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>
>
>
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-participate wic 0
!
!
ip cef
ip tcp synwait-time 10
!
!
no ip bootp server
no ip domain lookup
ip domain name
ip name-server x.x.x.1
ip name-server x.x.x.2
ip name-server x.x.x.1
ip name-server x.x.x.2
!
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 xxxx password xxxx
username xxx privilege 15 password xxxx
!
!
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 xxxxx ipaddr x.x.x.x 1718
h323-gateway voip h323-id xxxxx
h323-gateway voip tech-prefix 258#
!
interface GigabitEthernet0/0
ip address pr.pr.pr.pr 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 pl.pl.pl.pl 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 60000
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 pl.pl.pl.pl
!
!
ip http server
ip http authentication local
no ip http secure-server
ip nat pool mmmmm x.x.x.x x.x.x.x netmask 255.255.255.248
ip nat inside source list 1 pool mmmmm overload
!
!
access-list 1 permit x.x.x.0 0.0.0.255
no cdp run
!
!
!
!
!
!
control-plane
!
!
!
voice-port 1/0:23
timeouts interdigit 4
timeouts call-disconnect 2
timeouts wait-release 10
bearer-cap speech
!
voice-port 1/0/0:23
timeouts interdigit 4
timeouts call-disconnect 2
timeouts wait-release 10
bearer-cap speech
!
!
!
!
!
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 ipv4:x.x.x.x
dtmf-relay sip-notify rtp-nte
signal-type ext-signal
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:x.x.x.x expires 3600
sip-server ipv4:x.x.x.x:5060
!
!
banner login ^CCCCAuthorized access only!
Disconnect IMMEDIATELY if you are not an authorized user!^C
!
line con 0
password fffffffffff
stopbits 1
line aux 0
stopbits 1
line vty 0 4
exec-timeout 0 0
password fffffffffffff
!
scheduler allocate 30000 4000
!
end
Catembe#
Catembe#
Catembe#
Catembe#
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