*bump* On Fri, Mar 26, 2010 at 11:04 AM, ronald teng <[email protected]> wrote:
> *****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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