Re: [Asterisk-Users] grandstream transfer, park and conference
Hi! I have 2 grandstream budgetone 100 series. I can call allright, but I can™t do call transfer, park and call conference. (all features works with tdm devices) the 1. Check if Asterisk is always in the media path, i.e. you need the t or T option (or something similar) in your Dial statement. Alternatively you could introduce a canreinvite=no in sip.conf for the GS phones. 2. Check your context setup in extensions.conf and make sure that in call cases your GS phone has the parkedcalls context available Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream transfer, park and conference
1. Check if Asterisk is always in the media path, i.e. you need the t or T option (or something similar) in your Dial statement. Alternatively you could introduce a canreinvite=no in sip.conf for the GS phones. 2. Check your context setup in extensions.conf and make sure that in call cases your GS phone has the parkedcalls context available Philipp I have an update for this problem, and I discovered strange problems. I can do transfer, call parking nicely now except one thing: * only recognize one dtmf only (for example when I press # on my budgetone, it will say transferring, and put my caller on music, but when I press 234, * only catch 2 (in my budgetone, it will say there's no valid extension .), but if I transfer it to one digit extension first (when the call is received, then I want to do transferring/parking/meetme, I need to transfer the call to extension that has only 1 digit, then it will work perfectly (I can transfer anywhere I want (2/3/4 digits)) Is it a bug? If it is, from budgetone, or *? And how to deal with it? Thanks Isianto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream transfer, park and conference
Your English is just fine. :) What's your extensions.conf and sip.conf for your Grandstreams look like? What are your options in the GS config webpage for: 1) NAT transversal (and are you behind a NAT firewall) 2) Send Flash event 3) Send DTMF Best regards, Ryan Thrash On May 3, 2004, at 8:51 PM, Ing Isianto Istiadi wrote: I have 2 grandstream budgetone 100 series. I can call allright, but I cant do call transfer, park and call conference. (all features works with tdm devices) the The budgetone using 1.0.4.55. 1. If I called using sip to sip (from phone a to phone b), I cant transfer it at all or parking it or dial to conference. 2. if the call come from pstn, then the first phone who answer can park the call, and be picked up by the second phone, but after that the parking stuff wont work anymore. (it seems asterisk doesnt recognize #) 3. Ive already set dtmf to info 4. It seems on case 2 above, that even the # works for the first call from pstn to sip, but asterisk only recognize at most 2 digit after # being pressed (for example, I have ext 700 to park the call, when I look at * console, it only receive 70) Thanks and forgive my English ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream transfer, park and conference
What's your extensions.conf and sip.conf for your Grandstreams look like? I'm not in my machine right now, but here's the relevant configs Extensions.conf [ext] Ignorepat=9 Exten=_9XX,1,Dial(zap/1,tTr,20) Exten=_9XX,2,hangup [sip] Include=ext Include=parkedcalls Sip.conf Posrt = 5060 Bindaddr = 0.0.0.0 [Isianto] Type=friend Username=Isianto Secret=xxx Host=dynamic Qualify=50 Context=sip Mailbox=22 Disallow=all Allow=gsm Allow=ulaw Dtmfmode=info [Istiadi] Type=friend Username=istiadi Secret=xxx Host=dynamic Qualify=50 Context=sip Mailbox=23 Disallow=all Allow=gsm Allow=ulaw Dtmfmode=info What are your options in the GS config webpage for: 1) NAT transversal (and are you behind a NAT firewall) Set to no 2) Send Flash event Set to no 3) Send DTMF Dtmf = info (I tried rfcxxx, I forgot) Thanks Isianto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
