Already disabled both acd and sip trunking (w/c were on by default). Calls still not able to go thru to a user extension. As to whether Cisco is able to handle "sip refer"... i assumed it does as some articles on the sip foundry site talks about configuring cisco routers as a gateway. I'll try to check on this further. If anyone here knows whether a cisco 2651xm using a vic-2fxo (running firmware C2600 ADVENTERPRISEK9-M Version 12.4(15)T11) can handle 'SIP REFER', i would really appreciate some info.
update: found an interesting article about this sip refer thing http://www.ciscopartner.info/en/US/docs/ios/12_2t/12_2t15/feature/guide/ftsipcal.html#wp1063962 will check on this further and update ya'll. Again if anyone already knows the answer, pls do respond. thnx -Ron On Wed, Mar 31, 2010 at 5:36 PM, Tony Graziano <tgrazi...@myitdepartment.net > wrote: > You go to the server, choose configure, and enable only the roles you need. > > If you have a PSTN only connection, you do not need SIP trunking enabled. I > think this was mentioned to you before. > > Realize 4.1.7 is develoipment code and may not be upgradeable to 4.2 > either. > > If the Cisco is not setup or able to handle "SIP REFER", it means it cannot > successfuly transfer any calls routing through it. > ============================ > 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: sipx-users-boun...@list.sipfoundry.org > <sipx-users-boun...@list.sipfoundry.org> > To: Staffan Kerker <ietf-li...@kerker.se> > Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> > Sent: Wed Mar 31 05:29:16 2010 > Subject: Re: [sipx-users] calls from PSTN to operator(AA)=ok but > transferfromoperator to an extension=failed > > I tried doing a debug ccsip for when i am using plar 100 ....refer to > attached file 'plar100'...cant find anything odd here > > I also tried debugging while i'm using plar 210 (an actual user extension, > imaginary sexy receptionist *grin*) and the results are on attached file > 'plar210'....this one shows 404 error...don't know why. I have adjusted the > dialpeers accordingly. If you need to see my cisco gateway config, i can > post it. > > not sure about this Sip Refer you mentioned (im totally new to sipX and to > sip itself) > as for the sipXbridge....how do i use that for my sipX connection to my > gateway? I'm currently using sipX 4.1.7 and it seems sip trunking is > running > by default unlike in 4.0.4 where i have to turn it on manually. > > > > On Wed, Mar 31, 2010 at 3:22 PM, Staffan Kerker <ietf-li...@kerker.se > >wrote: > > > On 31 mar 2010, at 08.56, ronald teng wrote: > > > > > I have a problem w/ calls from pstn not being able to go through > to > > a user extension. It will get to the operator/AA/ext 100 just fine but > > after > > dialling a user extension (as per the AA's instruction) the AA says it > > will > > be transferring but nothing happens...it just stays silent for around 10 > > secs then gives a busy signal. I've attached the log sipX generated for > > further info on the problem. (thnx to todd for explaining how find the > > logs). > > > > > > Could it be the case that SipX is trying to use SIP REFER to transfer the > > call and the Cisco Router is not supporting this? You can verify this > > using Wireshark or the debug ccsip messages command on the router. > > > > I haven't tried this, but isn't this one of the features of the > > SipXBridge, > > to "translate" REFER to SIP Re-INVITES? Maybe you can use > > the SipXBridge on your SIP connection to the gateway. > > > > Regards > > /Staffan > > > > -- > > Staffan Kerker > > mail/sip/xmpp: staf...@kerker.se > > > > "Don't get involved in politics man, just play the gig..." /Sgt Floyd, > > Electric Mayhem Band > > > > >
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