login | register
Wed 01 of Aug, 2007 [09:14 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.44s
  • Memory usage: 2.24MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 3.72

Asterisk timer

Zaptel timers for Asterisk


There are at least two Asterisk applications that need support of a timer to work properly:

It may also be required with Music on Hold, i.e. to improve sound quality.

For Linux, several solutions exist to provide a timer, for other operating systems there is nothing, yet.

How to get a working timer

  • A Digium ZAPTEL INTERFACE has working timers.
  • If you don't have Digium hardware, there are three replacements:

    • ztdummy (in the standard zaptel distribution on the Asterisk CVS) on 2.4 Linux kernel uses USB-UHCI timers in USB drivers on platforms with UHCI USB support. On kernel version 2.6 it uses internal high-resolution kernel timer and do not require any additional hardware. Ztdummy is a kernel module that you load with the Linux command modprobe. Read your Linux documentation on how to load and unload kernel modules. If you are using 2.4 kernel please note that usb-uhci must be loaded as a module and may not be compiled into the kernel for ztdummy to work.

Zaprtc works fine on SMP wth kernel 2.6.

Note: Zaprtc is actually a replacement for the standard RTC module. It provides the same facilities, but includes extra parts for Zaptel use. You will need to unload standard RTC module (rmmod rtc) or re-compile the kernel without RTC support (in your kernel source dir: "make menuconfig" --> Character Devices --> uncheck Enhanced Real Time Clock; now re-compile the kernel) in order to be able to use zaprtc.


Note

  • If you are on a 2.6 kernel, be sure to read /usr/src/zaptel/README.udev

For FreeBSD


For OpenWRT



Created by oej, Last modification by Hans Zandbelt on Tue 21 of Nov, 2006 [21:23 UTC]

Comments Filter

by Mat on Friday 09 of December, 2005 [19:50:35 UTC]
Why a kernel module is needed ?
Why can't we use /dev/rtc ?

For imformation mplayer use /dev/rtc @1024 if it can from years.

PS : I am not sure POSIX timer could provide a such resolution.

Re: Clarification usage on ztdummy

by Real name on Monday 17 of October, 2005 [03:25:45 UTC]
Woah guys... this is a very bad joke. Besides being barely documented...

Linux provides many types of real-time timers. You don't need to maintain a crappy little non-standard and non-portable driver. The asterisk server should be using POSIX timers.

This is just oh-my-god bad. What a hack!

Error message - is this caused by a timing problem?

by alakon on Friday 11 of March, 2005 [23:23:18 UTC]
I get the following error - is this caused by a timing problem (some diigts change to protect the innocent)?

   — Accepting AUTHENTICATED call from 217.160.244.186, requested format = 4, actual format = 4
   — Executing Answer("IAX2/livevoip@217.160.244.186:4569/3", "") in new stack
   — Executing Wait("IAX2/livevoip@217.160.244.186:4569/3", "2") in new stack
   — Executing AGI("IAX2/livevoip@217.160.244.186:4569/3", "areskicc.php") in new stack
   — Launched AGI Script /var/lib/asterisk/agi-bin/areskicc.php
 areskicc.php: 'agi_request' => 'areskicc.php'
 areskicc.php: 'agi_channel' => 'IAX2/livevoip@217.160.244.186:4569/3'
 areskicc.php: 'agi_language' => 'en'
 areskicc.php: 'agi_type' => 'IAX2'
 areskicc.php: 'agi_uniqueid' => '1110514027.0'
 areskicc.php: 'agi_callerid' => '"8005550000" '
 areskicc.php: 'agi_dnid' => 'unknown'
 areskicc.php: 'agi_rdnis' => 'unknown'
 areskicc.php: 'agi_context' => 'livevoipinbound'
 areskicc.php: 'agi_extension' => '8005550001'
 areskicc.php: 'agi_priority' => '3'
 areskicc.php: 'agi_enhanced' => '0.0'
 areskicc.php: 'agi_accountcode' => ''
 areskicc.php: 'dig' => '/usr/bin/dig'
 areskicc.php: 'debug' => 'true'
 areskicc.php: >> ANSWER
 areskicc.php: string(82) ""8005550000"  ; IAX2/livevoip@217.160.244.186:4569/3 ; 1110514027.0 ; "n
 areskicc.php: string(26) "Requesting DTMF ::> Len-10"n
 areskicc.php: >> GET DATA prepaid-enter-pin-number 10000 10
   — Playing 'prepaid-enter-pin-number' (language 'en')
Mar 10 23:07:09 NOTICE954: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!
Mar 10 23:07:09 WARNING962: file.c:1058 ast_waitstream_full: Wait failed (No such file or directory)
 == Spawn extension (livevoipinbound, 8005550001, 3) exited non-zero on 'IAX2/livevoip@217.160.244.186:4569/3'
   — Hungup 'IAX2/livevoip@217.160.244.186:4569/3'

Edit

AutoStart Kernel 2.6

by Anonymous on Thursday 06 of January, 2005 [01:43:13 UTC]
What I need to do in order to start the ztdummy with kernel 2.6?

Re: Clarification usage on ztdummy

by PoWeRKILL on Thursday 07 of October, 2004 [18:54:08 UTC]
The following is not true, if Conference is not used you don't have to use any ZaptelRTC or ZTDUMMY.
Edit

Clarification usage on ztdummy

by Anonymous on Saturday 10 of July, 2004 [03:07:42 UTC]
It really really needs to be noted here that if you're using a pure VoIP setup (e.g. No digium hardware at all) you MUST have either ztdummy or zaprtc loaded for ANYTHING that requires asterisk to stream audio and not just for MeetME and IAX. If you do not load it then anything that asterisk streams over the internet, VoiceMail especially, will sound like CRAP...
Edit

more about ztdummy

by Anonymous on Tuesday 23 of March, 2004 [17:55:37 UTC]
it doesn't require/use /etc/zapata.conf
after successfully modprobing ztdummy, you should now have /proc/zaptel
upon asterisk startup in the logs you might see:
Mar 23 11:27:36 WARNING16384: Unable to open IAX timing interface: Permission denied
this refers to /dev/zap/pseudo. make it read/writable by the process running asterisk.
The "Unable to load config iax1.conf" message in the log file is unrelated, as are the "ignoring port for now", and "ignoring rxwink". You can rid yourself of these warnings by copying iax.conf into iax1.conf.

ztdummy

by dg1nsw on Monday 02 of February, 2004 [15:07:21 UTC]
I had UHCI on my USB so i gave it a quick try.
I used this ztdummy-module and got scratched up sound from asterisk. (worked before)
With zaprtc it worked better however for all hardcore linuxers: You HAVE to execute the rtcsetup. I am not completely clear what it does but if you dont execute it asterisk will not respond. (asterisks scheduler will have no timing instead of a bad one :)
And if you get messages about missing symbols when loading one of the kernel modules you should do "lsmod" and look if the module is loaded. You find it in your sourcedirectory after compiling called "zaptel.o".

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver