Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, 2011-01-24 at 01:09 -0500, RR wrote: On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger pabelan...@digium.com wrote: On 11-01-23 10:24 PM, RR wrote: email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp. coz I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I see, the following If you read CHANGES, you will also see you kernel 2.6.25+ *and* glibc 2.8+. Lenny ships with 2.7-1 yep, had read that too, just very new to debian so was fearing I'll have to do a manual install / upgrade of glibcI guess that's what I have to do :( will figure out how to do that. Just an FYI. Be sure to test it to a non production system, trying to replace glibc from source is not an easy task. *MANY* things need tweaking and lots of apps can break with the wrong glibc version. -- _ -- 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 on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 3:11 AM, Stelios Koroneos skoron...@digital-opsis.com wrote: On Mon, 2011-01-24 at 01:09 -0500, RR wrote: On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger pabelan...@digium.com wrote: On 11-01-23 10:24 PM, RR wrote: email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp. coz I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I see, the following If you read CHANGES, you will also see you kernel 2.6.25+ *and* glibc 2.8+. Lenny ships with 2.7-1 yep, had read that too, just very new to debian so was fearing I'll have to do a manual install / upgrade of glibcI guess that's what I have to do :( will figure out how to do that. Just an FYI. Be sure to test it to a non production system, trying to replace glibc from source is not an easy task. *MANY* things need tweaking and lots of apps can break with the wrong glibc version. Thanks for the warning Stelios. Yes, This is a VM which I snapshot every step of the way to revert back to if I break something too bad. it's a lot easier to just revert to snapshot in 20 secs, then trying to fix whatever broke :) -- _ -- 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 on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote: In the meantime, does anyone have a nice way to update a stable/stock lenny installation with the updated glibc as well as the latest kernel At this point the easiest option will be to upgrade to squeeze. R -- _ -- 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] Crossover cable for E1 ?
On Saturday 22 Jan 2011, Tim Panton wrote: I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card. Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable? If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to stock such a thing. It's easy to make an ISDN crossover cable. Cut one end of a standard network cable. Get a new RJ45 plug, rubber boot (not strictly necessary, but makes it look neater) and crimping tool. Push the rubber boot onto the end of the cable first. Strip about 2cm. of outer sheath and separate the inner pairs, then arrange in this order from left to right with the brass contacts uppermost: white/blue blue white/green orange white/orange green white/brown brown (i.e., wrap the green pair around the orange pair). Now trim the wires so there is just over 1cm. protruding from the outer sheath. Insert the wires into the new plug, crimp, and pull the rubber boot down over the plug. If you have an AVO, test for continuity as follows: pin 1 to pin 5, pin 2 to pin 4, pin 4 to pin 2, pin 5 to pin 1. (Pins 3, 6, 7 and 8 are not used in this configuration.) Finally, as this cable is a special one, BE SURE TO LABEL IT to prevent mistakes. For completeness' sake I ought to say, if you want or need to crimp the other end yourself, the wires should be arranged as follows: white/orange orange white/green blue white/blue green white/brown brown You did push the rubber boot on the end before you crimped the plug, didn't you? -- AJS Answers come *after* questions. -- _ -- 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 on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West ro...@firedrake.orgwrote: On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote: In the meantime, does anyone have a nice way to update a stable/stock lenny installation with the updated glibc as well as the latest kernel At this point the easiest option will be to upgrade to squeeze. R Umm yeah that might not be a smart thing to do since eventually all of this needs to run in a production environment and Squeeze is still in a RC mode. Would be nice if I could go to it though but don't think it'll be that smart esp. all other software that needs to work along with it might break too...who knows -- _ -- 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] Crossover cable for E1 ?
On Monday 24 January 2011 03:46:18 A J Stiles wrote: On Saturday 22 Jan 2011, Tim Panton wrote: I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card. Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable? If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to stock such a thing. It's easy to make an ISDN crossover cable. Cut one end of a standard network cable. Get a new RJ45 plug, rubber boot (not strictly necessary, but makes it look neater) and crimping tool. Push the rubber boot onto the end of the cable first. Strip about 2cm. of outer sheath and separate the inner pairs, then arrange in this order from left to right with the brass contacts uppermost: white/blue blue white/green orange white/orange green white/brown brown This is incorrect. The pairs should be: blue white/blue white/green white/orange orange green white/brown brown Wire 1 MUST swap with 4 and Wire 2 MUST swap with 5. To do as you have shown above switches the polarity on each electrical circuit. It is especially important that you do not switch the polarity, as some equipment does not auto-correct for reversed polarity. -- Tilghman -- _ -- 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 1.6 SSH Console Colors Debian Lenny
2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote: On Thursday 20 Jan 2011, JR Richardson wrote: Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I'm running Debian but have been running Asterisk since before there was a proper Debian package, and so I ended up writing my own init.d script. See attached. No guarantees or anything :) A number of things I did not like about it: 1. I don't trust safe_asterisk to properly handle being run twice and such. 2. Likewise with daemonization. safe_asterisk is still at the console. 3. You run asterisk as root. And use /var/run/asterisk.pid . Please use a non-root user and /var/run/asterisk/asterisk.pid . 4. On 'restart' you do nothing if the process was not running. That's not the standard semantics. 5. Even if a pid file exists, it does not mean that the process listed in it is your process. In short: A. Don't re-invent start-stop-daemon. B. Let's just move to upstart/systemd so there won't be a need for this stupid guardian safe asterisk. All these reasons seem fine for me. So the remaining question is how can we still get colors with ssh console ?. Is it compliant with start-stop-daemon, for instance ? Cheers -- _ -- 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 1.6 SSH Console Colors Debian Lenny
On Monday 24 January 2011 04:09:31 Olivier wrote: 2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote: On Thursday 20 Jan 2011, JR Richardson wrote: Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. In short: A. Don't re-invent start-stop-daemon. B. Let's just move to upstart/systemd so there won't be a need for this stupid guardian safe asterisk. All these reasons seem fine for me. So the remaining question is how can we still get colors with ssh console ?. Is it compliant with start-stop-daemon, for instance ? Why not just use the start script included with Asterisk? I solved this exact problem a while back, so unless somebody has broken the script since, it should still be working. -- Tilghman -- _ -- 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] Info on using LDAP with Asterisk?
On 11-01-23 02:56 PM, Jeff B wrote: There does not seem to be very much info out there about using LDAP to create asterisk configurations. Does anyone have some information that they would suggest I start with? We've tried to document some of it here: http://ofps.oreilly.com/titles/9780596517342/ch18.html#ExternalServices_id291590 Thanks! Leif. -- _ -- 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] Crossover cable for E1 ?
On Monday 24 Jan 2011, Tilghman Lesher wrote: On Monday 24 January 2011 03:46:18 A J Stiles wrote: white/blue blue white/green orange white/orange green white/brown brown This is incorrect. The pairs should be: blue white/blue white/green white/orange orange green white/brown brown Wire 1 MUST swap with 4 and Wire 2 MUST swap with 5. To do as you have shown above switches the polarity on each electrical circuit. It is especially important that you do not switch the polarity, as some equipment does not auto-correct for reversed polarity. I stand corrected -- though, the arrangement I described has definitely worked for me in the past (I'm looking right now at a homebrew crossover cable I replaced with a longer one when we installed a new server), so maybe our WCT410P is just more forgiving than some kit out there? Anyway, Tilghman is the expert, I'm just a satisfied user :) -- AJS Answers come *after* questions. -- _ -- 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 stops responding
Am 23.01.2011 18:38, schrieb Carlos Chavez: On Sat, 22 Jan 2011 19:47:43 -0500, Mark Deneen wrote On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez cur...@telecomabmex.com wrote: On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote On 22 Jan 2011, at 18:02, Carlos Chavez wrote: Cannot allocate memory Have you tried looking at memory? S The server has 8gb of ram and 8gb of swap. Free indicates that there are at least two free gb of memory and swap remains at 0 use. Just asking the obvious, but, x86-64? How big is the asterisk process? 4650 root 15 0 3153M 132M 10420 S 0.0 1.6 3:24.40 /usr/sbin/asterisk -f -vvvg -c 2.6.18-194.32.1.el5 #1 SMP Wed Jan 5 17:52:25 EST 2011 x86_64 x86_64 x86_64 GNU/Linux Plase show us the header of the used ps-command. -- _ -- 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] Info on using LDAP with Asterisk?
Thanks, That's more info than I've been able to find to date. I'll work on digesting it now. On Mon, Jan 24, 2011 at 7:37 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 11-01-23 02:56 PM, Jeff B wrote: There does not seem to be very much info out there about using LDAP to create asterisk configurations. Does anyone have some information that they would suggest I start with? We've tried to document some of it here: http://ofps.oreilly.com/titles/9780596517342/ch18.html#ExternalServices_id291590 Thanks! Leif. -- _ -- 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] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com wrote: On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West ro...@firedrake.orgwrote: On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote: In the meantime, does anyone have a nice way to update a stable/stock lenny installation with the updated glibc as well as the latest kernel At this point the easiest option will be to upgrade to squeeze. R Umm yeah that might not be a smart thing to do since eventually all of this needs to run in a production environment and Squeeze is still in a RC mode. Would be nice if I could go to it though but don't think it'll be that smart esp. all other software that needs to work along with it might break too...who knows Wow, alright, after an all-nighter, I was able to get timerfd.so compiled in Asterisk 1.8.2.2 under Debian Lenny 5.0.7 with Kernel 2.6.26-2-amd64. Of course, due to the glibc requirement of 2.8+, a lot of dodgey upgrades had to be performed. I have no idea how stable this is going to be in production but I am going to write a quick How-To and stick it on the Wiki if someone can point me to the correct location this should go to. A lot of components needs to get upgraded in the correct order to have this work well, but it might save someone else the time and effort. Will respond to this email again, with the link to the Wiki page once I am done with the HowTo and people tell me where it needs to go. Cheers, \RR -- _ -- 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] Info on using LDAP with Asterisk?
On Mon, Jan 24, 2011 at 10:22 AM, Jeff B jeffb.l...@gmail.com wrote: Thanks, That's more info than I've been able to find to date. I'll work on digesting it now. Please add you comments and findings to https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver to help us have a good source of information. ~~~ Andrew lathama Latham lath...@gmail.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
[asterisk-users] Unable to insert cdr-data into mysql-DB
Hello list, I keep on getting the error : ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server 127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost' (using password: YES) I have a 'cdr' table in my MySQL-DB. On this table the user 'asteriskcdr' has select, insert, update privileges. GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO 'asteriskcdr'@'127.0.0.1'; cdr_mysql.conf : [global] hostname=127.0.0.1 dbname=Asterisk table=cdr password=mysecret user=asteriskcdr port=3306 sock=/tmp/mysql.sock userfield=1 I really don't know why Asterisk cannot connect to the table.. Kind regards, Jonas. -- _ -- 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] Voicemail hangs up
Hello. I am running Asterisk v1.6 on Ubuntu 10.10 with FreePBX v2.8. When I call the voicemail for any of my extensions, the call just dies. On a softphone, I get no sound whatsoever; it just hangs up after a couple of seconds. On my handset attached to my SPA-3102, it get a sound like when you leave an analogue phone off the hook. I have three extensions setup and they all do the same thing. Everything was configured through FreePBX. It did work, initially, but I think I changed something and now it is not. Here is the asterisk log: --- START --- == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [*97@from-internal:1] Answer(SIP/203-0003, ) in new stack -- Executing [*97@from-internal:2] Wait(SIP/203-0003, 1) in new stack -- Executing [*97@from-internal:3] Macro(SIP/203-0003, user-callerid,) in new stack -- Executing [s@macro-user-callerid:1] Set(SIP/203-0003, AMPUSER=203) in new stack -- Executing [s@macro-user-callerid:2] GotoIf(SIP/203-0003, 0?report) in new stack -- Executing [s@macro-user-callerid:3] ExecIf(SIP/203-0003, 1?Set(REALCALLERIDNUM=203)) in new stack -- Executing [s@macro-user-callerid:4] Set(SIP/203-0003, AMPUSER=203) in new stack -- Executing [s@macro-user-callerid:5] Set(SIP/203-0003, AMPUSERCIDNAME=Test) in new stack -- Executing [s@macro-user-callerid:6] GotoIf(SIP/203-0003, 0?report) in new stack -- Executing [s@macro-user-callerid:7] Set(SIP/203-0003, AMPUSERCID=203) in new stack -- Executing [s@macro-user-callerid:8] Set(SIP/203-0003, CALLERID(all)=Test 203) in new stack -- Executing [s@macro-user-callerid:9] GotoIf(SIP/203-0003, 0?continue) in new stack -- Executing [s@macro-user-callerid:10] Set(SIP/203-0003, __TTL=64) in new stack -- Executing [s@macro-user-callerid:11] GotoIf(SIP/203-0003, 1?continue) in new stack -- Goto (macro-user-callerid,s,18) -- Executing [s@macro-user-callerid:18] NoOp(SIP/203-0003, Using CallerID Test 203) in new stack -- Executing [*97@from-internal:4] Macro(SIP/203-0003, get-vmcontext,203) in new stack -- Executing [s@macro-get-vmcontext:1] Set(SIP/203-0003, VMCONTEXT=default) in new stack -- Executing [s@macro-get-vmcontext:2] GotoIf(SIP/203-0003, 0?200:300) in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing [s@macro-get-vmcontext:300] NoOp(SIP/203-0003, ) in new stack -- Executing [*97@from-internal:5] Set(SIP/203-0003, VMBOXEXISTSSTATUS=SUCCESS) in new stack -- Executing [*97@from-internal:6] GotoIf(SIP/203-0003, 1?mbexist) in new stack -- Goto (from-internal,*97,106) -- Executing [*97@from-internal:106] VoiceMailMain(SIP/203-0003, 203@default) in new stack -- Executing [*97@from-internal:107] GotoIf(SIP/203-0003, 0?playret) in new stack -- Executing [*97@from-internal:108] Macro(SIP/203-0003, hangupcall,) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/203-0003, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/203-0003, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/203-0003, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/203-0003, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/203-0003' in macro 'hangupcall' == Spawn extension (from-internal, *97, 108) exited non-zero on 'SIP/203-0003' -- Executing [h@from-internal:1] Macro(SIP/203-0003, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/203-0003, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/203-0003, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/203-0003, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/203-0003, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/203-0003' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/203-0003' --- END --- Here is my sip_additional.conf entry for the extension I used above (302): --- START --- [203] deny=0.0.0.0/0.0.0.0 secret= (blanked for security reasons) dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic type=friend nat=yes port=5060 qualify=yes callgroup= pickupgroup= dial=SIP/203 mailbox=203@default permit=0.0.0.0/0.0.0.0 callerid=device 203 callcounter=yes faxdetect=no --- END --- Here is the entry for extension 203 in extensions_additional.conf: --- START --- exten = 203,1,Macro(exten-vm,203,203) exten = 203,n,Goto(vmret,1) exten = 203,hint,SIP/203 exten =
Re: [asterisk-users] Unable to insert cdr-data into mysql-DB
Hi, maybe the error is on this line: sock=/tmp/mysql.sock if you use CentOS the correct line is: sock=/var/lib/mysql/mysql.sock if you use Debian/ubuntu: sock=/var/run/mysqld/mysqld.sock 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] Unable to insert cdr-data into mysql-DB
Jonas Kellens wrote: Hello list, I keep on getting the error : ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server 127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost' (using password: YES) I have a 'cdr' table in my MySQL-DB. On this table the user 'asteriskcdr' has select, insert, update privileges. GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO 'asteriskcdr'@'127.0.0.1'; [snip] Grant rights to 'asteriskcdr'@'localhost' instead of @'127.0.0.1'? (re the error message above) Regards, Marius -- _ -- 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] Unable to insert cdr-data into mysql-DB
On 01/24/2011 07:43 AM, Jonas Kellens wrote: Hello list, I keep on getting the error : ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server 127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost' (using password: YES) I have a 'cdr' table in my MySQL-DB. On this table the user 'asteriskcdr' has select, insert, update privileges. GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO 'asteriskcdr'@'127.0.0.1'; cdr_mysql.conf : [global] hostname=127.0.0.1 dbname=Asterisk table=cdr password=mysecret user=asteriskcdr port=3306 sock=/tmp/mysql.sock userfield=1 I really don't know why Asterisk cannot connect to the table.. Kind regards, Jonas. -- _ -- 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 redo your grant like this: GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO 'asteriskcdr'@'127.0.0.1' IDENTIFIED by 'mysecret'; follow it with: flush privileges; -- _ -- 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 on Debian Lenny with timerfd
On 01/24/2011 07:29 AM, RR wrote: On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com mailto:ranjt...@gmail.com wrote: On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West ro...@firedrake.org mailto:ro...@firedrake.org wrote: On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote: In the meantime, does anyone have a nice way to update a stable/stock lenny installation with the updated glibc as well as the latest kernel At this point the easiest option will be to upgrade to squeeze. R Umm yeah that might not be a smart thing to do since eventually all of this needs to run in a production environment and Squeeze is still in a RC mode. Would be nice if I could go to it though but don't think it'll be that smart esp. all other software that needs to work along with it might break too...who knows This a statement we hear from people periodically that just confuses me... they say they can't update to an 'RC' release of something (Linux distro, Asterisk, etc.) because they need to run in production mode, but they're willing to consider replacing something as fundamental as the Linux kernel (a bit scary) or glibc (very scary) instead. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.org mailto:ro...@firedrake.org wrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a
stable/stock lenny
installation with the updated glibc as well as the latest kernel
At this point the easiest option will be to upgrade to squeeze.
R
Umm yeah that might not be a smart thing to do since eventually all
of this needs to run in a production environment and Squeeze is
still in a RC mode. Would be nice if I could go to it though but
don't think it'll be that smart esp. all other software that needs
to work along with it might break too...who knows
This a statement we hear from people periodically that just confuses me...
they say they can't update to an 'RC' release of something (Linux distro,
Asterisk, etc.) because they need to run in production mode, but they're
willing to consider replacing something as fundamental as the Linux kernel
(a bit scary) or glibc (very scary) instead.
haha touché Kevin :) Mate, the response to that is one word: Ignorance :)
people like me, who're not developers nor experts of the platform have
absolutely no clue what glibc actually does or the impact it actually has.
Nor do I know, as a user, how stable Squeeze RC2 really is at this stage of
its development. If I had more people in the community say that they're
running it in production, then maybe I'll just believe them and start
working with Squeeze directly instead of wasting my time like I did trying
to have it compiled in Lenny. I just believed when the developers of Debian
say that Squeeze RC2 is in testing and Lenny is stable and decide that
it's probably not a good idea to run RC2 in production. I guess part of the
thinking was that other software besides {*} that needs to run on this
machine may not even build or run or be stable on Squeeze RC till the
authors/users of that other software state that it's been tested with it and
it's stable or even builds on it. So, people like me believe that if I
upgrade ALL components that depend on glibc and that glibc depends on to the
current version, then we'll be ok but we wouldn't have touched anything else
in the system, not realising or understanding that satsisfying dependencies
doesn't mean anything and something somewhere could just break because of
this unsolicited upgrade thus making the system more unstable. I have really
no explanation for you as to why people (incl. myself) say these things
other than just lack of insight and knowledge about the intricacies of
things like glibc and the impact it can have on the stability of the system
when upgraded out of context. *sigh* :(
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Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
In the meantime, does anyone have a nice way to update a stable/stock lenny installation with the updated glibc as well as the latest kernel Scary and risky, as others have noted! There is an official backports release kit associated with Debian, which contains newer versions of many packages which have been back-ported to be mostly-drop-in-compatible with current Debian stable distribution. You can find information about it at http://backports.debian.org/ However, it does not appear to contain an updated release of glibc - likely for the reasons that other folks have alluded to (the stability risks outweigh the benefits). I suspect that unless you're willing to put a lot of blood, sweat, tears, and toil into the effort of getting the newer glibc into Lenny, you're either going to have to switch to the testing distribution (Squeeze) or wait until Squeeze is officially released as the new stable distribution -- _ -- 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] ReceiveFAX issue.
On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman brya...@zktech.com wrote: I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. Record the call with Monitor() or MixMonitor(). Listen to the call. See if you can figure out anything obvious. Why are you doing a Wait(2)? -- _ -- 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] U-verse DTMF tuning for Zaptel
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Monday, January 24, 2011 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] U-verse DTMF tuning for Zaptel On Mon, Jan 24, 2011 at 3:37 PM, Steve Edwards asterisk@sedwards.com wrote: One of my clients is complaining that their customers that use U-verse (and other cable providers) for telephone service cannot enter credit card numbers reliably. The issue not all digits are received in my dialplan. The calls come in on PRI. It's an old 1.2 install, so the only tweak available is 'relaxdtmf.' Any clues on how to proceed? Would jumping to 1.6 help? Can you record a few calls just to confirm the problem? I've had 'dtmf issues' that really were cases of people entering data during recordings rather than at the appropriate time in the prompts. I know this is old school thinking, but do the users need a beep prompt to tell them when to start hitting their keys? -- _ -- 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] U-verse DTMF tuning for Zaptel
On Mon, Jan 24, 2011 at 3:37 PM, Steve Edwards asterisk@sedwards.com wrote: One of my clients is complaining that their customers that use U-verse (and other cable providers) for telephone service cannot enter credit card numbers reliably. The issue not all digits are received in my dialplan. The calls come in on PRI. It's an old 1.2 install, so the only tweak available is 'relaxdtmf.' Any clues on how to proceed? Would jumping to 1.6 help? On Mon, 24 Jan 2011, David Backeberg wrote: Can you record a few calls just to confirm the problem? We know the problem exists -- the boss just installed U-verse at his house :) Recording may have some value to see if DTMF is being received in the audio stream and to measure tone duration. I've had 'dtmf issues' that really were cases of people entering data during recordings rather than at the appropriate time in the prompts. The customers enter their card number while a 'preamble' is backgrounding. It works fine from cell and copper, just not from U-verse and their ilk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] U-verse DTMF tuning for Zaptel
On Mon, Jan 24, 2011 at 4:51 PM, Steve Edwards asterisk@sedwards.com wrote: We know the problem exists -- the boss just installed U-verse at his house :) It works fine from cell and copper, just not from U-verse and their ilk. Well, I would say more data samples are needed then. It could certainly be the boss's uverse connection and not other uverse connections. If every SIP connection failed wouldn't you know it by 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] U-verse DTMF tuning for Zaptel
On Mon, 24 Jan 2011, David Backeberg wrote: Well, I would say more data samples are needed then. It could certainly be the boss's uverse connection and not other uverse connections. They've had complaints from off-site agents and from long term customers who noticed that 'it stopped working' when they switched. If every SIP connection failed wouldn't you know it by now? If it were SIP... My client guestimates it may affect up to 20% of their customers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] U-verse DTMF tuning for Zaptel
On 01/24/2011 03:54 PM, David Backeberg wrote: Well, I would say more data samples are needed then. It could certainly be the boss's uverse connection and not other uverse connections. Have you ruled out a problem on the U-Verse side? In particular, do U-Verse customers only experience this problem with your client's system? I assume than any cable- or IP-based telephony provider is using some- thing very much like an ATA to connect their customer's analog phones to their digital network. If they're compressing the audio stream, they can't use in-band DTMF, so the ATA has to listen to the audio, attempt to recognize DTMF and send it on via SIP, AVT, etc. As I found out with the PAP2T in my home office, it's hard to get this 100% right. If the DTMF is being missed on the U-Verse end and never sent to your client, there's nothing that you can do. (A recording should very quickly establish whether or not this is the case.) -- Ian Pilcher arequip...@gmail.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
[asterisk-users] extconfig, realtime, and SIP
I'm confused about a few things relating to realtime, SIP and config in general. As I understand it, with the exception of extensions.conf, I can either have a config file completely in text or completely in a database. Is that correct? I can't find documentation for exactly what switch = does but is that only in the dialplan and a way to have it partly from a file and partly from a database? For SIP, do I understand it correctly that I can have sip tables both via realtime AND in sip.conf? For the LDAP realtime, how can I implement setvar? -- _ -- 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 on Debian Lenny with timerfd
On 01/24/2011 12:46 PM, RR wrote:
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com
mailto:ranjt...@gmail.com mailto:ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.org mailto:ro...@firedrake.org
mailto:ro...@firedrake.org mailto:ro...@firedrake.org wrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a
stable/stock lenny
installation with the updated glibc as well as the latest kernel
At this point the easiest option will be to upgrade to
squeeze.
R
Umm yeah that might not be a smart thing to do since
eventually all
of this needs to run in a production environment and Squeeze is
still in a RC mode. Would be nice if I could go to it though but
don't think it'll be that smart esp. all other software that
needs
to work along with it might break too...who knows
This a statement we hear from people periodically that just confuses
me... they say they can't update to an 'RC' release of something
(Linux distro, Asterisk, etc.) because they need to run in
production mode, but they're willing to consider replacing something
as fundamental as the Linux kernel (a bit scary) or glibc (very
scary) instead.
haha touché Kevin :) Mate, the response to that is one word: Ignorance
:) people like me, who're not developers nor experts of the platform
have absolutely no clue what glibc actually does or the impact it
actually has. Nor do I know, as a user, how stable Squeeze RC2 really is
at this stage of its development. If I had more people in the community
say that they're running it in production, then maybe I'll just believe
them and start working with Squeeze directly instead of wasting my time
like I did trying to have it compiled in Lenny. I just believed when the
developers of Debian say that Squeeze RC2 is in testing and Lenny is
stable and decide that it's probably not a good idea to run RC2 in
production. I guess part of the thinking was that other software
besides {*} that needs to run on this machine may not even build
or run or be stable on Squeeze RC till the authors/users of that other
software state that it's been tested with it and it's stable or even
builds on it. So, people like me believe that if I upgrade ALL
components that depend on glibc and that glibc depends on to the current
version, then we'll be ok but we wouldn't have touched anything else in
the system, not realising or understanding that satsisfying dependencies
doesn't mean anything and something somewhere could just break because
of this unsolicited upgrade thus making the system more unstable. I have
really no explanation for you as to why people (incl. myself) say these
things other than just lack of insight and knowledge about the
intricacies of things like glibc and the impact it can have on the
stability of the system when upgraded out of context. *sigh* :(
And you've made my point: You chose a specific version of Debian to run,
which you are happy running in 'production'. Given that you have made
that choice, you can *only* install packages that distribution provides
on your system. Any other packages you install are not part of that
version, and thus have not gone through the same testing/qualification
processes (whatever they may be). Discussing installation of packages
(any packages) from a later Debian release, or installation of a package
from source that overwrites the Debian package, seems totally
inconsistent with being 'in production', no matter how small or large
the package may be. Each such decision must be thoroughly researched and
the possible ramifications understood before any changes are made, so as
to keep the system as stable as possible.
In essence, this is somewhat like buying a car with a high efficiency
powertrain because you want to save fuel, but then later complaining
that it doesn't accelerate as fast as you'd like... so you make plans to
replace the engine. Sure, you can do it, but you've defeated the purpose
of the choice you made in the first place :-)
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com www.asterisk.org
--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To
[asterisk-users] Lots of warnings: SUBSCRIBE failure: no Accept header: pvt
Does anyone know how to get rid of these warnings? ~ [Jan 24 18:10:04] WARNING[2629]: chan_sip.c:15898 handle_request_subscribe: SUBSCRIBE failure: no Accept header: pvt: stateid: -1, laststate: 0, dialogver: 0, subscribecont: 'local-extensions', subscribeuri: '' ~ Version: Asterisk 1.4.26.1 Using Polycom phones primarily. Any ideas? -- _ -- 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] RTCP packets when on hold
I have continued searching but still haven’t found any way to get asterisk to send RTCPs when on hold. This issue seems to have come up several times and has been reported by several people but nothing seems to have come of it. Should I be filing a bug report? Or are there any workarounds available on the asterisk side? Ryan Tucker SERVICE DESK ENGINEER [cid:image001.jpg@01CBBC7A.FC1AD500] p +61 (0) 7 3018 0280 f +61 (0) 7 3018 0282 w www.rgtech.com.auhttp://www.rgtech.com.au/ P Please consider the environment before printing CAUTION - This message may contain privileged and confidential information intended only for the use of the addressee named above. If you are not the intended recipient of this message you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify R G Technologies immediately. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of R G Technologies. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Tucker Sent: Sunday, 23 January 2011 8:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RTCP packets when on hold Hi, It seems that asterisk doesn't send RTCP packets when a call is on hold. Is there any way to get asterisk to send these packets? I'm in the process of setting up a Lync (microsoft voice) server which will use an asterisk box as a gateway. The trunking between asterisk and lync is 'working' however when a call is put on hold asterisk stops sending RTCP packets to lync, and after 30 seconds of not receiving RTCP packets, lync drops the call due to timeout. I did find two posts from people who came accross this same issue back in 2008, however there was no solution given. Thanks in advance, Ryan. inline: image001.jpg-- _ -- 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] Unknow T callerid
Hi List. Have any of you guys ever see an incoming call throught Dahdi channel which has an callerid T. I know whenever is a private call, it shows as callerid, but what does it mean a T callerid? Best Regards -- Jose Flores Galicia floj...@gmail.com BriefCode Code Based Training -- _ -- 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 on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 7:07 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/24/2011 12:46 PM, RR wrote:
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com
mailto:ranjt...@gmail.com mailto:ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.org mailto:ro...@firedrake.org
mailto:ro...@firedrake.org mailto:ro...@firedrake.org
wrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a
stable/stock lenny
installation with the updated glibc as well as the latest kernel
At this point the easiest option will be to upgrade to
squeeze.
R
Umm yeah that might not be a smart thing to do since
eventually all
of this needs to run in a production environment and Squeeze is
still in a RC mode. Would be nice if I could go to it though
but
don't think it'll be that smart esp. all other software that
needs
to work along with it might break too...who knows
This a statement we hear from people periodically that just confuses
me... they say they can't update to an 'RC' release of something
(Linux distro, Asterisk, etc.) because they need to run in
production mode, but they're willing to consider replacing something
as fundamental as the Linux kernel (a bit scary) or glibc (very
scary) instead.
haha touché Kevin :) Mate, the response to that is one word: Ignorance
:) people like me, who're not developers nor experts of the platform
have absolutely no clue what glibc actually does or the impact it
actually has. Nor do I know, as a user, how stable Squeeze RC2 really is
at this stage of its development. If I had more people in the community
say that they're running it in production, then maybe I'll just believe
them and start working with Squeeze directly instead of wasting my time
like I did trying to have it compiled in Lenny. I just believed when the
developers of Debian say that Squeeze RC2 is in testing and Lenny is
stable and decide that it's probably not a good idea to run RC2 in
production. I guess part of the thinking was that other software
besides {*} that needs to run on this machine may not even build
or run or be stable on Squeeze RC till the authors/users of that other
software state that it's been tested with it and it's stable or even
builds on it. So, people like me believe that if I upgrade ALL
components that depend on glibc and that glibc depends on to the current
version, then we'll be ok but we wouldn't have touched anything else in
the system, not realising or understanding that satsisfying dependencies
doesn't mean anything and something somewhere could just break because
of this unsolicited upgrade thus making the system more unstable. I have
really no explanation for you as to why people (incl. myself) say these
things other than just lack of insight and knowledge about the
intricacies of things like glibc and the impact it can have on the
stability of the system when upgraded out of context. *sigh* :(
And you've made my point: You chose a specific version of Debian to run,
which you are happy running in 'production'. Given that you have made that
choice, you can *only* install packages that distribution provides on your
system. Any other packages you install are not part of that version, and
thus have not gone through the same testing/qualification processes
(whatever they may be). Discussing installation of packages (any packages)
from a later Debian release, or installation of a package from source that
overwrites the Debian package, seems totally inconsistent with being 'in
production', no matter how small or large the package may be. Each such
decision must be thoroughly researched and the possible ramifications
understood before any changes are made, so as to keep the system as stable
as possible.
In essence, this is somewhat like buying a car with a high efficiency
powertrain because you want to save fuel, but then later complaining that it
doesn't accelerate as fast as you'd like... so you make plans to replace the
engine. Sure, you can do it, but you've defeated the purpose of the choice
you made in the first place :-)
I know right? I wish I could have those hours of the night back that I
wasted in trying to get it working on Lenny ... wish I'd done some homework
and realised that all sorts of Squeeze installation ISOs are in fact
available for Sparc. I thought currently only Lenny was available for Sparc
so needed to stick with it. Oh well, that's a lesson for me right there. But
hopefully not all was a wasted effort,
[asterisk-users] DNS A queries
Hi I want to solve ipaddress with DNS A queries. There is the problem that ipddress resolution of the DNS is not correct. Because Ａsterisk is not put sip. before a message of XX.com I could see in the log that asterisk send A queries without sip. Is this problem for asterisk configuration or hosts setting? Any help would be really appreciated DNSStandard query SRV _sip._UDP.xx.com DNSStandard query response SRV 0 0 5060 sip.xx.com DNSStandard query SRV _sip._UDP.xx.com DNSStandard query response SRV 0 0 5060 sip.xx.com DNSStandard query SRV _sip._UDP.xx.com DNSStandard query response SRV 0 0 5060 sip.xx.com DNSStandard query A xx.com DNSStandard query response A 126.**.**.** DNSStandard query A xx.com DNSStandard query response A 126.**.**.** version 1.6.0.28 Global Signalling Settings: DNS SRV lookup: Yes CentOS 5.5 thanks -- _ -- 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 on Debian Lenny with timerfd
I know this is an {*} list but does anyone know if simply adding the Squeeze
repository to my sources.lst and running an 'aptitude
upgrade/safe-upgrade/full-upgrade will just upgrade Lenny - Squeeze
without me having to rebuild the system from scratch?
In my experience: you're likely to run into a few things which
need some amount of manual fiddling, after an upgrade of this sort,
but it's usually quite manageable.
The Debian people seem to be very good about making sure that
stable-version-to-stable-version upgrades go smoothly... the
process isn't perfect (from what I've seen) but it's usually
quite close. The upgrade path is usually tested out quite well
before the release team throws The Big Switch, and there normally
are good release notes which describe the corner cases which may
need manual intervention.
I have several systems which have been through multiple major
Debian upgrades, without having to be slagged down and rebuilt
from the ground up. That's better than I ever achieved with (e.g.)
Red Hat, which (in my experience) really didn't take at all well to
in-place upgrades... I usually had to do a fresh install and then
port my personal files over.
Things may not be as smooth when jumping from Stable to Testing,
precisely because this isn't an official-release pathway, and
the packages in Testing are usually in somewhat of a state of
flux. Even upgrades *within* the Testing distribution can leave
you with a system which doesn't fly right... this isn't common but
it does happen. For example, a recent upgrade within Stable pulled
a bunch of the firmware files out of the kernel package and moved
them to a separate non-free package - if I hadn't noticed an error
message during RAMdisk rebuilt, my next boot would have left me
with a non-functioning wired Ethernet adapter.
If you decide to follow this route, follow the Debian instructions
for upgrading... back up your package configurations, and (I suggest)
everything in the /etc/ directory hierarchy, as well as all of your
personal files. This will give you a much better chance to invoke
the spirit of the ancient pagan god DoOver, if something goes wrong
during the upgrade.
--
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Re: [asterisk-users] U-verse DTMF tuning for Zaptel
On Mon, 24 Jan 2011, David Backeberg wrote: Can you record a few calls just to confirm the problem? If I record via mixmonitor(), I just get a bunch of clicks where the DTMF should be. I'm assuming this is because the DSP has already taken the DTMF out of the audio stream. If I record via 'sudo ztmonitor 89 -R rx -T tx' and then convert to wav with '/usr/bin/sox -r 8000 -s -w -c 1 -t raw tx tx.wav' all I get is a steady 'high pitched tone' which Audacity says has peaks at 16hz, 1khz, 2khz, 3khz and 4khz. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] U-verse DTMF tuning for Zaptel
On Mon, 24 Jan 2011, David Backeberg wrote: Can you record a few calls just to confirm the problem? On Mon, 24 Jan 2011, Steve Edwards wrote: If I record via mixmonitor(), I just get a bunch of clicks where the DTMF should be. I'm assuming this is because the DSP has already taken the DTMF out of the audio stream. If I record via 'sudo ztmonitor 89 -R rx -T tx' and then convert to wav with '/usr/bin/sox -r 8000 -s -w -c 1 -t raw tx tx.wav' all I get is a steady 'high pitched tone' which Audacity says has peaks at 16hz, 1khz, 2khz, 3khz and 4khz. ztmonitor -vv shows rx values peaking around 6000 and tx values peaking around 7000. Isn't that rather 'hot?' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 8:57 PM, Dave Platt dpl...@radagast.org wrote:
I know this is an {*} list but does anyone know if simply adding the
Squeeze
repository to my sources.lst and running an 'aptitude
upgrade/safe-upgrade/full-upgrade will just upgrade Lenny - Squeeze
without me having to rebuild the system from scratch?
In my experience: you're likely to run into a few things which
need some amount of manual fiddling, after an upgrade of this sort,
but it's usually quite manageable.
The Debian people seem to be very good about making sure that
stable-version-to-stable-version upgrades go smoothly... the
process isn't perfect (from what I've seen) but it's usually
quite close. The upgrade path is usually tested out quite well
before the release team throws The Big Switch, and there normally
are good release notes which describe the corner cases which may
need manual intervention.
I have several systems which have been through multiple major
Debian upgrades, without having to be slagged down and rebuilt
from the ground up. That's better than I ever achieved with (e.g.)
Red Hat, which (in my experience) really didn't take at all well to
in-place upgrades... I usually had to do a fresh install and then
port my personal files over.
Things may not be as smooth when jumping from Stable to Testing,
precisely because this isn't an official-release pathway, and
the packages in Testing are usually in somewhat of a state of
flux. Even upgrades *within* the Testing distribution can leave
you with a system which doesn't fly right... this isn't common but
it does happen. For example, a recent upgrade within Stable pulled
a bunch of the firmware files out of the kernel package and moved
them to a separate non-free package - if I hadn't noticed an error
message during RAMdisk rebuilt, my next boot would have left me
with a non-functioning wired Ethernet adapter.
If you decide to follow this route, follow the Debian instructions
for upgrading... back up your package configurations, and (I suggest)
everything in the /etc/ directory hierarchy, as well as all of your
personal files. This will give you a much better chance to invoke
the spirit of the ancient pagan god DoOver, if something goes wrong
during the upgrade.
Thanks Dave. Sounds like a man who's not had his hand soaking in ivory
liquid and been through the toils and tortures of various upgrades over the
years. Very insightful though. Goof thing this discussion ensued as I am
learning a lot about what to be wary of not least of all, the truth about
testing, RC and stable distribution. Which is why, despite eating humble
pie re: the RC vs Stable discussion, I was going to wait till the status on
RC changes to stable and maybe even help out a bit in the upgrade path
testing. Good thing is that I don't necessarily need to muck around with the
Production machines at the moment as all development is being done in the
Lab, and some of that is in VMs, so I have the power of snapshots with me
along with physical access to machines should anything break badly. The
production machines are sitting 10,000 miles away so the best I have is
console access to them.
Speaking of in-place upgrades, does adding the Squeeze repo. in the
sources.lst conf and running 'aptitude safe-upgrade/full-upgrade'
automaticaly begins the upgrade or is there more to it? You mentioned about
backing up configs and data etc so it doesn't sound like it's that simple
eh?
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Re: [asterisk-users] Unknow T callerid
I guess that was the CallerID transmitted by the calling channel. On Mon, Jan 24, 2011 at 7:31 PM, Jose Flores Galicia floj...@gmail.com wrote: Hi List. Have any of you guys ever see an incoming call throught Dahdi channel which has an callerid T. I know whenever is a private call, it shows as callerid, but what does it mean a T callerid? Best Regards -- Jose Flores Galicia floj...@gmail.com BriefCode Code Based Training -- _ -- 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] SIP, IAX2 and ISDN ISUP data
Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the voicemail DDI) - Originally Called Number (e.g. So if Adam phones Bob, Bob is diverted to Charlie, and Charlie is diverted to Voicemail, then Adam probably doesn't want Charlie's Voicemail). I believe this information should be in SIP Divert headers, can someone confirm this? Do I get the same information if I use an IAX2 connection to connect a local Asterisk server to an external one? Does IAX2 route GSM/ISDN SMS between servers, and if so, would the remote/ISDN connected server need to explicitly support this, or do the remote cards look local? Any help greatly appreciated, Phil -- _ -- 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 and Kamailio integration on cloud EC2 amazon no voice.
Hi All, i am stuck in NAT issue on ec2 cloud computing from last 2-3 days , may be some of you are doing setup and integration on cloud. below is my setup details which may help you to suggest me solution. Asterisk version : 1.6.2.6 1) Kamailio server having public_ip as well local ip .i am using mediaproxy [also tried rtpproxy] . 2) Asterisk server having public_ip as well local ip. setup: *UAC - KAMAILIO - ASTERISK* UAC registered to kamailio registration is successful. once it dial PSTN number i forwarded a call to asterisk server and then is created problem because i am not getting any media from asterisk server. so basically UAC sends a registered request to kamailio public ip and kamailio and asterisk works on private ip , it sends data to asterisk private ip, i am getting sip signaling and it looks okay. i can provide it too if we required. here is my asterisk sip.conf kamailio context looks like [vmserver] type=friend context=default host=***local_ip_of_kamailio*** ; for below three i have tried all available options *directmedia=nonat directrtpsetup=yes nat=yes * t1min=500 disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm qualify=yes let me know how to solve this nating issue also i opened all required ports for sip. and rtp regards Dhaval -- _ -- 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
