Re: [asterisk-users] centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
Hi, I'm also interested in rpm packages including chan_gtalk and res_jabber because I do not want to have a build environment on my productive server. Does anybody knows the reason why this is not available via rpm? best regards Christoph Am 28.11.2011 05:30, schrieb Vladimir Mikhelson: I just go through the whole process. * ./configure * make menu * make * make install I tried building pieces but then ran into the problem where Asterisk was not happy with different version of my modules. I tried to inquire what specific flags or other parameters I needed to use while compiling so that my modules would be accepted into the RPM delivered Asterisk, but ran into the same wall of silence. As I maintain all other components by yum update I need to install Asterisk and then overwrite it by make install. The same with DAHDI. Not very elegant or convenient. -Vladimir On 11/27/2011 10:23 PM, Gaurav P wrote: Do you build from source and copy res_jabber.so and chan_gtalk.so to the rpm installed directories? Or have you just given up on the packages and instead build from source? On Sun, Nov 27, 2011 at 8:35 PM, Vladimir Mikhelson v...@mikhelson.com mailto:v...@mikhelson.com wrote: It has been almost a year since I suggested to consider including these into the RPM build. There was no friction ever since, and I am building from sources too... It seems the RPM maintainers think that Google Voice connectivity is an experimental feature and thus it should not be included in the RPM. Or maybe their logic is different. The end result is the same. -Vladimir On 11/27/2011 7:22 PM, Gaurav P wrote: Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -- _ -- Bandwidth and Colocation Provided byhttp://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 byhttp://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] Update problem | CLI commands missing
Hi List, is there somebody how is able to help me here? Or at least to get more details why this occurs? best regards Christoph Am 08.06.2011 18:00, schrieb Christoph Timm: Hi List, I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW. Currently I'm running Asterisk 1.8.3.3. I have the following problem, if I do the update to the actual 1.8.4.2. There are several commands on the CLI which are not working or even not present like core show uptime (not working) core restart (not present) core show version (not present) my Skype for Asterisk is also not loaded correctly. 190 modules are loaded, if I do a 'module show'. I miss also some messages in the log like [Jun 7 21:21:31] VERBOSE[3449] loader.c: func_version.so = (Get Asterisk Version/Build Info). Does anyone know something about this problem? best regards Christoph -- _ -- 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] Update problem | CLI commands missing
Hi List, I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW. Currently I'm running Asterisk 1.8.3.3. I have the following problem, if I do the update to the actual 1.8.4.2. There are several commands on the CLI which are not working or even not present like core show uptime (not working) core restart (not present) core show version (not present) my Skype for Asterisk is also not loaded correctly. 190 modules are loaded, if I do a 'module show'. I miss also some messages in the log like [Jun 7 21:21:31] VERBOSE[3449] loader.c: func_version.so = (Get Asterisk Version/Build Info). Does anyone know something about this problem? best regards Christoph -- _ -- 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] IPv6 and IPv4 NAT not working
Hi All, I tried to play a little bit with IPv6 to test our VoIP quality software with IPv6 RTP streams. I add bindaddr=:: to the general section of the sip.conf and netstat shows that Asterisk is listing also on IPv6. My Asterisk server is behind a IPv4 NAT and was working absolutely perfect. But after my bindaddr change I got a problem with external calls. I spend some time to investigate this issue and found out the outbound calls are working. The externaddr is used in the SIP INVITE. If I received a inbound call the externaddr isn't used any more in SDP part of the answer from the Asterisk. The result is one way audio. In addition I saw the following message in the Asterisk log: [May 25 19:18:18] WARNING[3674] chan_sip.c: Address remapping activated in sip.conf but we're using IPv6, which doesn't need it. Please remove localnet and/or externaddr settings. I think that is a bug because the externaddr is used correct during the outbound calls. My Asterisk version is 1.8.3.3! Is somebody able to help me with that issue? Should I write a bug ticket for that? best regards Christoph -- _ -- 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
