Switching the dtmf mode to RFC2833 solved my problem, thanks a lot Sam Good work everyone
-----Messaggio originale----- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Lutgring, Sam Inviato: giovedì 14 febbraio 2008 13.55 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9 Try switching your DTMF mode on both asterisk and the phone to RFC2833. I have not seen the issue that you are describing, but I had some very strange issues like call hang-ups when I was using INFO. After switching the issues were gone and I have had no further troubles. Hope this helps you. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: Thursday, February 14, 2008 3:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9 Thanks Henry, anyway the phone is always registered when i get the busy tone * Name : 502 Secret : <Set> MD5Secret : <Not set> Context : local Language : it FromUser : FromDomain : Callgroup : 1 (2) Pickupgroup : 1 (2) Mailbox : LastMsgsSent : -1 Dynamic : Yes Expire : 703 seconds Expiry : 900 Insecure : No Nat : No ACL : No CanReinvite : No PromiscRedir : No DTMFmode : info LastMsg : 0 ToHost : Addr->IP : 192.168.13.171 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Username : 502 Codecs : 0x8010f (g723|gsm|ulaw|alaw|g729|h263) Codec Order : (alaw|ulaw|gsm|g729|g723) Status : OK (22 ms) Useragent : Grandstream GXP2000 1.1.5.15 Full Contact : sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Any idea? Thanks again to all -----Messaggio originale----- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Henry Devito Inviato: mercoledì 13 febbraio 2008 22.01 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Is your phone actually registered to the switch. go to the CLI and do a 'sip show peers' see if extension 502 is registered. Making an outbound call does not prove that the phone is registered. ----- Original Message ----- From: "C F" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, February 13, 2008 2:09 PM Subject: 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 _______________________________________________ -- 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 20.00 No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.21.7/1332 - Release Date: 17/03/2008 10.48 _______________________________________________ -- 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