

Circuit is activated in SS and number range is ready for customer use. xxxxxxx100-xxxxx999

 

Customer IP address=  172.29.X.X/30

STC IP Address =            172.29.X.X/30

Protocol= SIP 

SIP Port = 5060 

Transport Protocol=UDP

Voice Codec= G711 A-Law 

DTMF = IN-Band DTMF with RFC2833 

SIP Server IP address = 10.205.x.x

 

Thanks in Advance.

 


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#2 talent pk

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Posted 20 May 2014 - 01:34 AM

I can give you the CME side configs on Router

 

 

Step 1 Registering to the Remote SIP SERVER as User Agent

=====================================================

global conf > 

sip-ua

sip-server x.x.x.x:5060

registrar x.x.x.x

transport udp

authentication realm if provided from VoIP Provider

 

 

Step 2 Create Extension or DN Phones as per your Provider

====================================================

 

voice register dn x

name xxxx

label xxx

mwi xx

 

voice register pool 1

codec g711alaw /g711ulaw what ever

number your number from VoIP Provider

username xxxx password xxxxx username password provided from VoIP provider

 

 

Step 3 Create Call Routing Rule or Dial-peer to Remote SIP Server

======================================================

dial-peer voice xx  voip

destination-pattern xxxxxxx10

session-target ipv4:sipserver:5060

no digit-strip

codec g711alaw

session protocol sipv2

dtmf-relay rtp-nte or whatever

 

 

for Cisco Call Manager please refer to your proper Cisco Call Manager Admin Guide of how to setup up SIP 


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