[asterisk-users] Re: notransfer local channel on redirect
2006/9/21, Benko [EMAIL PROTECTED]: notransfer-option(\n) on redirected calls? sorry, it is called no release quote:(the n stands for no release) so is there a way to tell asterisk to not release a local channel on a redirect so the billsec and duration is written to it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr records on transfer
On Mon, 6 Mar 2006 18:53:30 +0100 (CET) Christian Benke [EMAIL PROTECTED] wrote: Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the RDNIS-variable i've been able to set seperate records for each call-leg with the correct accountcodes, but the billsec are still written to the first callleg, the second callleg(originated by callee) receives 0 billsec, which is not what i want. the callee(the one who forwards the call), should be billed. since the local-channel is passed to the originating channel, it is clear that the billsec are added to the callers record. but is there any way to influence this??? since the phones have this functionality built-in, why should i ask my clients to use some keycombination to transfer calls and prevent transfer-by-button? As far as i've understood, the /n-option for the local-channel would do the behaviour i want - but how could i add it on a moved temporarily? kind regards christian can i assume this is a known problem? Can anyone at least confirm it? Or is my report unclear? I really appreciate any comments, this is a huge problem for me as my whole concept depends on it! Thanks!!! Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr records on transfer
Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the RDNIS-variable i've been able to set seperate records for each call-leg with the correct accountcodes, but the billsec are still written to the first callleg, the second callleg(originated by callee) receives 0 billsec, which is not what i want. the callee(the one who forwards the call), should be billed. since the local-channel is passed to the originating channel, it is clear that the billsec are added to the callers record. but is there any way to influence this??? since the phones have this functionality built-in, why should i ask my clients to use some keycombination to transfer calls and prevent transfer-by-button? As far as i've understood, the /n-option for the local-channel would do the behaviour i want - but how could i add it on a moved temporarily? kind regards christian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cmd set with multiple values
hello!
has this made it into 1.2.3 already:
http://bugs.digium.com/view.php?id=6128 ?
i'm trying to set a variable that should be used as a dialstring in the
dial-command, including parameters seperated with the respective
delimiter, e.g. like:
exten = 907,1,Set(DESTINATION1=Zap/G1/4989123456789|10|gh)
exten = 907,n,Set(DIALSTRING=${DESTINATION1})
exten = 907,n,Dial(${DIALSTRING})
asterisk complains:
Feb 3 12:39:40 WARNING[26200]: pbx.c:6010 pbx_builtin_setvar: Ignoring
entry '10' with no = (and not last 'options' entry)
i've tried several of the resolution-proposals mentioned in the
bugnotices, but none of them seems to work yet.
the best fit was exten =
907,1,Set(DESTINATION1='Zap/G1/4989123456789,10,gh') but then the value
included in the quotes seems to be set as a string that is not parsed when
dialing ${DIALSTRING}, resulting in
Called G1/4989123456789,10,gh
is there any workaround?
thanks
christian
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[Asterisk-Users] instant fallback to zap in case of sip/iax/xyz-failure
i would like to carry some oversea pstn-destinations via sip to providers like stanaphone, however, in case of a network-failure or if the provider is not available, i want to fallback to the zap-channels so the call is carried out to the pstn directly. the usual approach would be to check the dialstatus(e.g.NOANSWER). however, asterisk tries 60seconds to reach that peer(even when the ip i'm sending the call too is a dead end(no host)). i could limit a call by setting a timeout but this limit would also apply if a final destination doesn't pick up within the timeout. so basically, when i send a call via a sip-channel, i would like to know the network-status of the foreign host immediately(at least within 5 seconds) so i can reroute the call without having to wait for a host that is probably dead... this seems to be possible with iax and CHANUNAVAIL, (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=3360history=1), though i haven't tried it. also i _need_ to use sip, iax (currently) is not an option. is there any mechanism in asterisk that allows to get the vital sip-status of a foreign host?! thanks for your input!!! ;-) regards christian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agi debug - unable to set normal priority
Hello! In my agi-debug i get the following error-message: AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Unable to set normal priority AGI Tx 510 Invalid or unknown command AGI Rx SET VARIABLE MODCLI 00434345452 the agi i call is a very simple shellscript that simply removes wrong charakters: #!/bin/bash modcli=`echo $1 | sed -e 's/#//g' -e 's/*//g'` #echo $modcli echo SET VARIABLE MODCLI $modcli the script works as expect, sending the modified variable back to asterisk... anyone knows what this error-message means? regards christian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] avoided deadlock/channel already in use
Hello! After upgrading my production machine to 1.2.1(used to be 1.2.0) on friday i experienced strange behaviour yesterday, i received deadlock-avoided-messages and channels refusing to hangup on span1(used for inbound calls), both messages in all cases paired: Jan 9 17:40:01 WARNING[30003] chan_zap.c: Ring requested on channel 0/17 already in use on span 1. Hanging up owner. Jan 9 17:40:01 WARNING[21571] channel.c: Avoided deadlock for '0xb6e89798', 10 retries! first they appeared twice at 17:40 and once at 18:07 but at 19:30 span 1 refused to accept any call, spitting out the ring requested-message every second for different channels on span1(about 60 times) and 4 avoided deadlock messages, until i restarted asterisk 1 minute later. not only that this is very annoying and severe, also most of the concurrent calls are hungup when that happens(drop from 20 calls to 2). Both error-messages(ring requested, Avoided deadlock) didn't yield much on google, 2 recommendations to ask the tech [EMAIL PROTECTED] i use a digium te410p but i guess the tech support will ask me to send them a debugging trace - as this is a production machine(which used to work with the same configuration for at least 2 months without much fuss) i can't recompile asterisk with debugging support atm. When doing ps faux|grep asterisk i see daughter processes of asterisk - this didn't appear on installations prior to 1.2.1 and my second machine has only one process running as it used to be(same version 1.2.1), also the times when the daughter processes were started fit the times when the problems above happend, though there are other daughter processes started at times were no problems occured, some were started on 6.January(the day of the upgrade). i rebooted the machine yesterday in the night(since 19:40 no more problems occured), but there are some daughter process again: # ps fauxw|grep asterisk root 5882 0.0 0.0 2344 1100 ?S00:53 0:00 /bin/sh /usr/sbin/monit_asterisk asterisk 5891 1.8 0.9 35228 19744 ?Sl 00:53 16:43 \_ /usr/sbin/asterisk -U asterisk -G asterisk -p asterisk 5940 0.0 0.2 5608 4376 ?S00:54 0:02 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 asterisk 5944 0.0 0.0 3712 412 ?S00:54 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 asterisk 5941 0.0 0.0 4328 1836 ?S00:54 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 tvm128.mp3 asterisk 5943 0.0 0.0 3712 480 ?S00:54 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 tvm128.mp3 asterisk 5942 0.0 0.2 11848 5820 ?S00:54 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 06_-_Massive_Attack_-_Prayer_for_england.mp3 asterisk 5963 0.0 0.0 3712 424 ?S00:54 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 06_-_Massive_Attack_-_Prayer_for_england.mp3 asterisk 11894 0.0 0.5 30356 11676 ?S12:41 0:00 \_ /usr/sbin/asterisk -U asterisk -G asterisk -p asterisk 13179 0.0 0.5 31012 12272 ?S12:49 0:00 \_ /usr/sbin/asterisk -U asterisk -G asterisk -p asterisk 13657 0.0 0.5 31012 12272 ?S12:51 0:00 \_ /usr/sbin/asterisk -U asterisk -G asterisk -p asterisk 2394 0.0 0.7 35048 15144 ?S15:53 0:00 \_ /usr/sbin/asterisk -U asterisk -G asterisk -p root 21099 0.0 0.0 4104 1592 pts/0S+ 13:55 0:01 | \_ rasterisk r root 24101 0.0 0.0 4104 1592 pts/1S+ 14:20 0:01 | \_ rasterisk r root 6009 0.0 0.0 1500 476 pts/2S+ 16:18 0:00 | \_ grep asterisk root 29100 0.0 0.0 4104 1576 pts/3S+ 14:58 0:00 \_ rasterisk r asthost log # are daughter processes a new behaviour? there are none on my other box, which is identical in hard- and software(a harddisk-clone) despite that it has a sangoma-card... on my box(intel xeon dual se7520jr2) the raid-controller and the te410p share irq11 - but since there were no problems at all since 6 months i assumed that it works. hope this is still not related! i send a lot of calls via sip to the other machine(first machine inbound, second machine outbound) and yesterday i had 2 avoided deadlock messages on the second machine as well(but no corresponding ringing-message as on the first machine), the first one was 80minutes before the first messages on the first machine, the second one was at the same time at 18:07 when i had two pairs(avoided deadlockringing) on the first machine... my final assumption would be that this is related to my telco, some pri-problem on their side maybe... i can't reproduce the problem and yesterday there was not more traffic on the machine than usual, it is running fine again since 15h... i hope my report is clear enough,
[Asterisk-Users] Virtual Memory Usage
Good Morning List! I have a instance of asterisk running since 2 weeks under relative heavy zap usage(mostly disa-customers), about 3000calls/day. last time(2 weeks ago) it had been shut down by oom-killer for some reason and since then i keep a jealous watch over the asterisk process. I know that linux is keeping old pages in the cache and erases them when free memory is needed, though i'm concerned about the virtual memory usage of asterisk. it is rising every day and is at 56488kB at the moment(it never declines) with a processor time of 192:19.77. Is this normal behaviour and will the virtual memory be erased at some point when there really is insufficient memory? Or should i be concerned about it? The free memory has always been between 50 and 70 MB in this two weeks (cache is at 1969MB so enough spare memory yet). I also suspect my two agi-bash-scripts for producing the memory usage, i'm not sure if the results of the scripts are kept in (virt)memory, though i'm not a specialist how linux manages this and would like to hear a professional opinion about it(see attachments for my - simple - bash-scripts that are called for every call, authentication and dialstring cosmetics, the list-usergroup1 that is beeing called is a simple list of numbers) In the meantime i've installed monit(tildeslash.com/monit/) which is a quite nice tool for monitoring and restarting of processes in case of failure, though i'd like to find the real problem, not just fight the consequences... disa-usergroup1.sh Description: Binary data disacut.sh Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
