On Thu, Feb 14, 2008 at 10:12 AM, Henry Devito <[EMAIL PROTECTED]> wrote: > I had GXP-2000's running on 1.0 versions of asterisk even earlier. So I > know it does work. I upgraded one of my customers GXP's to the latest
I'm not sure you are right, since I have had Polycoms that didn't work, it's quite possible you should have GPXs that do work. > firmware in it still works. Can you post the output of the CLI with verbose > set to 99 and the the output from the asterisk log file that has the call in > it. You can usually do a 'tail /var/log/asterisk/full -n 400' right after > the call fails. > > I will be glad to help, just need a little more info to narrow down the > issue. > > Thanks > Henry > > > ----- Original Message ----- > From: "Giordano Grandis" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Thursday, February 14, 2008 2:15 AM > Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9 > > > 1. The phone has not the DND active, i checked it several times > 2. Outbound calls always success, the problem is when the phone receive a > call, it repsnds with busy signalling. > 3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade > asterisk. > > Thanks for all > > -----Messaggio originale----- > Da: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Per conto di C F > Inviato: mercoledì 13 febbraio 2008 21.09 > A: Asterisk Users Mailing List - Non-Commercial Discussion > Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 > > Just check DND if it's on on the phone or not. > What is the CLI output when you try making a phone call? > Why don't you try it with a later version of astrisk and a Phone? > > On Feb 13, 2008 10:58 AM, Giordano Grandis <[EMAIL PROTECTED]> wrote: > > > > > > Hi all gusy, > > i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a > > few > > go in "busy" state, if you call it get the busy tone but the phone can > > male > > any type of call. > > This is my sip.conf > > > > [502] > > language = it > > username = 502 > > secret = <password> > > host = dynamic > > type = friend > > context = local > > canreinvite = yes > > dtmfmode = info > > callgroup = 1 > > pickupgroup = 1 > > callerid = 502 <502> > > > > Under Grandstream's support suggest, I set "Use randmom port" to yes and > > "Nat traversal (STUN)" to "No, but send keep alive" but without success. > > This is the firmware version: Program-- 1.1.5.15 Bootloader-- 1.1.5.6 > > > > Anyone can help me ? > > > > Thanks in advance > > > > Giordano > > > > > > No virus found in this outgoing message. > > Checked by AVG Free Edition. > > Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: > > 12/02/2008 > > 15.20 > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 > 15.20 > > > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 > 20.00 > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users