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.ms thing, 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!
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