And beware of those Cisco RV series 'firewalls'. In the past with 1-to-1 NAT I've noted that they actually just open up ALL ports. Scary as hell...
Run away... run towards pfSense. Mike On Fri, Oct 12, 2012 at 2:39 AM, Todd Hodgen <[email protected]> wrote: > 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 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/ > -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square**** Suite 201**** Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher <http://twitter.com/mpicher> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro> www.ezuce.com ------------------------------------------------------------------------------------------------------------ There are 10 kinds of people in the world, those who understand binary and those who don't.
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
