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: <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 '[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] 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=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 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
