Re: [c-nsp] SIP VoIP Config
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//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//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 08 password no remote-party-id registrar ipv4:1.2.3.4 expires 3600
Re: [c-nsp] SIP VoIP Config
Hi Ben, Done it already. Thanks Pedro Matusse -Original Message- From: Ben Steele [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 08, 2008 3:58 AM To: [EMAIL PROTECTED] Cc: cisco-nsp@puck.nether.net Subject: Re: [c-nsp] SIP VoIP Config If you haven't already, try posting this in the cisco-voip mailing list, they are very active, [EMAIL PROTECTED] Ben 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 cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/ ___ cisco-nsp mailing list cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/
Re: [c-nsp] SIP VoIP Config
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 08 password 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 cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/ ___ cisco-nsp mailing list cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/
Re: [c-nsp] SIP VoIP Config
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 1:14 pm Subject: Re: [c-nsp] SIP VoIP Config 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 08 password 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 cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/ ___ cisco-nsp mailing list cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/
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 08 password 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 cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/ SIP Call Debug.TXTSIP Call Debug 2.TXT Catembe# Catembe# Catembe# Catembe# Catembe# Catembe#sh run Building configuration... Current configuration : 4895 bytes ! 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 . ! aaa new-model ! ! ! ! aaa session-id common clock timezone PCTime 2 no network-clock-participate slot 1 network-clock
[c-nsp] SIP VoIP Config
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 sh run Building configuration... Current configuration : 4612 bytes ! 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 ! aaa new-model ! ! ! ! aaa session-id common clock timezone PCTime 2 no network-clock-participate slot 1 network-clock-participate wic 0 ! ! ip cef ip tcp synwait-time 10 ! ! no ip bootp server no ip domain lookup ip domain name ? ip name-server ip name-server ? ip name-server ? ip name-server ? ! 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 g711ulaw codec preference 2 g729r8 ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! username ? password ? username ? privilege 15 password ? ! ! controller E1 0/0/0 clock source line primary channel-group 0 timeslots 1-31 ! controller E1 0/0/1 ! controller T1 1/0 framing esf clock source line primary linecode b8zs cablelength short 133 pri-group timeslots 1-24 description Dialogic Production IVR Board (D/240SC-T1) ! controller T1 1/0/0 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 ? ipaddr ? 1718 h323-gateway voip h323-id h323-gateway voip tech-prefix 258# ! interface GigabitEthernet0/0 ip address 192.168.4.254 255.255.255.0 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 ? 255.255.255.252 ip nat outside ip virtual-reassembly ! 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 6 isdn timer T321 0 isdn incoming-voice voice isdn T309-enable isdn sending-complete no cdp enable ! interface Serial1/0/0:23 no ip address encapsulation hdlc autodetect encapsulation ppp v120 lapb-ta no snmp trap link-status isdn switch-type primary-ni isdn timer T321 0 isdn incoming-voice voice no fair-queue 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 ? ! ! ip http server ip http authentication local no ip http secure-server ip nat pool ? ? netmask 255.255.255.248 ip nat inside source list 1 pool ? overload ! ! access-list 1 permit 0.0.0.255 no cdp run ! ! ! ! ! ! control-plane ! ! ! voice-port 1/0:23 bearer-cap 3100Hz ! voice-port 1/0/0:23 bearer-cap 3100Hz ! ! ! ! ! dial-peer voice 123 pots service session answer-address 8882785987 destination-pattern 888... port 1/0:23 forward-digits all ! dial-peer voice 234 pots answer-address 888... destination-pattern 888... port 1/0/0:23 forward-digits all ! dial-peer voice 100 voip service session destination-pattern .T redirect ip2ip voice-class codec 1 session protocol sipv2 session target sip-server no vad ! ! gateway timer receive-rtp 1200 ! sip-ua disable-early-media 180 retry invite 4 retry response 2 retry bye 2 retry cancel 2 retry notify 2 retry options 0 oli sip-server ipv4:? ! ! banner login ^Authorized access only! Disconnect IMMEDIATELY if you are not an authorized user!^C ! line con 0 password stopbits 1 line aux 0 stopbits 1 line vty 0 4 exec-timeout 0 0 password ! scheduler allocate 3 4000 ! end ___ cisco-nsp mailing list cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/
Re: [c-nsp] SIP VoIP Config
If you haven't already, try posting this in the cisco-voip mailing list, they are very active, [EMAIL PROTECTED] Ben 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 cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/ ___ cisco-nsp mailing list cisco-nsp@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-nsp archive at http://puck.nether.net/pipermail/cisco-nsp/
