Really the best thing you can do is put your log with sipx (proxy) to debug, and grab whatever best level of detail/logging you can from your gateway. I don't think this happens with others and people probably arent answering you because either it doesnt work well for them or the MFR simply doesnt provide an adequate sip stack or support.
If you see something in the logs, post it here, but you need to discern WHERE the BYE is coming from. Since the RTP is established between the UA (phone) and the gateway, sipx is mostly out of the picture except recording the BYE to cut the CRD record. This is why it is important to use a good network infrastructure along with the gateway and handset, of course. There are a couple of easy gateways to use: AudioCodes and Patton. For less detailed configuration options and ease of configuration a lot of people choose Audiocodes. (not me). Good luck. 2012/4/23 Nitin Mirchandani <nitin_mirchand...@hotmail.com> > I have one suggestion for you - Dont use Grandstream. I dont know which > stack they use - But be it gateway or phone - Its simply unstable (gave up > trying) > > ------------------------------ > Date: Mon, 23 Apr 2012 11:54:14 -0700 > From: branderso...@msn.com > To: sipx-users@list.sipfoundry.org > Subject: Re: [sipx-users] Grandstream GXW4104 > > > Could Problem number two be caused by incorrect Refresher, or timer > settings? If so, what should they be? > > On the gateway: > > *Session Expiration: * (in seconds. default 180 seconds) * > Min-SE: * (in seconds. default and minimum 90 seconds) * > Caller Request Timer: * Yes No (Request for timer when making > outbound calls) > *Callee Request Timer: * Yes No (When caller supports timer but did > not request one) * > Force Timer: * Yes No (Use timer even when remote party does not > support) > *UAC Specify Refresher: * UAC UAS Omit (Recommended) * > UAS Specify Refresher: * UAC UAS (When UAC did not specify refresher > tag) > > > > -Bryan Anderson > > > > On Wed, Apr 18, 2012 at 10:37 AM, Bryan Anderson <branderso...@msn.com>wrote: > > I have been having issues with a new Grandstream GXW4104 fxo gateway and > was wondering if anyone could help. > > We have 4 pstn lines from qwest going into the gateway. All calls go to > an Auto Attendant when answered. > > the two problems we have experienced are: > > 1) After about 1-1.5 hours the call hit the Auto Attendant but wont > transfer out. Some dials and extension they just get dead air. (this is > fixed by rebooting the gateway.) > > 2) The external uses (either some one who called it, or some one we have > called) stop hearing audio, but we can still here them. This happens > anywhere from 1-10 minutes into the call. > > sipXecs (4.4.0- 2011-08-14EDT19:47:12 domU-12-31-39-04-D4-A1) > > Grandstream firmware: Program--1.3.4.13 Loader--1.1.3.4 Boot--1.1.3.2 > > The phones are 1 Polycom IP 550 and 7 Polycom SoundPoint IP 331 > > -Bryan Anderson > > > > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > _______________________________________________ sipx-users mailing list > sipx-users@list.sipfoundry.org List Archive: > http://list.sipfoundry.org/archive/sipx-users/ > > _______________________________________________ > 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 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- 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
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