Hi all:
I�ve [EMAIL PROTECTED] with public ip and one cisco ATA 186 at home behind a windows pc with two etherned cards and doing NAT. The problem is that I can dial an extention from the natted ata but can hear audio.
The same natted cisco ATA can connect to the test calls at http://billing.mutualphone.com/phpBB2/viewtopic.php?t=40 and can receive voice with the cisco ata186 behind the nat box.
This is my config:
/etc/asterisk/sip.conf ; Note: If your SIP devices are behind a NAT and your Asterisk ; server isn't, try adding "nat=1" to each peer definition to ; solve translation problems.
[general]
port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=g729 allow=g723 allow=gsm allow=g726 allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown
#include sip_nat.conf #include sip_additional.conf
/etc/asterisk/sip_additional.conf [500] username=500 type=friend secret=500 qualify=200 port=5060 pickupgroup= nat=1 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid="Guillermo Salas M. HOME" <500> allow=all
This is my Cisco ATA 186 settings: ata000ccee57310 Version: v3.1.0 atasip (Build 040211A) MAC: 0.12.206.229.115.16 SerialNumber: INM071965T5 ProductId: ATA186I1 Features: 0x00000000 HardwareVersion: 0x0006 0x0000
Cisco ATA 186 (SIP) Configuration
StaticIP: 192.168.0.2
StaticRoute: 192.168.0.1
StaticNetMask: 255.255.255.0
UID0: 500
PWD0: 500
UID1: 0
PWD1: 0
GkOrProxy: my.asterisk.box
UseLoginID: 1
LoginID0: 500
LoginID1: 0
AltGk: 0
AltGkTimeOut: 0
SIPRegInterval: 240
MaxRedirect: 2
SIPRegOn: 1
NATIP: the box running kerio winroute firewall
SIPPort: 5060
MediaPort: 16384
OutBoundProxy: my.asterisk.box
NatServer: 0
NatTimer: 0x00000007
MsgRetryLimits: 0x00000000
SessionTimer: 0x00000002
SessionInterval:1800
MinSessionInterval: 60
DisplayName0: 0
DisplayName1: 0
LBRCodec: 3
AudioMode: 0x00140014
RxCodec: 3
TxCodec: 3
NumTxFrames: 2
CallFeatures: 0xffffffff
PaidFeatures: 0xffffffff
CallerIdMethod: 0x00019e60
FeatureTimer: 0x00000000
FeatureTimer2: 0x0000001e
Polarity: 0x00000000
ConnectMode: 0x00060401
TimeZone: 17
NTPIP: my.ntp.box
AltNTPIP: 0.0.0.0
DNS1IP: my.dns.provider
DNS2IP: my.dns.provider.2
TOS: 0x0000a8b8
SigTimer: 0x05400564
OpFlags: 0x0000006a
VLANSetting: 0x0000002b
FXSInputLevel: -1
FXSOutputLevel: -4
NPrintf: 192.168.0.1.9001
TraceFlags: 0x00000001
SyslogIP: 0.0.0.0.514
SyslogCtrl: 0x00000000
RingOnOffTime: 2,4,25
IPDialPlan: 1
DialPlan: *St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.-
DialPlanEx: 0
DialTone: 2,31538,30831,1380,1740,1,0,0,1000,0,0
BusyTone: 2,30467,28959,1191,1513,0,4000,4000,0,0,0
ReorderTone: 2,30467,28959,1191,1513,0,2000,2000,0,0,0,0,0,0,0,0,0
RingBackTone: 2,30831,30467,1943,2111,0,16000,32000,0,0,0
CallWaitTone: 1,30831,0,5493,0,0,2400,2400,4800,0,0
AlertTone: 1,30467,0,5970,0,0,480,480,1920,0,0
SITone: 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
CallCmd: Af;AH;BS;NA;CS;NA;Df;EB;Ff;EP;Kf;EFh;HH;Jf;AFh;HQ;I*67;gA*82;fA#90v#;OI;H#72v#;bA#74v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;xA;Uh;GQ;
CFGID: 0x00000000
The log when I put a call to ext 100 (x-lite with public ip):
May 5 09:04:58 DEBUG[8283]: ##### Testing 127.0.0.1 with 0.0.0.0
May 5 09:04:58 DEBUG[8283]: ##### Testing 127.0.0.1 with 127.0.0.0
May 5 09:04:58 VERBOSE[8283]: == Manager 'admin' logged on from 127.0.0.1
May 5 09:04:58 DEBUG[8283]: Manager received command 'command'
May 5 09:04:58 DEBUG[8283]: Manager received command ''
May 5 09:04:58 DEBUG[8283]: Manager received command 'Logoff'
May 5 09:04:58 VERBOSE[8283]: == Manager 'admin' logged off from 127.0.0.1
May 5 09:04:58 VERBOSE[8283]: dialparties.agi: Dial string is SIP/102|15|tr
May 5 09:04:58 VERBOSE[8283]: -- AGI Script dialparties.agi completed, returning 0
May 5 09:04:58 VERBOSE[8283]: -- Executing Dial("SIP/500-34f2", "SIP/102|15|tr") in new stack
May 5 09:04:58 DEBUG[8283]: SIMPLE DIAL (NO URL)
May 5 09:04:58 DEBUG[8283]: Setting NAT on RTP to 0
May 5 09:04:58 DEBUG[8283]: Setting NAT on VRTP to 0
May 5 09:04:58 NOTICE[8283]: Unable to create channel of type 'SIP'
May 5 09:04:58 VERBOSE[8283]: == Everyone is busy/congested at this time
May 5 09:04:58 DEBUG[8283]: Exiting with DIALSTATUS=CHANUNAVAIL.
May 5 09:04:58 VERBOSE[8283]: -- Executing Wait("SIP/500-34f2", "1") in new stack
May 5 09:04:59 VERBOSE[8283]: -- Executing VoiceMail("SIP/500-34f2", "[EMAIL PROTECTED]") in new stack
May 5 09:04:59 DEBUG[8283]: voicemail/default/102/unavail doesn't exist, doing what we can
May 5 09:04:59 DEBUG[8283]: Ooh, format changed from unknown to g729
May 5 09:04:59 VERBOSE[8283]: -- Playing 'vm-theperson' (language 'en')
May 5 09:04:59 DEBUG[8283]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found
May 5 09:05:01 VERBOSE[8283]: -- Playing 'digits/1' (language 'en')
May 5 09:05:01 VERBOSE[8283]: -- Playing 'digits/0' (language 'en')
May 5 09:05:02 VERBOSE[8283]: -- Playing 'digits/2' (language 'en')
May 5 09:05:03 VERBOSE[8283]: -- Playing 'vm-isunavail' (language 'en')
May 5 09:05:04 VERBOSE[8283]: -- Playing 'vm-intro' (language 'en')
May 5 09:05:07 WARNING[8283]: Failed to write frame
May 5 09:05:07 VERBOSE[8283]: == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'SIP/500-34f2' in macro 'exten-vm'
May 5 09:05:07 VERBOSE[8283]: == Spawn extension (from-internal, 102, 1) exited non-zero on 'SIP/500-34f2'
May 5 09:05:07 VERBOSE[8283]: -- Executing Macro("SIP/500-34f2", "hangupcall") in new stack
May 5 09:05:07 VERBOSE[8283]: -- Executing ResetCDR("SIP/500-34f2", "w") in new stack
May 5 09:05:07 DEBUG[8283]: cdr_mysql: inserting a CDR record.
May 5 09:05:07 DEBUG[8283]: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) VALUES ('2005-05-05 09:04:57','\"Guillermo Salas M. HOME\" <500>','500','102','from-internal', 'SIP/500-34f2','','ResetCDR','w',10,8,'ANSWERED',3,'')
May 5 09:05:07 VERBOSE[8283]: -- Executing NoCDR("SIP/500-34f2", "") in new stack
May 5 09:05:07 WARNING[8283]: CDR on channel 'SIP/500-34f2' not posted
May 5 09:05:07 WARNING[8283]: CDR on channel 'SIP/500-34f2' lacks end
May 5 09:05:07 VERBOSE[8283]: -- Executing Wait("SIP/500-34f2", "5") in new stack
May 5 09:05:07 VERBOSE[8283]: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/500-34f2' in macro 'hangupcall'
May 5 09:05:07 VERBOSE[8283]: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/500-34f2'
May 5 09:05:07 DEBUG[8283]: update_user_counter(500) - decrement inUse counter
May 5 09:05:59 DEBUG[8283]: Manager received command 'Command'
May 5 09:05:59 DEBUG[8283]: Manager received command 'Command'
May 5 09:06:14 DEBUG[8283]: Setting NAT on RTP to 4
May 5 09:06:14 DEBUG[8283]: Setting NAT on VRTP to 4
May 5 09:06:27 DEBUG[8283]: Auto destroying call '[EMAIL PROTECTED]'
Best regards,
-- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net
Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net
Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
