Re: [asterisk-users] Asterisk and QoS
For ingress - yes, but not quite correct. No you cant directly control the QoS on someone elses interface, but you can do something none the less. There is a queue on the interface facing you (and if an ISP is quite likely been made very large) - then when you get a lot of packets coming in such as when you have an ftp session going it will queue everything. The trick is to force the other end to keep a minimal queue buffer by using a "police" filter. It drops packets to keep the buffer to a small size thus helping latency and minimising the effects a large buffer has on your traffic. The queue is to help the other end manage their flows - not yours! - so policing helps you! Tricky, but it appears to work. BillK On Fri, 2010-07-30 at 12:29 -0600, Tim Densmore wrote: > There's no real way of shaping or applying QoS on inbound interfaces > on any device. You can affect how that traffic behaves once it's > entered your device, but not how it's queued on its way to that > device. Think of lit like trying to stanch the flow of water at the > end of a hose rather than simply turning the pressure down at the > spigot. To properly queue, it has to be done on egress, so you'd be > better off looking at applying QoS to whatever moves traffic to your > astersk box if "input" traffic on the asterisk box is the issue. You > can, of course, effectively setup queuing outbound return traffic > *from* the asterisk box. > -- _ -- 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] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Well for the best test you can call in on that line and fire Echo() app and then you'll see if the lines "hangup by themselves" ... is you use fxsks/fxs_ks signaling and it's supported by your lines then it's that that makes remote hangup possible regards Martin On Fri, Jul 30, 2010 at 9:12 AM, bruce bruce wrote: > Thank Martin, > That makes absolute sense. I have turned busy detect off for now and haven't > heard complains or lines remaining open for a Day. I am in Canada. I just > checked chan_dahdi.conf and I don't see callprogress there at all. So, I > guess the lines are fine for hanging up by themselves. Hope this doesn't > give me probs in future. > Thanks, > Bruce > On Fri, Jul 30, 2010 at 6:18 AM, Martin wrote: >> >> Either turn off busydetect or increase the busycount to 5-7 or even >> more ... (10 should be conservative) >> busydetect looks for cadence or patterns of the same length ... beep >> on [X ms] beep off [Y ms] >> so you can afford to increase busycount and have a few second longer >> calls / the line is kept longer offhook >> but you don't get false busy detections >> >> Also in US/Canada callprogress will do a better job then busydetect >> since it looks for specific frequencies of the busy signal >> and not just noise/beep then silence ... If you're somewhere else then >> you can hire a coder to tweak callprogress algorithm >> to your country's busy signal frequencies ... Just record the busy >> signal with ztmonitor and send to someone for code patch... >> >> regards >> Martin >> >> On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce wrote: >> > Hmmwhat about call waiting? >> > You mean, when a call comes in on that specific line, it generate two >> > beep >> > tones and hence the system hangs up thinking it's end of the call? >> > Interesting!!! >> > If it is call-waiting do I have to set all of the following off for it >> > to >> > not give me problem again: >> > callwaiting=yes >> > usecallingpres=yes >> > callwaitingcallerid=yes >> > threewaycalling=yes >> > transfer=yes >> > canpark=yes >> > cancallforward=yes >> > busydetect=yes >> > busycount=3 >> > Please elaborate a bit if I am off-topic. >> > Regards, >> > Bruce >> > On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas >> > wrote: >> >> >> >> From: asterisk-users-boun...@lists.digium.com >> >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce >> >> bruce >> >> Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden? >> >> Couldbusy detect be the problem? >> >> >> >> >> >> >> >> I am getting a complain that call on analogue lines (Sangoam A400D) >> >> drops >> >> all of a sudden. Here is what I see in logs: >> >> >> >> >> >> >> >> Could be callwaiting? >> >> >> >> -- >> >> _ >> >> -- 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 >> > >> >> -- >> _ >> -- 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 > -- _ -- 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] agi macro problem
> On Fri, 30 Jul 2010, Zarko Zivanovic wrote: > >> I need simple whopicked.agi (instead of .rb) which will simply take the >> value 30086 (that I pass to macro) On Fri, 30 Jul 2010, Steve Edwards wrote: > While ".rb" suggests a "Ruby source file," ".agi" suggests nothing. On Fri, 30 Jul 2010, Danny Nicholas wrote: > I'm sure this is "not a good practice", but I actually use the .agi > suffix on my AGI's or AGI launch stubs to "keep it straight" that even > though the actual program is bash or PERL, that the function of the > program is to generally be launched by the Asterisk AGI. That way, if > the program gets moved from /var/lib/asterisk/agi-bin, I know it's not > just a run-of-the-mill bash or Perl script. Since the #! Line tells you > what it actually is, it is (IMO) a "no harm" proposition. I don't have an issue with the practice. The phrasing of the OP implied [s]he thought that .agi meant something "instead of" .rb. Naming your Ruby script .agi would be perfectly acceptable to Asterisk, the shell, and Ruby. -- 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] app_swift.c:338 engine: Failed to set voice
On 07/28/2010 08:20 PM, Landy Landy wrote: > Jeremy, > > Thanks a lot that helped and solved the problem. I had it as: > voice=Marta-8kHz before and that didn't work and now changed it to > voice=Marta. That's because you only have the Marta-16kHz voice installed. -- 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
[asterisk-users] Aastra ignore call button hangs up call instead of going to voicemail
I have a Asterisk server (PBX in a Flash) with Aastra 57i phones. When
there is an incoming call the phone will display two buttons "answer"
and "ignore". If you press "ignore" the call is dropped instead of sent
to voice mail. The following is the log:
-- Called 111
-- SIP/111-1c14 is ringing
-- Got SIP response 486 "Busy Here" back from 192.168.3.126
-- SIP/111-1c14 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [...@macro-dial:8] Set("DAHDI/10-1", "DIALSTATUS=BUSY") in
new stack
-- Executing [...@macro-dial:9] GosubIf("DAHDI/10-1", "1?BUSY,1") in new
stack
== Spawn extension (macro-dial, s, 10) exited non-zero on 'DAHDI/10-1'
in macro 'dial'
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on
'DAHDI/10-1' in macro 'exten-vm'
== Spawn extension (from-did-direct, 111, 1) exited non-zero on
'DAHDI/10-1'
-- Hungup 'DAHDI/10-1'
The extensions.conf file has this macro-dial in it:
; Rings one or more extensions. Handles things like call forwarding and
DND
; We don't call dial directly for anything internal anymore.
; ARGS: $TIMER, $OPTIONS, $EXT1, $EXT2, $EXT3, ...
; Use a Macro call such as the following:
; Macro(dial,$DIAL_TIMER,$DIAL_OPTIONS,$EXT1,$EXT2,$EXT3,...)
[macro-dial]
exten => s,1,GotoIf($["${MOHCLASS}" = ""]?dial)
exten => s,n,SetMusicOnHold(${MOHCLASS})
exten => s,n(dial),AGI(dialparties.agi)
exten => s,n,NoOp(Returned from dialparties with no extensions to call
and DIALSTATUS: ${DIALSTATUS})
exten => s,n+2(normdial),Dial(${ds}) ;
dialparties will set the priority to 10 if $ds is not null
exten => s,n,Set(DIALSTATUS=${IF($["${DIALSTATUS_CW}"!
="" ]?${DIALSTATUS_CW}:${DIALSTATUS})})
exten => s,n,GosubIf($["${SCREEN}" != ""]?${DIALSTATUS},1)
exten => s,20(huntdial),NoOp(Returned from dialparties with hunt groups
to dial )
exten => s,n,Set(HuntLoop=0)
exten => s,n(a22),GotoIf($[${HuntMembers} >= 1]?a30) ; if this is from
rg-group, don't strip prefix
exten => s,n,NoOp(Returning there are no members left in the hunt group
to ring)
; dialparties.agi has setup the dialstring for each hunt member in a
variable labeled HuntMember0, HuntMember1 etc for each iteration
; and The total number in HuntMembers. So for each iteration, we will
update the CALLTRACE Data.
;
exten => s,n+2(a30),Set(HuntMember=HuntMember${HuntLoop})
exten => s,n,GotoIf($[$["${CALLTRACE_HUNT}" != "" ] &
$[$["${RingGroupMethod}" = "hunt" ] | $["${RingGroupMethod}" =
"firstavailable"] | $["${RingGroupMethod}" =
"firstnotonphone"]]]?a32:a35)
exten => s,n(a32),Set(CT_EXTEN=${CUT(FILTERED_DIAL,,$[${HuntLoop} +
1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Goto(s,a42)
;Set Call Trace for each hunt member we are going to call "Memory groups
have multiple members to set CALL TRACE For" hence the loop
;
exten => s,n(a35),GotoIf($[$["${CALLTRACE_HUNT}" != "" ] &
$["${RingGroupMethod}" = "memoryhunt" ]]?a36:a50)
exten => s,n(a36),Set(CTLoop=0)
exten => s,n(a37),GotoIf($[${CTLoop} > ${HuntLoop}]?a42) ; if this is
from rg-group, don't strip prefix
exten => s,n,Set(CT_EXTEN=${CUT(FILTERED_DIAL,,$[${CTLoop} + 1])})
exten => s,n,Set(DB(CALLTRACE/${CT_EXTEN})=${CALLTRACE_HUNT})
exten => s,n,Set(CTLoop=$[1 + ${CTLoop}])
exten => s,n,Goto(s,a37)
exten => s,n(a42),Dial(${${HuntMember}}${ds})
exten => s,n,Set(HuntLoop=$[1 + ${HuntLoop}])
exten => s,n,GotoIf($[$[$["foo${RingGroupMethod}" !=
"foofirstavailable"] & $["foo${RingGroupMethod}" !=
"foofirstnotonphone"]] | $["foo${DialStatus}" = "fooBUSY"]]?a46)
exten => s,n,Set(HuntMembers=0)
exten => s,n(a46),Set(HuntMembers=$[${HuntMembers} - 1])
exten => s,n,Goto(s,a22)
exten => s,n(a50),DBdel(CALLTRACE/${CT_EXTEN})
exten => s,n,Goto(s,a42)
; For call screening
exten => NOANSWER,1,Macro(vm,${SCREEN_EXTEN},BUSY,${IVR_RETVM})
exten => NOANSWER,n,GotoIf($["${IVR_RETVM}" != "RETURN" |
"${IVR_CONTEXT}" = ""]?bye)
exten => NOANSWER,n,Return
exten => NOANSWER,n(bye),Macro(hangupcall)
exten => TORTURE,1,Goto(app-blackhole,musiconhold,1)
exten => TORTURE,n,Macro(hangupcall)
exten => DONTCALL,1,Answer
exten => DONTCALL,n,Wait(1)
exten => DONTCALL,n,Zapateller()
exten => DONTCALL,n,Playback(ss-noservice)
exten => DONTCALL,n,Macro(hangupcall)
; make sure hungup calls go here so that proper cleanup occurs from call
confirmed calls and the like
;
exten => h,1,Macro(hangupcall)
Which unfortunately doesn't make much sense to me. I do see a
macro-exten-vm with a comment that it is where the call should be routed
is the extension is busy or doesn't answer. But I'm not sure how to
modify the macro-dial to make it happen.
I appreciate any help that anyone can give thanks in advance,
Jeremy
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Re: [asterisk-users] VUC Friday: Twilio OpenVBX
On Fri, Jul 30, 2010 at 11:54 AM, Alex Bell wrote: > /r, > r u not on talkshoe anymore? This is 2 weeks in a row that I've clicked > in, but no one was home? At least last week I was one of 2 guests, today I > was all by my lonesome... :( Hi Alex, When I'm not in my own place, I can't bridge the call, so we are not on Talkshoe. You can use the widget on http://vuc.me, skype:vuc.me or sip:200...@login.zipdx.com when the conference bridge is up. The next scheduled conference is found here: http://vuc.me/next Today's is done. In part two, an interesting discussion about the Do Not Call LIst happened and I will post a recording of that as well, perhaps tomorrow. /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] VUC Friday: Twilio OpenVBX
/r, r u not on talkshoe anymore? This is 2 weeks in a row that I've clicked in, but no one was home? At least last week I was one of 2 guests, today I was all by my lonesome... :( /al On Thu, Jul 29, 2010 at 8:42 PM, Randy R wrote: > Interesting offering, free from Twilio, this is php you install on > your own server to build a brandable "VBX". Worth checking out! > Listen to tomorrow for more about this and talk to lead engineer or > Twilio CEO if you have any questions; > > sip:200...@login.zipdx.com or Skype:vuc.me > > IRC: #vuc on Freenode.net or http://vuc.me/irc > > Info about VUC is htp://vuc.me > > Best, > > /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 > -- _ -- 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 and QoS
There's no real way of shaping or applying QoS on inbound interfaces on any device. You can affect how that traffic behaves once it's entered your device, but not how it's queued on its way to that device. Think of lit like trying to stanch the flow of water at the end of a hose rather than simply turning the pressure down at the spigot. To properly queue, it has to be done on egress, so you'd be better off looking at applying QoS to whatever moves traffic to your astersk box if "input" traffic on the asterisk box is the issue. You can, of course, effectively setup queuing outbound return traffic *from* the asterisk box. On 07/30/2010 11:37 AM, Jonas Kellens wrote: My problem is that my Asterisk server is sometimes also FTP-server for uploading of MoH-files. I don't want this FTP-traffic to interfere with ongoing VoIP-calls. Therefore I would like to give priority to the RTP-traffic. I read that there is not really a way of shaping incoming traffic on Linux (ingress). Anyone on this list know how to deal with other packets coming in on the same interface ?! I have a gigabit link on a gigabit network... but don't know if this is enough. Kind regards, Jonas. On 07/30/2010 03:36 PM, William Kenworthy wrote: HTB is a bad choice for VoIP. When it "borrows" bandwidth, according to the docs it doesnt release it back until its finished so if its using all the bandwidth for a download before the VoIP call starts, VoIP gets starved even if you reserve an excess of bandwidth as it still queues. When I tried it sort of worked but didnt have the effect I expected on a busy link, probably for this reason. HTB tries to be fair about sending packets but with VoIP being fair sucks :) A better way is to use a "prio" filter at the root, the priority 1 branch having a plain fifo on it - send VoIP and acks only this way. The priority 2 branch has a HTB hierarchy with sfq leaves for the rest of the traffic. This seems to work much better, but I have not tested well yet. I am also using a police filter for incoming (on ADSL) and have not noticed any problems - but it is only lightly limiting (to try and keep the queues at the ISP end short.) Lastly, test to make sure the packets are flowing where you expect them to - I had to correct a few miss-understandings I had on how it all worked before everything went where I wanted it to :) TC and TCNG do seem dead, but I think thats partly because its relatively mature and doesnt need much work. BillK -- _ -- 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 and QoS
The Astlinux project has been using the HTB queue and a shaper based on Wondershaper for several years. Recently, we ported the work to Arno's Firewall as a plugin. That work would make it usable on generic Linux distribution. To be effective, you need to have traffic classified properly by the applications. You can find Arno's iptables firewall on the following page: http://rocky.eld.leidenuniv.nl/joomla/ Darrick On 07/30/2010 03:06 AM, Jonas Kellens wrote: > Hello list, > > anyone here using Asterisk together with HTB for queing incoming and > outgoing packets ? > > I've tried to subscribe myself to the Mailinglist of the Linux Advanced > Routing & Traffic Control project, but I get no confirmation. This list > seems dead. > > It seems my test case with HTB is not giving any noticeable results. Can > I ask questions on this mailinglist ? > > Perhaps you can give my other QoS-implementations like MasterShaper, if > it works well together with a firewall that uses iptables. > > > > Kind regards, > > Jonas. > -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and QoS
My problem is that my Asterisk server is sometimes also FTP-server for uploading of MoH-files. I don't want this FTP-traffic to interfere with ongoing VoIP-calls. Therefore I would like to give priority to the RTP-traffic. I read that there is not really a way of shaping incoming traffic on Linux (ingress). Anyone on this list know how to deal with other packets coming in on the same interface ?! I have a gigabit link on a gigabit network... but don't know if this is enough. Kind regards, Jonas. On 07/30/2010 03:36 PM, William Kenworthy wrote: HTB is a bad choice for VoIP. When it "borrows" bandwidth, according to the docs it doesnt release it back until its finished so if its using all the bandwidth for a download before the VoIP call starts, VoIP gets starved even if you reserve an excess of bandwidth as it still queues. When I tried it sort of worked but didnt have the effect I expected on a busy link, probably for this reason. HTB tries to be fair about sending packets but with VoIP being fair sucks :) A better way is to use a "prio" filter at the root, the priority 1 branch having a plain fifo on it - send VoIP and acks only this way. The priority 2 branch has a HTB hierarchy with sfq leaves for the rest of the traffic. This seems to work much better, but I have not tested well yet. I am also using a police filter for incoming (on ADSL) and have not noticed any problems - but it is only lightly limiting (to try and keep the queues at the ISP end short.) Lastly, test to make sure the packets are flowing where you expect them to - I had to correct a few miss-understandings I had on how it all worked before everything went where I wanted it to :) TC and TCNG do seem dead, but I think thats partly because its relatively mature and doesnt need much work. BillK -- _ -- 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] Please test: STUN patch for Asterisk behind NAT
Hi there! David has put up a patch to fix the STUN issues that has plagued Asterisk 1.6 ever since that feature was introduced. Now we need testers to verify the patch, so please grab the patch (or check out the SVN branch) and add your comments: https://issues.asterisk.org/view.php?id=17622 Thanks, Philipp -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
Tnx, now it's working fine. :) On Fri, Jul 30, 2010 at 5:16 PM, Sean Bright wrote: > On 7/26/2010 4:05 AM, Andraž wrote: > > I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from > > sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, > > freetds-bin, but, when I run configure and then make menuconfig in > > section "Call Detail Recording" -> "cdr_tds" it's "disabled". > The packaged version of FreeTDS on 10.04 is 0.82 which is too recent for > the Asterisk 1.4. cdr_tds module. The latest version of FreeTDS > supported by Asterisk 1.4 is 0.64. So you will have to compile from > source (or find an ancient package) to get this working. > > -- > _ > -- 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 Crashes - Segmentation Fault
Manmohan wrote: > There was on very silly mistake and i missed to check that properly. Really > apologize for that. > Following change was done to get the conf-recording into the proper path: > chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings > following is the output: > [r...@linuxtest sounds]# ll > total 6416 > drwxrwxr-x 2 asterisk asterisk 4096 Jul 30 08:29 conf-recordings > [r...@linuxtest sounds]# ll conf-recordings/ > total 4060 > -rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27 > meetme-conf-rec-74438-1280463795.8.wav > The only thing now is no speaker icon onto the webpage when i click to past > conference link. The web interface cannot find the recording. The reason it cannot is that the name is wrong. By wrong, I mean it contains information that the database and program is not aware of (1280463795.8). To make this clear, if this conference was the 3rd one you ever scheduled on this system the correct file name would be- meetme-conf-rec-74438-3.wav using the format meetme-conf-rec-%PIN%-%BOOKID%.wav The database knows the pin and bookid, so it can construct the file name and test if it exists. > Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of getting > speaker icon in past conference)? I was not able to get the change into app_meetme to use the bookid in the filename, even though it has access to bookid. I gave up and now store the filename in the database, which app_meetme will use if it exists. Other that a handful of bug-fixes, this is the major difference between 4.0.1 and 4.0.2 Dan -- _ -- 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 Crashes - Segmentation Fault
Hi Dan,
There was on very silly mistake and i missed to check that properly. Really
apologize for that.
Following change was done to get the conf-recording into the proper path:
chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings
following is the output:
[r...@linuxtest sounds]# ll
total 6416
drwxrwxr-x 2 asterisk asterisk4096 Jul 30 08:29 conf-recordings
[r...@linuxtest sounds]# ll conf-recordings/
total 4060
-rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27
meetme-conf-rec-74438-1280463795.8.wav
The only thing now is no speaker icon onto the webpage when i click to past
conference link.
Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of
getting speaker icon in past conference)?
--Manmohan Singh
On Fri, Jul 30, 2010 at 8:16 PM, Dan Austin wrote:
> Manmohan wrote:
> > I did added the record option in user options as well.
>
> > $Mod_Options = array(array(_("Announce"), "I"), array(_("Record"), "r"));
> > $User_Options = array(array(_("Announce"), "I"), array(_("Listen Only"),
> "m"), array(_("Wait for Leader"), "w"),
> > array(_("Record"), "r"));
>
> > And the gre8 news is, it did worked this time. But it saved the recorded
> file in the following path:
> That is good to hear.
>
> > /var/lib/asterisk/sounds/with the name as
> "meetme-conf-rec-74438-1280463795.8.wav
>
> > Than i tried to move the file to
> /var/lib/asterisk/sounds/conf-recordings/ just to see that it gives me a
> > speaker icon when i click to past conferences.
>
> > Unfortunately i couldnt see this speaker icon to hear this recorded
> conference .wav file.
> I am not surprised. By default MeetMe creates unique file names by
> appending
> pin-uniqueid, but uniqueid is not know until the conference starts, so the
> web interface
> does not know what to look for. Part of the changes to app_meetme included
> setting the
> realtime filename to use.
>
> > I tried to download the .wav file into my windows machine and the filed
> played well..
>
> > like i mentioned in my earlier mail that following line i had added in
> lib/define.php, please correct me if i am wrong:
>
>
> > define ("RECORDING_PATH", "/var/lib/asterisk/sounds/conf-recordings/");
>
> > Do you think This recording path is taking the effect here?
>
> That setting effect where the WMM interface looks for recordings and not
> where Asterisk puts
> them. Looking back at your email history, I see you are on 4.0.1. After
> all of the suggestions,
> I remembered that I too found problems with recordings and addressed them
> in 4.0.2
>
> That version adds a field to the database and stores the recording names in
> the database. I
> recommend using that version instead of 4.0.1. You can move your copy of
> lib/defines.php to
> the 4.0.2 install and keep your changes.
>
> Dan
>
>
> --
> _
> -- 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
>
--
Thanks & Regards
Manmohan Singh Jandu
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Manmohan wrote:
> I did added the record option in user options as well.
> $Mod_Options = array(array(_("Announce"), "I"), array(_("Record"), "r"));
> $User_Options = array(array(_("Announce"), "I"), array(_("Listen Only"),
> "m"), array(_("Wait for Leader"), "w"),
> array(_("Record"), "r"));
> And the gre8 news is, it did worked this time. But it saved the recorded file
> in the following path:
That is good to hear.
> /var/lib/asterisk/sounds/ with the name as
> "meetme-conf-rec-74438-1280463795.8.wav
> Than i tried to move the file to /var/lib/asterisk/sounds/conf-recordings/
> just to see that it gives me a
> speaker icon when i click to past conferences.
> Unfortunately i couldnt see this speaker icon to hear this recorded
> conference .wav file.
I am not surprised. By default MeetMe creates unique file names by appending
pin-uniqueid, but uniqueid is not know until the conference starts, so the web
interface
does not know what to look for. Part of the changes to app_meetme included
setting the
realtime filename to use.
> I tried to download the .wav file into my windows machine and the filed
> played well..
> like i mentioned in my earlier mail that following line i had added in
> lib/define.php, please correct me if i am wrong:
> define ("RECORDING_PATH", "/var/lib/asterisk/sounds/conf-recordings/");
> Do you think This recording path is taking the effect here?
That setting effect where the WMM interface looks for recordings and not where
Asterisk puts
them. Looking back at your email history, I see you are on 4.0.1. After all
of the suggestions,
I remembered that I too found problems with recordings and addressed them in
4.0.2
That version adds a field to the database and stores the recording names in the
database. I
recommend using that version instead of 4.0.1. You can move your copy of
lib/defines.php to
the 4.0.2 install and keep your changes.
Dan
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Re: [asterisk-users] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
DNSMasq has always been enabled. It's only one check box and if I didn't have it enabled phones won't work. So, that is set. Any other suggestions? including things regarding DNSMasq? Thanks On Fri, Jul 30, 2010 at 11:04 AM, Dave Cotton wrote: > On 30/07/10 16:15, bruce bruce wrote: > > Adria, > > > > How can I build a dns cache into my lan? I am using a Linksys 48 port > > POE switch and running a micro DD-WRT firmware on a linksys router. > > > > DD-WRT supports DNSMasq which would do exactly what you need. > > DC > > > > -- > _ > -- 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] DUNDi questions
Hi all, I have two questions regarding DUNDi and Asterisk Realtime. I have successfully set up DUNDi on my two Asterisk boxes, which means "dundi show peers" on each box shows the other box as known and "dialplan show dundiextens" shows the extensions on each box configured in sip.conf. 1. But when i switch my config to use sip in realtime, my extensions are only visible to DUNDi if i set rtcachefriends in sip.conf to yes. Am I forced to set rtcachefriends to use DUNDi with realtime or do I miss something, maybe an additional column in my database table? 2. How can I use DUNDi within my dialplan to determine if an extension is reachable and then establish a call to it and if not, pass the call to my PSTN device? Thanks in advance, Oliver -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
On 7/26/2010 4:05 AM, Andraž wrote: > I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from > sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, > freetds-bin, but, when I run configure and then make menuconfig in > section "Call Detail Recording" -> "cdr_tds" it's "disabled". The packaged version of FreeTDS on 10.04 is 0.82 which is too recent for the Asterisk 1.4. cdr_tds module. The latest version of FreeTDS supported by Asterisk 1.4 is 0.64. So you will have to compile from source (or find an ancient package) to get this working. -- _ -- 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] agi macro problem
>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards >Subject: Re: [asterisk-users] agi macro problem >You are confusing a language with a protocol. An AGI is a program that >complies with the AGI protocol. It can be written in any language that >reads from stdin and writes to stdout. I don't know of any language that >fails this requirement. >Ruby is not a popular language for writing AGIs, at least on this list. >Most AGI compliant programs are written in Perl, PHP, c, or shell. I'm sure this is "not a good practice", but I actually use the .agi suffix on my AGI's or AGI launch stubs to "keep it straight" that even though the actual program is bash or PERL, that the function of the program is to generally be launched by the Asterisk AGI. That way, if the program gets moved from /var/lib/asterisk/agi-bin, I know it's not just a run-of-the-mill bash or Perl script. Since the #! Line tells you what it actually is, it is (IMO) a "no harm" proposition. -- _ -- 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] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
On 30/07/10 16:15, bruce bruce wrote: > Adria, > > How can I build a dns cache into my lan? I am using a Linksys 48 port > POE switch and running a micro DD-WRT firmware on a linksys router. > DD-WRT supports DNSMasq which would do exactly what you need. DC -- _ -- 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] agi macro problem
On Fri, 30 Jul 2010, Zarko Zivanovic wrote: I need simple whopicked.agi (instead of .rb) which will simply take the value 30086 (that I pass to macro) While ".rb" suggests a "Ruby source file," ".agi" suggests nothing. This should be simple – no ruby - just agi. You are confusing a language with a protocol. An AGI is a program that complies with the AGI protocol. It can be written in any language that reads from stdin and writes to stdout. I don't know of any language that fails this requirement. Ruby is not a popular language for writing AGIs, at least on this list. Most AGI compliant programs are written in Perl, PHP, c, or shell. -- 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] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
Adria, How can I build a dns cache into my lan? I am using a Linksys 48 port POE switch and running a micro DD-WRT firmware on a linksys router. Gareth, I think the registration time is part of the reason. I have lowered it less than 10 seconds. Thanks On Fri, Jul 30, 2010 at 8:21 AM, Adrià Vidal wrote: > try to have a dns cache into your LAN, Aastra phone are prone to fail when > have any little DNS error. > > > -- > -- > Adrià Vidal > > > > -- > _ > -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Is it easy to install along with FreePBX as well? Thanks On Fri, Jul 30, 2010 at 5:49 AM, Lenz Emilitri wrote: > QueueMetrics is actually free (as in beer) for very small call centers and > individual hackers. > l. > > 2010/7/28 Zeeshan Zakaria > > There is none for free. >> >> Zeeshan A Zakaria >> >> -- >> www.ilovetovoip.com >> >> On 2010-07-27 6:12 PM, "bruce bruce" wrote: >> >> :-) I knew someone would bring up FreePBX. I have FreePBX installed and >> it's not good for Queues at all. It's using the reporting tool from Areski >> and Areski has recently released an upgrade to it which again is not what I >> want. >> >> There are few other programs that do this but really none that are neat in >> interface or useful in features. >> >> I guess no one else has any thoughts on this? Maybe there is none >> available? >> >> Thanks, >> Bruce >> >> >> >> On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg >> wrote: >> > >> > On Mon, Jul 26... >> >> -- >> _ >> -- 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 >> > > > > -- > Loway - home of QueueMetrics - http://queuemetrics.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 > -- _ -- 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] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Thank Martin, That makes absolute sense. I have turned busy detect off for now and haven't heard complains or lines remaining open for a Day. I am in Canada. I just checked chan_dahdi.conf and I don't see callprogress there at all. So, I guess the lines are fine for hanging up by themselves. Hope this doesn't give me probs in future. Thanks, Bruce On Fri, Jul 30, 2010 at 6:18 AM, Martin wrote: > Either turn off busydetect or increase the busycount to 5-7 or even > more ... (10 should be conservative) > busydetect looks for cadence or patterns of the same length ... beep > on [X ms] beep off [Y ms] > so you can afford to increase busycount and have a few second longer > calls / the line is kept longer offhook > but you don't get false busy detections > > Also in US/Canada callprogress will do a better job then busydetect > since it looks for specific frequencies of the busy signal > and not just noise/beep then silence ... If you're somewhere else then > you can hire a coder to tweak callprogress algorithm > to your country's busy signal frequencies ... Just record the busy > signal with ztmonitor and send to someone for code patch... > > regards > Martin > > On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce wrote: > > Hmmwhat about call waiting? > > You mean, when a call comes in on that specific line, it generate two > beep > > tones and hence the system hangs up thinking it's end of the call? > > Interesting!!! > > If it is call-waiting do I have to set all of the following off for it to > > not give me problem again: > > callwaiting=yes > > usecallingpres=yes > > callwaitingcallerid=yes > > threewaycalling=yes > > transfer=yes > > canpark=yes > > cancallforward=yes > > busydetect=yes > > busycount=3 > > Please elaborate a bit if I am off-topic. > > Regards, > > Bruce > > On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas > wrote: > >> > >> From: asterisk-users-boun...@lists.digium.com > >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce > bruce > >> Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden? > >> Couldbusy detect be the problem? > >> > >> > >> > >> I am getting a complain that call on analogue lines (Sangoam A400D) > drops > >> all of a sudden. Here is what I see in logs: > >> > >> > >> > >> Could be callwaiting? > >> > >> -- > >> _ > >> -- 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 > > > > -- > _ > -- 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] Problem with Sangoma card...
El 30/07/10 00:14, Carlos Chavez escribió: > On Thu, 29 Jul 2010 20:47:58 -0500 (CDT), Tim Nelson wrote > >> - "Carlos Chavez" wrote: >> >>> I have a problem with a Sangoma card. It worked until yesterday. >>> Now >>> I keep getting this error: >>> >>> Jul 29 17:45:17 pbxacura kernel: wanpipe1: Enable E1 CAS signalling >>> mode! >>> Jul 29 17:45:17 pbxacura kernel: wanpipe1:w1g1: Rx Error: No >>> 'DeviceSelect' from target: pci fatal error! (0x000FA000) >>> Jul 29 17:45:48 pbxacura last message repeated 30946 times >>> >>> It is using Wanpipe 3.5.6 and Zaptel 1.4.12.1 with Asterisk 1.4.30 >>> and >>> MFC/R2. The E1 shows Tx side blocked and no calls in or out. I >>> reinstalled Wanpipe and Zaptel but I keep getting the same error. >>> >>> >> Sounds like your card needs to be: >> >> 1. Reseated in it's PCI slot >> 2. RMA'ed to Sangoma >> >> Either way, contact Sangoma's top notch tech support and they'll >> either help you fix the issue or get you a replacement. >> >> > I already moved the card to another PCI slot and contacted tech support > at Sangoma. Just covering all the bases in case someone here had the same > experience. > > -- > Carlos Chavez > Director de Tecnología > Telecomunicaciones Abiertas de México S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > > > My guess is that you need to upgrade the firmware of the Sangoma card. We had problems with a Dell server and a Sangoma card, solved by an upgraded firmware that fixed some PCI parity errors. This could be a long shot, but I'm pretty sure the Sangoma tech support will ask you you upgrade the firmware to the latest version as a first measure. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- 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] Disconnect supervision tone detection
Your best bets are going to be #1 hanguponpolarityswitch=yes Or #2 callprogress=yes I'd hang my hat on #1 personally -- _ -- 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] SEPMAC.xml for Ciscp 7970 IP Phone
On Thu, Jul 29, 2010 at 4:15 AM, zeynep yildirim wrote: > Hi All, > > I upgraded 7970 from SCCP to SIP. But the phone isn't registering. > Have you got any working XML file for 7970 phones. Isn't registering with what? If you're registering that with CallManager, you have to change the phone config after your firmware change. If you're registering with asterisk, you have to tell the phone where to try registering. -- _ -- 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] perform tasks outside a dial-plan (not during acall)
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Harel Cohen Subject: [asterisk-users] perform tasks outside a dial-plan (not during acall) Can the Asterisk do "things" not during a call? For example I would like to change my dial plan during certain hours\dates or I would like to check some information in the astdb (e.g. counters of al sort) and handle it as required and so on. All of this is not call-related therefore I don't know if I can somehow do it using the dial-plan applications\functions. I know I can do chron jobs on the Linux level but for maintenance and readability I would prefer to do these tasks from within the Asterisk. Is it possible to configure the Asterisk to perform routine tasks on certain times or certain intervals? By definition, all dial-plan actions/functions have to be done from within a "call". This does not mean that you have to actually make a call at 3 in the morning. You can set up contexts to do these functions and use "Local" calls from AMI or cron to perform these functions. Let's use a simple example from your post: I want to see the Asterisk DB keys at a given point in time. In cron I could set up '15 4 * * * /usr/sbin/asterisk -rx "database show" ' to show me what the database contained at 4:15 am each day. But I don't get up until 6 and I want this in a file to look at later. So I make an AGI to do this instead. [dialplan-snapshot] Exten => s,1,answer Exten => s,n,AGI(snapshot.agi) Exten => s,n,hangup Now in cron I do this instead 15 4 * * * /usr/sbin/asterisk -rx "dial lo...@dialplan-snapshot" And Asterisk runs this context obediently just like I had woke up and dialled to this point. This may not be 100% correct due to mail-reformatting or guy at keyboard, but the concepts have been discussed in this list this month. -- _ -- 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 and QoS
HTB is a bad choice for VoIP. When it "borrows" bandwidth, according to the docs it doesnt release it back until its finished so if its using all the bandwidth for a download before the VoIP call starts, VoIP gets starved even if you reserve an excess of bandwidth as it still queues. When I tried it sort of worked but didnt have the effect I expected on a busy link, probably for this reason. HTB tries to be fair about sending packets but with VoIP being fair sucks :) A better way is to use a "prio" filter at the root, the priority 1 branch having a plain fifo on it - send VoIP and acks only this way. The priority 2 branch has a HTB hierarchy with sfq leaves for the rest of the traffic. This seems to work much better, but I have not tested well yet. I am also using a police filter for incoming (on ADSL) and have not noticed any problems - but it is only lightly limiting (to try and keep the queues at the ISP end short.) Lastly, test to make sure the packets are flowing where you expect them to - I had to correct a few miss-understandings I had on how it all worked before everything went where I wanted it to :) TC and TCNG do seem dead, but I think thats partly because its relatively mature and doesnt need much work. BillK On Fri, 2010-07-30 at 10:06 +0200, Jonas Kellens wrote: > Hello list, > > anyone here using Asterisk together with HTB for queing incoming and > outgoing packets ? > > I've tried to subscribe myself to the Mailinglist of the Linux > Advanced Routing & Traffic Control project, but I get no confirmation. > This list seems dead. > > It seems my test case with HTB is not giving any noticeable results. > Can I ask questions on this mailinglist ? > > Perhaps you can give my other QoS-implementations like MasterShaper, > if it works well together with a firewall that uses iptables. > > > > 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 -- William Kenworthy Home in Perth! -- _ -- 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] perform tasks outside a dial-plan (not during a call)
Harel Cohen wrote: > Hi all, > > Can the Asterisk do “things” not during a call? For example I would like > to change my dial plan during certain hours\dates or I would like to > check some information in the astdb (e.g. counters of al sort) and > handle it as required and so on. All of this is not call-related > therefore I don’t know if I can somehow do it using the dial-plan > applications\functions. I know I can do chron jobs on the Linux level > but for maintenance and readability I would prefer to do these tasks > from within the Asterisk. > > Is it possible to configure the Asterisk to perform routine tasks on > certain times or certain intervals? > > Thanks, > > Harel > It would depend on exactly what you wanted to do. If you wanted to change the dialplan then you would normally just call an AGI program and have that do diffeent things depending on the time of the day. If you wanted to check 'counters' then you would normally not store them in the built in internal database but store them in a sql database instead which can be monitored via an external program via a cron job. If you want asterisk to do things at particular times then you would generally have a program which connects to the asterisk manager interface and isue commands when required. -- _ -- 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] agi macro problem
In theory this snippet will do the trick
Save as updatech.pl
#!/usr/local/bin/perl -w
$ENV{PATH} = '/usr/sbin:/:/usr/bin:/usr/local/apache/bin'; # reasonable path
$ENV{ENV} = "/etc/bash.bachrc";
use strict;
use warnings;
use File::Find;
use DBI;
use Date::Calc qw(:all);
use Asterisk::AGI;
$|=1;
my $agi = Asterisk::AGI->new();
my %input = $agi->ReadParse();
my ($chanval) = @ARGV;
my $agi_channel = $input{'agi-channel'};
## db vars
my $data_source = "dbi:Pg:dbname=Asterisk";
my $username = "root";
my $auth = "xxx";
#
##-- connect to the db --
##establish the DBI connection with transaction processing
my $dbh = DBI->connect($data_source, $username, $auth, {
AutoCommit => 0,
RaiseError => 1,
} ) or die "Can't connect to database: ", $DBI::errstr, "\n";
# read the password file to get account type
my $upd_sh = $dbh->prepare( "UPDATE call_log SET local='CHANNEL' WHERE
id='$chanval' AND channel='$agi_channel'");
$upd_sh->execute();
$dbh->commit();
$dbh->rollback();
exit;
and change line 2 of the macro to exten => s,2,AGI(updatech.pl,$ARG1)
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[asterisk-users] perform tasks outside a dial-plan (not during a call)
Hi all, Can the Asterisk do “things” not during a call? For example I would like to change my dial plan during certain hours\dates or I would like to check some information in the astdb (e.g. counters of al sort) and handle it as required and so on. All of this is not call-related therefore I don’t know if I can somehow do it using the dial-plan applications\functions. I know I can do chron jobs on the Linux level but for maintenance and readability I would prefer to do these tasks from within the Asterisk. Is it possible to configure the Asterisk to perform routine tasks on certain times or certain intervals? Thanks, Harel -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
I removed freetds which I installed from apt-get. Run what you said and stil doesn't work. :( I just hit reply, so I don't touch the subject line. On Fri, Jul 30, 2010 at 2:12 PM, A J Stiles wrote: > On Friday 30 Jul 2010, Andraž wrote: > > >From source also doesn't work. :( > > If you ran ldconfig to force update of library configuration after you > installed the freetds you compiled, and re-ran ./configure in the asterisk > build directory, and it still doesn't want to let you use freeTDS, then > something else must be the problem. > > It's possible that the configure script is getting confused with the > FreeTDS > installed using apt-get and the FreeTDS installed from source code. So try > $ ./configure --with-tds=/usr/local/lib > to force it to use the freeTDS files found in /usr/local/lib (make > appropriate substitutions if required). > > > By the way: Your reply belongs *under* whatever you are replying to. > Then, > somebody reading the messages in future can see the proper flow of the > conversation. > > -- > AJS > > -- > _ > -- 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] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
try to have a dns cache into your LAN, Aastra phone are prone to fail when have any little DNS error. -- -- Adrià Vidal -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
On Friday 30 Jul 2010, Andraž wrote: > >From source also doesn't work. :( If you ran ldconfig to force update of library configuration after you installed the freetds you compiled, and re-ran ./configure in the asterisk build directory, and it still doesn't want to let you use freeTDS, then something else must be the problem. It's possible that the configure script is getting confused with the FreeTDS installed using apt-get and the FreeTDS installed from source code. So try $ ./configure --with-tds=/usr/local/lib to force it to use the freeTDS files found in /usr/local/lib (make appropriate substitutions if required). By the way: Your reply belongs *under* whatever you are replying to. Then, somebody reading the messages in future can see the proper flow of the conversation. -- AJS -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
>From source also doesn't work. :( On Fri, Jul 30, 2010 at 1:15 PM, Fred Posner wrote: > On Jul 30, 2010, at 5:04 AM, Andraž wrote: > > > Ok, problem is another, when I run configure, it write this: > > checking for tds_version in -ltds... no > > configure: *** > > configure: *** The FreeTDS installation on this system appears to be > broken. > > configure: *** Either correct the installation, or run configure > > configure: *** without explicitly specifying --with-tds > > ODBC is not a good solution, only if I can change the names of CDR > fields. > > > > How can I "repair" the installlation? > > > > On Wed, Jul 28, 2010 at 2:58 PM, Andraž wrote: > > I resolved this isue using odbc. > > > > > > On Mon, Jul 26, 2010 at 11:27 AM, Tzafrir Cohen < > tzafrir.co...@xorcom.com> wrote: > > On Mon, Jul 26, 2010 at 10:05:27AM +0200, Andraž wrote: > > > Hi, > > > > > > I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from > > > sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, > > > freetds-bin, but, when I run configure and then make menuconfig in > section > > > "Call Detail Recording" -> "cdr_tds" it's "disabled". It only writes > that > > > "Depends on: freetds(E)". On another server (same configuration) I > installed > > > the same packages, and it's working fine. Any suggestions, what I did > wrong? > > > > Have you re-ron ./configure #? > > > > -- > > Tzafrir Cohenv > > Have you tried installing freetds from source? > > wget > ftp://ftp.ibiblio.org/pub/Linux/ALPHA/freetds/stable/freetds-stable.tgz > tar -zxvf freetds-stable.tgz > cd freetds-0.82 > ./configure --prefix=/usr --with-tdsver=7.0 --with-unixodbc=/usr/lib >or ./configure --prefix=/usr --with-tdsver=7.0 --with-unixodbc=/usr > make && make install > > > ---fred > http://qxork.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 > -- _ -- 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] GoToIfTime problem
Jonas Kellens wrote: > Hello list, > > how come when the time is 12:31:18, the GoToIfTime-statement evaluates > to true ?? > > As noted in the Wiki: "Times before Asterisk 1.6.2 are only accurate down to the 2-minute interval. So 12:01 is treated the same as 12:00. Starting with 1.6.2, times are accurate down to the minute. " Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- 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] agi macro problem
I am trying this approach to see who picked the line:
Here is what i am doing:
EXEC DIAL SIP/ vaso &Zap/35||M(testing^30086)
Macro:
[macro-testing]
exten => s,1,DumpChan()
exten => s,2,AGI(whopicked.rb)
exten => s,3,Hangup()
>From console:
-- SIP/ vaso -e26c answered Zap/14-1
-- Executing DumpChan("SIP/ vaso -e26c", "") in new stack
-- Executing DumpChan("SIP/vaso-e26c", "") in new stack
Dumping Info For Channel: SIP/vaso-e26c:
Info:
Name= SIP/vaso-e26c
Type= SIP
UniqueID= 1280487752.1809
CallerID= 8221
CallerIDName= (N/A)
DNIDDigits= (N/A)
State= Up (6)
Rings= 0
NativeFormat= 2
WriteFormat=4
ReadFormat= 4
1stFileDescriptor= 74
Framesin= 3
Framesout= 0
TimetoHangup= 0
ElapsedTime=0h0m0s
Context=macro-testing
Extension= s
Priority= 1
CallGroup=
PickupGroup=
Application=DumpChan
Data= (Empty)
Blocking_in=(Not Blocking)
Variables:
MACRO_DEPTH=1>
ARG1=30086
MACRO_PRIORITY=1
MACRO_CONTEXT=siptest
-- Executing AGI("SIP/vaso-e26c", "whopicked.rb") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/whopicked.rb
AGI Tx >> agi_request: whopicked.rb
AGI Tx >> agi_channel: SIP/vaso-e26c
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1280487752.1809
AGI Tx >> agi_callerid: 8221
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 3
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 33
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: macro-testing
AGI Tx >> agi_extension: s
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
-- AGI Script whopicked.rb completed, returning 0
-- Executing Hangup("SIP/vaso-e26c", "") in new stack
I need simple whopicked.agi (instead of .rb) which will simply take the
value 30086 (that I pass to macro)
And do this:
UPDATE call_log SET local = 'CHANNEL' WHERE id = '30086'
Where channel is agi_channel: SIP/vaso-e26c
This should be simple - no ruby - just agi.
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Re: [asterisk-users] FreeTDS (Microsoft MsSQL 2008) and CDR
On Jul 30, 2010, at 5:04 AM, Andraž wrote: > Ok, problem is another, when I run configure, it write this: > checking for tds_version in -ltds... no > configure: *** > configure: *** The FreeTDS installation on this system appears to be broken. > configure: *** Either correct the installation, or run configure > configure: *** without explicitly specifying --with-tds > ODBC is not a good solution, only if I can change the names of CDR fields. > > How can I "repair" the installlation? > > On Wed, Jul 28, 2010 at 2:58 PM, Andraž wrote: > I resolved this isue using odbc. > > > On Mon, Jul 26, 2010 at 11:27 AM, Tzafrir Cohen > wrote: > On Mon, Jul 26, 2010 at 10:05:27AM +0200, Andraž wrote: > > Hi, > > > > I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from > > sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, > > freetds-bin, but, when I run configure and then make menuconfig in section > > "Call Detail Recording" -> "cdr_tds" it's "disabled". It only writes that > > "Depends on: freetds(E)". On another server (same configuration) I installed > > the same packages, and it's working fine. Any suggestions, what I did wrong? > > Have you re-ron ./configure #? > > -- > Tzafrir Cohenv Have you tried installing freetds from source? wget ftp://ftp.ibiblio.org/pub/Linux/ALPHA/freetds/stable/freetds-stable.tgz tar -zxvf freetds-stable.tgz cd freetds-0.82 ./configure --prefix=/usr --with-tdsver=7.0 --with-unixodbc=/usr/lib or ./configure --prefix=/usr --with-tdsver=7.0 --with-unixodbc=/usr make && make install ---fred http://qxork.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] GoToIfTime problem
Hello list,
how come when the time is 12:31:18, the GoToIfTime-statement evaluates
to true ??
[Jul 30 12:31:18] -- Executing [...@macro-hours:42]
GotoIfTime("SIP/TELin-0067", "9:00-12:30|fri|*|*?exit") in new stack
[Jul 30 12:31:18] -- Goto (macro-hours,s,58)
The macro jumps to step 58, namely "exit".
I would expect the GoToIfTime-statement evaluates to False and goes to
the next step, no ?!
Kind regards,
Jonas.
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Re: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Either turn off busydetect or increase the busycount to 5-7 or even more ... (10 should be conservative) busydetect looks for cadence or patterns of the same length ... beep on [X ms] beep off [Y ms] so you can afford to increase busycount and have a few second longer calls / the line is kept longer offhook but you don't get false busy detections Also in US/Canada callprogress will do a better job then busydetect since it looks for specific frequencies of the busy signal and not just noise/beep then silence ... If you're somewhere else then you can hire a coder to tweak callprogress algorithm to your country's busy signal frequencies ... Just record the busy signal with ztmonitor and send to someone for code patch... regards Martin On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce wrote: > Hmmwhat about call waiting? > You mean, when a call comes in on that specific line, it generate two beep > tones and hence the system hangs up thinking it's end of the call? > Interesting!!! > If it is call-waiting do I have to set all of the following off for it to > not give me problem again: > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > canpark=yes > cancallforward=yes > busydetect=yes > busycount=3 > Please elaborate a bit if I am off-topic. > Regards, > Bruce > On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas wrote: >> >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce >> Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden? >> Couldbusy detect be the problem? >> >> >> >> I am getting a complain that call on analogue lines (Sangoam A400D) drops >> all of a sudden. Here is what I see in logs: >> >> >> >> Could be callwaiting? >> >> -- >> _ >> -- 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 > -- _ -- 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] How can i switch to samba server omitting sshfs
Hi, When the record file method is called by FAGI, the Asterisk server saves the file on its localmachine. This needs to be sent to the ASR server machine so that the ASR can decode the file. Similarly, when Festival synthesizes speech, the wav file is stored on the Festival server machine and needs to be sent to the Asterisk server machine so that Asterisk can play it back.For now, we have been mapping drives so that the ASR and Asterisk server share a folder where the recorded files are stored and the Asterisk server and Festival server share a folder where the synthesized files are stored. We do this using sshfs. I now want to go with samba server. Because there can be so many machines connected to asterisk server. So i now have to create directory specific to a user and store the files. This is the requirement. How can i achieve this. Please help me in this regard. Thanks & Regards, Jahnavi. -- _ -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
QueueMetrics is actually free (as in beer) for very small call centers and individual hackers. l. 2010/7/28 Zeeshan Zakaria > There is none for free. > > Zeeshan A Zakaria > > -- > www.ilovetovoip.com > > On 2010-07-27 6:12 PM, "bruce bruce" wrote: > > :-) I knew someone would bring up FreePBX. I have FreePBX installed and > it's not good for Queues at all. It's using the reporting tool from Areski > and Areski has recently released an upgrade to it which again is not what I > want. > > There are few other programs that do this but really none that are neat in > interface or useful in features. > > I guess no one else has any thoughts on this? Maybe there is none > available? > > Thanks, > Bruce > > > > On Tue, Jul 27, 2010 at 11:41 AM, David Backeberg > wrote: > > > > On Mon, Jul 26... > > -- > _ > -- 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 > -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 82
Hi! > i want to get channel-id before dialing so that i can dial using that > channel id. I am afraid that is not going to work. Maybe you should take a step back and describe what it actually is that you are trying to accomplish. Philipp -- _ -- 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] Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
bruce bruce wrote: > Hi Everyone, > > I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The > phones occasionally go into "No Service" mode. The POE switch doesn't > seem to be the problem as it's tested fine. I think the router sometimes > gives up and comes back quickly. Or something of that nature. However, > the connections are maintained if a call is going on because there are > peer to peer connections between the phones in a network. Anyhow, if the > phones are restarted they work fine. > > So, I was looking around the Aastra Admin UI to find any timer to lower > it to 1 second to check and make sure the device always has an ip but I > can't seem to find anything other than LLDP which is set at 30 and I > don't think that will be of any help. > > I did a test where I would disconnect the router from the switch and > after a while phones go into "No Service" but if I plug it back into the > switch the phones do not come back right away. Maybe something should be > dialed on the phone or wait long time or restart it to work again. > > Any work around? > > Thanks a lot > Try changing the registration period so that they perform a regular re-register. That way if something happens and they fail to register when they notice a problem they will try again a bit later. -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
Ok, problem is another, when I run configure, it write this: checking for tds_version in -ltds... no configure: *** configure: *** The FreeTDS installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** without explicitly specifying --with-tds ODBC is not a good solution, only if I can change the names of CDR fields. How can I "repair" the installlation? On Wed, Jul 28, 2010 at 2:58 PM, Andraž wrote: > I resolved this isue using odbc. > > > On Mon, Jul 26, 2010 at 11:27 AM, Tzafrir Cohen > wrote: > >> On Mon, Jul 26, 2010 at 10:05:27AM +0200, Andraž wrote: >> > Hi, >> > >> > I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from >> > sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, >> > freetds-bin, but, when I run configure and then make menuconfig in >> section >> > "Call Detail Recording" -> "cdr_tds" it's "disabled". It only writes >> that >> > "Depends on: freetds(E)". On another server (same configuration) I >> installed >> > the same packages, and it's working fine. Any suggestions, what I did >> wrong? >> >> Have you re-ron ./configure #? >> >> -- >> Tzafrir Cohen >> icq#16849755 >> jabber:tzafrir.co...@xorcom.com >> +972-50-7952406 mailto:tzafrir.co...@xorcom.com >> http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- _ -- 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 QoS
Hello list, anyone here using Asterisk together with HTB for queing incoming and outgoing packets ? I've tried to subscribe myself to the Mailinglist of the Linux Advanced Routing & Traffic Control project, but I get no confirmation. This list seems dead. It seems my test case with HTB is not giving any noticeable results. Can I ask questions on this mailinglist ? Perhaps you can give my other QoS-implementations like MasterShaper, if it works well together with a firewall that uses iptables. 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
Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 82
thanks for your reply but i think ${BRIDGEPEER} will work only when both
channels are connected. i want to get channel-id before dialing so that i
can dial using that channel id.
> ${BRIDGEPEER} is probably a good way to do what you want.. if Channel
> A calls Channel B, and you want Channel A to "get" the channelID of
> Channel B, as long as the two channels are bridged, ${BRIDGEPEER} will
> do what you want
>> perhaps ${CALLERID(DNID)}
>>
>>> my question is how can i get channel-id of a user or peer. I tried using
>>> ChanIsAvail(username). this works correctly when user and asterisk are
on
>>> Local LAN. But my asterisk server is on public ip and users are behind
nat,
>>> and so this method says unknow host when used on public asterisk server.
>>> I also tried built-in variable ${CHANNEL}, but this returns the
channel-id
>>> of the calling channel. but i want channel-id of called user.
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Re: [asterisk-users] Registering 2 phone numbers to same router
That helped. I can now register both. Looks like I need to forward all traffic from the second asterisk instance to the main one for all the users to successfully register and talk to each other. Is forwarding all traffic from one instance to the main one possible? How can I do that? Thanks JW -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kyle Kienapfel Sent: Friday, July 30, 2010 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Registering 2 phone numbers to same router On Thu, Jul 29, 2010 at 4:05 PM, jwexler wrote: > On Thu, Jul 29, 2010 at 10:15 PM, Paul Belanger wrote: >> MAC Address? Are you sure? Why would your ISP care about level 2? I >> could understand IP address (level 3). If this is the case, you will >> need to spoof your MAC. > > Actually, it is mind boggling that the isp even cares about restricting > phone registrations per device which is apparently what they are trying to > do. Without a work around, I would need to have 3 separate machines just to > register the three phone numbers. That would be a real mess. On their ip > phone settings page, there is a column labeled "mac address". They do not > display the mac addresses that they populate there but they do restrictions > by info received on the nic from which the registration was sent. > Unfortunately, simply spoofing the mac address would be insufficient because > there is no way to specify which nic to use in the 2 register statements in > sip.conf. I have not been able to use iptables or ip route to make up an > additional address to the router that asterisk can use successfully. I can > do so such that firefox can access, login to, and update the router but not > asterisk for some strange reason. The router is at 192.168.40.1. I set up > 192.168.40.3 as a new ip that just routes to 192.168.40.1 which firefox is > happy with. Asterisk chokes. Maybe because of the rt200ne patch? Link is in > Japanese but it patches sip.c so that I can register with the router: > http://voip-info.jp/index.php/RT-200NE%E5%AF%BE%E5%BF%9C%E3%83%91%E3%83%83%E > 3%83%81 > Or some other cause? Suggestions on some kind of workaround be really > appreciated? I hope some day, Asterisk will provide the option to specify > registrations by nic interface. > Thanks > JW > So asterisk registers with the router that your isp gave you? I'd try multiple asterisks with the same ip address, just different ports for SIP and RTP. Are you sure its limited by mac address? a quicker test to probe for that would be to use two softphones on the same computer, one for each sip accounts -- _ -- 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-users Digest, Vol 72, Issue 81
thanks for your reply but i did not meant that. ${CALLERID(DNID)} will
return then number which i don't want. what i want is channel-id like if we
have a user named "nasir", then we dial it as follows
Dial(SIP/nasir)
but actual channel-id that asterisk uses is something like " nasir-2b487e9".
and on the asterisk cli we can check this when call is answered or hangup,
asterisk attaches some random id with username.
i am dialing sip uri using "Dial(SIP/119.26.18.235:5062)" which causes
changed INVITE adn TO headers, so i want to get the channel-id that asterisk
internally uses do dial it.
if we use ChanIsAvail(SIP/nasir) or ChanIsAvail(SIP/192.168.0.10:5062) this
works on Local LAN and it returns "SIP/192.168.0.10:5062-3fe934f4" , but
when asterisk is on Live Ip and users are behind Router then this function
gives error of unknow host. so i want to know if there is any other function
that does this job.
so what is want is to get this channel-id ( like nasir-2487e9) and dial it
like
Dial(SIP/nasir-2487e9) or Dial(SIP/119.26.18.235:5062-34e984b)"
hope this clears what i wanna do.
> Message: 8
> Date: Thu, 29 Jul 2010 10:37:07 -0500
> From: "Danny Nicholas"
> Subject: Re: [asterisk-users] How to extract channel-id of a user or
>peer
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>
> Message-ID: <201007291515.o6tffv8t025...@mail.debsinc.com>
> Content-Type: text/plain; charset="us-ascii"
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Javaid
> Subject: [asterisk-users] How to extract channel-id of a user or peer
>
> my question is how can i get channel-id of a user or peer. I tried using
> ChanIsAvail(username). this works correctly when user and asterisk are on
> Local LAN. But my asterisk server is on public ip and users are behind nat,
> and so this method says unknow host when used on public asterisk server.
> I also tried built-in variable ${CHANNEL}, but this returns the channel-id
> of the calling channel. but i want channel-id of called user.
>
>
> --
> perhaps ${CALLERID(DNID)}
>
>
--
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Re: [asterisk-users] MOH distorted voice in Native and MP3 format
>> Something you may want to try (its fixed it for us) is putting an I >> (uppercase I) on the asterisk invocation line. >> >> We run servers in the cloud and can't get reliable timing from ISDN >> cards etc so this instructs asterisk to generate its own internal >> timing. If you have ISDN you probably don't want to do this as they >> "should" provide better timing. >> >> Its probably worth a try anyway. >> >> eg. >> asterisk -vvvg >> change to >> asterisk -vvvgI > > Something is better than nothing, I have configured the init.d script > to start asterisk with option -I, and restarted the asterisk. > Thanks Kevin for your tip. > Hello, I just found that setting option -I doesn't resolve the issue. Perhaps it is reducing the chances of occurrence, but issue is still persist. -- -MohammedShehzad -- _ -- 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
