I doubt that will make a difference. Chicago is a brand new switch, it has the same issue with voicemail, just tested.
I think it ought to be determined if this is a sipxbridge issue or voip.msissue. Alternatively, your voicemail should say "please dont talk over 60 seconds, make it quick! <BEEP>". It's strange that its always at 60 seconds, which makes me wonder if sipxbridge stops sending RTP keepalive during a freeswitch session. If so, that would not be good, but strange it only happens with some voip.ms POP's. On Fri, Nov 11, 2011 at 5:01 PM, Todd Hodgen <[email protected]> wrote: > I believe VOIP.ms is in the process of updating their switches. I was > told several months ago that Seattle and one other were the two newest. > Makes me wonder if only the old ones work correctly, and their new platform > does not?**** > > ** ** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Tony Graziano > *Sent:* Friday, November 11, 2011 1:55 PM > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] 1 Minute Voicemails with Voip.ms**** > > ** ** > > Nevermind. I decided that would not work.**** > > ** ** > > I thought the real question is... when the call comes into the Media > Server (Freeswitch), is sipxbridge STILL sending any keepalive back on RTP > packets. I suspect it is not, because if it did, it probably wouldn't > hangup at the other end on you.**** > > ** ** > > Now the real question is -- If sipxbridge is handling the call and the > caller hits the voicemail system, does sipxbridge continue to handle rtp > keepalive (if so, its not working) or does freeswitch. if freeswitch is (i > can't imagine it does, but...) then sipxbridge ought to be able to know > this and handle the keepalive or forward it.**** > > ** ** > > Since voip.ms evidently watched a call not get sent the keepalive, and I > registered a DID to atlanta just now and tried EVERY rtp keepalive setting > in sipxbridge, we ought to zero in on "what is sipxbridge not doing > correctly" or "what is sipxbridge sending properly that voip.ms is not > interpreting correctly".**** > > ** ** > > Have you performed a call trace (media/bridge at debug) and got a packet > capture? If you do that and think that sipxbridge is not doing something > properly, a JIRA ought to be opened. Since noone else has these issues with > other carriers (that have been reported), it might just be a voip.msthing, > and if it is their problem, they need to be made aware of it too. > **** > > On Fri, Nov 11, 2011 at 4:26 PM, Tony Graziano < > [email protected]> wrote:**** > > Have you set the gateway in sipx to "Use Dummy RTP payload" and registered > to one of their red headed stepchildren POP's? I say stepchild, because > they seem to treat all pop's differently or with favoritism. I'm sure those > POP's got what they deserve though.**** > > ** ** > > On Fri, Nov 11, 2011 at 4:13 PM, Gerald Drouillard < > [email protected]> wrote:**** > > Does anyone know of a sipx rtp "keepalive" setting during voicemail? > I just wanted to share this experience for anyone in the future having > this problem:**** > > 02:37:02 PM *[Gerald Drouillard]* Yes, and the voicemail message is > capped at exactly 1 min everytime with the 2 dids on this account. > 02:37:57 PM *[Gerald Drouillard]* Calling in through a different ITSP to > the same extension does not have that problem. > 02:38:31 PM *[Albert]* I see. Please hold on a moment. > 02:38:41 PM *[Gerald Drouillard]* Under this same account the other > subaccount does not have the problem > 02:40:22 PM *[Albert]* You're using the same PBX and the same network? > 02:41:04 PM *[Gerald Drouillard]* Same version of pbx different location > 02:41:22 PM *[Gerald Drouillard]* I actually have 2 other locations > working fine > 02:42:33 PM *[Albert]* Ok. Let me check your settings one more time. > 02:46:44 PM *[Gerald Drouillard]* I see a rate limiting entry on the > firewall. I disabled and will try again. > 02:48:41 PM *[Albert]* Sure let me know if that helps. > 02:49:18 PM *[Gerald Drouillard]* Nope. > 02:52:31 PM *[Gerald Drouillard]* The "bye" does come from your system. > 02:53:43 PM *[Albert]* Ok, hold on a moment. > 02:55:22 PM *[Gerald Drouillard]* Is there a "hang up on 1 min of silence > coming from my pbx" setting on your side? > 02:59:11 PM *[Gerald Drouillard]* I am rebooting the system now to > disable one other iptables rule. > 02:59:59 PM *[Gerald Drouillard]* Keep in mind though it does work coming > in from another ITSP > 03:00:57 PM *[Albert]* Yes, we understand. We are currently reviewing the > trace. Please make any test that you try convenient and let us know if that > helps. > 03:09:22 PM *[Albert]* Gerald, can you let me know if the call that goes > to your voicemail is put in hold or something like that. > 03:10:26 PM *[Albert]* Because we don't have any timeout setting if the > call remain in silence during a period of time. > 03:14:22 PM *[Albert]* Also notice that at the moment our server have a > timeout of 60 seconds if we don't receive any RTP packages. In that case if > your system/device doesn't send any RTP package during that time the call > is ended. > 03:15:26 PM *[Gerald Drouillard]* I doubt it, the trace shows ringing, > then IVR. > 03:15:54 PM *[Gerald Drouillard]* When leaving a voicemail over 60 sec? > 03:16:27 PM *[Albert]* Gerald can you please make a test using our > newyork or Los Angeles, we have that setting to 15 minutes in those servers. > 03:17:17 PM *[Gerald Drouillard]* ok. Do I have to switch the subaccount > setting or can I just log into newyork? > 03:18:06 PM *[Albert]* You need to change the server in your system and > also change the Point of Presence in the DID number. > 03:21:32 PM *[Albert]* Gerald, once you change those settings, please > test again. > 03:22:28 PM *[Gerald Drouillard]* The settings are changed. The pbx is > logging into newyork > 03:22:44 PM *[Gerald Drouillard]* I'll make the call now > 03:22:50 PM *[Albert]* Ok. Sure. > 03:25:45 PM *[Gerald Drouillard]* That seems to be working. Got past 1 min > 03:25:59 PM *[Gerald Drouillard]* What pop's have the 60 sec rule? > 03:27:00 PM *[Albert]* Only Newyork and los Angeles have a rule to 15 > minutes. For the other servers is set to 60 seconds. > 03:28:10 PM *[Albert]* What seems to be happening is that the device > stops sending RTP packages when enters to voicemail. Please try to find a > settings that avoids this behavior and that issue should not occur. > 03:28:57 PM *[Gerald Drouillard]* Why would it send RTP if it does not > have to send anything in vm? > 03:29:11 PM *[Albert]* In order to keep the connection alive. > 03:29:40 PM *[Albert]* I can also suggest that you perform a firmware > update to see if that helps. > 03:30:16 PM *[Gerald Drouillard]* All systems are up to date. > 03:30:36 PM *[Albert]* Please notice, that this is a security measure in > order to avoid the calls keep connected when the device is not sending any > information. > 03:31:13 PM *[Gerald Drouillard]* There is a keep alive setting that is > set to none at the moment. I am guessing that needs to be RTP for you? > 03:33:36 PM *[Albert]* That settings usually works for the registration. > 03:34:28 PM *[Gerald Drouillard]* I don't think there is a setting during > voicemail to "keepalive" > 03:34:32 PM *[Albert]* The keep alive setting, it's to prevent a router > from closing it's NAT External port. > 03:35:06 PM *[Albert]* In that case Gerald, I can suggest that you use > our Newyork or Los Angeles server for the meantime. > 03:35:35 PM *[Gerald Drouillard]* There is a NAT setting in voip.ms. > Would turning that off work? > 03:36:56 PM *[Albert]* No, that would not work. Please notice that you > need to find any setting related with the RTP packages.**** > > > > **** > > -- **** > > Regards**** > > --------------------------------------**** > > Gerald Drouillard**** > > Technology Architect**** > > Drouillard & Associates, Inc.**** > > http://www.Drouillard.biz**** > > ** ** > > _______________________________________________ > 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 > > Email: [email protected] > > 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 > > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services!**** > > > > **** > > ** ** > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > > Email: [email protected] > > 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 > > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services!**** > > _______________________________________________ > 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 Email: [email protected] 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 Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services!
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