I would use the model 4524 instead of 4114. On Apr 24, 2012 2:48 PM, "Bryan Anderson" <[email protected]> wrote:
> Sorry I sent this yesterday from the wrong email account. Since I tried > sending that though, It looks like we might be going with a patton. I was > looking at the SN4114/JO/EUI? Is that a good model to go with for a 4port > FXO or is there any issues with this one? > > Thanks, > Bryan > > > ------------------------ > > Thank you for your responses. It was not my choice to go with the > grandstream, it was what was given to me for the office. I will attach a > sip trace for a call where I called in, and wasn't able to be transferred > from the front desk. I have noticed that when the calls stop transferring > the gateway's web interface shows no calls on the lines, but the telnet > interface show's all 8 channels use, but it is a 4 line FXO. The gateway > isn't releasing the channels, and sends a "No channels available" message. > I have sent this same trace to grandstream and am awaiting a reply. Not > all calls get stuck just some, like my cell phone, but I don't know the > other numbers. I did notice some past posts of people using this gateway > previously but none of them have said anything as to if they still use it > or not. I realize this is not an optimal gateway but have been told to do > everything I can and have a clear reason why it doesn't work before they > will go with a Patton. > > I did notice the Soundpoint 331's are using the 3.3.3 firmware which I > believe I have been seeing some concerns about, these arrived last week > with that firmware installed. > > > > -Bryan Anderson > > > > > On Mon, Apr 23, 2012 at 2:00 PM, Tony Graziano < > [email protected]> wrote: > >> 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 <[email protected]> >> >>> 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: [email protected] >>> To: [email protected] >>> 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 >>> <[email protected]>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 >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >>> >>> >>> _______________________________________________ sipx-users mailing list >>> [email protected] List Archive: >>> http://list.sipfoundry.org/archive/sipx-users/ >>> >>> _______________________________________________ >>> 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 >> ~~~~~~~~~~~~~~~~~~ >> 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: [email protected].**net<[email protected]> >> >> Helpdesk Customers: >> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> >> Blog: http://blog.myitdepartment.net >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- 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
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
