Primus is also LINGO. Primus is a large aggregator and also runs a residential service (Lingo). The Lingo service does not support invite without sdp, unless the reinvite is to one of their services and typically only from one of their ATA's.
I think you would do well to ask them if they support this AND it is very important to make sure the invite for the incoming call comes to your server on port 5080. I don't think your issue is unusual and usually stems from one of 3 core misconfiguration types: 1. Incompatible ITSP - Does a transfer from the AA to a user work? Does a call from a user to another user work? (both as inbound calls via the trunk). Is the original invite coming on port 5080. 2. Does the phone ring? If so, how was it configured (manually of by sipx)? Please tell me you didn't register the line manually using the sipx ip address. DNS is VERY important for the refer to voicemail. IF you registered by IP, make sure you add the IP as a domain alias, but really you should NEVER register by IP and expect all things to work well. 3. Firewall - Unless your firewall is doing 1:1 NAT, you need to make sure it can do Manual AON (static port nat). With pfsense this is easy, but YOU CANNOT create port forward rules until you do this for SIPX becuase they will follow the original NAT type. I sent you a link of how to do this earlier, its pretty straightforward. You should be able to use the pcap tool in pfsense and have it listen on WAN port 5080 and do a capture and see if the ITSP sends the call in on the right port. If not, it will never work right (no matter what version) and you need to ask them if they support this. Good luck! On Fri, Oct 19, 2012 at 1:50 AM, Joegen Baclor <[email protected]> wrote: > Transferring ITSP originated calls requires that your ITSP supports INVITE > without SDP. Before barking on something on the system, check first if your > ITSP supports this. If not, there is no way your ITSP will work with sipx > initiated transfers. > > > > On 10/19/2012 01:19 PM, Tony Graziano wrote: >> >> On Fri, Oct 19, 2012 at 1:07 AM, Henry Kwan <[email protected]> wrote: >>> >>> My installation was right from the 4.4 ISO. I did try without updating >>> at >>> all but to no avail. >>> >>> My ITSP is Primus Canada. >>> >>> Well I have to admit that I am not knowledgeable in setting up pfSense. >>> In >>> fact I am not knowledgeable on how to produce a pcap or produce a >>> siptrace >>> as Tony suggested. Having said that, I'll continue to play with 4.4 and >>> look into how to perform the tasks suggested when time permits. >>> >> Pfsense >> http://blog.myitdepartment.net/?p=297 >>> >>> In the mean time, 4.2.1 will have to suffice until I can figure out what >>> I >>> did wrong. >>> >>> By the way, my observation regarding the inconsistent behaviour on >>> restarts >>> for both 4.2.1 and 4.4.0 is not an attack on sipXecs. It is just that - >>> an >>> observation. Maybe someone can comment if this observation is also only >>> experienced by me. If that's the case, I must be a jinx or have a unique >>> ability to bring out the worst in sipXecs. >>> >> I can set up a new system each day and don't experience this behavior. >> It's really important to observe how much RAM you have installed (I >> prefer 8GB minimum but for very small installs 4GB on 4.4 is enough, >> though 8GB should be the minimum for 4.6). >>> >>> Best regards to all, >>> >>> Henry Kwan >>> >>> ________________________________ >>> From: George Niculae <[email protected]> >>> >>> To: Discussion list for users of sipXecs software >>> <[email protected]> >>> Cc: Henry Kwan <[email protected]> >>> Sent: Thursday, October 18, 2012 6:29:01 PM >>> >>> Subject: Re: [sipx-users] External calls cannot be transferred to voice >>> mail >>> (sipXecs 4.4.0) >>> >>> On Fri, Oct 19, 2012 at 3:26 AM, Tony Graziano >>> <[email protected]> wrote: >>>> >>>> Rather than use an old unsupportable version, produce a pcap from your >>>> firewall or produce a siptrace from sipx itself. >>>> >>>> I don't think your off the cuff observation is exactly right on targetm >>>> . >>>> Version 4.2 used its own media server while 4.4 uses FreeSWITCH and >>>> there >>>> are significant close changes. >>>> >>>> You could also indicate whether or not you followed a tutorial on how to >>>> properly configure pfsense and who the itsp is. >>>> >>> Additionally, if you could try scenario with 4.4 built from ISO, >>> without yum updating to latest, and report back, will help identifying >>> if issue in latest patches >>> >>> Thanks >>> George >>> >>> >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> > -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
