I don't know your configuration, but the (new) wiki says to set up each system as an unmanaged gateway for site-to-site calling.
In order to transfer the calls you need a site to site dialing plan . What you are not saying is: Whether or not you can natively dial between locations. Whether or not the call gets dropped, silence or lost/hung. You should also describe the phone type at both sides involved and the firmware (if not a softphone). On Sun, Apr 18, 2010 at 10:20 AM, Francis Tinio <[email protected]> wrote: > sonicwall, but ports are opened and calls actually work. I mean Location A > can take and receive calls, and forward calls within the same Location. The > same for Location B. The issue happens when Location A transfers to Location > B or Location A calls Location B. > > > On Apr 18, 2010, at 8:33 AM, Picher, Michael wrote: > >> Hmmm... what kind of firewalls at the remote sites? >> >>> -----Original Message----- >>> From: Francis Tinio [mailto:[email protected]] >>> Sent: Sunday, April 18, 2010 8:17 AM >>> To: Picher, Michael >>> Subject: RE: [sipx-users] cannot transfer between 2 remote locations >>> >>> Nope internet calling is currently disabled. >>> >>> >>> >>> -----Original Message----- >>> From: Picher, Michael <[email protected]> >>> Sent: April 18, 2010 7:33 AM >>> To: Francis Tinio <[email protected]>; Scott Lawrence >>> <[email protected]> >>> Cc: sipx-users <[email protected]> >>> Subject: RE: [sipx-users] cannot transfer between 2 remote locations >>> >>> Sounds like maybe you have Internet Calling enabled... it should be >>> disabled. >>> >>> Mike >>> >>>> -----Original Message----- >>>> From: [email protected] [mailto:sipx-users- >>>> [email protected]] On Behalf Of Francis Tinio >>>> Sent: Friday, April 16, 2010 11:45 AM >>>> To: Scott Lawrence >>>> Cc: sipx-users >>>> Subject: Re: [sipx-users] cannot transfer between 2 remote locations >>>> >>>> basically, this is how we're setup with this client. >>>> >>>> The sipx server is in our Philly datacenter. It's not behind NAT >> and >>>> it has a public IP with firewall turned on. All ports closed except >>>> the ones I said earlier. Then we have 2 locations. Our client's >>>> office, and our office. We have phones in both locations connected >>> to >>>> the sipx server. >>>> >>>> As for firewall, both offices are behind firewalls and the polycom >>>> phones behind NAT. Location A has extensions 5001, 5004. Location >> B >>>> has 5002, 5003. >>>> >>>> Incoming calls go through without issue for both locations. THey >> can >>>> receive and make calls. We can receive and make calls. However, >>>> Location A cannot contact Location B. But we can call from within >>> the >>>> same location. 5001 can call 5004 but not 5002. 5002 can call >> 5003 >>>> but not 5001. Although the extensions would ring, except that >> there >>>> is no audio when the call is picked up. But within the same >> location >>>> audio is fine. >>>> >>>> Also, if 5001 calls 5002 and 5002 does not pickup, the call >>>> successfully gets redirected to the VM system with audio working. >>>> >>>> >>>> >>>> On Apr 16, 2010, at 11:36 AM, Scott Lawrence wrote: >>>> >>>>> On Fri, 2010-04-16 at 10:59 -0400, Francis Tinio wrote: >>>>>> Hi. >>>>>> >>>>>> Our issue with calls suddenly disconnecting after 18 secs has >> been >>>>>> fixed. >>>>>> >>>>>> However, it appears another problem that I initially thought was >>>>>> linked to this is not. We cannot seem to call another extension >>> on >>>> a >>>>>> different location. Nor can we transfer calls between phones on >>>>>> different locations. We can however call extensions located in >>> the >>>>>> same location and transfer calls in the same location. >>>>>> >>>>>> Are there specific ports needed for call transferring? I know we >>>> have >>>>>> enabled ports 21, 8443, 12000, 5060, 5070 5080, 30000-31000 >>>>> >>>>> There's no way to answer your question without a careful >>> description >>>> how >>>>> your different locations are configured (dial plans, extension >>>>> numbering, local network definitions), what phones you are using >>>>> (including version numbers), and how your network is configured >>> (are >>>>> there NATs between the locations, private VPNs...). >>>>> >>>>> After you've described all that... a careful description of what >>> call >>>> is >>>>> made that does not work, and specifically what failure you see >> (sip >>>>> traces please, not narrative about what noises are heard). >>>>> >>>>> >>>> >>>> >>>> _______________________________________________ >>>> 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/ >>> >> > > > _______________________________________________ > 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. _______________________________________________ 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/
