(4500 series have two ethernet portsand are typically more flexible in the event a basic sip connection has to be used from it (i.e. some people use them to connect one ethernet port to a sip provider as it has a nat function and bring in trunks for non-sipx related stuff too). Unless you have a need for more than 4 FXO ports, the 4500 series is the way to go (IMO).
On Tue, Apr 24, 2012 at 3:07 PM, Tony Graziano <[email protected] > wrote: > 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/ >> > -- ~~~~~~~~~~~~~~~~~~ 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] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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