RE: [Asterisk-Users] cisco AS5300 : problem configuration
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 [EMAIL PROTECTED] 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: 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 '[EMAIL PROTECTED]' of Response 101: Found Sep 29 18:01:23 DEBUG[8201]: File chan_sip.c, Line 861 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED] > > Sent: 29 September 2003 17:08 > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] ci
RE: [Asterisk-Users] cisco AS5300 : problem configuration
I realized when I sent it ;( ME = Stupid It's already changed, no way guy ;) 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:[EMAIL PROTECTED] > > Sent: 29 September 2003 17:08 > > To: [EMAIL PROTECTED] > > 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
RE: [Asterisk-Users] cisco AS5300 : problem configuration
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:[EMAIL PROTECTED] > Sent: 29 September 2003 17:08 > To: [EMAIL PROTECTED] > 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 >
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:[EMAIL PROTECTED] > > Sent: 29 September 2003 14:02 > > To: [EMAIL PROTECTED] > > 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=u
RE: [Asterisk-Users] cisco AS5300 : problem configuration
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:[EMAIL PROTECTED] > Sent: 29 September 2003 14:02 > To: [EMAIL PROTECTED] > 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 > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
