Re: [Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution
Hi, no, it is http://www.snom.com/download/share !!! Sven On Monday 11 October 2004 17:18, Alex Barnes wrote: > Someone pointed me here >> > > http://www.snom.com/downloads/share (had to guess at URL as the Snom > site appears to be down or uber slow but if that's not it its damn close > > :-P ) > > Which lists all versions of firmware for all their phones. Handy if you > have a specific version in mind but don't know the correct URL. Tho I > haven't had problems with the auto-update so far. > > HTH > > alex > > -Original Message- > From: Mike Meyer [mailto:[EMAIL PROTECTED] > Sent: 11 October 2004 16:12 > To: Asterisk Users Group > Subject: [Asterisk-Users] Re: Dial group continues to ring after answer > -SNOM phones and solution > > > Asterisk Users; > > Just wanted to let you know I fixed my problem. > > To follow up on my own testing of the situation, I find that the > continued ringing after pickup only occured on the SNOM phones in the > group. The Grandstream phones stop ringing when another phone picks up. > > Having turned on SIP debugging I have verified that the cancel message > is sent to the SNOM phone (and others in the group) when one of the > group phones is picked up, as expected. > > It appears that the SNOMs don't handle the cancel message the same as > the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the > latest official release. > > It seems that these phones even though they are set to do automatic > update, they do not. Or perhaps that capability was broken in the > firmware version I had last updated to. > > THE SOLUTION: > To remedy the problem I upgraded to version 3.52 beta version. Also > 2.04g fixes this problem as well. > > I had to create my own internal TFTP server and flash update to 3.52. > The standard update process did not work to go beyond 2.03y or 2.04g. I > tried 2.05e & f and these would never come out of boot. > > MORAL TO THE STORY: Keep your SIP phone firmware up to date. > > SNOM support is telling me to upgrade to 3.54. I don't see this one > listed on the standard update URL. I am a little leery about moving to > that one. > > Now to upgrade my GrandStream's. They seem to be stuck at an old version > as well. > > Thanks, > Mike Meyer > > On Tue, 2004-10-05 at 16:47, Mike Meyer wrote: > > Asterisk Users: > > > > We have our * dial plan set up to ring 5 phones in the office > > and it > > > delivers the call to the first that picks up their receiver. > > The problem is that after the pickup, everyone else's SIP phone > > keeps > > > ringing at least once and sometimes twice. This interferes with the > > conversation, while others pick up the phone and get nothing. > > > > Does anyone else have similar problems or have a solution to > > stop the > > > ring once answered? My dial statement looks like the following and has > > > > a timeout of 15 seconds. > > > > exten => MainTeam,1,Dial(${MainTeamChannels},15,tr) > > exten => MainTeam,2,Voicemail(u${MainTeam_EXT}) > > ... > > note the variables MainTeamChannels define the SIP phone channels > > defined in sip.conf and MainTeam_EXT is the voicemail box for this > > group extension. > > > > As an alternative, I am considering doing a round robin on a > > call > > > group or pickup group and implementing call pickup. > > > > Any ideas welcome. > > > > Mike Meyer > > ___ > 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 > > > Dear Friends of Ubiquity Software: > > As you may have noticed, Ubiquity Software began using the web domain > ubiquity.com earlier this year in addition to the previously established > ubiquity.net for our website and email communications to you. However, > since that time, a dispute has emerged with respect to actual ownership of > the ubiquity.com domain. > > As an international software company founded over decade ago, you can > always reach Ubiquity Software under the website www.ubiquity.net > <http://www.ubiquity.net/> and via email at @ubiquity.net. However, we > have also chosen to expand our domain to the more specific > www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> for web and > @ubiquitysoftware.com for email communications. > > Please use either the historical ubi
Re: [Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution
Hi, On Monday 11 October 2004 19:12, Dave Cotton wrote: > On Mon, 2004-10-11 at 11:51 -0500, Mike Meyer wrote: > > >Someone pointed me here >> > > > > > >http://www.snom.com/downloads/share > > http://www.snom.com/download/share > > > ! > > That where the SNOM support team sent me. Seems that they may be > > suggesting a different process or URL do update from. My concern is > > whether the latest version 3.54 has been tested and is an official > > release. I hate to put something out that hasn't been through a > > sufficient QA process. I don't want to risk getting my user's mad at me > > with a bad version of software. > > I've been working though the 3.5x series and haven't noticed any real > nasties yet. > > Out of interest has anyone worked out how to use the Action URL > settings? Some few specific events on the phone can trigger web get requests to the configured URLs. Like lifting the handset is triggering "there is some action going on on the phone" etc... Regards, Sven ___ 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] Re: Dial group continues to ring after answer -SNOM phones and solution
On Mon, 2004-10-11 at 11:51 -0500, Mike Meyer wrote: > >Someone pointed me here >> > > >http://www.snom.com/downloads/share > http://www.snom.com/download/share > > > ! > That where the SNOM support team sent me. Seems that they may be > suggesting a different process or URL do update from. My concern is > whether the latest version 3.54 has been tested and is an official > release. I hate to put something out that hasn't been through a > sufficient QA process. I don't want to risk getting my user's mad at me > with a bad version of software. > I've been working though the 3.5x series and haven't noticed any real nasties yet. Out of interest has anyone worked out how to use the Action URL settings? -- Dave Cotton <[EMAIL PROTECTED]> ___ 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] Re: Dial group continues to ring after answer -SNOM phones and solution
>Someone pointed me here >> >http://www.snom.com/downloads/share >... Yup! That where the SNOM support team sent me. Seems that they may be suggesting a different process or URL do update from. My concern is whether the latest version 3.54 has been tested and is an official release. I hate to put something out that hasn't been through a sufficient QA process. I don't want to risk getting my user's mad at me with a bad version of software. Mike ___ 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] Re: Dial group continues to ring after answer -SNOM phones and solution
Someone pointed me here >> http://www.snom.com/downloads/share (had to guess at URL as the Snom site appears to be down or uber slow but if that's not it its damn close :-P ) Which lists all versions of firmware for all their phones. Handy if you have a specific version in mind but don't know the correct URL. Tho I haven't had problems with the auto-update so far. HTH alex -Original Message- From: Mike Meyer [mailto:[EMAIL PROTECTED] Sent: 11 October 2004 16:12 To: Asterisk Users Group Subject: [Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution Asterisk Users; Just wanted to let you know I fixed my problem. To follow up on my own testing of the situation, I find that the continued ringing after pickup only occured on the SNOM phones in the group. The Grandstream phones stop ringing when another phone picks up. Having turned on SIP debugging I have verified that the cancel message is sent to the SNOM phone (and others in the group) when one of the group phones is picked up, as expected. It appears that the SNOMs don't handle the cancel message the same as the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the latest official release. It seems that these phones even though they are set to do automatic update, they do not. Or perhaps that capability was broken in the firmware version I had last updated to. THE SOLUTION: To remedy the problem I upgraded to version 3.52 beta version. Also 2.04g fixes this problem as well. I had to create my own internal TFTP server and flash update to 3.52. The standard update process did not work to go beyond 2.03y or 2.04g. I tried 2.05e & f and these would never come out of boot. MORAL TO THE STORY: Keep your SIP phone firmware up to date. SNOM support is telling me to upgrade to 3.54. I don't see this one listed on the standard update URL. I am a little leery about moving to that one. Now to upgrade my GrandStream's. They seem to be stuck at an old version as well. Thanks, Mike Meyer On Tue, 2004-10-05 at 16:47, Mike Meyer wrote: > Asterisk Users: > > We have our * dial plan set up to ring 5 phones in the office and it > delivers the call to the first that picks up their receiver. > The problem is that after the pickup, everyone else's SIP phone keeps > ringing at least once and sometimes twice. This interferes with the > conversation, while others pick up the phone and get nothing. > > Does anyone else have similar problems or have a solution to stop the > ring once answered? My dial statement looks like the following and has > a timeout of 15 seconds. > > exten => MainTeam,1,Dial(${MainTeamChannels},15,tr) > exten => MainTeam,2,Voicemail(u${MainTeam_EXT}) > ... > note the variables MainTeamChannels define the SIP phone channels > defined in sip.conf and MainTeam_EXT is the voicemail box for this > group extension. > > As an alternative, I am considering doing a round robin on a call > group or pickup group and implementing call pickup. > > Any ideas welcome. > > Mike Meyer > > > ___ 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 Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net <http://www.ubiquity.net/> and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> [EMAIL PROTECTED] ___ 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] Re: Dial group continues to ring after answer - SNOM phones and solution
Asterisk Users; Just wanted to let you know I fixed my problem. To follow up on my own testing of the situation, I find that the continued ringing after pickup only occured on the SNOM phones in the group. The Grandstream phones stop ringing when another phone picks up. Having turned on SIP debugging I have verified that the cancel message is sent to the SNOM phone (and others in the group) when one of the group phones is picked up, as expected. It appears that the SNOMs don't handle the cancel message the same as the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the latest official release. It seems that these phones even though they are set to do automatic update, they do not. Or perhaps that capability was broken in the firmware version I had last updated to. THE SOLUTION: To remedy the problem I upgraded to version 3.52 beta version. Also 2.04g fixes this problem as well. I had to create my own internal TFTP server and flash update to 3.52. The standard update process did not work to go beyond 2.03y or 2.04g. I tried 2.05e & f and these would never come out of boot. MORAL TO THE STORY: Keep your SIP phone firmware up to date. SNOM support is telling me to upgrade to 3.54. I don't see this one listed on the standard update URL. I am a little leery about moving to that one. Now to upgrade my GrandStream's. They seem to be stuck at an old version as well. Thanks, Mike Meyer On Tue, 2004-10-05 at 16:47, Mike Meyer wrote: > Asterisk Users: > > We have our * dial plan set up to ring 5 phones in the office and it > delivers the call to the first that picks up their receiver. > The problem is that after the pickup, everyone else's SIP phone keeps > ringing at least once and sometimes twice. This interferes with the > conversation, while others pick up the phone and get nothing. > > Does anyone else have similar problems or have a solution to stop the > ring once answered? My dial statement looks like the following and has a > timeout of 15 seconds. > > exten => MainTeam,1,Dial(${MainTeamChannels},15,tr) > exten => MainTeam,2,Voicemail(u${MainTeam_EXT}) > ... > note the variables MainTeamChannels define the SIP phone channels > defined in sip.conf and MainTeam_EXT is the voicemail box for this group > extension. > > As an alternative, I am considering doing a round robin on a call group > or pickup group and implementing call pickup. > > Any ideas welcome. > > Mike Meyer > > > ___ 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