I will reiterate it would be plainly simple for you verify if the initial invite is being sent to you on port 5080. On Oct 20, 2012 9:30 AM, "Henry Kwan" <[email protected]> wrote:
> Hi Todd, > > I've done that originally, that is, use the sipxecs template (without > making the changes suggested in the wiki) for the SP492 to configure them. > I only did the changes suggested in the wiki because I encountered the > mentioned problem. > > I think there is a good possibility that Primus does not support INVITE > without SDP, and if so my setup will never work as Tony and Joegen stated. > I'll ask Primus that question. > > In the mean time, I'll try to learn more on setting up pfSense, including > producing a pcap trace and producing a siptrace on sipxecs to aid me > identifying the root of the problem. > > Thanks a bunch to all, > > Henry Kwan > > > ------------------------------ > *From:* Todd Hodgen <[email protected]> > *To:* 'Henry Kwan' <[email protected]>; 'Discussion list for users of > sipXecs software' <[email protected]> > *Sent:* Saturday, October 20, 2012 1:10:52 AM > *Subject:* RE: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > Henry, Try allowing the existing sipxecs template configure the phone, > without making the changes in the wiki to the profiles. > > I have a system with approximately 20 of those phones working perfectly > with the system managing the templates for the phones completely. > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Henry Kwan > *Sent:* Friday, October 19, 2012 4:40 PM > *To:* Tony Graziano; Joegen Baclor > *Cc:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > Hi Tony, > > I really appreciate that you took the time to elaborate in detail below. > I shall follow-up and perform your suggestions when time permits. Please > also see my response below. > > Best regards, > > Henry Kwan > > ------------------------------ > *From:* Tony Graziano <[email protected]> > *To:* Joegen Baclor <[email protected]> > *Cc:* Discussion list for users of sipXecs software < > [email protected]>; Henry Kwan <[email protected]> > *Sent:* Friday, October 19, 2012 2:36:25 AM > *Subject:* Re: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > 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. > > >> OK, I'll ask Primus about this. > > 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've confirmed with Primus that they could accept signalling on 5060 on > their side and sent signalling to us on 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. > > >> Did not test a transfer from the AA to a user, will try that. > >> Call from a user (internal phone) to another user through local dialing > plan (i.e. 9-...) worked. > >> I think the original invite must come on port 5080 as that was the port > that was forwarded. 5060 was not forwarded. > > 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. > > >> Yes, the phone rang. The phone, Linksys SPA942, was configured via the > sipX web pages. I then reset the phone and have the configuration > downloaded to the phone via TFTP (I think this is the mechanism). > >> I re-installed sipXecs 4.4 a number of times. Sometimes IP as a domain > alias would appear automatically. I've also manually done that. In any > event, that did not help. > >> I also run the tests on the configuration test page and everything > passed, including DNS checks. I've also downloaded Flight Test (I think > that was the name) and everything passed. > > 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. > > >> I did not do 1:1 NAT as I was not sure how to do that properly. I've > read up on it now and will try that out in the future. For port > forwarding, I did do Manual AON with static port checked on pfSense. I > also needed to create rules to pass this traffic. Same thing was done for > port range 30000 to 31000. With this setup on pfSense, I could call in > from an external phone but still could not transfer to voice mail when no > one answered. It behaved exactly the same as using other routers - fast > busy when the attempt of transfer was made. > >> I have not had time to follow the link that you sent me earlier but > will definitely read up on it. > > 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/ > -- 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
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