We have been using this for a week now or so. Just did my "required reading" and noticed that it would be useful to put an internal SBC (sipxbridge) in between the gateways and ITSP/Media Gateway and voila......we now fly straight. I think I stumbled upon this http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration#2._Configure_SipXbridge right before you sent me this mail.
Best to all of you! -- Trevor G. Francis President [email protected] Ph. +1 888.616.0662 Fx. +1 214.722.1318 4608 Abbott Ave, #111 Dallas, TX 75205 MSN: [email protected] Personal emails should be addressed to: [email protected] -- On Feb 4, 2010, at 3:37 PM, Tony Graziano wrote: > If you have an SBC, that's your gateway. If you register or create a > siptrunk in sipx that traverses a firewall, you use sipxbridge. > > I'm getting the impression you haven't done your homework young man. Please > see the wiki for your reading assignments... > > > ============================ > 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: Francis Trevor <[email protected]> > To: Tony Graziano <[email protected]> > Cc: Todd Hodgen <[email protected]> > Sent: Thu Feb 04 16:27:17 2010 > Subject: Re: [sipx-users] Dropped Calls > > What should I use to trace? Im sorry, I have never done any meaningful > traces. > > We have gateways with ITSP's AND with a PRI Gateway (AS5350). > Both are producing the same results. The only thing that isnt producing the > same is extension to extension calls. They work fine, either going over the > local network or over the internet to an extension behind NAT. > > Regards. > > -- > Trevor G. Francis > President > [email protected] > > Ph. +1 888.616.0662 > Fx. +1 214.722.1318 > 4608 Abbott Ave, #111 > Dallas, TX 75205 > MSN: [email protected] > Personal emails should be addressed to: [email protected] > -- > > On Feb 4, 2010, at 3:22 PM, Tony Graziano wrote: > >> Dunno. Are all of your gateways with itsp's or do you have an analog or >> pri >> gateway? >> >> Station to station and Internet calls ([email protected]) might go through >> sipxbridge, so do remote users. >> >> You are not really providing much information. It could be a lot of >> things. >> How about a call trace? >> ============================ >> 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: [email protected] >> <[email protected]> >> To: Todd Hodgen <[email protected]> >> Cc: [email protected] <[email protected]> >> Sent: Thu Feb 04 16:15:04 2010 >> Subject: Re: [sipx-users] Dropped Calls >> >> I will see what I can do with pcap. What is strange is that the same RTP >> and >> signaling apply for extension to extension calls and they arent affected >> by >> the 5 minute drop mark. Only calls made to gateways......inbound or >> outbound. Is this a sipxbridge issue? >> >> >> -- >> Trevor G. Francis >> President >> [email protected] >> >> Ph. +1 888.616.0662 >> Fx. +1 214.722.1318 >> 4608 Abbott Ave, #111 >> Dallas, TX 75205 >> MSN: [email protected] >> Personal emails should be addressed to: [email protected] >> -- >> >> On Feb 4, 2010, at 2:46 PM, Todd Hodgen wrote: >> >>> Trevor, I believe the question here is where is the media being lost – >>> which will help to identify why you are receiving the bye. Is it lost >>> between the sipxbridge and the SBC, is it lost between the phone and the >>> sipxbridge, is it lost between the proxie and the bridge, or any number >>> of >>> other places. A trace will identify EXACTLY where it is being lost, >>> which >>> is a critical part of giving you any valuable feedback to find the >>> solution. >>> >>> If you can run a pcap, there are some great resources on the list that >>> can >>> probably identify where the issue is very quickly for you. >>> >>> From: [email protected] >>> [mailto:[email protected]] On Behalf Of Trevor >>> Francis >>> Sent: Thursday, February 04, 2010 12:29 PM >>> To: Dale Worley >>> Cc: [email protected] >>> Subject: Re: [sipx-users] Dropped Calls >>> >>> That is correct, the Bye is coming from the user's end because the media >>> disappears. We can make it between 5 and 6 minutes on calls. All calls >>> die >>> between 5-6 minutes. However, extension to extension calls, even ones >>> going over the open internet, can exist perpetually. These "Bye" messages >>> arent random. But I think the issue is not "Bye" messages, but rather the >>> media disappearing after 5 minutes. Again, this is happening from all >>> providers including a AS5350 that is on the same subnet as the SipX box. >>> >>> By the say, a Sansay is a SBC. But our issues arent just related to >>> SBC's. >>> >>> >>> Any ideas fellas? >>> >>> >>> -- >>> Trevor G. Francis >>> President >>> [email protected] >>> >>> Ph. +1 888.616.0662 >>> Fx. +1 214.722.1318 >>> 4608 Abbott Ave, #111 >>> Dallas, TX 75205 >>> MSN: [email protected] >>> Personal emails should be addressed to: [email protected] >>> -- >>> >>> On Feb 4, 2010, at 2:18 PM, Dale Worley wrote: >>> >>> >>> On Thu, 2010-02-04 at 13:39 -0600, Josh Patten wrote: >>> >>> Same thing happened to me. Random BYE from various devices. Only >>> happened when using sipXbridge. >>> >>> In Trevor Francis' case, the device generating the BYE calls itself >>> "Sansay VSX 2.1". I assume that it is a user's phone, and that the user >>> is hanging up because he isn't getting any audio. >>> >>> Dale >>> >>> >>>
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