Re: [asterisk-users] RHEL6 packages - SRTP support? [patch]
On 03/06/13 23:04, Daniel Pocock wrote: On 03/06/13 19:18, Jason Parker wrote: On 06/03/2013 12:03 PM, Daniel Pocock wrote: I tried building manually from the source RPM Before running rpmbuild, I installed libsrtp-devel and I notice that res_srtp.so is generated during the build However, the rpmbuild fails for other reasons (see the other email I sent to the list about mISDNutils-devel and other spec file errors) Can you confirm the exact procedure you recommend for rpmbuild on a CentOS6 system rpmbuild --rebuild --without tds --without misdn somepackage.src.rpm Now trying it on a fresh VM (CentOS 6 + EPEL6, freshly built) I have these errors during installation of build dependencies: Installing : kmod-dahdi-linux-fwload-vpmadt032-2.6.2-1_centos6.2.6.32 20/73 WARNING: /lib/modules/2.6.32-358.6.2.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko needs unknown symbol vpmadtreg_unregister WARNING: /lib/modules/2.6.32-358.6.2.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko needs unknown symbol vpmadtreg_register WARNING: /lib/modules/2.6.32-358.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko needs unknown symbol vpmadtreg_unregister WARNING: /lib/modules/2.6.32-358.el6.x86_64/weak-updates/dahdi-linux-fwload-vpmadt032/dahdi_vpmadt032_loader.ko needs unknown symbol vpmadtreg_register and then rpmbuild fails with: checking for gcc... no checking for cc... no configure: error: in `/root/rpmbuild/BUILD/asterisk-11.4.0': configure: error: no acceptable C compiler found in $PATH See `config.log' for more details error: Bad exit status from /var/tmp/rpm-tmp.isB97h (%build) RPM build errors: Bad exit status from /var/tmp/rpm-tmp.isB97h (%build) so I think to be added to the build dependencies in the spec file. Then there is a more cryptic failure: checking how to run the C++ preprocessor... /lib/cpp configure: error: in `/root/rpmbuild/BUILD/asterisk-11.4.0': configure: error: C++ preprocessor /lib/cpp fails sanity check See `config.log' for more details error: Bad exit status from /var/tmp/rpm-tmp.q7wr5n (%build) config.log reveals that g++ is missing. The build dependency is gcc-c++ - I install that and it fails due to missing make Later, there is another failure due to missing subversion. Altogether, these are the missing lines for the spec file: BuildRequires: gcc BuildRequires: gcc-c++ BuildRequires: make BuildRequires: subversion although I would recommend not having a build dependency on SVN or network access, some people like to build on secured machines without network access. Eventually, I end up with the same failure I had before: RPM build errors: File not found: /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cdr_adaptive_odbc.so File not found: /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cdr_odbc.so File not found: /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/cel_odbc.so File not found: /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/func_odbc.so File not found: /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/res_config_odbc.so File not found: /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64/usr/lib64/asterisk/modules/res_odbc.so Now I've tried: rpmbuild --rebuild \ --without tds \ --without misdn \ --without odbc \ asterisk-11.4.0-1_centos6.src.rpm and on the rebuild, I get gzip: ./usr/share/man/man8/autosupport.8 already exists; do you wish to overwrite (y or n)? gzip: ./usr/share/man/man8/astgenkey.8 already exists; do you wish to overwrite (y or n)? which suggests that `make clean' didn't really clean up after the last attempt Finally, I get: Checking for unpackaged file(s): /usr/lib/rpm/check-files /root/rpmbuild/BUILDROOT/asterisk-11.4.0-1_centos6.x86_64 error: Installed (but unpackaged) file(s) found: /usr/lib64/asterisk/modules/func_speex.so /usr/lib64/asterisk/modules/res_srtp.so RPM build errors: Installed (but unpackaged) file(s) found: /usr/lib64/asterisk/modules/func_speex.so /usr/lib64/asterisk/modules/res_srtp.so so I added those two items to the spec file and finally I have a build. I attach a diff for fixing the spec file, it fixes all these issues except the `make clean' Now I tried to repeat the build on the original CentOS6 box using the spec file that I patched on the fresh VM The build failed again on the original box, complaining about unpackaged files I removed the following -devel packages from the box: rpm -e mISDN-devel radiusclient-ng-devel openldap-devel spandsp-devel freetds-devel and then I was able to run the build successfully. So it appears that the build is sensitive to the
[asterisk-users] offline builds - mp3 [patch]
As mentioned in the thread about MP3, I found that the rpmbuild process
demands network access, e.g. to access the mp3 code in SVN.
Some people need to build on isolated networks though
I've attached a patch that allows the MP3 code to be placed in /tmp
before the build starts, then svn will not be used during the build. If
it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead
of going to SVN
I'm not sure if there are other build steps that access the network,
this one was more obvious because I was trying to build on a fresh VM
without any svn client
--- contrib/scripts/get_mp3_source.sh.orig 2013-06-04 12:41:08.222602824 +0200
+++ contrib/scripts/get_mp3_source.sh 2013-06-04 12:40:45.218602846 +0200
@@ -9,6 +9,15 @@
exit 1
fi
+LOCAL_COPY=/tmp/asterisk-contrib-mp3.tar.gz
+if [ -f ${LOCAL_COPY} ]; then
+echo ***
+echo Found ${LOCAL_COPY} - unpacking it, not downloading
+echo ***
+tar xzf ${LOCAL_COPY}
+exit 0
+fi
+
svn export http://svn.digium.com/svn/thirdparty/mp3/trunk addons/mp3 $@
exit 0
--
_
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[asterisk-users] Google/XMPP and Asterisk/XMPP
Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk? In particular, I notice many of the references to XMPP on the wiki link to https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google which seems to suggest that XMPP support and Google Talk support are one and the same. Is the XMPP support only tuned for Google variation of XMPP/ICE/TURN, or is it supported for all open Jabber servers? I currently run 1.8 (before chan_motif) against ejabberd -- _ -- 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] Google/XMPP and Asterisk/XMPP
Daniel Pocock wrote: Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk? In particular, I notice many of the references to XMPP on the wiki link to https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google which seems to suggest that XMPP support and Google Talk support are one and the same. Is the XMPP support only tuned for Google variation of XMPP/ICE/TURN, or is it supported for all open Jabber servers? I currently run 1.8 (before chan_motif) against ejabberd The XMPP support is not tuned for Google Talk by any means, and the voice part (chan_motif) supports the two Google derivatives and the actual Jingle spec. -- Joshua Colp Digium, Inc. | Senior Software Developer 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
[asterisk-users] Skinny directmedia
Asterisk 11 CentOS 6.4 Cisco 7971 phones Does chan_skinny support directmedia? Jacob Miles Software Engineer jacob.e.mi...@l-3com.com 903.457.4422 image001.png-- _ -- 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] Asterisk T.38 Pass-Through doesn't work
On 4/06/2013 4:53 AM, Andrey Polovov wrote: On 06/03/2013 05:03 PM, Larry Moore wrote: Have you checked the installed version of firmware against the latest available from Cisco? Oh! I didn't guess to check. The firmware was not fresh, but upgrading doesn't help. Looking at your SIP information when your ITSP initiated a T.38 session it did not indicate a maxmimum bitrate, it would appear your spa112 attempted to negotiate a connection at 2400bps. Whether there is a way to force my provider to indicate maximum bitrate? Do you have a sip debug session when you sent a fax from your Asterisk box to the PSTN, it would be interesting to see if it sends it as a t.38 or reverts to G711 audio. I have collect a set of debugs (with fresh SPA112 firmware) and actual config files: I would suggest you test the SPA112 directly against your SIP provider however ensure you have the latest firmware, there appear to have been some FAX related fixes in recent versions. To do this change the following (based upon the screen shot you made available); Nat Mapping Enable: yes Call Waiting Serv: no MWI Serv: no Proxy: 80.75.130.136 Register: no Make Call Without Reg: yes Ans Call Without Reg: yes User ID: 7495777 Password: remotesecret FAX T38 Redundancy: 3 FAX Tone Detect Mode: callee FAX T38 Return to Voice: no When you get this working you can then look at making it work through Asterisk - this is how I got my SPA8800 working. I am assuming your network configuration is set up correctly on the spa112. You may want to look at enabling the following options, on my SPA8800 they are located under the SIP tab in the section headed NAT Parameters: Handle via rport: yes Insert via rport: yes Send Resp to Src Port: yes In addition, once it is working and providing your SIP provider permits ECM through a T.38 service I would encourage one to enable options such as FAX T38 ECM Enable. If you are still experiencing problems you may want to perform a packet capture (set the snap size to 1500) of the communications between the spa112 and the other end point and run it through Wireshark and examine the VoIP calls in the captured packets. Good luck. Larry. -- _ -- 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] Dial-App / Feature Disconnect
Hi, I configured in features.conf, that the Dial-App may be cancelled by pressing the pound key. That works fine. The caller can cancel the bridged call. BUT can I configure it that way, that the dialing itself can NOT be cancelled? My dial should only be cancelled by the timeout or by the gangup of the caller. Asterisk 11.3.0 Best regards, -Thorsten- -- _ -- 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] blog about WebRTC + TLS + Asterisk 11
I've now prepared a blog about my experience setting up Asterisk 11 with repro as a SIP proxy for WebSocket clients: http://danielpocock.com/using-resiprocate-to-connect-asterisk-webrtc In particular, the focus is on the use of packages because that makes it faster for more people to deploy identical working systems. To get the demo running for the WebSocket client, I really only needed to change about 5 lines in sip.conf - all other configuration is the default - the more painful step is rebuilding the packages with SRTP support. Regards, 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
Re: [asterisk-users] G.729 codec in pass-thru mode
Kamlesh Kumar wrote: SIP.conf [100] username=100 secret=password type=friend host=dynamic nat=yes canreinvite=no insecure=port disallow=all allow=ulaw allow=alaw allow=g729 context=asterisk qualify=no Is there also an allow=g729 line in sip.conf for the ITSP's SIP peer? SIP Trace: 201.xxx.xxx.xxx = SIP Softphone which originates the call xxx.xxx.xxx.xxx = Asterisk server yyy.yyy.yyy.yyy = ITSP ... --- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060 From: 100 sip:1...@xxx.xxx.xxx.xxx;tag=as643c20b1 To: sip:12127773...@yyy.yyy.yyy.yyy;tag=gK029aaa8c Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx CSeq: 102 INVITE Contact: sip:12127773...@yyy.yyy.yyy.yyy:5060 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Content-Length: 234 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy s=SIP Media Capabilities c=IN IP4 zzz.zzz.zzz.zzz t=0 0 m=audio 21996 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - [Jun 3 13:11:31] --- (11 headers 11 lines) --- [Jun 3 13:11:31] Found RTP audio format 0 [Jun 3 13:11:31] Found RTP audio format 101 [Jun 3 13:11:31] Found audio description format PCMU for ID 0 [Jun 3 13:11:31] Found audio description format telephone-event for ID 101 [Jun 3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jun 3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jun 3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996 [Jun 3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-34d9 is making progress passing it to SIP/100-34d8 [Jun 3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042 [Jun 3 13:11:31] Adding codec 0x4 (ulaw) to SDP [Jun 3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP This response from the ITSP says that only u-law may be used for the call. Please contact the ITSP and confirm that they actually support the G.729 codec. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Codec Mismatch
Sometimes in huge call volume am facing this type of error, [Jun 4 08:42:46] WARNING[8459][C-79fa]: channel.c:5075 ast_write: Codec mismatch on channel Local/8038@xss-call-out-4774;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:04] WARNING[8285][C-79da]: channel.c:5075 ast_write: Codec mismatch on channel Local/6513@xss-call-out-4775;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:10] WARNING[8790][C-7a2c]: channel.c:5075 ast_write: Codec mismatch on channel Local/18002662279@xss-call-out-4778;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:23] WARNING[8355][C-79e6]: channel.c:5075 ast_write: Codec mismatch on channel Local/2896@xss-call-out-4779;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:25] WARNING[7577][C-798a]: channel.c:5075 ast_write: Codec mismatch on channel Local/2896@xss-call-out-477a;1 setting write format to slin from ulaw native formats (ulaw) basically Asterisk will do the slin to ulaw, hope there should not be any problem... But am not sure why am getting this error? will this affect my call? -- _ -- 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] offline builds - mp3 [patch]
On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: As mentioned in the thread about MP3, I found that the rpmbuild process demands network access, e.g. to access the mp3 code in SVN. Some people need to build on isolated networks though I've attached a patch that allows the MP3 code to be placed in /tmp before the build starts, then svn will not be used during the build. If it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead of going to SVN I'm not sure if there are other build steps that access the network, this one was more obvious because I was trying to build on a fresh VM without any svn client I'm sure you're aware of: http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.13.1~dfsg-3/mpglib -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] offline builds - mp3 [patch]
On 04/06/13 18:37, Tzafrir Cohen wrote: On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: As mentioned in the thread about MP3, I found that the rpmbuild process demands network access, e.g. to access the mp3 code in SVN. Some people need to build on isolated networks though I've attached a patch that allows the MP3 code to be placed in /tmp before the build starts, then svn will not be used during the build. If it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead of going to SVN I'm not sure if there are other build steps that access the network, this one was more obvious because I was trying to build on a fresh VM without any svn client I'm sure you're aware of: http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.13.1~dfsg-3/mpglib The notes suggest that MP3 patent issues are a factor so I guessed that's why it is excluded from the tarball When building with rpmbuild the tarball is usually not unpacked manually, hence my own proposed patch looks in /tmp for the mp3 code - it could just as easily use your the patch from Debian as an input though, as long as it can be found in /tmp or some other predefined location. -- _ -- 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] Implementing G729 Passthrough - VM recordings, maybe even a2billing
We would like implement G729 passthrough for our calls and get rid of the encoding overhead, and a little confused as to how to do this, and some unanswered questions. Do we need the open source G729? If so, do we still need the patent license. Not so much of an issue, just checking. Finally, a recent howto of how to enforce pci passthrough and disable encoding would be greatly appreciated. Oh, and there is also the issue with VM recordings, message etc.. On a slightly unrelated, we are using a2billing on some of our machines, and I think that we would have to convert that media from slink to G729 if not done so already... Thanks in Advance, Nick. -- _ -- 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] offline builds - mp3 [patch]
On Tue, Jun 04, 2013 at 06:44:43PM +0200, Daniel Pocock wrote: On 04/06/13 18:37, Tzafrir Cohen wrote: On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: As mentioned in the thread about MP3, I found that the rpmbuild process demands network access, e.g. to access the mp3 code in SVN. Some people need to build on isolated networks though I've attached a patch that allows the MP3 code to be placed in /tmp before the build starts, then svn will not be used during the build. If it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead of going to SVN I'm not sure if there are other build steps that access the network, this one was more obvious because I was trying to build on a fresh VM without any svn client I'm sure you're aware of: http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.13.1~dfsg-3/mpglib The notes suggest that MP3 patent issues are a factor so I guessed that's why it is excluded from the tarball When building with rpmbuild the tarball is usually not unpacked manually, hence my own proposed patch looks in /tmp for the mp3 code - it could just as easily use your the patch from Debian as an input though, as long as it can be found in /tmp or some other predefined location. How would you do that in a proper chrooted build? The proper fix would be to applow to use a newer version of mpglib that is included with some distributions. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] problem to install asterisk on vps digitalocean
Hi list, I try to install asterisk on vps server , but fails when I want to install dahdi [root@shark dahdi-linux-2.6.3-rc1]# make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.32-358.6.2.el6.x86_64 kernel installed. make: *** [modules] Error 1 I have the libraries installed kernel rpm -qa | grep -i kernel kernel-firmware-2.6.32-358.6.2.el6.noarch dracut-kernel-004-303.el6.noarch kernel-devel-2.6.32-358.6.2.el6.x86_64 kernel-2.6.32-358.6.2.el6.x86_64 kernel-headers-2.6.32-358.6.2.el6.x86_64 [root@shark dahdi-linux-2.6.3-rc1]# uname -r 2.6.32-358.6.2.el6.x86_64 regardss -- rickygm http://gnuforever.homelinux.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] problem to install asterisk on vps digitalocean
On Tue, Jun 04, 2013 at 12:09:24PM -0600, troxlinux wrote: You do not appear to have the sources for the 2.6.32-358.6.2.el6.x86_64 kernel installed. make: *** [modules] Error 1 What is the output of: ls -lat /lib/modules/`uname -r`/build and ls -lat /lib/modules/`uname -r`/build/ -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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
Re: [asterisk-users] problem to install asterisk on vps digitalocean
ls -lat /lib/modules/`uname -r`/build lrwxrwxrwx 1 root root 50 May 23 13:06 /lib/modules/2.6.32-358.6.2.el6.x86_64/build - ../../../usr/src/kernels/2.6.32-358.6.2.el6.x86_64 ls -lat /lib/modules/`uname -r`/build/ total 8 drwxr-xr-x 3 root root 4096 May 24 22:02 .. drwxr-xr-x 2 root root 4096 May 23 19:50 . lrwxrwxrwx 1 root root 48 May 23 19:50 linux-headers-2.6.32-358.6.2.el6.x86_64 - /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 lrwxrwxrwx 1 root root 34 May 23 19:34 2.6.32-358.6.2.el6.x86_64 - kernels/2.6.32-358.6.2.el6.x86_64/ 2013/6/4 Russ Meyerriecks rmeyerrie...@digium.com On Tue, Jun 04, 2013 at 12:09:24PM -0600, troxlinux wrote: You do not appear to have the sources for the 2.6.32-358.6.2.el6.x86_64 kernel installed. make: *** [modules] Error 1 What is the output of: ls -lat /lib/modules/`uname -r`/build and ls -lat /lib/modules/`uname -r`/build/ -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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 -- rickygm http://gnuforever.homelinux.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] problem to install asterisk on vps digitalocean
On Tue, Jun 04, 2013 at 12:50:41PM -0600, troxlinux wrote: ls -lat /lib/modules/`uname -r`/build lrwxrwxrwx 1 root root 50 May 23 13:06 /lib/modules/2.6.32-358.6.2.el6.x86_64/build - ../../../usr/src/kernels/2.6.32-358.6.2.el6.x86_64 This looks good. ls -lat /lib/modules/`uname -r`/build/ total 8 drwxr-xr-x 3 root root 4096 May 24 22:02 .. drwxr-xr-x 2 root root 4096 May 23 19:50 . lrwxrwxrwx 1 root root 48 May 23 19:50 linux-headers-2.6.32-358.6.2.el6.x86_64 - /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 lrwxrwxrwx 1 root root 34 May 23 19:34 2.6.32-358.6.2.el6.x86_64 - kernels/2.6.32-358.6.2.el6.x86_64/ Oops. It looks like your build directory didn't get setup correctly. I'd take a look at your -devel and -headers rpms to see what's up. As a workaround, you may be able to specify your headers directly with: KSRC=/usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 make Also, thanks for using -rc! It really helps us out. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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
Re: [asterisk-users] G.729 codec in pass-thru mode
1. Your softphone is not sending g729 [Jun 3 13:11:27] Capabilities: us - 0x10c (ulaw|alaw|g729), *peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing)*, combined - 0x4 (ulaw) I think free version of eyebeam doesn't come with g729, try Microsip or some other with g729 codec. If it is full version, check in the advanced sip settings and allow g729 2. canreinvite should be set to yes for using pass-thru mode check this interesting article Just FYI: Can we bypass Asterisk for RTP session?http://techyatwork.blogspot.ae/2010/10/can-asterisk-bypass-rtp-and-work-like.html Regards, -- _ -- 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] problem to install asterisk on vps digitalocean
thnk Russ , I have seen the Makefile, but I see many KSRC, where exactly would put KVERS:=$(shell uname -r) endif ifndef KSRC ifneq (,$(wildcard /lib/modules/$(KVERS)/build)) KSRC:=/lib/modules/$(KVERS)/build else KSRC_SEARCH_PATH:=/usr/src/linux KSRC:=$(shell for dir in $(KSRC_SEARCH_PATH); do if [ -d $$dir ]; then echo $$dir; break; fi; done) endif endif 2013/6/4 Russ Meyerriecks rmeyerrie...@digium.com On Tue, Jun 04, 2013 at 12:50:41PM -0600, troxlinux wrote: ls -lat /lib/modules/`uname -r`/build lrwxrwxrwx 1 root root 50 May 23 13:06 /lib/modules/2.6.32-358.6.2.el6.x86_64/build - ../../../usr/src/kernels/2.6.32-358.6.2.el6.x86_64 This looks good. ls -lat /lib/modules/`uname -r`/build/ total 8 drwxr-xr-x 3 root root 4096 May 24 22:02 .. drwxr-xr-x 2 root root 4096 May 23 19:50 . lrwxrwxrwx 1 root root 48 May 23 19:50 linux-headers-2.6.32-358.6.2.el6.x86_64 - /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 lrwxrwxrwx 1 root root 34 May 23 19:34 2.6.32-358.6.2.el6.x86_64 - kernels/2.6.32-358.6.2.el6.x86_64/ Oops. It looks like your build directory didn't get setup correctly. I'd take a look at your -devel and -headers rpms to see what's up. As a workaround, you may be able to specify your headers directly with: KSRC=/usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 make Also, thanks for using -rc! It really helps us out. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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 -- rickygm http://gnuforever.homelinux.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] problem to install asterisk on vps digitalocean
On Tue, Jun 04, 2013 at 01:33:56PM -0600, troxlinux wrote: thnk Russ , I have seen the Makefile, but I see many KSRC, where exactly would put KVERS:=$(shell uname -r) endif ifndef KSRC ifneq (,$(wildcard /lib/modules/$(KVERS)/build)) KSRC:=/lib/modules/$(KVERS)/build else KSRC_SEARCH_PATH:=/usr/src/linux KSRC:=$(shell for dir in $(KSRC_SEARCH_PATH); do if [ -d $$dir ]; then echo $$dir; break; fi; done) endif endif No need to modify the Makefile. (Unless you really want to). Just use the following command at your prompt. Bash will pass in the KSRC variable into the make system. KSRC=/usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 make Also, you might try forcing a reload of the -devel package, then seeing if that fixes the build directory. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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
Re: [asterisk-users] problem to install asterisk on vps digitalocean
On Tue, Jun 04, 2013 at 01:33:56PM -0600, troxlinux wrote: thnk Russ , I have seen the Makefile, but I see many KSRC, where exactly would put KVERS:=$(shell uname -r) endif ifndef KSRC ifneq (,$(wildcard /lib/modules/$(KVERS)/build)) KSRC:=/lib/modules/$(KVERS)/build else KSRC_SEARCH_PATH:=/usr/src/linux KSRC:=$(shell for dir in $(KSRC_SEARCH_PATH); do if [ -d $$dir ]; then echo $$dir; break; fi; done) endif endif make KSRC=/full/path/to/kernel/headers/directory Makefiles allow you to override variables in the command line: if you run 'make KSRC=whatever', the variable KSRC is defined, and hence make will not get into the part between 'ifndef KSRC' and the corresponding 'endif'. BTW: this is mentioned in the README of dahdi-linux: http://docs.tzafrir.org.il/dahdi-linux/#_kernel_source_headers -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] offline builds - mp3 [patch]
On 04/06/13 19:13, Tzafrir Cohen wrote: On Tue, Jun 04, 2013 at 06:44:43PM +0200, Daniel Pocock wrote: On 04/06/13 18:37, Tzafrir Cohen wrote: On Tue, Jun 04, 2013 at 12:49:35PM +0200, Daniel Pocock wrote: As mentioned in the thread about MP3, I found that the rpmbuild process demands network access, e.g. to access the mp3 code in SVN. Some people need to build on isolated networks though I've attached a patch that allows the MP3 code to be placed in /tmp before the build starts, then svn will not be used during the build. If it finds /tmp/asterisk-contrib-mp3.tar.gz then it will be used instead of going to SVN I'm not sure if there are other build steps that access the network, this one was more obvious because I was trying to build on a fresh VM without any svn client I'm sure you're aware of: http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.13.1~dfsg-3/mpglib The notes suggest that MP3 patent issues are a factor so I guessed that's why it is excluded from the tarball When building with rpmbuild the tarball is usually not unpacked manually, hence my own proposed patch looks in /tmp for the mp3 code - it could just as easily use your the patch from Debian as an input though, as long as it can be found in /tmp or some other predefined location. How would you do that in a proper chrooted build? The proper fix would be to applow to use a newer version of mpglib that is included with some distributions. I'm not claiming that this was a proper fix - it is just a bare minimum to allow offline builds with rpmbuild. Although it has the feeling of a hack, it doesn't prevent anybody implementing a more elegant solution in future. On the other hand, I was thinking about simply making up my own branch of the code and a repackaged tarball and maybe even publishing some convenient binary RPMs for everybody who wants to try this. I realise that asterisk-11.deb packages are a work in progress too, I didn't want to put pressure on people to finish them, that's why I've just been talking about the RPMs today. -- _ -- 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] problem to install asterisk on vps digitalocean
Hi here again, I've tried both ways [root@shark dahdi-linux-2.6.3-rc1]# KSRC=/usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.32-358.6.2.el6.x86_64 kernel installed. make: *** [modules] Error 1 [root@shark dahdi-linux-2.6.3-rc1]# make KSRC=/usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.32-358.6.2.el6.x86_64 kernel installed. make: *** [modules] Error 1 2013/6/4 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Jun 04, 2013 at 01:33:56PM -0600, troxlinux wrote: thnk Russ , I have seen the Makefile, but I see many KSRC, where exactly would put KVERS:=$(shell uname -r) endif ifndef KSRC ifneq (,$(wildcard /lib/modules/$(KVERS)/build)) KSRC:=/lib/modules/$(KVERS)/build else KSRC_SEARCH_PATH:=/usr/src/linux KSRC:=$(shell for dir in $(KSRC_SEARCH_PATH); do if [ -d $$dir ]; then echo $$dir; break; fi; done) endif endif make KSRC=/full/path/to/kernel/headers/directory Makefiles allow you to override variables in the command line: if you run 'make KSRC=whatever', the variable KSRC is defined, and hence make will not get into the part between 'ifndef KSRC' and the corresponding 'endif'. BTW: this is mentioned in the README of dahdi-linux: http://docs.tzafrir.org.il/dahdi-linux/#_kernel_source_headers -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 -- rickygm http://gnuforever.homelinux.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] problem to install asterisk on vps digitalocean
t == troxlinux xserverli...@gmail.com writes: t I try to install asterisk on vps server , but fails when I want to t install dahdi There is no hardware for dahdi to use; you shouldn't need to install it. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- 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] problem to install asterisk on vps digitalocean
if it is true I have not any hardware but I need help to solve it and I think it could serve other future 2013/6/4 James Cloos cl...@jhcloos.com t == troxlinux xserverli...@gmail.com writes: t I try to install asterisk on vps server , but fails when I want to t install dahdi There is no hardware for dahdi to use; you shouldn't need to install it. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- rickygm http://gnuforever.homelinux.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] G.729 codec in pass-thru mode
Mark Henry wrote: 1. Your softphone is not sending g729 This was a SIP trace of a successful u-law call. In an earlier post Kamlesh provided a trace of a failed G.729 call which did not include the dialog between the Asterisk server and the ITSP. I asked for this trace so that I could see the codecs offered by the ITSP. 2. canreinvite should be set to yes for using pass-thru mode I believe that by pass-thru mode [1] Kamlesh means he wants to avoid transcoding from G.729 to another codec since that requires a license per channel. Pass- thru mode can be achieved with canreinvite=no as shown by the following line from the successful u-law SIP trace and Mark Michelson's asterisk-dev post: [Jun 3 13:11:32] -- Packet2Packet bridging SIP/100-34d8 and SIP/yyy.yyy.yyy.yyy-34d9 From [asterisk-dev] Native Bridging: terminology [2]: ...within SIP, native bridging has two subcategories. One, typically referred to as SIP native bridging is used when reINVITEs are enabled. The endpoints send their media directly to one another. The other subcategory is called Packet 2 Packet or P2P bridging. If reINVITEs are not enabled, but there are also no features that require the Asterisk core to be in the voice path, then the bridging will be done at the RTP layer of Asterisk. [1] http://www.voip-info.org/wiki/view/Asterisk+G.729+pass-thru [2] http://lists.digium.com/pipermail/asterisk-dev/2010-March/043053.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] problem to install asterisk on vps digitalocean
On Tue, Jun 04, 2013 at 01:59:05PM -0600, troxlinux wrote: Hi here again, I've tried both ways [root@shark dahdi-linux-2.6.3-rc1]# KSRC=/usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.32-358.6.2.el6.x86_64 kernel installed. make: *** [modules] Error 1 Let us see the following: ls -lat /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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
Re: [asterisk-users] problem to install asterisk on vps digitalocean
ls -lat /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 ls: cannot access /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64: No such file or directory 2013/6/4 Russ Meyerriecks rmeyerrie...@digium.com On Tue, Jun 04, 2013 at 01:59:05PM -0600, troxlinux wrote: Hi here again, I've tried both ways [root@shark dahdi-linux-2.6.3-rc1]# KSRC=/usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 make make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.32-358.6.2.el6.x86_64 kernel installed. make: *** [modules] Error 1 Let us see the following: ls -lat /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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 -- rickygm http://gnuforever.homelinux.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] problem to install asterisk on vps digitalocean
On Tue, Jun 04, 2013 at 02:53:07PM -0600, troxlinux wrote: ls -lat /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 ls: cannot access /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64: No such file or directory Oops. I got turned around by that broken symlink. Okay, i'm pretty sure your -devel package isn't installed correctly. Try to use yum to remove the -devel package and re-install it. Then check if the /lib/modules/`uname -r`/build directory was installed correctly. As a workaround, the following might work: Check to make sure that this file exists: /usr/src/kernels/2.6.32-358.6.2.el6.x86_64/.config If so, then try: KSRC=/usr/src/kernels/2.6.32-358.6.2.el6.x86_64/ make -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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
Re: [asterisk-users] problem to install asterisk on vps digitalocean
excelente works fine ... thnk russ 2013/6/4 Russ Meyerriecks rmeyerrie...@digium.com On Tue, Jun 04, 2013 at 02:53:07PM -0600, troxlinux wrote: ls -lat /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64 ls: cannot access /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64: No such file or directory Oops. I got turned around by that broken symlink. Okay, i'm pretty sure your -devel package isn't installed correctly. Try to use yum to remove the -devel package and re-install it. Then check if the /lib/modules/`uname -r`/build directory was installed correctly. As a workaround, the following might work: Check to make sure that this file exists: /usr/src/kernels/2.6.32-358.6.2.el6.x86_64/.config If so, then try: KSRC=/usr/src/kernels/2.6.32-358.6.2.el6.x86_64/ make -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 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 -- rickygm http://gnuforever.homelinux.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] Is uniqueid/sequence a safe CDR table primary key ?
OK, then I'll go with linkedid, uniqueid and sequence number. Thanks for sharing this on this list 2013/6/3 Matthew Jordan mjor...@digium.com On 06/03/2013 11:20 AM, Olivier wrote: Hi, When dealing with CDR SQL tables, I always added an auto-incremented cdr_id key as a primary key, just in case provided uniqueid key went wrong. Now I'm facing a situation where I need to insert into a database's table and from the dialplan, a reference to the CDR record which is currently processed. So my questions are: 1. Can uniqueid/sequence (or uniqueid/sequence/calldate) bundle be safely used as CDR's table primary key (ie I cannot have any uniqueid/sequence combination from one CDR record to match a past uniqueid/sequence combination) ? Possibly. Things to keep in mind: * You can run into uniqueid collisions across multiple systems if you do not specify a system name in asterisk.conf or do not specify a unique system name in asterisk.conf. * You can run into uniqueid collisions if your system clock goes backwards for any reason (the uniqueid for a channel happens to use a timestamp for its uniqueness) Whether or not this is unique enough will be completely dependent on your overall system configuration. In general, the recommended combination that *should* uniquely specify a CDR (when configured correctly) is linkedid (which should be enabled and added to your schema), uniqueid, and sequence number, with the asterisk system name specified. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
