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. On Oct 18, 2012 8:12 PM, "Henry Kwan" <[email protected]> wrote: > First of all, allow me to thank everyone who had given me advice on the > problem that I have been encountering: The problem of the inability to have > external calls transferred to voice mail when external calls were not > answered, using version 4.4.0. I am, somewhat, happy that I've resolved > the problem. The resolution has nothing to do with actually identifying > the problem but, rather, avoiding it. Let me explain. > > All the advices in suggesting that my router, WRVS4400N or the RV016, may > have been the root of the problem turned out to be wrong. My ITSP was also > not the source of the problem. I downloaded, installed, used pfSense as my > firewall/router and the same problem persisted. After much reading and > searching on the net, I've decided to use 4.2.1 instead of 4.4.0. Viola > this version worked just fine with the same settings/configuration and same > hardware, including the routers. All the routers, WRVS4400N, RV016, and > pfSense, worked. > > My obvious conclusion is that one, or more, bug was introduced into 4.4.0 > that caused this behaviour but regression testing of the release did not > catch it. I am, however, surprised that no one else is reporting this > behaviour, or bug. Perhaps someone already did but I simply missed it. > > It is all good that this problem is out of my way. > > I have another observation that I'd like to seek advice. This observation > is applicable to both 4.2.1 and 4.4.0. I've observed that sometimes after > making changes to configurations and restarted the required processes, as > prompted by sipXecs, I could not make external calls but internal calls and > receiving external calls were just fine. Then I did "Send Profile" to my > server to restart everything but that was also a hit and miss (meaning > sometimes the problem of not able to make external calls went away but > sometime not). I then did "service sipxecs restart" on the command line > but that was also a hit and miss. This problem was also observed even if > no configuration changes were made but simply restarting the sipxecs > processes using methods mentioned above would cause the same observed > problem. There were no changes on internal and external hardware either. > So the observed problem had nothing to do with configuration changes. When > this problem occurred, my phone (SPA942) would show "Calling" then quickly > show "Forbidden". > > My off-the-cuff conclusion is that there must be some race conditions or > out-of-order events, for lack of a better term, that sipXecs encountered > but could not consistently resolve or handle properly, leading to this > condition. I may be totally wrong here but I cannot explain why restarting > the application without changing any configuration and hardware will cause > this inconsistent behaviour on the side of the application. > > Please excuse my long submission and thank you for your attention. > > Best regards, > > Henry Kwan > > > ------------------------------ > *** > From:* Richard Bruce <[email protected]> > *To:* 'Discussion list for users of sipXecs software' < > [email protected]> > *Sent:* Wednesday, October 17, 2012 6:00:23 AM > *Subject:* Re: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > You should probably check the DNS settings on the gateway. I have had > this problem on multiple analog gateways, having forgotten to set this. > > > > Richard Bruce > Dimensional Communications > 7915 S. Emerson Ave, Suite 131 > Indianapolis, IN 46237 > (317) 215-4199- office > (317) 946-1899 - cell > ------------------------------ > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Tony Graziano > *Sent:* Friday, October 12, 2012 10:31 AM > *To:* Henry Kwan; Sipx-users list > *Subject:* Re: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > Are you configuring the spa942 manually? If so, do t do that and let sipx > configure it. Resist the urge to change the configuration for the phone > within sipx. Explain how you are configured (is sipx DNS and dhcp server), > etc. > On Oct 12, 2012 10:27 AM, "Henry Kwan" <[email protected]> wrote: > Hi Todd, > > Thank you for your response and your assurance that the combination of > SPA942 and SipXecs 4.4 works. > > I am just curious regarding the transfer to voice mail since I am not > knowledgeable on the sequence of operation. How is the signalling > different between transfer to voice mail from an internal call and that for > an external call? Is it correct to say that for an internal call to voice > mail transfer, only the phone and the SIP server are involved; for an > external call, the ITSP, SIP server, and phone are involved (therefore the > router and ITSP may affect this operation)? But the call has already been > handed to the SIP server, so why does the ITSP need to get into the scene? > If the ITSP is not involved, what is the difference in handling transfer to > voice mail between an internal and external call? > > I apologize for all these questions but I just am mystified by my > encounters and observations. > > Thanks and best regards, > > Henry Kwan > > *From:* Todd Hodgen <[email protected]> > *To:* 'Henry Kwan' <[email protected]>; ' Discussion list for users of > sipXecs software ' <[email protected]> > *Sent:* Friday, October 12, 2012 12:39:02 AM > *Subject:* RE: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > Henry, I can’t speak to the router, or your ITSP provider. I can > state that I have a site running on 4.4 with a single server, server > provides DHCP and DNS, and works with SPA942 phones. I did not use the > wiki recommendations. I simply provisioned them via the management > templates and they work perfectly. > > Trunks are provided via a PRI gateway – I’ve used Epygi and Patton > gateways at this site with great results from both of them. > > I would suggest router or ITSP are your issue, as others have. > > VOIP.ms is a low cost ITSP provider that for a minimum investment you > can use to test. We know they work, and for a few bucks you can save > yourself some time in troubleshooting. > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Henry Kwan > *Sent:* Thursday, October 11, 2012 8:24 PM > *To:* Tony Graziano > *Cc:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > The router, Linksys WRVS4400N, that I am using is not a home router. > It is a small business router. Having said that it still may not mean it > is a suitable router for SipX. > > I managed to obtain another router and do more testing tonight. The > router is a Linksys/Cisco RV016. It has one-to-one NAT and I set it up to > have a one-to-one NAT entry between my internal sipx server and the > router's external interface. > > Using the RV016, the following test results were obtained (please note > that I had to port forward 5080, and 30000 to 31000, otherwise external > calls would come through with just one-to-one NAT setup and enabled): > > All the previous test results remained exactly the same. That is to say > internal calls could be transferred to voice mail when no one answer the > calls but external calls could not. > > I then setup forwarding directly to voice mail by calling the external > voice mail DID number that I setup. That worked!! > > I am beginning to think that it may have to do with how the SPA942 > operates or it was not setup properly via the sipxecs web interface. But I > am not knowledgeable enough to examine and change the settings on the > SPA942. > > If anyone can give me suggestions to troubleshoot this problem, I'd much > appreciate it. > > Best regards, > > Henry Kwan > *From:* Tony Graziano <[email protected]> > *To:* Henry Kwan <[email protected]> > *Cc:* Discussion list for users of sipXecs software < > [email protected]> > *Sent:* Thursday, October 11, 2012 11:35:38 AM > *Subject:* Re: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > Tested by who? Just because it works as a home router for voip doesn't > mean it will probably work for your office hosting a PBX, BIG FAT > difference. > > On Thu, Oct 11, 2012 at 12:18 PM, Henry Kwan <[email protected]> wrote: > > Do I need one-to-one NAT, or symmetric NAT? I bought this router > because it > > had been tested to work with VoIP, whatever that means, but I forgot the > > source of this information. > > > > From: Tony Graziano <[email protected]> > > To: Henry Kwan <[email protected]>; Discussion list for users of sipXecs > > software <[email protected]> > > Sent: Thursday, October 11, 2012 9:28:30 AM > > Subject: Re: [sipx-users] External calls cannot be transferred to voice > mail > > (sipXecs 4.4.0) > > > > I don't think the router is compatible with the ability to 1:1 NAT or > > do NAT without changing (randomizing) the source port. I would get > > thee to a router that will do thusly. Even if you do all of the above, > > you will likely have frequent or all the time broken audio. > > > > On Thu, Oct 11, 2012 at 10:37 AM, Henry Kwan <[email protected]> wrote: > >> I am a total newbie on SipXecs. I am also green when it comes to the > SIP > >> and VoIP PBX scene. Please excuse my seemingly simple question. > >> > >> The problem that I am encountering, essentially, is that external calls > >> cannot be transferred to voice mail when a call is not answered. > Internal > >> calls that were not answered were transferred to voice mail without a > >> problem. > >> > >> My setup: > >> - SipXecs 4.4.0 installed from the download ISO and updated to the > latest > >> patches with yum. OS is also updated to Centos 5.8, with the latest > >> patches. > >> - Phones are Linksys SPA942 only, no other phones are on the system. > Only > >> 3 > >> phones are on the system. > >> - Domain: mydomain.company.com. company.com is registerd but > >> mydomain.company.com is local/internal and the DNS server is the Sipx > PBX. > >> - Sipx PBX is the DHCP server, dhcp.conf file is edited to assign a > >> limited > >> range of IP addresses. No other dhcp servers are on the subnet. > >> - The workarounds stated on the sipfoundry wiki for the SPA942 are > >> implemented, i.e.: > >> a. MOH Server: [email protected] > >> b. Message Waiting: checked > >> c. Mailbox ID: $USER_ID > >> d. Voice Mail Server: [email protected]. I have > >> also changed mydomain.company.com to the IP address of the sipx server. > >> - Use internal sipXbridge to connect to my SIP trunk. SIP trunk > >> authenticated successfully and works. > >> - Router used is Linksys WRVS4400N. Port 5080 and 30000 to 31000 are > >> forwarded to the SipX PBX. > >> - Aliases are setup for these 3 phones are set for DID. > >> > >> With the above setup, I can dial extensions and have their respective > >> voice > >> mail kick-in when a call is not answered. Dial out and DID work as > well. > >> The problem that I am encountering now is that voice mail does not > kick-in > >> when an external call is not answered. Voice mail does work for > internal > >> calls, though. > >> > >> I've also added domain aliases of the IP address of the PBX and > >> PBX.mydomain.company.com <http://pbx.mydomain.company.com/> to the > setup but that did not help. > >> > >> I also setup one of the phones to call forward to another phone, then > >> voice > >> mail. The call forwart to another extension worked but call forward to > >> voice mail did not. > >> > >> In desperation, I also added an A record for mydomain.company.com in my > >> DNS > >> server but that did not help. > >> > >> Lacking the experience of sipX, VoIP PBX, SIP, and network debug tools, > I > >> hope experienced SipXecs users can shed some on my plight. > >> > >> Thank you. > >> > >> Henry Kwan > >> > >> _______________________________________________ > >> 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/ > > > > > > > > -- > ~~~~~~~~~~~~~~~~~~ > 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]. net<[email protected]> > > Helpdesk Customers: http://myhelp.myitdepartment. > net<http://myhelp.myitdepartment.net/> > Blog: http://blog.myitdepartment.net > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > _______________________________________________ > 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
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
