*****resending to include the users-list....sorry, my first time using
mailing lists :P *****
Hi Staffan,
thnx for replying below is my router config....just heads up though, i
only copied most of the configs (mainly the dialpeers) here as im not
exactly sure how dial peers work in SIP. I only ever used it with SCCP :P
copied it from
http://sipx-wiki.calivia.com/index.php/HowTo_configure_Cisco_SIP_Gateway_with_sipX
.
UPDATE: still cant make outgoing calls but im now able to receive calls from
the pstn...plar doesnt seem to work though. i have to go through a 2 step
process meaning ... when i call 8888888 and get a dialtone, i have to dial
the 100 extension (operator) and then dial a user extension from there. I
cant directly call a user extension when i get a dialtone as i only get a
busy signal that way...weird. There's also a really long delay for the line
to be released after ending a call.....so when you try calling the 8888888
number again after ending a call...you'll get either a busy tone or a
message saying the line is temporarily not available.
there are also some lines here which i don't know what it's for...if you can
also shed some light on it, that would be awesome
1.voice call carrier capacity active
voice rtp send-recv
2. voice source-group secured
disconnect-cause call-reject
3. (under the fxo port line)
supervisory disconnect dualtone pre-connect
supervisory answer dualtone
4. service session (on the link...it says "application session" but when i
type it in on the cli, i get a message saying the command has been
deprecated)
hope to hear from ya soon,
Ron
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname HQRTR-3651
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$SRYA$TK6vTyEzaX5owgjE7XiQ2.
!
no aaa new-model
no network-clock-participate slot 1
no network-clock-participate wic 0
ip cef
!
!
!
!
no ip domain lookup
ip domain name ourcompany.lan
ip name-server 10.9.20.254
!
multilink bundle-name authenticated
!
!
!
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
sip
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
!
!
!
!
!
!
!
!
!
!
!
!
voice source-group secured
access-list 1
disconnect-cause call-reject
!
!
!
crypto pki trustpoint TP-self-signed-2586099690
enrollment selfsigned
subject-name cn=IOS-Self-Signed-
Certificate-2586099690
revocation-check none
rsakeypair TP-self-signed-2586099690
!
!
crypto pki certificate chain TP-self-signed-2586099690
certificate self-signed 01
30820242 308201AB A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32353836 30393936 3930301E 170D3130 30333130 30343039
34355A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 35383630
39393639 3030819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
81009D86 97762ED4 38413BE3 92AA4A68 785F50C1 7AE134C8 4C302987 E6E4AF27
4E21C914 07FF958F 4D021E7C AE22B4CB 01A4597B 33D7D81A DE5453F0 9F78C7F6
A0BAC4DC 843D795F CF08BC4A E63986AB 1C2FBFC3 190CEE12 9BA1281A D852AECD
3853A629 62AEF558 67D8D724 3ED2D832 11E02534 AA8773E2 A13AF72C 0A304934
3DAF0203 010001A3 6A306830 0F060355 1D130101 FF040530 030101FF 30150603
551D1104 0E300C82 0A485152 54522D33 36353130 1F060355 1D230418 30168014
B3CF27AE 88736988 2B0BB5AD B8A5F93D 4115BE7B 301D0603 551D0E04 160414B3
CF27AE88 7369882B 0BB5ADB8 A5F93D41 15BE7B30 0D06092A 864886F7 0D010104
05000381 8100796B 08980E24 07D57AF5 76544976 D098E414 F42CEAAE B2BD8C75
6B51B87C 11924376 5EBA5466 404177C0 AA10BF6D 1441F138 7BF951AC 6C8EC1EC
EE30EECD 5A527D8B 662C6D1A E9EBA8F7 DB5264AA D9C70287 12AB8B9F ADA8C149
552435F9 512DC915 616796ED A37101D1 06C1676B 49C87E4C 11D8D1A1 AFF70B8B
605DD70D 53A6
quit
!
!
username ourcompany privilege 15 password 0 pass
archive
log config
hidekeys
!
!
!
!
ip tcp path-mtu-discovery
!
!
!
!
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 10.1.1.1 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1.10
encapsulation dot1Q 10
ip address 10.9.10.1 255.255.255.0
!
interface FastEthernet0/1.20
encapsulation dot1Q 20
ip address 10.9.20.1 255.255.255.0
!
interface FastEthernet0/1.30
encapsulation dot1Q 30
ip address 10.9.30.1 255.255.255.0
!
router rip
version 2
network 10.0.0.0
network 192.168.1.0
!
ip forward-protocol nd
!
!
ip http server
ip http authentication local
ip http secure-server
!
snmp-server packetsize 4096
snmp-server enable traps tty
!
!
!
!
!
control-plane
!
!
!
voice-port 1/1/0
supervisory disconnect dualtone pre-connect
supervisory answer dualtone
input gain 8
no vad
cptone PH
connection plar 101
!
voice-port 1/1/1
!
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 100 voip
huntstop
service session
destination-pattern ...
rtp payload-type nte 98
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
!
dial-peer voice 10 pots
huntstop
service session
destination-pattern 9.......
port 1/1/0
forward-digits 7
!
dial-peer voice 101 voip
huntstop
service session
destination-pattern ........
rtp payload-type nte 98
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
!
dial-peer voice 130 pots
service session
destination-pattern 0$
port 1/1/0
forward-digits all
!
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 1200
!
sip-ua
max-forwards 15
sip-server ipv4:10.9.20.254
!
!
!
line con 0
exec-timeout 0 0
password pass
logging synchronous
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
password pass
logging synchronous
login local
transport input telnet ssh
!
ntp server 10.9.20.254
!
end
On Thu, Mar 25, 2010 at 9:48 PM, Staffan Kerker <[email protected]>wrote:
>
> On 25 mar 2010, at 08.42, ronald teng wrote:
>
> I configured my gateway as per instructions from
> http://sipx-wiki.calivia.com/images/d/db/Cisco-SipX-TDM-SIP-GW.pdf(although
> im not using a T1/E1 line) but im neither able to make nor receive pstn
> calls. When i call our number 8888888(not our real number) using a different
> line, i just get a dial tone. I tried dialling an extension after i get the
> dialtone and i get a busy tone. When i try calling out, it just gives me a
> busy tone. I tried doing a debug on my router and got the results posted
> below. I can also send my router script if necessary. Pls help
>
>
> Without having seen your configuration or knowing what voice interface
> cards you are using it's hard to give any help. If you send me your
> configuration I'm happy to give it a try.
>
> If you just get a dial tone or busy tones it's usually due to misconfigured
> dial-peers and destination-patterns. If you are using different voice cards
> than PRI (FXO, FXS, BRI) the configuration in obove referenced guide has to
> be modified.
>
> Best regards
> /Staffan
>
>
> --
> Staffan Kerker
> mail/sip/xmpp: [email protected]
>
> "There is absolutely no money above the 5th fret..." /Donald "Duck" Dunn
>
>
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