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Gamma SIP Configuration

saunders69
Beginner
Beginner

Hi Can anyone help.

I have been put into a new role that has to get the VoIP phone working in this new company - there is no other communications for them so it is important that they get this up an running. The company has decided to go with 'Gamma' telecom UK.

There is no PBX here, all calls will be handled by a 3925 router running 15.1 T5 IOS.

I have asked Gamma's help desk and they sent an email with settings in and kept repeating 'All customer equipment is different' when asking about some configuration examples for a Cisco router, for such a big company it would be nice if  they could give some examples with main devices like Cisco.

Gamma want to install this information on the router and I am not sure where:

Endpoint Name : DC2N16BJXXXXX
Your CPE IP Address : 89.213.XX.XX
Channels Created : 10

DDIs Allocated to This Endpoint : 0161YYYYY - 0161YYYYY

*** The examples I have come across do not say where I should put the below ip/port information ***

Signalling:UDP port 5060 egress/ingress to:- 88.215.55.97
Media: All UDP ports between 6000 - 40000 egress/ingress to:- 88.215.55.98


Request-URI Header
The number within the Request-URI is used for routing calls and should contain the called number. The format of this number can either be UK national format (with leading zero), UK international format (with two leading zeros and a country code), or full e.164 format (with leading +).

Examples
INVITE sip:+441618777148@100.100.100.29 SIP/2.0
INVITE sip:00441618777148@100.100.100.29 SIP/2.0
INVITE sip:01618777148@100.100.100.29 SIP/2.0

TO Header
Header must contain SIP Trunking SIP gateway address and the called number. The format of this number can either be UK national format (with leading zero), UK international format (with two leading zeros and a country code), or full e.164 format (with leading +).

Examples:
To: <sip:+441618777148@100.100.100.29>
To: <sip:00441618777148@100.100.100.29>
To: <sip:01618777148@100.100.100.29>

FROM Header
The user part of the SIP URI within the 'From' header must contain the Calling Line Identity of the originating device. This identity must be a number registered to the endpoint. The format of this number can either be national significant (with or without a leading zero), or in full e.164 format. The domain part of the SIP URI within the 'From' header can contain either the IP address of the public facing interface or a customer-defined FQDN. Gamma will not use this FQDN for any purpose

Examples
From: <sip:+441618777148@100.100.100.29> or using FQDN From: <sip:+441618777148@your.domain.local>
From: <sip:01618777148@100.100.100.29> or using FQDN From: <sip:01618777148@your.domain.local>
From: <sip:1618777148@100.100.100.29> or using FQDN From: <sip:1618777148@your.domain.local>

PAID Header
The format of the PAID number can either be national significant (with or without a leading zero), or in full e.164 format. If no PAID header is provided, Gamma will insert a PAID header with the default network CLI.

Examples
P-Asserted-Identity: <sip:+441618777148@100.100.100.29>
P-Asserted-Identity: <sip:01618777148@100.100.100.29>
P-Asserted-Identity: <sip:1618777148@100.100.100.29>

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

Any help to point in the right directions will be good - I have been searching for the last 3 days and now I am out of options.

Thanks in advance.

Paul

1 Accepted Solution

Accepted Solutions

Mohammed al Baqari
VIP Advisor VIP Advisor
VIP Advisor

Hi,

With Gamma and based on your notes, the below basic config will get the SIP trunk working. You still need to configure CME or another trunk to CUCM. Also, you need to configure more security feature to avoid getting your SIP Gateway compromised .

voice service voip
 ip address trusted list
  ipv4 88.215.55.97
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
  transport switch udp tcp
  asserted-id pai
  early-offer forced
  midcall-signaling passthru
  g729 annexb-all
  no call service stop

!!! ... Assuming that you use 9 as access code

voice translation-rule 1
 rule 1 /^9\(.+\)/ /\1/
voice translation-rule 2
 rule 1 /^00\(.+\)/ /+\1/

voice translation-profile DIGIT-MANIPULATION
 translate calling 2
 translate called 1

voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g729br8
 codec preference 3 g711ulaw
 codec preference 4 g711alaw

dial-peer voice 10 voip
 translation-profile outgoing DIGIT-MANIPULATION
 destination-pattern 9.T
 session protocol sipv2
 session target ipv4:88.215.55.97
 incoming called-number 0161.....
 voice-class codec 1  
 voice-class sip rel1xx supported "100rel"
 voice-class sip early-offer forced
 voice-class sip bind control source-interface Loopback0    !!! ... This is your interface IP which will connect to Gamma. You can change it as needed
 voice-class sip bind media source-interface Loopback0
 dtmf-relay rtp-nte

 sip-ua
 nat symmetric check-media-src
 retry invite 3
 timers trying 200

View solution in original post

6 Replies 6

Jonathan Els
Contributor
Contributor

Hi Paul

Please provide a sh run of your CME configuration, and let's look at how we can set up your SIP Trunk.

 - Jonathan

Mohammed al Baqari
VIP Advisor VIP Advisor
VIP Advisor

Hi,

With Gamma and based on your notes, the below basic config will get the SIP trunk working. You still need to configure CME or another trunk to CUCM. Also, you need to configure more security feature to avoid getting your SIP Gateway compromised .

voice service voip
 ip address trusted list
  ipv4 88.215.55.97
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
  transport switch udp tcp
  asserted-id pai
  early-offer forced
  midcall-signaling passthru
  g729 annexb-all
  no call service stop

!!! ... Assuming that you use 9 as access code

voice translation-rule 1
 rule 1 /^9\(.+\)/ /\1/
voice translation-rule 2
 rule 1 /^00\(.+\)/ /+\1/

voice translation-profile DIGIT-MANIPULATION
 translate calling 2
 translate called 1

voice class codec 1
 codec preference 1 g729r8
 codec preference 2 g729br8
 codec preference 3 g711ulaw
 codec preference 4 g711alaw

dial-peer voice 10 voip
 translation-profile outgoing DIGIT-MANIPULATION
 destination-pattern 9.T
 session protocol sipv2
 session target ipv4:88.215.55.97
 incoming called-number 0161.....
 voice-class codec 1  
 voice-class sip rel1xx supported "100rel"
 voice-class sip early-offer forced
 voice-class sip bind control source-interface Loopback0    !!! ... This is your interface IP which will connect to Gamma. You can change it as needed
 voice-class sip bind media source-interface Loopback0
 dtmf-relay rtp-nte

 sip-ua
 nat symmetric check-media-src
 retry invite 3
 timers trying 200

Thank you -- the problem has been sorted now and I will keep your info for further VoIP configurations that I need to do.

Hi Paul, Any chance you could send me a copy of the config as I have the same problem.

Could anyone help me with this, I have Gamma as a Sip TP and I am experiencing similar issue with incoming call. Gamma does not provide CUBE template. I followed the suggested configuration but with no success. Debug is showing me disconnecting cause SIP 400 and disconnect cc 100.

 

Anyone has suggestion?

 

p.s if someone could explain me on why there is few config line in sip-ua would help me

 

 

Thx

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