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14
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8
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SIP trunk from CUBE without auth\reg

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sSiDs
Level 1
Level 1

‎09-08-2016 12:41 AM - edited ‎03-17-2019 08:04 AM

Hi team! after long time struggling with dropped sip-ua registration authentication, our voice ISP advise to create "straight" trunk without auth...but with "binding over our public IP"

actually, i don't now how to to do it...where and what needs to be changed.

below is out dial-peer for CUCM and Lync and outside calls

dial-peer voice 1 voip 
 description ***outbound NATIONAL*** 
 translation-profile outgoing calerid 
 destination-pattern 8[2-9]......... 
 progress_ind setup enable 3 
 session protocol sipv2 
 session target sip-server 
 session transport udp 
 voice-class codec 1  
 voice-class sip profiles 1 
 voice-class sip bind control source-interface FastEthernet0/1 
 voice-class sip bind media source-interface FastEthernet0/1 
 dtmf-relay rtp-nte 
 no vad 
! 
dial-peer voice 101 voip 
 description ***to->CUCM*** 
 destination-pattern [1,2]... 
 progress_ind setup enable 3 
 session protocol sipv2 
 session target ipv4:10.110.240.10 
 session transport udp 
 voice-class codec 1  
 voice-class sip bind control source-interface FastEthernet0/0.8 
 voice-class sip bind media source-interface FastEthernet0/0.8 
 dtmf-relay rtp-nte 
 no vad 
! 
dial-peer voice 2000 voip 
 description **Incoming Call from SIP Trunk** 
 service sip_ivr 
 destination-pattern 1003 
 session protocol sipv2 
 session target sip-server 
 session transport udp 
 incoming called-number 111111 
 dtmf-relay rtp-nte 
 codec g711ulaw 
 no vad 
! 
dial-peer voice 102 voip 
 description System-Incoming-Dial-Peer 
 translation-profile incoming calerid 
 answer-address 749511111111 
 session protocol sipv2 
 session target sip-server 
 session transport udp 
 voice-class sip bind control source-interface FastEthernet0/1 
 voice-class sip bind media source-interface FastEthernet0/1 
 dtmf-relay rtp-nte 
 codec g711ulaw 
 no vad 
! 
dial-peer voice 2 voip 
 description ***outbound INTERNATIONAL*** 
 translation-profile outgoing calerid 
 destination-pattern 810.T 
 progress_ind setup enable 3 
 session protocol sipv2 
 session target sip-server 
 session transport udp 
 voice-class codec 1  
 voice-class sip profiles 1 
 voice-class sip bind control source-interface FastEthernet0/1 
 voice-class sip bind media source-interface FastEthernet0/1 
 dtmf-relay rtp-nte 
 no vad 
! 
dial-peer voice 3 voip 
 destination-pattern 0[1-3] 
 progress_ind setup enable 3 
 session protocol sipv2 
 session target sip-server 
 session transport udp 
 voice-class sip bind control source-interface FastEthernet0/1 
 voice-class sip bind media source-interface FastEthernet0/1 
 dtmf-relay rtp-nte 
 codec g711ulaw 
! 
dial-peer voice 1199 pots 
 destination-pattern 1199 
 port 0/3/0 
 no sip-register 
! 
dial-peer voice 2001 voip 
 description 8-800-Incoming-Dial-Peer 
 translation-profile incoming 88 
 session protocol sipv2 
 session transport udp 
 incoming called-number 222222 
 voice-class codec 1  
 dtmf-relay rtp-nte 
 no vad 
! 
dial-peer voice 5000 voip 
 tone ringback alert-no-PI 
 description ***Outgoing Call to LYNC*** 
 translation-profile outgoing 5000 
 destination-pattern 5... 
 progress_ind setup enable 3 
 rtp payload-type comfort-noise 13 
 session protocol sipv2 
 session target ipv4:10.110.1.32:5068 
 session transport tcp 
 voice-class codec 1  
 voice-class sip block 183 sdp present 
 voice-class sip bind control source-interface FastEthernet0/0.10 
 voice-class sip bind media source-interface FastEthernet0/0.10 
 dtmf-relay rtp-nte 
 fax protocol none 
 no vad 
! 
dial-peer voice 5001 voip 
 description ***Incoming Call from LYNC*** 
 translation-profile incoming Lync_OUT 
 session protocol sipv2 
 session transport tcp 
 incoming called-number +5...$ 
 voice-class codec 1  
 voice-class sip bind control source-interface FastEthernet0/0.10 
 voice-class sip bind media source-interface FastEthernet0/0.10 
 dtmf-relay rtp-nte 
 no vad 
! 
! 
num-exp 9 1199 
gateway 
 timer receive-rtp 1200 
R2c2801#sh run | s sip-ua
sip-ua 
 credentials username 111111 password 7 xxxxxxx realm voip.voiceisp.com
 credentials username 222222 password 7 xxxxxxx realm asterisk
 authentication username 111111 password 7 xxxxxxx realm REGISTRAR
 authentication username 222222 password 7 xxxxxxx realm asterisk
 no remote-party-id
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 retry register 5
 registrar 1 dns:voip.voiceisp.com:9060 expires 3600
 registrar 2 ipv4:10.10.10.1:9060 expires 3600
 sip-server ipv4:10.10.10.2:9060
 no suspend-resume
 host-registrar
 permit hostname dns:voip.voiceisp.com
Labels:
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1 Accepted Solution

Accepted Solutions

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Rajan
VIP Alumni
VIP Alumni
In response to sSiDs

‎09-27-2016 06:24 AM

Hi Sid,

If your SIP ISP do not need any authentication, then you could remove sip-ua configuration and directly point all the outbound dial-peers to the ITSP IP address instead of using " session target sip-server " as in the below dial-peer:

dial-peer voice 2 voip
description ***outbound INTERNATIONAL***
translation-profile outgoing calerid
destination-pattern 810.T
progress_ind setup enable 3
session protocol sipv2
session target ipv4:195.211.120.9
session transport udp
voice-class codec 1
voice-class sip profiles 1
voice-class sip bind control source-interface FastEthernet0/1
voice-class sip bind media source-interface FastEthernet0/1
dtmf-relay rtp-nte
no vad

Also make sure that these dial-peers are bound to FastEthernet0/1 which is used for accessing the outside network.

HTH

Rajan

View solution in original post

8 Replies 8

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sSiDs
Level 1
Level 1

‎09-08-2016 12:46 AM

forgot to add voice service voip settings

voice service voip
 ip address trusted list
 ipv4 195.111.1.234 //- voice ISP IP address
 ipv4 192.168.1.181
 ipv4 192.168.1.0 255.255.255.0
 ipv4 10.110.240.0 255.255.255.0 //-local voip netwotk for phones and CUCM
 address-hiding
 allow-connections sip to sip
 no supplementary-service h450.2
 no supplementary-service h450.3
 supplementary-service h450.12
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 redirect ip2ip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
 header-passing
 registrar server expires max 3600 min 120
 asserted-id pai
 midcall-signaling passthru
 no call service stop
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
!
voice class sip-profiles 1
 request INVITE sip-header Min-SE modify "1800" "6000"
0 Helpful

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Ahmed Adeyemi
Level 4
Level 4
In response to sSiDs

‎09-08-2016 08:14 AM

Hello Sid, they are several approaches to this you can take. If you are just looking to bind the traffic to your public IP, all you have to do is any traffic you are sending to session target sip-server, you would have to change the binding to go out whatever interface your public IP address is connected to. So for example change the underlined below:

another option would be to create a Voice class SIP profile and modify the from address to be that of the public IP.

Also remember that you would have to take of the sip-ua authentication since thats now want the ITSP would want.

dial-peer voice 1 voip 
 description ***outbound NATIONAL*** 
 translation-profile outgoing calerid 
 destination-pattern 8[2-9]......... 
 progress_ind setup enable 3 
 session protocol sipv2 
 session target sip-server 
 session transport udp 
 voice-class codec 1  
 voice-class sip profiles 1 
 voice-class sip bind control source-interface FastEthernet0/1 (Change to the interface of the public IP)
 voice-class sip bind media source-interface FastEthernet0/1 (Change to the interface of the public IP)
 dtmf-relay rtp-nte 
 no vad

Go to solution
sSiDs
Level 1
Level 1
In response to Ahmed Adeyemi

‎09-27-2016 05:05 AM

Could you please explain more detailed?

Does it mean what Voice ISP asked? "straight trunk" without SIP-UA

CUCM IP is 10.110.240.10

VoiceISP IP is 195.211.120.9

so I need to run "no SIP-UA"

and which dial-peer-voice need to be re-configued?

all of outgoing?

 my public interface is F0/1

FastEthernet0/1
ip address 193.106.XX.XXX 255.255.255.0
 ip access-group ANTISPOOFING in
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip nat outside
 ip virtual-reassembly in
 duplex auto
 speed auto
 no cdp enable
 no mop enabled

interface FastEthernet0/0
 no ip address
 speed 100
 full-duplex
!
interface FastEthernet0/0.2
 encapsulation dot1Q 2
 ip address 172.16.2.254 255.255.255.0
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip nat inside
 ip virtual-reassembly in
!
interface FastEthernet0/0.8
 description ***VOICE***
 encapsulation dot1Q 8
 ip address 10.110.240.254 255.255.255.0
 ip helper-address 10.110.1.22
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip nat inside
 ip virtual-reassembly in
!
interface FastEthernet0/0.10
 encapsulation dot1Q 10
 ip address 10.110.1.254 255.255.255.0
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip nat inside
 ip virtual-reassembly in
0 Helpful

Go to solution
Rajan
VIP Alumni
VIP Alumni
In response to sSiDs

‎09-27-2016 06:24 AM

Hi Sid,

If your SIP ISP do not need any authentication, then you could remove sip-ua configuration and directly point all the outbound dial-peers to the ITSP IP address instead of using " session target sip-server " as in the below dial-peer:

dial-peer voice 2 voip
description ***outbound INTERNATIONAL***
translation-profile outgoing calerid
destination-pattern 810.T
progress_ind setup enable 3
session protocol sipv2
session target ipv4:195.211.120.9
session transport udp
voice-class codec 1
voice-class sip profiles 1
voice-class sip bind control source-interface FastEthernet0/1
voice-class sip bind media source-interface FastEthernet0/1
dtmf-relay rtp-nte
no vad

Also make sure that these dial-peers are bound to FastEthernet0/1 which is used for accessing the outside network.

HTH

Rajan

Go to solution
devils_advocate
Level 7
Level 7

‎09-08-2016 02:08 AM

Presumably Fa0/1 has your Public IP?

Do you just have a single SIP provider or are there others?

Your config towards the ISTP is pointing at the following:

sip-server ipv4:10.10.10.2:9060

Is that the IP of your SIP provider?

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sSiDs
Level 1
Level 1
In response to devils_advocate

‎03-16-2018 05:43 AM

sorry for late reply and necro-posting)

10.10.10.2 it is ISP PBX address

fa0/1 has public i.e. 12.123.432.12

0 Helpful

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R0g22
Cisco Employee
Cisco Employee
In response to sSiDs

‎03-16-2018 06:12 AM

You are still facing this issue after 2 years ?

0 Helpful

Go to solution
sSiDs
Level 1
Level 1
In response to R0g22

‎04-25-2018 11:06 PM

no. now everything is OK.

0 Helpful
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