Hi Tom I've managed to get it working, tanks. The working config follow in attach.
Now I've a second issue. The outbound calls are supposed to come from a CT Server (with a Dialogic D/240SC-T1 card) that connects to the router via a T1. During the test phase I'm also using an FXS. >From the telephone connected to the FXS the call goes fine but from a telephone connected to the CT server there's a lot of noise added to the call channel. Any idea? Kind regards Pedro -----Original Message----- From: Tom Storey [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 08, 2008 3:39 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [c-nsp] SIP VoIP Config The only thing I can see wrong is the following: 001665: *Apr 8 14:41:45.225 PCTime: //-1/xxxxxxxxxxxx/SIP/Msg/ ccsipDisplayMsg: Sent: REGISTER sip:Destination_IP:5060 SIP/2.0 Via: SIP/2.0/UDP Source_IP:5060;branch=z9hG4bK5AC47 From: <sip:[EMAIL PROTECTED]>;tag=54447D0-DBD To: <sip:[EMAIL PROTECTED]> Date: Tue, 08 Apr 2008 12:41:45 GMT Call-ID: B9EFB396-48E11DD-A57D8CCE-6E567B30 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1207658505 CSeq: 43 REGISTER Contact: <sip:[EMAIL PROTECTED]:5060> Expires: 3600 Content-Length: 0 This is your router trying to register with your VoIP provider, but look at what your VoIP provider is sending back: 001667: *Apr 8 14:41:46.093 PCTime: //-1/xxxxxxxxxxxx/SIP/Msg/ ccsipDisplayMsg: Received: SIP/2.0 404 Not found Via: SIP/2.0/UDP Source_IP:5060;branch=z9hG4bK5AC47 From: <sip:[EMAIL PROTECTED]>;tag=54447D0-DBD To: <sip:[EMAIL PROTECTED]>;tag=as60705731 Call-ID: B9EFB396-48E11DD-A57D8CCE-6E567B30 CSeq: 43 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Since you do not specify an "authentication" command in your sip-ua configuration, the router is trying to register the number of your POTS dial-peer(s). Since the VoIP provider doesnt know about the numbers you are trying to register (888.......) they are sending back a 404 to indicate the number is not valid. You should check with your VoIP provider and see if you have a username (i.e. phone number) and password you need to specify when setting up a SIP client, and use an authentication line like I have in my config. Tom On 08/04/2008, at 9:56 PM, <[EMAIL PROTECTED]> <[EMAIL PROTECTED]> wrote: > Hi Tom, > > > In attach SIP messages. Note that I've replaced IP Addresses > with "Source_IP" and "Destination_IP" or "Destination_IP + n on the > last > octet". > > "Destination_IP + n on the last octet" means that on the SIP message > I'm getting de destination SIP gateway address and some oder IPs that > differ from the destination on the last octet. > > Pedro Wiliamo Matusse > Telecomunicações de Moçambique (TDM) > DSI > Tel. +258 21 482820 > Cell. +258 82 3080780 > Fax: +258 21 487812 > > ----- Original Message ----- > From: <[EMAIL PROTECTED]> > Date: Tuesday, April 8, 2008 1:58 pm > Subject: Re: [c-nsp] SIP VoIP Config > >> >> >> Going to send "debug ccsip messages" out put. >> >> "session >>> target sip-server". Is sip-server actually what you have in >> there, >>> or >>> do you normally have an IP address? >> >> Not sure, I'm in Africa and have SIP gateway in US. >> >> In attach the updated SIP config. >> >> >> Pedro Wiliamo Matusse >> Telecomunicações de Moçambique (TDM) >> DSI >> Tel. +258 21 482820 >> Cell. +258 82 3080780 >> Fax: +258 21 487812 >> >> ----- Original Message ----- >> From: Tom Storey <[EMAIL PROTECTED]> >> Date: Tuesday, April 8, 2008 1:35 pm >> Subject: Re: [c-nsp] SIP VoIP Config >> >>> Can you turn off all debugging, and then turn on "debug ccsip >>> messages" and forward that to me. >>> >>> I also notice that in your dial-peer 100 config you have >> "session >>> target sip-server". Is sip-server actually what you have in >> there, >>> or >>> do you normally have an IP address? >>> >>> Can you send through a more recent copy of your SIP configuration? >>> >>> >>> On 08/04/2008, at 8:44 PM, <[EMAIL PROTECTED]> >> <[EMAIL PROTECTED]> wrote: >>> >>>> Hi Tom, >>>> >>>> sending again >>>> >>>> >>>> Pedro Wiliamo Matusse >>>> Telecomunicações de Moçambique (TDM) >>>> DSI >>>> Tel. +258 21 482820 >>>> Cell. +258 82 3080780 >>>> Fax: +258 21 487812 >>>> >>>> ----- Original Message ----- >>>> From: Tom Storey <[EMAIL PROTECTED]> >>>> Date: Tuesday, April 8, 2008 1:22 pm >>>> Subject: Re: [c-nsp] SIP VoIP Config >>>> >>>>> I dont see any attached files ? >>>>> >>>>> On 08/04/2008, at 8:21 PM, <[EMAIL PROTECTED]> >>>> <[EMAIL PROTECTED]> wrote: >>>>> >>>>>> Hi Tom >>>>>> >>>>>> >>>>>> Thank you. Adapted you config but still no working. >>>>>> >>>>>> Can you please have a look on the debug output in attach. >>>>>> >>>>>> Kind Regards >>>>>> >>>>>> Pedro Wiliamo Matusse >>>>>> Telecomunicações de Moçambique (TDM) >>>>>> DSI >>>>>> Tel. +258 21 482820 >>>>>> Cell. +258 82 3080780 >>>>>> Fax: +258 21 487812 >>>>>> >>>>>> ----- Original Message ----- >>>>>> From: Tom Storey <[EMAIL PROTECTED]> >>>>>> Date: Tuesday, April 8, 2008 10:55 am >>>>>> Subject: Re: [c-nsp] SIP VoIP Config >>>>>> >>>>>>> Hi. >>>>>>> >>>>>>> If it helps, I recently configured a 1760 to connect to my ISPs >>>>>>> VoIP >>>>>>> service, and this is the config I used for my sip-ua: >>>>>>> >>>>>>> sip-ua >>>>>>> authentication username 08xxxxxxxx password xxxx >>>>>>> no remote-party-id >>>>>>> registrar ipv4:1.2.3.4 expires 3600 >>>>>>> sip-server ipv4:1.2.3.4:5060 >>>>>>> ! >>>>>>> >>>>>>> Initially I had issues where my calls didnt appear to be >> dialled> >>>> via >>>>>>> the VoIP provider, but with a bit of debugging from both >> ends we >>>>>>> figured out that I had to "no" the "remote-party-id" feature, >>>>>>> hence >>>>>>> you see "no remote-party-id" line in my config. >>>>>>> >>>>>>> The symptoms of my issue were I would dial the number, and it >>>>>>> would >>>>>>> sit there as if it were waiting for more characters, or it was >>>>>>> trying >>>>>>> to dial, and would eventually time out. It turns out it was >>>>>>> actually >>>>>>> dialling the number, but my VoIP provider was rejecting the >> call.> >>>> >>>>>>> You can use "debug ccsip" to see SIP messages to/from your >>>> router, >>>>>>> >>>>>>> this can help to get clues about what it going on (beware >>> that SIP >>>>>>> is >>>>>>> quite chatty, so a lot of output can be produced at times). >>>>>>> >>>>>>> For reference, my dial-peers/voice-ports look like this: >>>>>>> >>>>>>> voice-port 3/0 >>>>>>> cptone AU >>>>>>> timeouts interdigit 4 >>>>>>> timeouts call-disconnect 2 >>>>>>> timeouts wait-release 10 >>>>>>> description ** FXS right ** >>>>>>> ! >>>>>>> dial-peer voice 100 pots >>>>>>> destination-pattern 08........ >>>>>>> port 3/0 >>>>>>> ! >>>>>>> dial-peer voice 200 voip >>>>>>> destination-pattern [0,1][2-4,7,8]........ >>>>>>> session protocol sipv2 >>>>>>> session target ipv4:1.2.3.4 >>>>>>> dtmf-relay sip-notify rtp-nte >>>>>>> signal-type ext-signal >>>>>>> codec g711alaw >>>>>>> no vad >>>>>>> ! >>>>>>> >>>>>>> Other than the config above, I have zero other config >> related to >>>>>>> voice >>>>>>> on this router - no translation rules, codec profiles, etc - >> the >>>>>>> above >>>>>>> two snips of config are it! >>>>>>> >>>>>>> My setup is working 100% fine, inbound and outbound. >>>>>>> >>>>>>> Hope that helps. :-) >>>>>>> >>>>>>> Tom >>>>>>> >>>>>>> On 08/04/2008, at 6:38 AM, <[EMAIL PROTECTED]> >>>>>> <[EMAIL PROTECTED]> wrote: >>>>>>> >>>>>>>> Hi There, >>>>>>>> >>>>>>>> >>>>>>>> Trying to make calls from a POTS do VOIP in SIP setup in >> attach, >>>>>>> calls> from POTS are not beeing forwarded to VoIP port. >>>>>>>> >>>>>>>> Can any one help >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Pedro Wiliamo Matusse >>>>>>>> Telecomunicações de Moçambique (TDM) >>>>>>>> DSI >>>>>>>> Tel. +258 21 482820 >>>>>>>> Cell. +258 82 3080780 >>>>>>>> Fax: +258 21 487812 >>>>>>>> <config HJ3825 07 04 2008 23 >>>>>>>> >> 00h.TXT>_______________________________________________ >>>>>>>> cisco-nsp mailing list [email protected] >>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-nsp >>>>>>>> archive at http://puck.nether.net/pipermail/cisco-nsp/ >>>>>>> >>>>>>> >>>>>> >>>>> >>>>> >>>> <SIP Call Debug.TXT><SIP Call Debug 2.TXT> >>> >>> >> > <SIP Messages IP ADD Replaced.TXT>
sh run Building configuration... Current configuration : 4630 bytes ! ! Last configuration change at 00:29:10 GMT Wed Apr 9 2008 by john ! NVRAM config last updated at 00:29:13 GMT Wed Apr 9 2008 by john ! version 12.4 service tcp-keepalives-in service tcp-keepalives-out service timestamps debug datetime msec localtime show-timezone service timestamps log datetime msec localtime show-timezone service password-encryption service udp-small-servers service tcp-small-servers service sequence-numbers ! hostname Catembe ! boot-start-marker boot-end-marker ! card type t1 1 1 logging buffered 4096 no logging console enable secret 5 $1$vj4y$1o9WBE13g9XexuUWuK2nu. ! aaa new-model ! ! ! ! aaa session-id common clock timezone GMT 2 no network-clock-participate slot 1 network-clock-participate wic 0 no network-clock-participate wic 1 ! ! ip cef ip tcp synwait-time 10 ! ! no ip bootp server no ip domain lookup ip domain name howradjohnson.co.mz ip name-server xxx.xx.xx.xx ip name-server xxx.xx.xx.xx ip name-server xxx.xx.xx.xx ip name-server xxx.xx.xx.xx ! multilink virtual-template 1 multilink bundle-name authenticated ! isdn switch-type primary-ni voice-card 0 no dspfarm dsp services dspfarm ! voice-card 1 no dspfarm ! ! ! ! voice service voip redirect ip2ip sip bind control source-interface Serial0/0/0:0 bind media source-interface Serial0/0/0:0 ! ! voice class codec 1 codec preference 1 g729r8 ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! username catembe password 7 0453010B1F35545713495D username john privilege 15 password 7 060A062749041A0C15071D1918 ! ! controller E1 0/0/0 clock source line primary channel-group 0 timeslots 1-31 ! controller E1 0/0/1 ! controller T1 0/1/0 framing esf linecode b8zs ! controller T1 1/0 framing esf linecode b8zs cablelength short 133 pri-group timeslots 1-24 description Dialogic Production IVR Board (D/240SC-T1) ! translation-rule 1 Rule 1 1.......... 14050 ! ! ! ! ! ! ! interface Loopback0 no ip address h323-gateway voip interface h323-gateway voip id sip.thevoicemanager.com ipaddr xxx.xx.xx.xx 1718 h323-gateway voip h323-id [EMAIL PROTECTED] h323-gateway voip tech-prefix 258# ! interface GigabitEthernet0/0 ip address xxx.xx.xx.xx 255.255.255.0 ip nat inside ip virtual-reassembly duplex auto speed auto media-type rj45 no keepalive ! interface GigabitEthernet0/1 no ip address shutdown duplex auto speed auto media-type rj45 no keepalive ! interface Serial0/0/0:0 ip address xxx.xx.xx.xx 255.255.255.252 ip nat outside ip virtual-reassembly ip rtp header-compression passive ! interface Serial1/0:23 no ip address encapsulation ppp autodetect encapsulation ppp v120 lapb-ta no snmp trap link-status isdn switch-type primary-ni isdn timer T310 60000 isdn timer T321 0 isdn incoming-voice voice isdn T309-enable isdn bchan-number-order ascending isdn sending-complete no cdp enable ! interface Virtual-Template1 no ip address ppp multilink ppp multilink interleave ppp multilink fragment delay 20 ip rtp reserve 16384 100 64 ! ip route 0.0.0.0 0.0.0.0 xxx.xx.xx.xx ! ! ip http server ip http authentication local no ip http secure-server ip nat pool nnnnnnn xxx.xx.xx.xx xxx.xx.xx.xx netmask 255.255.255.248 ip nat inside source list 1 pool nnnnnnn overload ! ip access-list extended NAT-TRANSLATION permit ip any host xxx.xx.xx.xx ! access-list 1 permit xxx.xx.xx.xx 0.0.0.255 no cdp run ! ! ! ! ! ! control-plane ! ! ! voice-port 1/0/0 ! voice-port 1/0/1 ! voice-port 1/0:23 timeouts interdigit 4 timeouts call-disconnect 2 timeouts wait-release 10 ! ! ! ! ! dial-peer voice 123 pots service session destination-pattern 8.......... direct-inward-dial port 1/0:23 forward-digits all ! dial-peer voice 258 pots destination-pattern 258.T port 1/0/0 forward-digits all ! dial-peer voice 259 pots destination-pattern 259.T port 1/0/1 forward-digits all ! dial-peer voice 100 voip service session destination-pattern .T redirect ip2ip voice-class codec 1 session protocol sipv2 session target ipv4:xxx.xx.xx.xx no vad ! ! gateway timer receive-rtp 1200 ! sip-ua no remote-party-id disable-early-media 180 retry invite 4 retry response 2 retry bye 2 retry cancel 2 retry notify 2 retry options 0 oli registrar ipv4:xxx.xx.xx.xx expires 3600 sip-server ipv4:xxx.xx.xx.xx:5060 ! ! banner login ^CCCCAuthorized access only! Disconnect IMMEDIATELY if you are not an authorized user!^C ! line con 0 password 7 05030C023158561003554F stopbits 1 line aux 0 stopbits 1 line vty 0 4 exec-timeout 0 0 password 7 011B0C094B1F1E163B1C16 ! scheduler allocate 30000 4000 ! end Catembe# exit
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