Hi All, Here's a portion of the siptrace made by Cisco 7970 ip phone establishing call to a PSTN number.
Here the cisco phone were able to hit "Session Progress" and notice that the domain.com part (compared to polycom which is an ip) is ok. --------------------------------->>>> SIP/2.0 183* Session Progress* Record-Route: <sip:10.10.20.254:5060 ;lr;sipXecs-CallDest=LOCL;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMDAxNDY5MDYxOGNhMDAwMzI0MTM0MTgwLWQ1YTA2OWYw.900_ntap%2Aid%7EOTY3OS0y%21d634129c295aaee5117f070387c9a3a3;x-sipX-done> Via: SIP/2.0/UDP 10.10.20.146:5060;branch=z9hG4bK694467b0 From: "DEPT PROVISION" <sip:2...@domain.com <sip%3a...@domain.com> >;tag=0014690618ca000324134180-d5a069f0 To: <sip:98876...@domain.com <sip%3a98876...@domain.com> ;user=phone>;tag=1c413229785 Call-Id: 00146906-18ca0003-eb7e1260-6191b...@10.10.20.146 Cseq: 102 INVITE Contact: <*sip:unknow...@10.10.20.253 <sip%3aunknow...@10.10.20.253>* ;transport=tcp> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.60A.007.002 <<<--------------------------------------------------- SIP/2.0 500* Server Internal Error* Via: SIP/2.0/TCP 10.10.20.254;branch=z9hG4bK-XX-0053bYdJUfyDDc`_RHSEZ1rUWw Via: SIP/2.0/UDP 10.10.20.254;branch=z9hG4bK-XX-0050U22hom0XxbVcVFHFn_inwQ~N8bSyIZyczqlYsSkWoPNmg;id=9679-2 Via: SIP/2.0/UDP 10.10.20.146:5060;branch=z9hG4bK694467b0 From: "DEPT PROVISION" <sip:2...@domain.com <sip%3a...@domain.com> >;tag=0014690618ca000324134180-d5a069f0 To: <sip:98876...@domain.com <sip%3a98876...@domain.com> ;user=phone>;tag=1c413229785 Call-ID: 00146906-18ca0003-eb7e1260-6191b...@10.10.20.146 CSeq: 102 INVITE Record-Route: <sip:10.10.20.254:5060 ;lr;sipXecs-CallDest=LOCL;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMDAxNDY5MDYxOGNhMDAwMzI0MTM0MTgwLWQ1YTA2OWYw.900_ntap%2Aid%7EOTY3OS0y%21d634129c295aaee5117f070387c9a3a3> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.60A.007.002 Reason: Q.850 ;cause=111 ;text="local" Content-Length: 0 Thank you in advance. Rhon On Fri, Apr 9, 2010 at 11:04 PM, Rhon <c4rdi...@gmail.com> wrote: > 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 < > tgrazi...@myitdepartment.net> 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: 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/ >> >> ----- Original Message ----- >> From: Rhon <c4rdi...@gmail.com> >> To: Tony Graziano <tgrazi...@myitdepartment.net>; >> sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> >> 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 >> <tgrazi...@myitdepartment.net>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:8886...@10.10.20.253 <sip%3a8886...@10.10.20.253> >> <sip%3a8886...@10.10.20.253 <sip%253a8886...@10.10.20.253> >> >;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:2...@domain.com <sip%3a...@domain.com> < >> sip%3a...@domain.com <sip%253a...@domain.com>> >> >;tag=59ECBE7B-4DA70C06 >> To: <sip:98886...@domain.com <sip%3a98886...@domain.com> < >> sip%3a98886...@domain.com <sip%253a98886...@domain.com>>;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: 282f69ef-feacc0ca-92c66...@10.10.20.150 >> Contact: <sip:2...@10.10.20.150 <sip%3a...@10.10.20.150> < >> sip%3a...@10.10.20.150 <sip%253a...@10.10.20.150>>;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 <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 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 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. >> > >> > >> > >
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