Hi Everyone, We are still stuck with this problem. I followed the procedure ias indicated on the link http://wiki.sipfoundry.org/display/xecsuserV4r0/AudioCodes+5.4+Stand-Alone+Survivability, but still failed to initiate outgoing calls to the pstn.
Here's the sipXproxy.log "2010-04-15T04:18:21.691824Z":17:SIP:WARNING:sipxgw.company.com:SipRouter-11:B69FBB90:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction" "2010-04-15T04:18:21.783665Z":18:SIP:ERR:sipxgw.company.com:SipUserAgent-2:B6AFCB90:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction" "2010-04-15T04:18:42.133771Z":19:SIP:WARNING:sipxgw.company.com:SipSrvLookupThread21:B65F7B90:SipXProxy:"DNS query for name 'company.com', type = 1 (A): returned error" "2010-04-15T04:18:42.133911Z":20:SIP:WARNING:sipxgw.company.com:SipSrvLookupThread20:B66F8B90:SipXProxy:"DNS query for name '_sip._tls.company.com', type = 33 (SRV): returned error" "2010-04-15T04:18:42.138837Z":21:SIP:WARNING:sipxgw.company.com:SipRouter-11:B69FBB90:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction" "2010-04-15T04:18:42.230871Z":22:SIP:ERR:sipxgw.company.com:SipUserAgent-2:B6AFCB90:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction" I made a sipxecs dial plan: *Custom PSTN rule:* Dialed Number Prefix "9" and "Any number of digits" Resulting Call Dial blank and append "Entire Dialed Number" *AudioCodes setting:* AudioCodes IP -> Tel number manipulation: Destination Prefix: 9 Stripped Digits Number: 1 Prefix (Suffix) to Add: blank Manipulation Table: Source Prefix: * Source IP: * Stripped digits from left: 1 Number of Digits to Leave is blank. Any thoughts on how to fix our problems? Thanks in advance. Rhon On Mon, Apr 12, 2010 at 8:52 PM, Rhon <c4rdi...@gmail.com> wrote: > Hi Winson, > > Thank you for your reply, appreciate it. > > I will try your suggestion soon as I get in the office tom. Actually, we > are stuck with this problem. ;(( > > For our PSTN, we use FXO interface (RJ11) and not PRI. We're not using any > converter. > > > Thanks and best regards, > > Rhon > > > On Mon, Apr 12, 2010 at 1:08 PM, Winson (Elabram) <winson.k...@elabram.com > > wrote: > >> Dear Rhon, >> >> hi Rhon sorry i for late reply because i no in the office in this weekend >> >> So you can call properly right now? >> >> You can try use some of the softphone to trace the log, >> >> 1. install one softphone >> 2. install wireshark http://www.wireshark.org/download.html >> 3.open the wireshark after point to you network connection >> >> >> 4.try to use your softphone to PSTN call >> >> 5. after stop the services on wireshark and check all the incoming and >> outgoing in wireshark >> a) stop service >> b) telephony -> voip call >> c) click Graph >> d) check is it the phone number call is correct >> >> because sometime is the prefix is not correct will make the phone call >> wrong, >> 2nd is the your PSTN BNC gateway connect to your m1k issue (does you use >> any converter to converb you PRI to RJ48? ) since you can receive call >> should be ok. >> >> >> >> >> Rhon wrote: >> >> Hi, >> >> From the gui I cannot see the LOG you're referring to. Is it located in >> the Status/Diagnostic button? I don't see anything in there except what is >> posted earlier. >> >> @Winson >> We have 7digit number for our pstn number and in my dialplan I have these >> settings: >> >> Name: Local >> PSTN prefix: 9 >> External Number Lenght: Any no. of digits >> >> I tried stipping 1 digit but it does not make any difference. I still get >> a busy tone. >> >> Hope you can help me. >> >> Thanks >> >> Rhon >> >> On Fri, Apr 9, 2010 at 6:23 PM, Winson (Elabram) <winson.k...@elabram.com >> > wrote: >> >>> I agree what Tony said, better check you LOG file what is the number >>> going out? >>> after base on the number to setting out the Stripped Digits from >>> Manipulation >>> Tables > Dest Number IP->Tel >>> >>> Hope can help you :) >>> >>> >>> Tony Graziano wrote: >>> >>> None that I know of. If you cannot make the call from any phone >>> (Cisco/Polycom/softphone), I would ensure the call is getting to the gateway >>> first. >>> >>> The audiocodes has a logging function that will let you see the log in >>> realtime at the browser. If the call is getting to the AC, you should be >>> able to determine what number is being sent, and whether the is a visible >>> error message or reason. If you do not see the call getting to the AC, then >>> looking at your dialplan in the proxy and permissions would be the next >>> step. >>> >>> On Fri, Apr 9, 2010 at 5:42 AM, Rhon <c4rdi...@gmail.com> wrote: >>> >>>> Hi Tony, >>>> >>>> I also did that but it's still unable to make outgoing calls. >>>> >>>> Are there any settings that I have to manually configure on my >>>> Audiocodes Gateway? >>>> >>>> Thanks >>>> >>>> Rhon >>>> >>>> >>>> On Fri, Apr 9, 2010 at 4:53 PM, Tony Graziano < >>>> tgrazi...@myitdepartment.net> wrote: >>>> >>>>> Dont dial the first "9" at the phone. Just dial the 7 digit number. >>>>> >>>>> On Fri, Apr 9, 2010 at 4:13 AM, Rhon <c4rdi...@gmail.com> wrote: >>>>> >>>>>> Hi Everyone, >>>>>> >>>>>> I have another problem. I cannot call a PSTN number from a local >>>>>> extension (say 400). >>>>>> >>>>>> In my DialPlan I have the following settings: >>>>>> >>>>>> Name: Local >>>>>> PSTN prefix: 9 >>>>>> External Number Lenght: Any no. of digitsI >>>>>> >>>>>> On my cisco phone when I dial 98886655, I can see on the screen >>>>>> "Session Progress" and after a few seconds dropped the call. >>>>>> >>>>>> I can call any extension from the PSTN though. >>>>>> >>>>>> Please help. >>>>>> >>>>>> Rhon >>>>>> >>>>>> _______________________________________________ >>>>>> sipx-users mailing list sipx-users@list.sipfoundry.org >>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users >>>>>> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >>>>>> sipXecs IP PBX -- http://www.sipfoundry.org/ >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> ====================== >>>>> Tony Graziano, Manager >>>>> Telephone: 434.984.8430 >>>>> Fax: 434.984.8431 >>>>> >>>>> Email: tgrazi...@myitdepartment.net >>>>> >>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>> Telephone: 434.984.8426 >>>>> Fax: 434.984.8427 >>>>> >>>>> Helpdesk Contract Customers: >>>>> http://www.myitdepartment.net/gethelp/ >>>>> >>>>> Why do mathematicians always confuse Halloween and Christmas? >>>>> Because 31 Oct = 25 Dec. >>>>> >>>>> >>>> >>> >>> >>> -- >>> ====================== >>> Tony Graziano, Manager >>> Telephone: 434.984.8430 >>> Fax: 434.984.8431 >>> >>> Email: tgrazi...@myitdepartment.net >>> >>> LAN/Telephony/Security and Control Systems Helpdesk: >>> Telephone: 434.984.8426 >>> Fax: 434.984.8427 >>> >>> Helpdesk Contract Customers: >>> http://www.myitdepartment.net/gethelp/ >>> >>> Why do mathematicians always confuse Halloween and Christmas? >>> Because 31 Oct = 25 Dec. >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> sipx-users mailing list sipx-users@list.sipfoundry.org >>> List Archive: http://list.sipfoundry.org/archive/sipx-users >>> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >>> sipXecs IP PBX -- http://www.sipfoundry.org/ >>> >>> >>> >> >> >
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