[Asterisk-Users] grandstream budgetone 101
Maybe Im loosing my mind but Ive just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. Im using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone Im missing? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 budgetone 101
Just pick up the handset and the speakerphone will turn off automatically. If the handset isn't hung up already, just hang it up and pick it up again. Hanging up the handset won't hang up a speakerphone call... Paul - Original Message - From: dean collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, March 07, 2005 3:21 PM Subject: [Asterisk-Users] grandstream budgetone 101 Maybe Im loosing my mind but Ive just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. Im using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone Im missing? Cheers, Dean ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 budgetone 101
Do you have the handset still off the hook when you do this? If the handset is on hook and hit the speaker button it should hagup the call. If the handset is off hook, it should revert back to the handset. W From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collinsSent: Monday, March 07, 2005 3:22 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] grandstream budgetone 101 Maybe Im loosing my mind but Ive just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. Im using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone Im missing? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 budgetone 101
Just realized, if you dont hangup the handset and then press speakerphone a second time it disconnects the call. Thanks anyway, just need to make sure you place the handset in the cradle when you use speakerphone. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Monday, March 07, 2005 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] grandstream budgetone 101 Just pick up the handset and the speakerphone will turn off automatically. If the handset isn't hung up already, just hang it up and pick it up again. Hanging up the handset won't hang up a speakerphone call... Paul - Original Message - From: dean collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, March 07, 2005 3:21 PM Subject: [Asterisk-Users] grandstream budgetone 101 Maybe Im loosing my mind but Ive just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. Im using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone Im missing? Cheers, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Budgetone 101 channels don't disappear on hangup.
Hi Tom and Steve, Thanks for your replies. As it turns out, I didn't have an outbound SIP proxy defined in the phone's configuration. I don't know enough about SIP / see the relation between an outbound proxy and call status tracking, but there you go, that's what it was. Setting it to the hostname of the Asterisk gateway solved the problem. Thanks again, David. On Fri, Jul 23, 2004 at 08:55:00AM -0500, Tom Neville wrote: Weird, I was just having this same problem just yesterday. I doubt it's the same problem on your end, but it might indicate something similar? Filtering or firewalling of some kind? I installed * on a virtual server in our hosting environment. (The hosting people were complaining about latency, I figured there was no better test of latency than *.) I was using X-Lite to make calls. The calls would come up, then when I click hang up on X-Lite it would sit there for a while then drop. Looking at show channels, the call was still in place and ethereal was showing RTP packets still coming in. * was also throwing an error about max retries on sending a packet? Looking at the SDP packets, the server was telling the client to send the RTP packets back to 127.0.0.1. The virtual server has a weird network setup.. venet0Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 Mask:255.255.255.0 UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 RX packets:140238 errors:0 dropped:0 overruns:0 frame:0 TX packets:167187 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:30016231 (28.6 Mb) TX bytes:35546749 (33.9 Mb) venet0:0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:209.43.121.215 P-t-P:209.43.121.215 Bcast:209.43.121.215 Mask:255.255.255.255 UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1 RX packets:140238 errors:0 dropped:0 overruns:0 frame:0 TX packets:167187 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:30016231 (28.6 Mb) TX bytes:35546749 (33.9 Mb) From the looks of it, * was sending the IP from venet0 back to the client for the return of the RTP stream. In the sip.conf file, I put bindaddr=209.43.121.215. After that, calls come up normal and end completely normally. Like I said, this is probably not the problem in your situation.. but hopefully it'll lead you in the right direction? Tom On Jul 23, 2004, at 5:43 AM, David Wilson wrote: Hi there, I'm having problems with the Grandstream Budgetone 101 on hangup - show channels/show channels concise output is still showing the call's channels as active. The problem does not exist when I use SJPhone, so I'm assuming it isn't an Asterisk configuration issue. Has anyone seen this, or better, does anyone have a fix? :) Thanks, David. -- Before you judge a man, walk a mile in his shoes. After that, who cares? He's a mile away and you've got his shoes. -- Billy Connely ___ 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 ___ 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 -- The next great adventure of mankind is not for people who ask, What exactly is the point? They will never get it. -- http://news.bbc.co.uk/1/hi/sci/tech/3302375.stm ___ 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 Budgetone 101 channels don't disappear on hangup.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 28 July 2004 01:54 pm, David Wilson wrote: Hi Tom and Steve, Thanks for your replies. As it turns out, I didn't have an outbound SIP proxy defined in the phone's configuration. I don't know enough about SIP / see the relation between an outbound proxy and call status tracking, but there you go, that's what it was. Setting it to the hostname of the Asterisk gateway solved the problem. Thanks again, David. It's nice when it comes together... : ) I discovered that my problem was related to a bug in the GS web page where it does not keep the 2nd quad number of the IP address. So if you make a change without adding it back in, you're going to have to manually edit the phone to get it back up again. The new (firmware) version has a neat ringer which reads out the callers phone number. (At least it's new to me.) - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBCGk4ljK16xgETzkRAtNTAJ40XABR5NjcQJxOliBOgtzbAk5HcgCgwkL2 O9BgfHoAbjtQHsDxUvhWmbU= =btOq -END PGP SIGNATURE- ___ 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 Budgetone 101 channels don't disappear on hangup.
I fixed the problem some weeks ago. The extension that rings your bt needs an explicit hangup or congestion command following the dial command. Hope that helps, J. Goericke On Fri, 23 Jul 2004, David Wilson wrote: Hi there, I'm having problems with the Grandstream Budgetone 101 on hangup - show channels/show channels concise output is still showing the call's channels as active. The problem does not exist when I use SJPhone, so I'm assuming it isn't an Asterisk configuration issue. Has anyone seen this, or better, does anyone have a fix? :) Thanks, David. -- Before you judge a man, walk a mile in his shoes. After that, who cares? He's a mile away and you've got his shoes. -- Billy Connely ___ 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 ___ 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
[Asterisk-Users] Grandstream Budgetone 101 channels don't disappear on hangup.
Hi there, I'm having problems with the Grandstream Budgetone 101 on hangup - show channels/show channels concise output is still showing the call's channels as active. The problem does not exist when I use SJPhone, so I'm assuming it isn't an Asterisk configuration issue. Has anyone seen this, or better, does anyone have a fix? :) Thanks, David. -- Before you judge a man, walk a mile in his shoes. After that, who cares? He's a mile away and you've got his shoes. -- Billy Connely ___ 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
[Asterisk-Users] GrandStream BudgeTone 101 Question
I know there are several folks hanging around the list that resell VoIP products, I'd be interested in getting your info off-list as I'm trying to persuade someone to get me a budgetone 101 for Christmas ;) I've checked the yahoo site, and they seem to only carry it in white.. Ideally, I'd like blue, so if you have it, please let me know. Also, I've been watching the list for a couple of weeks now, and it seems the grandstream phone plays very well with Asterisk. Any particular gotchyas I should look for? Thank You, Patrick Cantwell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
