similar to: failed to extend from 512 to 676 message on console

Hello, We are using around 100 real time sip peers with phpagi. On asterisk cli, getting frequent message 'failed to extend from 512 to 676'. Once we execute 'sip reload', this message disappear for some time and then comes back. Please let us know the solution for this. asterisk version 1.6.2.9mysql 5.0server: Intel(R) Core(TM) i5-2500 CPU @ 3.30GHzRAM: 4 GB Thanks,Kamlesh

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Hello Asterisk version 1.6.2.9. I want to know is there any limitation on number of contexts or including external file (#include <filename>) which can be defined in extensions.conf. When I try to include around 40 external files, my dialplan doen't get reloaded. Regards, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Hi Folks, Mac OS 10.4.4 R 2.2.1(2005-12-20 r36812) Hmisc 3.0-10 acepack 1.3-2.2 I keep getting a "segmentation fault" when trying to run areg.boot in the Hmisc package. I include below output from two different attempts. Thank you in advance for any help. Hank Stevens The following is taken from the example in the areg.boot documentation, run inside Aquamacs Emacs: >

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Hi Folks, Mac OS 10.4.4 R 2.2.1(2005-12-20 r36812) Hmisc 3.0-10 acepack 1.3-2.2 I had R crashes while running areg.boot in Hmisc (see old message below), but now I realize that the problem appears to be in the avas function in acepack. I tried running running the avas example (in acepack package), and got an immediate crash. Any thoughts? The Apple crash report (from R GUI crash) follows.

Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: "xxxxxxxxxx" <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as23a29r59To: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060>Contact: <sip:xxxxxxxxxx at

I have a server with 6 gigs of memory. CentOS5/Xen sees only half of it. Could someone advice me how to get it recognize all my memory? I have installed 32bit CentOS5 and I'm running kernel-xen-2.6.18-8.1.8.el5. This kernel should have PAE support: $ grep PAE /boot/config-2.6.18-8.1.8.el5xen CONFIG_X86_PAE=y But xentop reports only 3 gigs of memory: Mem: 3145148k total, 2400512k used,

Hello, I'm using AGI scripting with asterisk and need to execute certain commands just after Dial(). But once dial command is executed, further commands/instructions are ignored. $agi->exec("Dial","SIP/100"); $dialstatus = $agi -> get_variable("DIALSTATUS"); if($dialstatus[data]=="ANSWER") { do something.......

helo I am running samba on an 800 Duron / 640MB PC2100 DDR Ram / 40 GB IBM Deskstar with slackware 8 / 2.2.17 Optimized for Duron. Whenever I copy files to the server, my load average gets to be about 6.0-8.0! On the 700 Duron we have at the office where I work, (same specs except it has 768MB of SDRAM and a RAID Array). Would the reason my box runs so slow be because its swap drive and the

Below is from make buildworld output on FreeBSD cvsup'd to RELENG_4 (so 4.8-STABLE) on the following hardware: Intel SHG2 Hodges Dual Xeon board 2 x Intel Xeon 2.4Ghz (512k) 2 x 512MB PC-2100 266Mhz ECC DDR Seagate 36GB 10K U320 LC SCSI Adaptec SCSI Raid 2000s 48MB SDRAM Intel Hudson 3 SC5200 base w 450W Any suggestions? Cheers, Carl. <snip> cc -o make_keys -O -pipe -I.

Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111214/b462516a/attachment.htm>

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Hello, voicemail password is not getting changed through phone handset while IVR indicates that password has been changed. During google I found that uniqueid column must not be changed so it is not changed. Please guide on this. During debug log I found below but in mysql db new password is not getting updated, [Jun 15 13:54:07] VERBOSE[6418] file.c: -- <SIP/123-00000005> Playing

Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456 at default:1] AGI("SIP/100-00000000", "call.php") in new stack -- Launched AGI Script

https://bugs.freedesktop.org/show_bug.cgi?id=63101 Priority: medium Bug ID: 63101 Assignee: nouveau at lists.freedesktop.org Summary: [Ubuntu 10.04.4 LTS 32-bit] NVIDIA GeForce 7300 GT AGP graphics card will not display any text characters on VIA Technologies Apollo MVP3-based mainboard QA Contact:

Hello, Asterisk version 11.13.1 I'm trying use realtime meetme application with c and q option. If both options are used together in meetme table under 'opts' field, c option (Announce user(s) count on joining a conference.) doesn't work i.e. system doesn't play user counting to caller. Is it bug or some configuration problem. Thanks, Kamlesh --------------

I have a Tyan Tiger S2466 MPX motherboard with Dual Atlon MP 2800+ CPUs and 1GB PC2100 DDR SDRAM. For disk drive I have an LSI53C1030 and 4 Seagate ST336607LWs in a software raid 5 configuration. I installed Centos 4.1 and everything was fine running kernel-smp-2.6.9-11.EL. However when I upated to Centos 4.2 I have run into problems. Namely after a finite amount of disk traffic the system

Hello, Running one asterisk server with below details. Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and we start getting problem in call quality like delay in connectivity, voice breakage etc.... Hardware: 2 Physical processor Intel(R) Xeon(R) CPU