further update: outgoing calls still dead but plar now up...seems i made a blunder....configured the wrong voice port...already corrected. I am now having problems w/ the auto attendant (extension 100)not working....the voicemail attendant (101) seems to be working fine though....anyone pls help. thnx.
On Mon, Mar 29, 2010 at 9:52 AM, ronald teng <[email protected]> wrote: > *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 >>> >>> >> >
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
