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: [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: [email protected] <[email protected]> To: Staffan Kerker <[email protected]> Cc: [email protected] <[email protected]> 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 <[email protected]>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: [email protected] > > "Don't get involved in politics man, just play the gig..." /Sgt Floyd, > Electric Mayhem Band > > _______________________________________________ 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/
