Cisco phones are quite finicky, as well as problematic. Please try this first...
1. Create a user without a phone and that does have voicemail permissions. 2. Call the AA from outside and ask for that extension. A. If the call promptly fails, this would indicate several things at once: 3. Calls from flowroute.com do not accept transfer via refer. As an example cal a DID that does ring an answer the call. 4. Now try to transfer that call to another user. If it fails, see "A" above. If your calls are failing, in the above example, you should either: 1. Use an SBC (I prefer Ingate with their sip trunking module), or 2. Use sipXbridge, To handle calls to/from your ITSP, both will support REFER and negotitate that as well as MOH. I do not see a template for flowroute, but I believe sipxbridge can work without nat to do what you want. Maybe someone else can chime in and share a configuration for flowroute if someone has been successful thus far. Are you really running it without a firewall in front of it? I think it might also be less problematic for your ITSP to send your calls on a specific port "different" from 5060 if you are using sipxbridge, to keep remote users from interfering with ITSP functionality. It might also be less complicated if they send to a static IP address instead of the current method you are using. sipXbridge does this externally via the port number, while Ingate, interestingly enough, uses two different internal IP addresses (because it does SIP firewalling too) to separate traffic to the ITSP from the traffic from remote users. My 2 cents. On Sun, Jan 17, 2010 at 1:49 PM, John Buswell <[email protected]> wrote: > Tony, > > I am using Flowroute.com as an ITSP, using call routing on their end by SIP > URI. How do I disable MOH on the phone and within sipXecs? I am using a > Cisco 7960 phone, there is nothing in the SIP config file for it that seems > to related to MOH. The only MOH menu option under features in the WebUI > seems to allow you to add / delete wav files. > > Thanks > > > On Sun, Jan 17, 2010 at 12:21 PM, Tony Graziano < > [email protected]> wrote: > >> The AA uses REFER to transfer calls. I'm assuming you are usig sipxbridge >> to >> connect to an ITSP. In that case sipxbridge handles the refer. Do you have >> MOH enabled on the phone or sipxbridge? >> >> I would start with MOH disabled everywhere, being mindful the caller would >> hear silence. If the call succeeds, enable MOH on sipxbridge only. >> >> What ITSP are you using? >> >> ============================ >> 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/ >> >> ----- Original Message ----- >> From: John Buswell <[email protected]> >> To: Tony Graziano <[email protected]> >> Cc: [email protected] <[email protected]> >> Sent: Sun Jan 17 11:27:40 2010 >> Subject: Re: [sipx-users] Indirect Extension Dialing (via Auto Attendent) >> not working >> >> Tony, >> >> I do see this in the sipXproxy log file when I try indirect extension >> dialing from my cell: >> >> 935:NAT:WARNING:sipx:SipClientUdp-8:B73B0B90:SipXProxy:"'yS3SFDNH8r2Xj': >> Received unexpected event FailureResponse while in state >> 'WaitingForInvite'" >> >> Which is making me think that something is not configured right somewhere >> or >> is not hitting the correct address. >> >> The server is not behind any NAT, the phones are but they seem to work ok. >> >> One other thing, two of the extensions are configured with DIDs, the >> direct >> dial to those DIDs works fine. >> >> Thanks >> >> >> On Sun, Jan 17, 2010 at 11:18 AM, John Buswell <[email protected]> >> wrote: >> >> > Tony, >> > >> > Yes. I can dial both the extensions and the extensions voicemail >> directly. >> > Yes, all the services are running and yes I did configure the services. >> > There is only 1 NIC in the system, and there is just 1 IP address bound >> to >> > the system. >> > >> > Any ideas or suggestions on where I should be looking in the logs to see >> > why the indirect dialing does not work properly? >> > >> > Thanks >> > >> > >> > On Sun, Jan 17, 2010 at 6:49 AM, Tony Graziano < >> > [email protected]> wrote: >> > >> >> If you dial the extensions voicemail directly (assume you created user >> >> 2000, can you dial 82000 from a registered phone and get the voicemail >> >> greeting)? >> >> >> >> Did you configure the services? Did you go back and verify all the >> >> services were running in sipxconfig? >> >> >> >> How many NIC's/ip addresses are bound to the system? >> >> >> >> On Sun, Jan 17, 2010 at 2:32 AM, John Buswell <[email protected]> >> wrote: >> >> >> >>> Hi, >> >>> >> >>> I have a fresh install of 4.0.4-017289 on ecs-centos5. Everything >> works >> >>> fine except for indirect extension dialing. I can dial extensions >> >>> directly >> >>> from one phone to another. However, if I dial in from outside, the >> >>> auto-attendent picks up, I dial the extension, but all I get is >> silence. >> >>> I'm >> >>> trying to figure out if I've done something stupid (like forgotten to >> >>> edit a >> >>> dial plan or an auto attendent configuration step) ? :) >> >>> >> >>> There are only two non-standard things about this setup: >> >>> >> >>> 1. It is using 4 digit extensions >> >>> 2. Its running on an FC12 system with the ECS-Centos5 install running >> >>> within a chroot. >> >>> >> >>> The reason for #2, is that the dedicated server provider we are using >> >>> did >> >>> not offer CentOS or an earlier version of FC that worked with the ECS >> >>> build. >> >>> So rather than trying to mess with it from source, we did a local >> >>> ECS-Centos5 install, tarred it up and installed it as a chroot. This >> >>> lead to >> >>> a few small things, like voicemail did not work because the Apache >> >>> configuration did not allow the external IP of the server to access >> the >> >>> voicemail cgi-bin scripts etc. >> >>> >> >>> So I'm hoping someone will either reply back with DOH, and tell me >> what >> >>> I >> >>> forgot, or hook me up with some tips to go troubleshoot this >> particular >> >>> issue! :) >> >>> >> >>> Thanks >> >>> >> >>> _______________________________________________ >> >>> 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. >> >> >> >> >> > >> > > -- ====================== 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.
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