[Asterisk-Users] cisco AS5300 : problem configuration
Areski
areski at e-group.org
Mon Sep 29 09:04:46 MST 2003
Hi again !!!
I commented out but still have the same problem.
I hear the first number of the Agi script "SAy digits 754546", I will
hear 7 but plof, it's disconnected after and I see the error below :
NOTICE[37899]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible
200 result=0
If I play a sound file, it's the same, that start but it's disconnected
after 1 seconds.
Here below what I m getting in the debug file :
Sep 29 18:01:21 DEBUG[8201]: File chan_sip.c, Line 1485 (sip_alloc):
Allocating new SIP call for
828D64BF-F1CA11D7-8A578158-AC69E3F9 at 213.232.105.12
Sep 29 18:01:21 DEBUG[8201]: File chan_sip.c, Line 4811
(handle_request): Check for res
Sep 29 18:01:21 DEBUG[8201]: File chan_sip.c, Line 932 (find_user): is
not a local user
Sep 29 18:01:21 DEBUG[8201]: File chan_sip.c, Line 3252 (build_route):
build_route: Contact hop: <sip:213.232.105.12:5060>
Sep 29 18:01:21 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:21 DEBUG[38923]: File pbx.c, Line 1143
(pbx_extension_helper): Launching 'Ringing'
Sep 29 18:01:21 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:21 DEBUG[38923]: File pbx.c, Line 1143
(pbx_extension_helper): Launching 'AGI'
Sep 29 18:01:21 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:21 DEBUG[38923]: File channel.c, Line 1456
(ast_set_write_format): Set channel SIP/-08102b70 to write format GSM
Sep 29 18:01:21 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:21 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:21 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:21 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:21 DEBUG[38923]: File rtp.c, Line 1007 (ast_rtp_write):
Ooh, format changed from UNKN to ALAW
Sep 29 18:01:21 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:22 DEBUG[38923]: File channel.c, Line 1456
(ast_set_write_format): Set channel SIP/-08102b70 to write format ALAW
Sep 29 18:01:22 DEBUG[38923]: File channel.c, Line 1456
(ast_set_write_format): Set channel SIP/-08102b70 to write format GSM
Sep 29 18:01:22 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:22 DEBUG[38923]: File channel.c, Line 1456
(ast_set_write_format): Set channel SIP/-08102b70 to write format ALAW
Sep 29 18:01:22 DEBUG[38923]: File channel.c, Line 1456
(ast_set_write_format): Set channel SIP/-08102b70 to write format GSM
Sep 29 18:01:22 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:23 DEBUG[38923]: File channel.c, Line 1456
(ast_set_write_format): Set channel SIP/-08102b70 to write format ALAW
Sep 29 18:01:23 DEBUG[38923]: File channel.c, Line 1456
(ast_set_write_format): Set channel SIP/-08102b70 to write format GSM
Sep 29 18:01:23 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:23 DEBUG[38923]: File channel.c, Line 1456
(ast_set_write_format): Set channel SIP/-08102b70 to write format ALAW
Sep 29 18:01:23 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:23 DEBUG[38923]: File pbx.c, Line 1143
(pbx_extension_helper): Launching 'Hangup'
Sep 29 18:01:23 DEBUG[38923]: File pbx.c, Line 1716 (ast_pbx_run): Spawn
extension (phoneenter,1879,3) exited non-zero on 'SIP/-08102b70'
Sep 29 18:01:23 DEBUG[38923]: File channel.c, Line 661 (ast_hangup):
Hanging up channel 'SIP/-08102b70'
Sep 29 18:01:23 DEBUG[38923]: File chan_sip.c, Line 973 (sip_hangup):
sip_hangup(SIP/-08102b70)
Sep 29 18:01:23 DEBUG[38923]: File chan_sip.c, Line 979 (sip_hangup):
find_user()
Sep 29 18:01:23 DEBUG[38923]: File chan_sip.c, Line 932 (find_user): is
not a local user
Sep 29 18:01:23 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
Sep 29 18:01:23 DEBUG[8201]: File chan_sip.c, Line 538 (__sip_ack):
Stopping retransmission on
'828D64BF-F1CA11D7-8A578158-AC69E3F9 at 213.232.105.12' of Response 101:
Found
Sep 29 18:01:23 DEBUG[8201]: File chan_sip.c, Line 861 (__sip_destroy):
Destorying call '828D64BF-F1CA11D7-8A578158-AC69E3F9 at 213.232.105.12'
Sep 29 18:01:23 DEBUG[1024]: File asterisk.c, Line 328 (urg_handler):
Urgent handler
On Mon, 2003-09-29 at 17:28, Low, Adam wrote:
> Areski,
>
> I would suggest you change the password on that 5300 right now, you provided the whole config file with the IP of AS5300 and the VTY password (although in very easy to break MD5) !!!
>
> Also in your sip.conf you have 'bindaddr = 0.0.0.0' so unless your running multiple NIC's on that box I'd suggest you comment out the bindaddr line altogether.
>
> > -----Original Message-----
> > From: Areski [mailto:areski at e-group.org]
> > Sent: 29 September 2003 17:08
> > To: asterisk-users at lists.digium.com
> > Subject: RE: [Asterisk-Users] cisco AS5300 : problem configuration
> >
> >
> > Hello,
> >
> > Below the IOS config file.
> > Should I disable RFC3389 ??? If yes HOW ??
> >
> >
> > Show running-config
> > -------------------------
> > version 12.2
> > service timestamps debug datetime msec
> > service timestamps log datetime msec
> > service password-encryption
> > service internal
> > !
> > hostname UK-GW01
> > !
> > enable secret 5 $1$Q7QI$wgMvyRdFRxalCmgcEv7A81
> > !
> > !
> > !
> > resource-pool disable
> > !
> > ip subnet-zero
> > no ip domain lookup
> > !
> > !
> > isdn switch-type primary-net5
> > !
> > voice call carrier capacity active
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > mta receive maximum-recipients 0
> > !
> > controller E1 0
> > clock source free-running
> > pri-group timeslots 1-31
> > !
> > controller E1 1
> > clock source line secondary 1
> > pri-group timeslots 1-31
> > !
> > controller E1 2
> > clock source line secondary 2
> > pri-group timeslots 1-31
> > !
> > controller E1 3
> > clock source line secondary 3
> > pri-group timeslots 1-31
> > !
> > !
> > !
> > interface Ethernet0
> > no ip address
> > shutdown
> > !
> > interface Serial0
> > no ip address
> > shutdown
> > no fair-queue
> > clockrate 2015232
> > !
> > interface Serial1
> > no ip address
> > shutdown
> > no fair-queue
> > clockrate 2015232
> > !
> > interface Serial2
> > no ip address
> > shutdown
> > no fair-queue
> > clockrate 2015232
> > !
> > interface Serial3
> > no ip address
> > shutdown
> > no fair-queue
> > clockrate 2015232
> > !
> > interface Serial0:15
> > no ip address
> > ip mroute-cache
> > isdn switch-type primary-net5
> > isdn incoming-voice modem
> > no cdp enable
> > !
> > interface Serial1:15
> > no ip address
> > ip mroute-cache
> > isdn switch-type primary-net5
> > isdn incoming-voice modem
> > no cdp enable
> > !
> > interface Serial2:15
> > no ip address
> > ip mroute-cache
> > isdn switch-type primary-net5
> > isdn incoming-voice modem
> > no cdp enable
> > !
> > interface Serial3:15
> > no ip address
> > ip mroute-cache
> > isdn switch-type primary-net5
> > isdn incoming-voice modem
> > no cdp enable
> > !
> > interface FastEthernet0
> > ip address 213.232.105.12 255.255.255.0
> > duplex auto
> > speed auto
> > !
> > ip classless
> > ip route 0.0.0.0 0.0.0.0 213.232.105.254
> > no ip http server
> > !
> > !
> > !
> > snmp-server community public RO
> > snmp-server enable traps tty
> > !
> > call rsvp-sync
> > !
> > voice-port 0:D
> > !
> > voice-port 1:D
> > !
> > voice-port 2:D
> > !
> > voice-port 3:D
> > !
> > !
> > mgcp profile default
> > !
> > dial-peer cor custom
> > !
> > !
> > !
> > dial-peer voice 100 pots
> > application session
> > direct-inward-dial
> > port 0:D
> > !
> > dial-peer voice 101 pots
> > application session
> > direct-inward-dial
> > port 1:D
> > !
> > dial-peer voice 102 pots
> > application session
> > direct-inward-dial
> > port 2:D
> > !
> > dial-peer voice 103 pots
> > application session
> > direct-inward-dial
> > port 3:D
> > !
> > dial-peer voice 300 voip
> > application session
> > destination-pattern 1879
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target ipv4:62.39.85.18:5060
> > dtmf-relay rtp-nte
> > codec g711alaw bytes 80
> > !
> > dial-peer voice 201 voip
> > application session
> > destination-pattern 1[6,7,9]..
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target sip-server
> > dtmf-relay rtp-nte
> > codec g711alaw bytes 80
> > !
> > dial-peer voice 204 voip
> > application session
> > destination-pattern 18[0-6,8,9].
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target sip-server
> > dtmf-relay rtp-nte
> > codec g711alaw bytes 80
> > !
> > dial-peer voice 206 voip
> > application session
> > destination-pattern 187[0-8]
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target sip-server
> > dtmf-relay rtp-nte
> > codec g711alaw bytes 80
> > !
> > gateway
> > timer receive-rtcp 1000
> > !
> > sip-ua
> > no oli
> > sip-server ipv4:62.39.85.19:5060
> > !
> > !
> > line con 0
> > line aux 0
> > line vty 0 4
> > password 7 094D4210160B
> > login
> > !
> > end
> >
> >
> > On Mon, 2003-09-29 at 14:17, Low, Adam wrote:
> > > I wouldn't expect you to be using RFC3389 if your using
> > A-law, can you include your IOS version and IOS config file ...
> > >
> > > I have not specified any allow's or disallow's in my *
> > config for the codecs with my 5300, I also use Cisco 79xx
> > phones and I use the option within the phones config file to
> > select the preffered codec and when I change this to
> > G.729/A-law/U-law all works perfectly for me.
> > >
> > > > -----Original Message-----
> > > > From: Areski [mailto:areski at e-group.org]
> > > > Sent: 29 September 2003 14:02
> > > > To: asterisk-users at lists.digium.com
> > > > Subject: [Asterisk-Users] cisco AS5300 : problem configuration
> > > >
> > > >
> > > > Hi all !!!
> > > >
> > > >
> > > >
> > > > I m trying to setup a cisco AS5300 and I ve got some problem !!!
> > > >
> > > > During a call test I m getting this error message all the time.
> > > >
> > > > NOTICE[15371]: File rtp.c, Line 263 (process_rfc3389):
> > RFC3389 support
> > > > incomplete. Turn off on client if possible
> > > >
> > > >
> > > >
> > > >
> > > > [general]
> > > > port = 5060 ; Port to bind to
> > > > bindaddr = 0.0.0.0 ; Address to bind to
> > > > context = kiki ; Default for incoming calls
> > > > allow=alaw ; Allow codecs in order
> > of preference
> > > > ;allow=ilbc
> > > > ;allow=all
> > > >
> > > >
> > > > [gw]
> > > > type=user
> > > > host=213.232.xxx.xx
> > > > dtmfmode=rfc2833 ; Choices are inband,
> > rfc2833, or info
> > > > context=kiki
> > > >
> > > >
> > > > ----------------------
> > > >
> > > > Also when I allow "all" for the codecs that's doesn't
> > work and in the
> > > > SIP trace, it seems that Asterisk doesn't choose the
> > > > appropriated codec.
> > > > WHY ??? I really see the GW asking to use ulaw !!!
> > > >
> > > >
> > > > ----------
> > > > When I try to setup a AGI script, for example:
> > > > SAY DIGITS 7565 ""
> > > > I can hear the first number 7 but nothing else !?!
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Any ideas about those problems ???
> > > > Thx for your helps,
> > > > Areski
> > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > >
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