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
Solved! Go to Solution.
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
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"
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
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
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
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?
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
03-16-2018 06:12 AM
04-25-2018 11:06 PM
no. now everything is OK.
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