Tony, 201 is a Polycom 650 ip phone. I'm not sure why it's using an ip rather than a domain.
Could it be the tftp = 10.10.20.254 in the dhcpd.conf is set? I cannot verify since I'm not in the office right now. Any idea how to fix this? Thanks On Fri, Apr 9, 2010 at 10:55 PM, Tony Graziano <[email protected] > wrote: > What is at line 201 and why does it use an ip instead of domain name? > ============================ > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > Fax: 434.984.8427 > > Helpdesk Contract Customers: > http://www.myitdepartment.net/gethelp/ > > ----- Original Message ----- > From: Rhon <[email protected]> > To: Tony Graziano <[email protected]>; > [email protected] <[email protected]> > Sent: Fri Apr 09 10:45:47 2010 > Subject: Re: [sipx-users] Cannot call from a local ext to PSTN > > Hi Tony, > > Thank you for your reply and patience. > > Kindly refer to my answers below. > > On Fri, Apr 9, 2010 at 5:49 PM, Tony Graziano > <[email protected]>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. > > > As posted earlier, our cisco-to-cisco phones can communicate each other > flawlessly, in the same manner's happening to polycom-to-polycom phones > internally. > But that's not the case if you call cisco to polycom and vice versa. > > > > > > 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. > > > > > I'm trying to see what's in the trace but honestly, all I can do is guess. > :( > > *Here's a portion of the siptrace:* > > INVITE sip:[email protected] <sip%[email protected]> > <sip%[email protected] <sip%[email protected]> > >;user=phone;sipxecs-lineid=2 > SIP/2.0 > Record-Route: <sip:10.10.20.254:5060 > > ;lr;sipXecs-CallDest=LOCL;sipXecs-rs=%2Aauth%7E.%2Afrom%7ENTlFQ0JFN0ItNERBNzBDMDY%60.900_ntap%2Aid%7EOTY3OS0x%2164cb5133a8c0212270a6ecebee818bf1> > Via: SIP/2.0/TCP 10.10.20.254;branch=z9hG4bK-XX-0049zB1VPMtcVj6Q3ApJe2U8Ow > Via: SIP/2.0/TCP > > 10.10.20.254;branch=z9hG4bK-XX-0046HBiqelyOFvAEWwyfPxdmLA~N8bSyIZyczqlYsSkWoPNmg;id=9679-1 > Via: SIP/2.0/UDP 10.10.20.150;branch=z9hG4bK7eb4f5d4E2BD9623 > From: "DEPT ACCOUNTING" <sip:[email protected] <sip%[email protected]> < > sip%[email protected] <sip%[email protected]>> > >;tag=59ECBE7B-4DA70C06 > To: <sip:[email protected] <sip%[email protected]> < > sip%[email protected] <sip%[email protected]>>;user=phone> * > <<---- I NOTICED HERE 9 WAS NOT STRIPPED AND LOOKS LIKE A SIP NUMBER? > 8886000 IS A PSTN NUMBER.* > Cseq: 2 INVITE > Call-Id: [email protected] > Contact: <sip:[email protected] <sip%[email protected]> < > sip%[email protected] <sip%[email protected]>>;x-sipX-nonat> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, > PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 > * > AC FXO Gateway* = 10.10.20.253 > *SIPX SERVER* = 10.10.20.254 > *DESTINATION (PSTN) NUMBER* = 8886000 > > Any suggestion/comment will be very appreciated. > > Thanks and have a good day! > > Rhon > > > > On Fri, Apr 9, 2010 at 5:42 AM, Rhon <[email protected]> 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 < > >> [email protected]> 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 <[email protected]> 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 digits > >>>> > >>>> 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 [email protected] > >>>> 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: [email protected] > >>> > >>> 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: [email protected] > > > > 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. > > > > >
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