Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
Announcements

Welcome to Cisco Support Community. We would love to have your feedback.

For an introduction to the new site, click here. And see here for current known issues.

New Member

Call Flow Between Two SIP Gateways ?

I have a little problem with SIP GATEWAY TO SIP GATEWAY. I have that topology of net:

FXS=========SIP GATEWAY====ISP====SIP GATEWAY========FXS

101                                                                                                 201

CME A

interface FastEthernet0/1

ip address 190.100.100.2 255.255.255.0

ip virtual-reassembly in

duplex auto

speed auto

ip route 0.0.0.0 0.0.0.0 190.100.100.1

!

voice-port 0/3/0

!

voice-port 0/3/1

!

!

!

mgcp profile default

!

!

dial-peer voice 2000 voip

translation-profile outgoing 4Digits2E164

preference 1

destination-pattern .T

session protocol sipv2

session target ipv4:190.100.101.2

dtmf-relay rtp-nte sip-notify

!

dial-peer voice 15 pots

translation-profile incoming 4Digits2E164

destination-pattern .T

port 0/3/0

!

!

gateway

timer receive-rtp 1200

!

sip-ua

authentication username cisco password 7 094F471A1A0A

retry invite 1

retry response 1

retry bye 1

retry cancel 1

registrar ipv4:190.100.100.2:5060 expires 3600

!

!

CME B

ip cef

no ipv6 cef

!

multilink bundle-name authenticated

!

!

!

!

!

voice call send-alert

voice rtp send-recv

!

voice service voip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

sip

  bind control source-interface FastEthernet0/1

  bind media source-interface FastEthernet0/1

!

!

interface FastEthernet0/1

ip address 190.100.101.2 255.255.255.0

ip virtual-reassembly in

duplex auto

speed auto

ip http server

ip http authentication local

ip http secure-server

ip http path flash:/gui

ip route 0.0.0.0 0.0.0.0 190.100.101.1

voice-port 0/3/0

!

voice-port 0/3/1

!

!

!

mgcp profile default

!

!

dial-peer voice 2000 voip

translation-profile outgoing 4Digits2E164

preference 1

destination-pattern .T

session protocol sipv2

session target ipv4:190.100.100.2:5060

dtmf-relay rtp-nte sip-notify

!

dial-peer voice 15 pots

translation-profile incoming 4Digits2E164

destination-pattern .T

port 0/3/0

!

!

gateway

timer receive-rtp 1200

sip-ua

authentication username cisco password 7 0822455D0A16

retry invite 1

retry response 1

retry bye 1

retry cancel 1

registrar ipv4:190.100.101.2:5060 expires 3600

!

ISP

interface FastEthernet0/0

ip address 190.100.101.1 255.255.255.0

duplex auto

speed auto

!

interface FastEthernet0/1

ip address 190.100.100.1 255.255.255.0

duplex auto

speed auto

##############################################

I have this debug when I put debug ccsip error

CME_A#debug ccsip error

SIP Call error tracing is enabled

CME_A#

SIP: (228) Group (a= group line) attribute, level 65535 instance 1 not found.

SIP: Attribute mid, level 1 instance 1 not found.

SIP: Attribute mid, level 1 instance 1 not found.

*Feb  6 17:35:39.674: //229/000000000000/SIP/Error/sipSPICheckFromToRequest:

Failed FROM/TO Request check - IGNORE IF HAIRPIN CALL

                old_from:       <sip:.*@190.100.100.2>;tag=E873B0-AF7

                old_to: <sip:.*@190.100.100.2>

                new_from:       <sip:.*@190.100.100.2>;tag=E873B0-AF7

                new_to: <sip:.*@190.100.100.2>

*Feb   6 17:35:39.686:   //229/000000000000/SIP/Error/ccsip_api_register_result_ind: Message  Code  Class 5xx Method Code 100 received for REGISTER

*Feb  6 17:35:39.686: //229/000000000000/SIP/Error/sipSPIRegPthruProcessResponse: Error NO RPCB

What is my problem?  I need your help

Can you give me some cofiguration like this, if you have ?

Thank you

Everyone's tags (3)
13 REPLIES
Hall of Fame Super Silver

Call Flow Between Two SIP Gateways ?

You do not need to define registar, etc between 2 CME systems, all you need to do is have the dial-peers and add the following global commands on both sides:

voice service voip

no ip address trusted authenticate

allow-connections sip to sip

HTH,

Chris

New Member

Re: Call Flow Between Two SIP Gateways ?

but I want to configure two  sip gateways, and with command 

registrar ipv4:190.100.101.2:5060 expires 3600 is necessary to register all points like fxs right?? And I don't have this comand

allow-connections sip to sip.

Are there any command to replace this?

Hall of Fame Super Silver

Call Flow Between Two SIP Gateways ?

No, nor do I think you could do this.  The only reason you would want to do this is for integration to SIP trunk provider. All you are doing is simply integrating 2 CME systems, again all you need is dial-peers on both sides pointing to each other and global "allow-connections sip to sip" which allows IP-IP GW connectivity.

HTH, please rate all useful posts!

Chris

New Member

Re: Call Flow Between Two SIP Gateways ?

well, I'm doing that demo in router 2801? But I have to realize this in AS5350. This is a getway so I want to integrate that gateway to other gateway (NET2PHONE). In this gateway there isn't that command what can I do?

thank you for your interest

New Member

Re: Call Flow Between Two SIP Gateways ?

I need registrar: because I want to register with a provider:

registrar {dns:address | ipv4:destination-address}[expires seconds][tcp][secondary]:Allows a gateway to register the E.164 numbers of non-SIP phones with a registrar or proxy server.

With that command I 'm to register my FXS rigth??

Hall of Fame Super Silver

Call Flow Between Two SIP Gateways ?

Do you have a SIP trunk service from your provider?  If not there is nothing you can register.

Chris

New Member

Re: Call Flow Between Two SIP Gateways ?

Are you ther

Chris??

New Member

Re: Call Flow Between Two SIP Gateways ?

So, with this command

registrar

Can I  register my fxs to make call or Isn't necesary realize this when I configure like this toplogy ?? I tried to simulate that

ok.

New Member

Re: Call Flow Between Two SIP Gateways ?

What is the correct configuration I need only dial peers ?

Hall of Fame Super Silver

Call Flow Between Two SIP Gateways ?

Simply having the "session target ipv4" pointing to the other side is what makes it work, also make sure you bind SIP protocol the IP address.

As to your 5350 GW are you saying you cannot enter the global allow command?  What IOS are you running?

Chris

New Member

Re: Call Flow Between Two SIP Gateways ?

I have that ios in the AS 5350

c5350-jk9s-mz.124-15.T15.bin

New Member

Re: Call Flow Between Two SIP Gateways ?

to produce like singanling in the two gateway. I need only put that right.

CME A

!

voice call send-alert

voice rtp send-recv

!

voice service voip

no ip address trusted authenticate

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

sip

  bind control source-interface FastEthernet0/1

  bind media source-interface FastEthernet0/1

!

!

ip http server

ip http authentication local

ip http secure-server

ip http path flash:/gui

ip route 0.0.0.0 0.0.0.0 190.100.100.1

!

!

dial-peer voice 2000 voip

translation-profile outgoing 4Digits2E164

preference 1

destination-pattern .T

session protocol sipv2

session target ipv4:190.100.101.2

dtmf-relay rtp-nte sip-notify

!

dial-peer voice 15 pots

translation-profile incoming 4Digits2E164

destination-pattern .T

port 0/3/0

!

!

gateway

timer receive-rtp 1200

!

sip-ua

retry invite 1

retry response 1

retry bye 1

retry cancel 1

!

!

!

CME B

interface FastEthernet0/1

ip address 190.100.101.2 255.255.255.0

ip virtual-reassembly in

duplex auto

speed auto

voice call send-alert

voice rtp send-recv

!

voice service voip

no ip address trusted authenticate

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

sip

  bind control source-interface FastEthernet0/1

  bind media source-interface FastEthernet0/1

!

ip route 0.0.0.0 0.0.0.0 190.100.101.1

dial-peer voice 2000 voip

translation-profile outgoing 4Digits2E164

preference 1

destination-pattern .T

session protocol sipv2

session target ipv4:190.100.100.2:5060

dtmf-relay rtp-nte sip-notify

!

dial-peer voice 15 pots

translation-profile incoming 4Digits2E164

destination-pattern .T

port 0/3/0

!

sip-ua

retry invite 1

retry response 1

retry bye 1

retry cancel 1

!

New Member

Re: Call Flow Between Two SIP Gateways ?

Yes, I tried to put the command allow-conn... sip to sip but in this IOS that command doesn't exist. There are other command that replace ???

2959
Views
0
Helpful
13
Replies