01-12-2012 04:37 AM - edited 03-16-2019 08:58 AM
Hello all,
I am trying to connect elastix with AS5300 cisco gateway via sip trunk
here is router config
sip-ua
authentication username XXX password XXX
retry invite 2
retry response 2
retry bye 2
retry cancel 2
retry options 2
registrar ipv4:192.168.113.211:5060 expires 3600
sip-server ipv4:192.168.113.211:5060
dial-peer voice 113 voip
destination-pattern 67520004.
session protocol sipv2
session target ipv4:192.168.113.211:5060
dtmf-relay rtp-nte
codec g729br8
elastix sip trunk
dial peer
host=192.168.113.6
username=XXX
secret=XXX
type=peer
canredirect=no
canreinvite=no
allow=g729br8
disallow=all
user detail
secret=XXX
type=user
context=from-trunk
username=XXX
fromuser=mena
qualify=no
canreinvite=no
allow=g729br8
disallow=all
always when i try to call any number give me a circuit busy message, is any document or help to such like this configuration between elastix and as5300 cisco gateway
Solved! Go to Solution.
01-12-2012 07:44 AM
No trace, no troubleshoot.
01-12-2012 04:39 AM
Router config is incomplete, it must have at least one dial-peer pots.
01-12-2012 04:52 AM
my network configuration as following :
as5300 working as h323 voice gateway on f0/0 and i want to add a sip gateway to as5300
here is the complete configuration of the as5300 VGW
Current configuration : 4393 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname XX
!
boot-start-marker
boot system flash c5350-is-mz.124-11.T4.bin
boot system flash c5350-is-mz.123-3g.bin
no boot startup-test
boot-end-marker
!
logging monitor critical
enable secret 5 $1$QhHi$sv47RYidJJvFcyRmv5ved1
!
!
!
resource-pool disable
no aaa new-model
clock timezone utc 2
spe default-firmware spe-firmware-1
!
!
ip cef
no ip domain lookup
!
!
multilink bundle-name authenticated
isdn switch-type primary-net5
!
voice call send-alert
voice rtp send-recv
!
voice service pots
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
!
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
h245 tunnel disable
h245 caps mode restricted
modem passthrough nse codec g711alaw
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server
!
!
voice class codec 1
codec preference 1 g729br8 bytes 60
!
!
!
!
!
!
!
!
voice translation-rule 1
rule 1 /^9#0/ /999/
!
!
voice translation-profile test
translate called 1
!
!
!
fax interface-type fax-mail
dial-control-mib retain-timer 1000
dial-control-mib max-size 1200
!
username x password XX
username X password XX
username X password XX
!
!
controller E1 3/0
framing NO-CRC4
pri-group timeslots 1-31
!
controller E1 3/1
framing NO-CRC4
pri-group timeslots 1-31
!
!
!
interface FastEthernet0/0
ip address 192.168.14.80 255.255.255.0
load-interval 30
duplex full
speed 100
h323-gateway voip interface
h323-gateway voip id gk-zone1 ipaddr 192.168.14.70 1719 priority 126
h323-gateway voip h323-id AlkanVGW
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 192.168.14.80
!
interface FastEthernet0/1
ip address 192.168.113.6 255.255.255.0
duplex auto
speed auto
!
interface Serial0/0
no ip address
shutdown
clock rate 2000000
!
interface Serial0/1
no ip address
shutdown
clock rate 2000000
!
interface Serial3/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn timer T310 400000
isdn incoming-voice modem
isdn sending-complete
no cdp enable
!
interface Serial3/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn timer T310 400000
isdn incoming-voice modem
isdn sending-complete
no cdp enable
!
interface Group-Async0
no ip address
encapsulation slip
group-range 1/00 1/59
!
!
ip route 0.0.0.0 0.0.0.0 192.168.14.33
ip route 10.0.0.0 255.0.0.0 192.168.14.254
ip route 10.20.1.0 255.255.255.240 192.168.14.33
no ip http server
!
!
snmp-server community xxx
!
!
!
control-plane
!
!
!
voice-port 3/0:D
cptone EG
bearer-cap Speech
!
voice-port 3/1:D
cptone EG
bearer-cap Speech
!
!
!
dial-peer voice 2 pots
destination-pattern 0.
direct-inward-dial
port 3/0:D
!
dial-peer voice 1 voip
tone ringback alert-no-PI
destination-pattern .
progress_ind setup enable 3
session target ras
codec g729br8 bytes 60
fax-relay ecm disable
fax rate 9600
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
!
dial-peer voice 3 pots
destination-pattern 0.
direct-inward-dial
port 3/1:D
!
dial-peer voice 1234 voip
destination-pattern 110.
session target ipv4:10.181.24.241
codec g729br8
!
dial-peer voice 122 voip
destination-pattern 122.
session target ipv4:10.181.28.245
codec g729br8
!
dial-peer voice 121 voip
destination-pattern 121.
session target ipv4:10.181.28.237
codec g729br8
!
dial-peer voice 113 voip
destination-pattern 67520004.
session protocol sipv2
session target ipv4:192.168.113.211:5060
dtmf-relay rtp-nte
codec g729br8
!
dial-peer voice 114 voip
!
!
gateway
timer receive-rtp 1200
!
sip-ua
authentication username xx password xx
retry invite 2
retry response 2
retry bye 2
retry cancel 2
retry options 2
registrar ipv4:192.168.113.211:5060 expires 3600
sip-server ipv4:192.168.113.211:5060
notify ignore substate
!
ss7 mtp2-variant Bellcore 0
ss7 mtp2-variant Bellcore 1
ss7 mtp2-variant Bellcore 2
ss7 mtp2-variant Bellcore 3
!
line con 0
line aux 0
line vty 0 4
password xx
login
line 1/00 1/59
modem InOut
!
scheduler allocate 10000 400
ntp clock-period 17180036
ntp server 192.168.14.33
end
i made this dial peer 113 and sip-ua in the previous code to add elstix as a sip proxy in the network and use a sip phone too beside h323
01-12-2012 05:17 AM
Your voip DPs need to have session protocol sipv2. Otherwise, they use H.323
Then you can take "debug ccsip message" with "term mon" to see what happens.
01-12-2012 06:30 AM
only i want apply sip gateway on int f0/1 to configure sip phone going out through pstn. and i configured session protocol in dial-peer 113 and the outage call (PSTN) should went out throurgh dial-peer 2 so should i configure sip session protocol in dial-peer 2
Thank you very match for your interest
01-12-2012 07:44 AM
No trace, no troubleshoot.
01-13-2012 05:33 AM
realy, thank you for your interest, i had a doubt that it will work. Can you further recommend me sip proxy or server working with as5300 and has a good billing system.
but any way thanks alot for your help.
02-22-2013 12:17 AM
Hi Mena,
could you please describe what hardware are you using with the AS5300? E1 card/ VoIP card/ DSPs?
Thanks a lot.
Regards
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide