Re: [asterisk-users] Force sip peers to re register
I think (since I opened this particular can-o-worms) that it depends on your bootrom/sip level. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Monday, March 05, 2012 10:26 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Force sip peers to re register On 03/05/2012 10:18 AM, Carlos Alvarez wrote: In addition, this NOTIFY request does not cause a Polycom phone to reset. It instructs the phone to check its provisioning server for any changes to its configuration, and if there are any then apply them (rebooting if necessary). If the configuration has not changed, sending the phone a check-conf NOTIFY should be a no-op. Make a small script that uses the touch command to update the Polycom's config file mod time/date. Then issue the standard CLI command for them to check config. No need to actually modify the file, it just looks at date/time. ... and since ANY change in a Polycom phone config file results in a reboot :-) (well, that's not as true as it once was) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Link2VoIP going out of business! Now what?
Last week I got an email from Link2VoIP saying that they are shutting it down in a few months. Sighting competition and the unethical changes being made to the Internet by special interest groups. I use Link2VoIP for termination, connecting my Asterisk servers to the regular old telephone company. I like Link2VoIP, I have a few numbers with them and many of my clients do to. Anyone else being affected by this? What are you doing for VoIP termination? I am in Canada, many popular VoIP providers do not work here. And soon that number will be one less. -- Easy, fast GUI development. http://PerlQt.wikidot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Link2VoIP going out of business! Now what?
I would suggest VoicePulse. They seem to have a wide presense. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Royce Souther Sent: Monday, March 05, 2012 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Link2VoIP going out of business! Now what? Last week I got an email from Link2VoIP saying that they are shutting it down in a few months. Sighting competition and the unethical changes being made to the Internet by special interest groups. I use Link2VoIP for termination, connecting my Asterisk servers to the regular old telephone company. I like Link2VoIP, I have a few numbers with them and many of my clients do to. Anyone else being affected by this? What are you doing for VoIP termination? I am in Canada, many popular VoIP providers do not work here. And soon that number will be one less. -- Easy, fast GUI development. http://PerlQt.wikidot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Link2VoIP going out of business! Now what?
On 05-03-12 17:41, Royce Souther wrote: Last week I got an email from Link2VoIP saying that they are shutting it down in a few months. Sighting competition and the unethical changes being made to the Internet by special interest groups. I use Link2VoIP for termination, connecting my Asterisk servers to the regular old telephone company. I like Link2VoIP, I have a few numbers with them and many of my clients do to. Anyone else being affected by this? What are you doing for VoIP termination? I am in Canada, many popular VoIP providers do not work here. And soon that number will be one less. One of the VoIP providers I use is voip.ms which is in Canada. They can port your and your client's numbers. Afaik they have a good reputation. Why don't you give them a call. http://voip.ms/contactus.php Regards, Patrick (no affiliation with voip.ms, just a customer) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Link2VoIP going out of business! Now what?
Voip.MS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Royce Souther Sent: Monday, March 05, 2012 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Link2VoIP going out of business! Now what? Last week I got an email from Link2VoIP saying that they are shutting it down in a few months. Sighting competition and the unethical changes being made to the Internet by special interest groups. I use Link2VoIP for termination, connecting my Asterisk servers to the regular old telephone company. I like Link2VoIP, I have a few numbers with them and many of my clients do to. Anyone else being affected by this? What are you doing for VoIP termination? I am in Canada, many popular VoIP providers do not work here. And soon that number will be one less. -- Easy, fast GUI development. http://PerlQt.wikidot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call notification on IP Telephone
I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display. Thanks. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:17, Administrator TOOTAI ad...@tootai.net wrote: Le 05/03/2012 15:54, Daniel Varella a écrit : Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm working with Siemens optiPoint 410 Economy and Yealink T22P phones. Use BLF if your phones support it. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Link2VoIP going out of business! Now what?
On Mon, Mar 5, 2012 at 9:41 AM, Royce Souther osgn...@gmail.com wrote: down in a few months. Sighting competition and the unethical changes being made to the Internet by special interest groups. Sounds a little whiney. No, a lot whiney. to. Anyone else being affected by this? What are you doing for VoIP termination? Never heard of them, but we're a VoIP company ourselves. We treat Canada the same as the US as far as rates and connectivity, and many other companies do too. It does cost more to work with Canada numbers and termination, but it's within a reasonable margin. Just do a little searching and you'll find many companies who don't see Canada any differently. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call notification on IP Telephone
I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display ?? -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:50, Daniel Varella dvare...@gmail.com wrote: I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display. Thanks. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:17, Administrator TOOTAI ad...@tootai.net wrote: Le 05/03/2012 15:54, Daniel Varella a écrit : Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm working with Siemens optiPoint 410 Economy and Yealink T22P phones. Use BLF if your phones support it. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call notification on IP Telephone
That depends on too many things to answer in a short reply, but if you do it the right way, yes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Varella Sent: Monday, March 05, 2012 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call notification on IP Telephone I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display ?? -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:50, Daniel Varella dvare...@gmail.com wrote: I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display. Thanks. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:17, Administrator TOOTAI ad...@tootai.net wrote: Le 05/03/2012 15:54, Daniel Varella a écrit : Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm working with Siemens optiPoint 410 Economy and Yealink T22P phones. Use BLF if your phones support it. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 06:34 AM, Eric Germann wrote: Does anyone have an idea on when 1.8.9.3 might show up in the RPM repositories? Thanks! EKG They should be available now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call notification on IP Telephone
I will try this tomorrow, that I will be with the phones near me. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 14:01, Danny Nicholas da...@debsinc.com wrote: That depends on too many things to answer in a short reply, but if you do it the right way, yes. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Varella Sent: Monday, March 05, 2012 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call notification on IP Telephone I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display ?? -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:50, Daniel Varella dvare...@gmail.com wrote: I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display. Thanks. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:17, Administrator TOOTAI ad...@tootai.net wrote: Le 05/03/2012 15:54, Daniel Varella a écrit : Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm working with Siemens optiPoint 410 Economy and Yealink T22P phones. Use BLF if your phones support it. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.10.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.10.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.10.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Prevent outbound SIP NOTIFY packets from displaying a port of 0 --- (Closes issue ASTERISK-19430. Reported by Schmooze Com) * --- Include iLBC source code for distribution with Asterisk --- (Closes issue ASTERISK-18943. Reported by Leif Madsen) * --- Fix callerid of originated calls --- (Closes issue ASTERISK-19385. Reported by ornix) * --- Fix outbound DTMF for inband mode of chan_ooh323 --- (Closes issue ASTERISK-19233. Reported, patched by Matt Behrens) * --- Create and initialize udptl only when dialog requests image media --- (Closes issue ASTERISK-16794. Reported by under, tested by Stefan Schmidt) * --- Don't prematurely stop SIP session timer --- (Closes issue ASTERISK-18996. Reported by Thomas Arimont) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.10.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.2.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Prevent outbound SIP NOTIFY packets from displaying a port of 0 --- (Closes issue ASTERISK-19430. Reported by Schmooze Com) * --- Include iLBC source code for distribution with Asterisk --- (Closes issue ASTERISK-18943. Reported by Leif Madsen) * --- Fix callerid of originated calls --- (Closes issue ASTERISK-19385. Reported by ornix) * --- Fix outbound DTMF for inband mode of chan_ooh323 --- (Closes issue ASTERISK-19233. Reported, patched by Matt Behrens) * --- Create and initialize udptl only when dialog requests image media --- (Closes issue ASTERISK-16794. Reported by under, tested by Stefan Schmidt) * --- Don't prematurely stop SIP session timer --- (Closes issue ASTERISK-18996. Reported by Thomas Arimont) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.2.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Monday, March 05, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available On 03/05/2012 06:34 AM, Eric Germann wrote: Does anyone have an idea on when 1.8.9.3 might show up in the RPM repositories? Thanks! EKG They should be available now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 01:49 PM, Eric Germann wrote: Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG ~20 minutes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
Well that's soon enough I guess :) Thanks for what you do! EKG -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Monday, March 05, 2012 2:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available On 03/05/2012 01:49 PM, Eric Germann wrote: Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG ~20 minutes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 09:52 PM, Jason Parker wrote: On 03/05/2012 01:49 PM, Eric Germann wrote: Will a 1.8.10.0 build be imminent or should we go ahead and push this in to production with testing? Thanks! EKG ~20 minutes Some packages seem to lag behind, eg asterisk18-addons-mysql is compiled against 1.8.7: asterisk18-addons-core-1.8.7.0-2_centos5 asterisk18-addons-mysql-1.8.7.0-2_centos5 Is this a problem with the repo? Are these packages obsolete/unmaintained or have been replaced by others? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Group write permissions /etc/asterisk/.
I notice that the installation of Asterisk 1.8.8 thru 1.8.10 (probably earlier versions too) remove the group write permissions from /etc/asterisk/. which is different than 1.4. And 1.6. Is this expected behavior? If so, what's the rationale? If not, I'll submit a bug report if someone hasn't beaten me to it. -K -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available
On 03/05/2012 06:00 PM, Lefteris Zafiris wrote: Some packages seem to lag behind, eg asterisk18-addons-mysql is compiled against 1.8.7: asterisk18-addons-core-1.8.7.0-2_centos5 asterisk18-addons-mysql-1.8.7.0-2_centos5 Is this a problem with the repo? Are these packages obsolete/unmaintained or have been replaced by others? They've been replaced. The latest packages are in new repositories and are now more appropriately named. See http://packages.asterisk.org/centos/5/asterisk-1.8/ as an example. Also, as of Asterisk 1.8, the -addons RPMs are now built from the same SRPM as the rest of the asterisk RPMs. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Group write permissions /etc/asterisk/.
On 03/05/2012 06:22 PM, Karl Fife wrote: I notice that the installation of Asterisk 1.8.8 thru 1.8.10 (probably earlier versions too) remove the group write permissions from /etc/asterisk/. which is different than 1.4. And 1.6. Is this expected behavior? If so, what's the rationale? If not, I'll submit a bug report if someone hasn't beaten me to it. -K The difference comes from using `install` rather than `mkdir`. mkdir defaults to a+rwx (777) - umask (likely 002 on your system), whereas install defaults to the much more sane u+rwx,g+rx,o+rx (755). I don't know if I would call it a bug since the switch to install was intentional, but I wouldn't say it's necessarily expected either. I don't really have a strong opinion either way though. If anything, I might be inclined to argue that 750 (or 770) would be more appropriate. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Group write permissions /etc/asterisk/.
On Tuesday 06 Mar 2012, Jason Parker wrote: I don't know if I would call it a bug since the switch to install was intentional, but I wouldn't say it's necessarily expected either. I don't really have a strong opinion either way though. If anything, I might be inclined to argue that 750 (or 770) would be more appropriate. Considering that (e.g.) sip.conf and iax.conf may contain passwords in clear-text, I'd agree that 770/750 for directories and 660/640 for files would be most appropriate. The g+w bit needs to be set only on those directories/files that ought to be writable from within the Asterisk process itself. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force sip peers to re register
As Kevin pointed out, it is obvious that there is no way of remote reset those phones since their registration status are unknown. SIP NOTIFY will only attempt to consult a registered phone and therefore no need, should it be that way Let me reconsult polyocm guide and see if there is a quicker way as Eric mentioned Sam On Mon, Mar 5, 2012 at 8:55 AM, Kevin P. Fleming kpflem...@digium.com wrote: As Alex pointed out, if the Asterisk server in question needs the phones to re-register in order to send them calls, then it probably cannot send them SIP NOTIFY requests either. This. I don't see how it would be possible to tell the phones to reboot unless you sent it from the server they are *currently* registered to. And if you can do that...you don't need to do that... In addition, this NOTIFY request does not cause a Polycom phone to reset. It instructs the phone to check its provisioning server for any changes to its configuration, and if there are any then apply them (rebooting if necessary). If the configuration has not changed, sending the phone a check-conf NOTIFY should be a no-op. Make a small script that uses the touch command to update the Polycom's config file mod time/date. Then issue the standard CLI command for them to check config. No need to actually modify the file, it just looks at date/time. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
