Staffan, I tried applying the changes but still cant make outgoing calls. btw, isn't sip trunking used only if i have an e1/t1 line to use it with? i online have a normals pots line....should i still enable it? Also, the auto attendants seems to have stopped working....any ideas? posted both debug results and updated config.....any further assistance would be greatly appreciated (if only i can send u beer via email.... :)
-Ron **************Debug results********************************** *Mar 12 11:53:35.398: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.9.20.254;branch=z9hG4bK-sipXecs-00ac37940c6f5c6672fad8f2e5fc 59307a94,SIP/2.0/TCP 10.9.20.254;branch=z9hG4bK-sipXecs-00a9d9cf0605eaecac753a4e cb794d5f3e12~f671e51cf1be4b37011e3481ad281e42,SIP/2.0/UDP 10.9.20.148;branch=z9h G4bKa85bb04f3E3300FC From: <sip:[email protected]>;tag=4C71031D-3B09BC1A To: <sip:[email protected];user=phone> Date: Fri, 12 Mar 2010 11:53:35 GMT Call-ID: [email protected] CSeq: 2 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 *Mar 12 11:53:35.418: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 404 Not Found Via: SIP/2.0/TCP 10.9.20.254;branch=z9hG4bK-sipXecs-00ac37940c6f5c6672fad8f2e5fc 59307a94,SIP/2.0/TCP 10.9.20.254;branch=z9hG4bK-sipXecs-00a9d9cf0605eaecac753a4e cb794d5f3e12~f671e51cf1be4b37011e3481ad281e42,SIP/2.0/UDP 10.9.20.148;branch=z9h G4bKa85bb04f3E3300FC From: <sip:[email protected]>;tag=4C71031D-3B09BC1A To: <sip:[email protected];user=phone>;tag=1377BE0-1369 Date: Fri, 12 Mar 2010 11:53:35 GMT Call-ID: [email protected] CSeq: 2 INVITE Allow-Ev HQRTR-3651#ents: telephone-event Server: Cisco-SIPGateway/IOS-12.x Reason: Q.850;cause=1 Content-Length: 0 *Mar 12 11:53:35.426: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: ACK sip:[email protected] <sip%[email protected]>;user=phone SIP/2.0 Contact: <sip:[email protected] <sip%[email protected]>;x-sipX-nonat> From: <sip:[email protected]>;tag=4C71031D-3B09BC1A To: <sip:[email protected];user=phone>;tag=1377BE0-1369 Call-Id: [email protected] Cseq: 2 ACK Max-Forwards: 20 Via: SIP/2.0/TCP 10.9.20.254;branch=z9hG4bK-sipXecs-00ac37940c6f5c6672fad8f2e5fc 59307a94 Content-Length: 0 ******************************************************************************************************** *******************Updated Config******************* ! 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 g711ulaw codec preference 2 g711alaw 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 3082024F 308201B8 A0030201 02020101 300D0609 2A864886 F70D0101 04050030 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 69666963 6174652D 32353836 30393936 3930301E 170D3130 30333130 30343039 34345A17 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 77307530 0F060355 1D130101 FF040530 030101FF 30220603 551D1104 1B301982 17485152 54522D33 3635312E 616C7374 65727068 2E6C616E 301F0603 551D2304 18301680 14B3CF27 AE887369 882B0BB5 ADB8A5F9 3D4115BE 7B301D06 03551D0E 04160414 B3CF27AE 88736988 2B0BB5AD B8A5F93D 4115BE7B 300D0609 2A864886 F70D0101 04050003 81810087 5BDE8173 31D82E63 835E1E5F F9A966CD 8892EF89 159A2CCF 8BDD4435 E23B0DAC D41CD525 E09CE1F8 C52CE870 2A7D6173 9CE21C59 71743B9E E9F8B02E F5CC8C8E BC895230 A7DC6F37 B50C07F5 24BE216B 004DDBD3 85085A5D 029BD891 92839089 CF3662D9 BE8FAD69 243F8691 3474CC9C 8B39819E 64F9DA9B 1367FB5B 380B11 quit ! ! username alsterph 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 20 voip description *** SIP trunk from SipX to router *** voice-class codec 1 session protocol sipv2 session target sip-server incoming called-number 9.T dtmf-relay rtp-nte fax protocol pass-through g711alaw no vad ! dial-peer voice 21 pots description *** Outbound to PSTN using FXO *** incoming called-number 9.T port 1/1/0 ! dial-peer voice 10 voip description *** SIP trunk from router to SipX *** destination-pattern 101 voice-class codec 1 session protocol sipv2 session target ipv4:10.9.20.254 dtmf-relay rtp-nte no vad ! ! 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 Mon, Mar 29, 2010 at 4:04 PM, Staffan Kerker <[email protected]>wrote: > Good to hear about progress! I currently don't have access to a Cisco > router with FXO interface, so there might be errors here. > > Now, if you only use _one_ FXO port, to my knowledge, you can only forward > it to _one_ extension. So, let's say that you configure > FXO-port 1/1/0 with plar 101, then all incoming PSTN-calls on that FXO > interface will end up with a SIP INVITE to sip:1...@sipserver. If > you wanna connect to another SipX "extension", you need multiple FXO > interfaces with different plar commands. However, forwarding > the call to an auto attendant on SipX should absolutely work... > > Now, for the outgoing part. You need a dial-peer that matches you outbound > call leg from SipX to the FXO interface. So, if you have > a SIP trunk configured in SipX that points to the Cisco router for a > specific dial plan rule, the digits you send on that trunk must have > a matching dial-peer on the router, connected to the FXO interface. > > Your dial-peer 100 voip matches the plar 101 and sends it to SipX. I would > configure the destination-pattern to explicitly match > the plar, but that's just me. > > Now, your dial-peer 10 is a pots dial-peer matching 8 digits starting with > a '9'. My guess is that this is what you send from > SipX on the trunk? Is this really the digits you send to SipX? pots > dial-peers usually (i'm not really sure about this) perform > 'pre-dot-strip', so the '9' would be stripped before sending out on the FXO > interface anyhow, even without the 'forward digits 7' command. > > Dial-peer 101 voip, this would be the dial-peer matching the incoming SIP > INVITE from SipX that is supposed to be sent out > on the FXO interface. Now, I would use the "incoming-called number" command > here to match instead of the "destination-pattern". > So, based on the previous dial-peer, "incoming-called number 9......." > > A good debug command to verify the SIP signaling in the router is: > > router# debug ccsip messages > > > ##### Rough out-of-the-head EXAMPLE ##### > PSTN: FXO 1/1/0, subscriber 555-12345 > Router IP: 10.10.10.2 > SipX IP: 10.10.10.1 > Prefix from SipX for PSTN access: '9' > Attendant extension on SipX, target for incoming PSTN calls: 100 > > ! > voice-port 1/1/0 > cptone SE > description *** FXO port 1, PSTN 555-12345 *** > port 1/1/0 > plar 100 > ! > ! > dial-peer voice 10 voip > description *** SIP trunk from router to SipX *** > destination-pattern 100 > voice-class codec 1 > session protocol sipv2 > sesstion target ip:10.10.10.1 > dtmf-relay rtp-nte > fax protocol pass-through g711alaw > no vad > ! > ! > dial-peer voice 20 voip > description *** SIP trunk from SipX to router *** > incoming-called number 9.T > voice-class codec 1 > sesstion protocol sipv2 > dtmf-relay rtp-nte > fax protocol pass-through g711alaw > no vad > ! > ! > dial-peer voice 21 pots > description *** Outbound to PSTN using FXO *** > incoming-called number 9.T > port 1/1/0 > ! > ! > ###### > > > Best regards, > /Staffan > > > -- > Staffan Kerker > mail/sip/xmpp: [email protected] > > "Don't get involved in politics man, just play the gig..." /Sgt Floyd, > Electric Mayhem Band > >
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