[Asterisk-Users] BudgeTone 101 doesn't register with FirmWare 1.5.23
Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? Best regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ 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] BudgeTone 101 doesn't register with FirmWare 1.5.23
On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. B ___ 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] BudgeTone 101 doesn't register with FirmWare 1.5.23
Bob Goddard a écrit : On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. Thank you Bob but I have just found what was wrong: two dhcp servers on the same network. Another question, Is the 1.0.6.3 stable enough for production use? Regards, Daniel ___ 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] BudgeTone 101 doesn't register with FirmWare1.5.23
they ever going to fix it? email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: Bob Goddard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 7:05 AM Subject: Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare1.5.23 On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. B ___ 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] BudgeTone 101 doesn't register with FirmWare1.5.23
On Tuesday 24 May 2005 18:16, hank smith wrote: they ever going to fix it? I sure as hell hope so. Such a bug is a show stopper. B - Original Message - From: Bob Goddard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 7:05 AM Subject: Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare1.5.23 On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. ___ 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] BudgeTone 101 doesn't register with FirmWare 1.5.23
On Tuesday 24 May 2005 17:07, Daniel ANDRE wrote: Bob Goddard a écrit : On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. Thank you Bob but I have just found what was wrong: two dhcp servers on the same network. Another question, Is the 1.0.6.3 stable enough for production use? I don't know. I only check for this single bug and until it's fixed no phone gets upgraded. B ___ 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] BudgeTone 101 doesn't register with FirmWare1.5.23
hey let me know when its fixed so I can upgrade mine to :) take care hank email: [EMAIL PROTECTED] gmail: [EMAIL PROTECTED] msn messenger: [EMAIL PROTECTED] aim: hanksmith5 skype: hanksmith5 - Original Message - From: Bob Goddard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 24, 2005 2:49 PM Subject: Re: [Asterisk-Users] BudgeTone 101 doesn't register with FirmWare1.5.23 On Tuesday 24 May 2005 17:07, Daniel ANDRE wrote: Bob Goddard a écrit : On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote: Hello, I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send a register staement (nothing in thertereal log). With the 1.0.3.81 version, the phone register properly. Is ther any know bug with the SW Version? There is a registration bug with the BT101s which has been around for years which is still in 1.0.6.3. If the phone is is rung but not answered, allowed to reregister as normal, then rung again and not answered, then the phone will never regiseter again until rebooted. Grandstream have known about this bug since 3rd March but have only acknowledged it since 15th May. Thank you Bob but I have just found what was wrong: two dhcp servers on the same network. Another question, Is the 1.0.6.3 stable enough for production use? I don't know. I only check for this single bug and until it's fixed no phone gets upgraded. B ___ 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] Budgetone 101 Hold/Xfer/Conf/Flash
Hold, transfer and flash only! the conference key is only for model 102-D Bill Michaelson escreveu: Is it possible to use the Hold/Transfer/Conference/Flash keys of the Budgetone-101 (FW 1.0.5.22) with Asterisk? ___ 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 -- Diego Aguirre FWD#: 459696 Tel/Enum: +55 21 2634-0968 ___ 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] Budgetone 101 Hold/Xfer/Conf/Flash
Is it possible to use the Hold/Transfer/Conference/Flash keys of the Budgetone-101 (FW 1.0.5.22) with Asterisk? ___ 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] Budgetone 101 Hold/Xfer/Conf/Flash
Go here... Type budgetone in the search box on left. http://www.voip-info.org/tiki-index.php W From: [EMAIL PROTECTED] on behalf of Bill Michaelson Sent: Sat 3/5/2005 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Budgetone 101 Hold/Xfer/Conf/Flash Is it possible to use the Hold/Transfer/Conference/Flash keys of the Budgetone-101 (FW 1.0.5.22) with Asterisk? ___ 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 winmail.dat___ 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] Budgetone 101
Everytime that I make a call to a Budgetone 101 phone. I always see the following:
-- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busy
I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this?
Josh
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Re: [Asterisk-Users] Budgetone 101
1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM
What firmware are you running on your 101?On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this? Josh ___ 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
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RE: [Asterisk-Users] Budgetone 101
1.0.5.22 is available for downloading here http://gs-firmware.gratissip.dk/ I dont know why these are available if Grandstream dont update their webpages to indicate newer versions are available. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Wilson Sent: Friday, February 18, 2005 10:56 AM To: Asterisk-Users@lists.digium.com Subject: Re: [Asterisk-Users] Budgetone 101 1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM What firmware are you running on your 101? On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial(SIP/1001-bac5, SIP/1000|20|tT) in new stack -- Called 1000 -- Got SIP response 302 Moved Temporarily back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this? Josh ___ 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] Budgetone 101
I am running version 1.0.5.22 on my 101 and am not having any problems. Robert On Fri, 18 Feb 2005 12:01:22 -0500 dean collins [EMAIL PROTECTED] wrote: 1.0.5.22 is available for downloading here http://gs-firmware.gratissip.dk/ I don't know why these are available if Grandstream don't update their webpages to indicate newer versions are available. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Wilson Sent: Friday, February 18, 2005 10:56 AM To: Asterisk-Users@lists.digium.com Subject: Re: [Asterisk-Users] Budgetone 101 1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM What firmware are you running on your 101? On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial(SIP/1001-bac5, SIP/1000|20|tT) in new stack -- Called 1000 -- Got SIP response 302 Moved Temporarily back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this? Josh ___ 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] Budgetone 101
dean collins wrote: 1.0.5.22 is available for downloading here http://gs-firmware.gratissip.dk/ I dont know why these are available if Grandstream dont update their webpages to indicate newer versions are available. Because, it's still in beta. It can be found on Grandstream's website. http://www.grandstream.com/BETATEST ___ 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] Budgetone 101
Can you send me that update? Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 9:54:24 AM
I'm running 1.0.5.22On Fri, Feb 18, 2005 at 08:16:20AM -0700, Josh Wilson wrote: 1.0.5.16 Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM What firmware are you running on your 101? On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following:-- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busyI can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this?Josh ___ 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
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Re: [Asterisk-Users] Budgetone 101
I have updated to 1.0.5.22 and I still get the same problem.
Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-a873 is busy [EMAIL PROTECTED] 2/18/2005 10:41:14 AM
Can you send me that update? Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 9:54:24 AM
I'm running 1.0.5.22On Fri, Feb 18, 2005 at 08:16:20AM -0700, Josh Wilson wrote: 1.0.5.16 Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM What firmware are you running on your 101? On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following:-- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busyI can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this?Josh ___ 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
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Re: [Asterisk-Users] Budgetone 101
looks like call forwarding is on localy on the phone. On Fri, 18 Feb 2005 08:55:53 -0700, Josh Wilson [EMAIL PROTECTED] wrote: 1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM What firmware are you running on your 101? On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial(SIP/1001-bac5, SIP/1000|20|tT) in new stack -- Called 1000 -- Got SIP response 302 Moved Temporarily back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this? Josh ___ 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 ___ 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] Budgetone 101
Fix the missing Contact field for SUBSCRIBE and INFO request Add support for upgrading firmware or modifying configuration via http. Support file path for http url. Add logic to detect and decline duplicate IP during DHCP application stage. Add call time ticking display for callee (BudgeTone 100 only) Support file content authentication checking using AES during firmware upgrade Support for release of IP upon detecting the link is down for more than 15 seconds and re-application for IP address as soon as the link is up again Support attended transfer and Replace header Support Proxy-Require header and its configurable content Support pre-scheduled firmware upgrade checking frequency and add control flag to allow or prohibit auto firmware upgrade. Support configurable PSTN access key string Support 2 different Web login screens (1 for end user and the other for admin). The login interface is shared between 2 different user modes but the edit screen is different. Add port forwarding, DMZ and DHCP server related configuration options to end user configuration screen Fix the loss of registration issue Fix the issue that a HOLD initiated by 1 party can be released by the other when the other party presses HOLD and then releases the HOLD. Fixed the extra @ character in From header when user ID is blank. Fix the issue related to negotiating and using the right MTU when remote end uses a smaller MTU (HT486 only) Fix the PPPoE link state monitoring issue if CHAP is used. Fix the issue where our RTP sequence ID is randomly changed when a 183 response is initially received and then a 200 OK response is received. Fixed layer 2 QoS (VLAN and 802.1p) issue Maintain the credential information for all subsequent REGISTER after the initial registration is successful, as opposed to restart challenge-authenticate cycle for each new REGISTER transaction Fix the reset to factory default which is recently broken Increase the timeout value for PPPoE call establishment. This will better accommodate some Chinese DSL modems' slow response. Also reset IP upon detecting the pppoe link is down for more than 15 seconds. Fix the issue where improperly deleting an un-initialized timer can cause timer malfunction Fix the issue that PPP PAP timer interferes with CHAP negotiation Fix the issue related to processing multiple IP addresses of DNS A record response Fix the issue that PCMU is always included in SDP even if it is never configured on HandyTone products Fix a bug to better handle very long Contact header, e.g., 500+ characters long Fix the ptime negotiation issue where we didn't use the default ptime when the remote end responds with a codec that is different from our first offered codec and which has no ptime in its SDP Fix the issue that after firmware upgrade the device should (but previously does not) reboot automatically. On Fri, 18 Feb 2005 12:01:22 -0500, dean collins [EMAIL PROTECTED] wrote: 1.0.5.22 is available for downloading here http://gs-firmware.gratissip.dk/ I don't know why these are available if Grandstream don't update their webpages to indicate newer versions are available. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Wilson Sent: Friday, February 18, 2005 10:56 AM To: Asterisk-Users@lists.digium.com Subject: Re: [Asterisk-Users] Budgetone 101 1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM What firmware are you running on your 101? On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following: -- Executing Dial(SIP/1001-bac5, SIP/1000|20|tT) in new stack -- Called 1000 -- Got SIP response 302 Moved Temporarily back from 172.22.5.4 -- SIP/1000-465e is busy I can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this? Josh ___ 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 -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk
Re: [Asterisk-Users] Budgetone 101
The Enable Call Features is set to NO!
C F [EMAIL PROTECTED] 2/18/2005 12:10:00 PM
looks like call forwarding is on localy on the phone.On Fri, 18 Feb 2005 08:55:53 -0700, Josh Wilson [EMAIL PROTECTED] wrote: 1.0.5.16 - the latest version. Michael 'Moose' Dinn [EMAIL PROTECTED] 2/18/2005 8:14:41 AM What firmware are you running on your 101? On Fri, Feb 18, 2005 at 08:04:51AM -0700, Josh Wilson wrote: Everytime that I make a call to a Budgetone 101 phone. I always see the following:-- Executing Dial("SIP/1001-bac5", "SIP/1000|20|tT") in new stack -- Called 1000 -- Got SIP response 302 "Moved Temporarily" back from 172.22.5.4 -- SIP/1000-465e is busyI can use X-Lite all the time to make a call without a problem, but any of the budgetone 101 phones I can not get to work anymore. Anybody know how to fix this?Josh ___ 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 ___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
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[Asterisk-Users] Budgetone 101 phones ? SIP through NAT ?
I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used for data and VOIP, do they put the asterisk machine behind a firewall or do they put the firewall on the asterisk machine ? Is anyone using QOS throttling when sharing VOIP and data on the same broadband connection ? Is it necessary ? Thanks. -- Kim Lux (Mr.) Diesel Research Inc ___ 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] Budgetone 101 phones ? SIP through NAT ?
Kim Lux wrote: I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm Cheap, but useable. I'd go for the 102 model though, since it has a 2nd ethernet port. It was listed as $124CA I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? I'm currently testing 1 phone behind NAT. Works fine. The server is running in the office, linked between an OpenVPN connection (TAP) and NATted behind the TAP adapter's IP. When people use asterisk on a broadband connection used for data and VOIP, do they put the asterisk machine behind a firewall or do they put I'm playing around with 1 Asterisk box on my Mandrake Firewall, I've opened up 1 port to allow the firewall to contact the Office Asterisk server(Plan on trying to get inter-Asterisk boxes taking). In a production environment though, it should be running on a separate box behind the firewall. Is anyone using QOS throttling when sharing VOIP and data on the same broadband connection ? Is it necessary ? I'm not yet. Doug ___ 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] Budgetone 101 phones ? SIP through NAT ?
I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used for data and VOIP, do they put the asterisk machine behind a firewall or do they put the firewall on the asterisk machine ? You will have a rather big problem doing that, you will likely end in the one-way audio scenario; to overcome that you may use STUN or TURN or an application level gateway (example: *). STUN is rather complicated to configure and I advise that only if you REALLY need it; never used TURN, but is almost the same. Now, you can put a multihomed machine (with an IP and posibly a NIC in each side of de fw), to do the passing of the calls to an external provider. But if you will link different offices each one with an * server, use IAX2 in the middle, it doesn't suffer the on-way-audio-problem, it's NAT friendly (if you have one or many in the middle), and if you enable trunking, saves a good bunch of bandwidth for each additional conversation. Is anyone using QOS throttling when sharing VOIP and data on the same broadband connection ? Is it necessary ? I use QOS to prevent de data traffic eating all the available bandwidth and render VoIP unusable. You can use a HTB with SFQ in each bucket, or a pfifo, or if you are brave and know what are about to do, use diffserv. Thanks. Miguel Ruiz Velasco = Miguel Ruiz Velasco Version: OpenKeyServer v1.2 Comment: Extracted from belgium.keyserver.net Signature: 0x59831109 __ Do you Yahoo!? All your favorites on one personal page Try My Yahoo! http://my.yahoo.com ___ 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] budgetone 101 and buttons
I just got a Budgetone 101 and I have it hooked to my * box. I thought I'd read somewhere that we can program the buttons on these phones to send DTMF tones, thereby effectively programming them. However, according to the user's manual, they have predefined SIP functionality. My dialplan implements the festures I want (transfer, message, stuff like that), so for uniformity, I'd just like the SIP to send DTMF like my analog lines in the Digium connectors. Is this possible? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] Budgetone-101 MWI
Hi Folks, Bit of a newbee here, so please be gentle. :) I'm trying to get the message waiting indication working on a budgetone-101. Is it as simple as putting `mailbox=n' where n is the mailbox number into sip.conf? Is there anything else I should check or set? -Cheers Max. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone-101 MWI
Max, That is what worked for me. if you want the MESSAGE button on the GS to dial the VM then put whatever extension you have defined for VM in the field Voice Mail UserID via the GS Admin Web Interface. Robert Hi Folks, Bit of a newbee here, so please be gentle. :) I'm trying to get the message waiting indication working on a budgetone-101. Is it as simple as putting `mailbox=n' where n is the mailbox number into sip.conf? Is there anything else I should check or set? -Cheers Max. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone-101 MWI
On Tue, 11 Nov 2003 22:19:09 +0100 (CET) rnc Info Lists [EMAIL PROTECTED] wrote: Max, That is what worked for me. Thanks Robert, I've got it setup so hitting message dials the VM app, and a mailbox entry in sip.conf but still no MWI light. I'll keep at it :) -Cheers Max. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone-101 MWI
Max Lock wrote: Hi Folks, Bit of a newbee here, so please be gentle. :) I'm trying to get the message waiting indication working on a budgetone-101. Is it as simple as putting `mailbox=n' where n is the mailbox number into sip.conf? Is there anything else I should check or set? Max, If you are using voicemail2 and multiple contexts then you need to specify mailbox = [EMAIL PROTECTED].. If you are using VM2 than you should probably use this syntax anyway.. Hope that helps. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Budgetone-101 MWI
Hi Didn't know there was a light under the message button, thought it just flashed the lcd display and gave a stuttered dial tone. This is how my mailboxes are setup in voicemail.conf [default] = ,Reception Mailbox 7001 = 7001,Office SIP Phone 7002 = 7002,Lounge SIP Phone first the extension number, then the password. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Max Lock Sent: 11 November 2003 20:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Budgetone-101 MWI On Tue, 11 Nov 2003 22:19:09 +0100 (CET) rnc Info Lists [EMAIL PROTECTED] wrote: Max, That is what worked for me. Thanks Robert, I've got it setup so hitting message dials the VM app, and a mailbox entry in sip.conf but still no MWI light. I'll keep at it :) -Cheers Max. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
