I also decided to test it with other media services (conferencing), and it only seems to be with voicemail. So I suspect sipx needs a freeswitch tweak to get around this. Since other media services (AA, conferencing, faxetc.) have rtp going the other way, it makes me suspect its a voicemail setting (freeswitch parameter), and I recall there was a "proxy media" setting for freeswitch invoked to handle fax, so I suspect the media settings in freeswitch need to be revisited.
On Fri, Nov 11, 2011 at 5:05 PM, Tony Graziano <tgrazi...@myitdepartment.net > wrote: > 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.msPOP's. > > > On Fri, Nov 11, 2011 at 5:01 PM, Todd Hodgen <thod...@frontier.com> 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:* sipx-users-boun...@list.sipfoundry.org [mailto: >> sipx-users-boun...@list.sipfoundry.org] *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 < >> tgrazi...@myitdepartment.net> 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 < >> gerryl...@drouillard.ca> 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 >> sipx-users@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-users/**** >> >> >> >> **** >> >> ** ** >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: tgrazi...@voice.myitdepartment.net >> Fax: 434.465.6833 >> >> Email: tgrazi...@myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpd...@voice.myitdepartment.net >> >> 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: tgrazi...@voice.myitdepartment.net >> Fax: 434.465.6833 >> >> Email: tgrazi...@myitdepartment.net >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: helpd...@voice.myitdepartment.net >> >> 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 >> sipx-users@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgrazi...@voice.myitdepartment.net > Fax: 434.465.6833 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpd...@voice.myitdepartment.net > > 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: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net 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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