Re: [asterisk-users] AEL and swap from macros to contexts
On Tue, Oct 7, 2008 at 2:20 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I
ignore this ael warnings? thanks
PJ
LOG: lev:3 file:pval.c line:2521 func: check_pval_item Warning: file
/etc/asterisk/extensions.ael, line 36-36: application call to Gosub
affects flow of control, and needs to be re-written using AEL if, while,
goto, etc. keywords instead!
Hi,
In definition use:
macro set_record(A,B) {
// do something
}
And for calling:
set_record(${CALLERID(NUM)},${EXTEN});
It will automatically be translated to GoSub in 1.6, but will remain
as Macro in 1.4.
yes, I know, but I hear on IRC, that macros will be deprecated and
suggestion was to move to contexts,
personaly I would like also move away from macros, because macros have
some limitations, eg. variable number of arguments isn't possible with
classic macros,
macros also require variable to be defined in macro definition (that is
needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
so I definitively agree with moving from macros to contexts, only one
bad thing is compiler warning, when I try to Gosub to context (as macro
replacement)
PJ
Pavel--
Yes, you can ignore the warnings and go ahead and hardcoded gosub calls
into your source. I didn't upgrade 1.4 to use gosub-instead-of-macro
because
the key element ended up being calling gosub with arguments, which
didn't
make it into 1.4.
Someday, when you upgrade from 1.4 to 1.6, you will have to change
all your gosub's to use the argument passing feature, if you hardcode
gosubs now. Or, you can backport the gosub-with-arguments feature to
1.4,
and use 1.6 AEL to compile... which will give you some future
portability
when you do move to 1.6...
Sorry to make simple things sound so complicated!
murf
murf, thank you for clear answer,
currently, I'm using asterisk trunk (and 1.6 also),
do you plan to remove quite confusing AEL warnings, that appears, when I
try to hardcode Gosub with arguments into ael dialplan?
Why would you still want to hardcode them?
because I would like to move completely away from using classic macros,
because it have some limitations, as I said, variable number of
arguments passed to macro is example,
so I moving from macros to contexts that do the same functionality and
haven't limitations that macros have
and if I will have only contexts in ael dialplan I must call it with
Gosub (I can't call context using )
I think you didn't understood, that declaring macro x and calling it
with x() would make AEL parser to do it for you. They are called
macro just in AEL, but internally they are GoSub's. Additionally you
will be ready for any other future changes.
For example You can use
$aelparse -d -n -w -q extensions.ael
and take a look at generated .conf file. In 1.6.0 it would be:
[set-record]
exten = s,1,Set(LOCAL(A)=${ARG1})
exten = s,2,Set(LOCAL(B)=${ARG2})
...
exten = s,20,Return()
And call to it:
Gosub(set_record,s,1(${CALLERID(num)},${EXTEN}))
Regards,
Atis
PJ
Please see above sample, you can use prefixing with and ().
Regards,
Atis.
PJ
Regards,
Atis
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Re: [asterisk-users] AEL and swap from macros to contexts
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I
ignore this ael warnings? thanks
PJ
LOG: lev:3 file:pval.c line:2521 func: check_pval_item Warning: file
/etc/asterisk/extensions.ael, line 36-36: application call to Gosub
affects flow of control, and needs to be re-written using AEL if, while,
goto, etc. keywords instead!
Hi,
In definition use:
macro set_record(A,B) {
// do something
}
And for calling:
set_record(${CALLERID(NUM)},${EXTEN});
It will automatically be translated to GoSub in 1.6, but will remain
as Macro in 1.4.
yes, I know, but I hear on IRC, that macros will be deprecated and
suggestion was to move to contexts,
personaly I would like also move away from macros, because macros have
some limitations, eg. variable number of arguments isn't possible with
classic macros,
macros also require variable to be defined in macro definition (that is
needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
so I definitively agree with moving from macros to contexts, only one
bad thing is compiler warning, when I try to Gosub to context (as macro
replacement)
PJ
Pavel--
Yes, you can ignore the warnings and go ahead and hardcoded gosub calls
into your source. I didn't upgrade 1.4 to use gosub-instead-of-macro
because
the key element ended up being calling gosub with arguments, which
didn't
make it into 1.4.
Someday, when you upgrade from 1.4 to 1.6, you will have to change
all your gosub's to use the argument passing feature, if you hardcode
gosubs now. Or, you can backport the gosub-with-arguments feature to
1.4,
and use 1.6 AEL to compile... which will give you some future
portability
when you do move to 1.6...
Sorry to make simple things sound so complicated!
murf
murf, thank you for clear answer,
currently, I'm using asterisk trunk (and 1.6 also),
do you plan to remove quite confusing AEL warnings, that appears, when I
try to hardcode Gosub with arguments into ael dialplan?
Why would you still want to hardcode them?
Please see above sample, you can use prefixing with and ().
Regards,
Atis.
PJ
Regards,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] asteriskt38.com
On Mon, Oct 6, 2008 at 7:39 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: I was going to write a blog once about the non-existent T.38 support in asterisk hence my purchase of the above domain. It expires in 10 days. T.38 support in asterisk still does not exist but I don't have any time. If someone wants this domain I will offer it for free and can send push it to your enom account since I was going to allow it to expire anyways. The only condition would be that you do not use it for a commercial use, i.e. you don't try to sell a t.38 module for asterisk. Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEL and swap from macros to contexts
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where people said, that
macros will be depracated, I tried to migrate from macros to contexts
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains,
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I
ignore this ael warnings? thanks
PJ
LOG: lev:3 file:pval.c line:2521 func: check_pval_item Warning: file
/etc/asterisk/extensions.ael, line 36-36: application call to Gosub
affects flow of control, and needs to be re-written using AEL if, while,
goto, etc. keywords instead!
Hi,
In definition use:
macro set_record(A,B) {
// do something
}
And for calling:
set_record(${CALLERID(NUM)},${EXTEN});
It will automatically be translated to GoSub in 1.6, but will remain
as Macro in 1.4.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] asteriskt38.com
On Mon, Oct 6, 2008 at 8:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote: That isn't real T.38 support, it's just Packet2Packet bridging that works correctly. Still need to use a Cisco gateway to support sending the faxes somewhere on the PSTN. But it does work and it is reliable, I use it every day. On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote: Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive. Hopefully it works. The one in CallWeaver doesn't. How do you mean - it doesn't? We currently use CallWeaver - Asterisk 1.4 - SIP Provider for sending and receiving faxes. Whenever we'll switch to 1.6, we plan to get rid of CallWeaver, as it has T.38 support in SendFax and ReceoveFax. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No reply to our critical packet
On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: The odd thing is on this particular phone it only happens when you call voicemail. It is certainly a bug in Asterisk, not the UA. Asterisk is trying to send to 192.168.1.x which obviously is not possible. Something in the NAT support is not working right. Hi, You should get SIP traces to see why Asterisk is trying to reply to 192.168.1.x. To do this, enter sip set debug on in asterisk CLI, and post us a log of call reaching voicemail and disconnecting. Regards, Atis On Mon, Oct 6, 2008 at 3:06 PM, SIP [EMAIL PROTECTED] wrote: This message is usually caused by Asterisk not receiving an ACK after about 30 seconds of attempts. There are countless misconfigured UAs and proxies out there that don't handle ACK well, so it would be nice to be able to turn this 'feature' off. What's annoying is that the explanation has always been If we can't get an ACK, we can't send any RTP data. This is patently false, as the RTP will often work fine even if ACK handling is misconfigured (we see it all the time). But alas. As far as I can tell, there's no way to disable this check. I suppose I could code around it, but not being the world's most proficient C coder, I'm always afraid I'll break something else. ;) N. Andrew Joakimsen wrote: I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 2 (Critical Response) [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. It seems to me that the problem is the way Asterisk is handling this critical packet -- of course it can not be sent to 192.168.1.54, the phone is at that IP behind a NAT and the Asterisk server is not. I can make any other phone call from this same phone as long as it is not voicemail and I can be on the line for hours with no problem. I am really at a loss here. I have searched a bit and come up with nothing other than blaming the UA. I know the Polycoms dont have the best NAT support but besides this it works problem-free. It's odd I can make a call anywhere else even for hours and not have any issues at all but 30 seconds into a voicemail call it just drops app5*CLI sip show peer 17865221569 app5*CLI * Name : 17865221569 Secret : Set MD5Secret: Not set Context : blended-lcr Subscr.Cont. : sla_stations Language : en AMA flags: Unknown Transfer mode: closed CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 17865221569 VM Extension : 14193016245 LastMsgsSent : 0/0 Call limit : 2 Dynamic : Yes Callerid : CENSORED MaxCallBR: 256 kbps Expire : 63 Insecure : no Nat : Always ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : Yes Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 74.CENSORED.213 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:[EMAIL PROTECTED] app5*CLI core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype
Re: [asterisk-users] Asterisk Queue question
On Thu, Oct 2, 2008 at 7:32 PM, voip crazy [EMAIL PROTECTED] wrote: When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy Fewest Calls working for a couple of mouths, and a new agent has been added this week in the queue and he is receiving all the incomings calls. How could I solve that? I do a nightly restart, however i suppose that module reload app_queue.so would do the trick :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we don't have software patents, they have been proposed to EU and reject few years ago. So by law - software is algorithm and can't be patented. In local laws we even are allowed to reverse-engineer software for needs of compatibility and interoperability. So, writing code for commercial codec and using it for interoperability with hardware devices (you purchased) is allowed by law. Damn, we even have a law that don't allow bittorrent trackers, as bittorrent file is considered breaking copyright law.. Ironic :p Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? I guess because it's completely legal here, and there's a disclaimer on page: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. It all depends on country and laws. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 1, 2008 at 10:12 AM, Steve Underwood [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we don't have software patents, they have been proposed to EU and reject few years ago. So by law - software is algorithm and can't be patented. In local laws we even are allowed to reverse-engineer software for needs of compatibility and interoperability. So, writing code for commercial codec and using it for interoperability with hardware devices (you purchased) is allowed by law. Damn, we even have a law that don't allow bittorrent trackers, as bittorrent file is considered breaking copyright law.. Ironic :p Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? I guess because it's completely legal here, and there's a disclaimer on page: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. It all depends on country and laws. There are a few algorithmic speedup patents around, what can accelerate codecs like G.729 and G.723.1, and which are purely software patents. Most of the relevant patents are *not* software patents. Don't confuse software patent with something running on a computer. Patents applicable to speech coding are perfectly valid in the vast majority of countries. Certainly in all the EU countries. It seems that this have been discussed numerous times. http://lists.digium.com/pipermail/asterisk-users/2004-October/058136.html Does anybody have some more legal experence with this? Any courts? Negotiations? NDA? :p From what i've found, there's an EU directive regarding software patents, but it's full of legal terms. Maybe anyone can comment? http://www.europarl.europa.eu/commonpositions/2005/pdf/c6-0058-05_en.pdf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 1, 2008 at 5:09 PM, Steve Underwood [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Wed, Oct 1, 2008 at 10:12 AM, Steve Underwood [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we don't have software patents, they have been proposed to EU and reject few years ago. So by law - software is algorithm and can't be patented. In local laws we even are allowed to reverse-engineer software for needs of compatibility and interoperability. So, writing code for commercial codec and using it for interoperability with hardware devices (you purchased) is allowed by law. Damn, we even have a law that don't allow bittorrent trackers, as bittorrent file is considered breaking copyright law.. Ironic :p Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? I guess because it's completely legal here, and there's a disclaimer on page: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. It all depends on country and laws. There are a few algorithmic speedup patents around, what can accelerate codecs like G.729 and G.723.1, and which are purely software patents. Most of the relevant patents are *not* software patents. Don't confuse software patent with something running on a computer. Patents applicable to speech coding are perfectly valid in the vast majority of countries. Certainly in all the EU countries. It seems that this have been discussed numerous times. http://lists.digium.com/pipermail/asterisk-users/2004-October/058136.html Does anybody have some more legal experence with this? Any courts? Negotiations? NDA? :p From what i've found, there's an EU directive regarding software patents, but it's full of legal terms. Maybe anyone can comment? http://www.europarl.europa.eu/commonpositions/2005/pdf/c6-0058-05_en.pdf You're back to talking about software patents again. People love to do that, in the same spirit that schoolboys cross their fingers in the hope it absolves them from something. Would you care to look through the patents which the G.729 patent pool licences, and try to find any software patents amongst them? Because it's one directive regulating software and mathematical/algorithmic patents. Personally, I don't use G.729 at all, i'm just curios about this. If you would point me, i would gladly take a look at this patent list, for now my searches were unsuccessful. I was also asking somebody for some legal experience with this, as theory and practical application of patent laws may differ :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable CDR?
On Mon, Sep 29, 2008 at 1:25 PM, Vincent [EMAIL PROTECTED] wrote:
Hello
I'm running Asterisk 1.4.21.2 on FreeBSD 6.3.
This part of extensions.conf...
;play a menu, and expect user to type any extension 1-4 or 9
exten = s,n,Wait(1)
exten = s,n,Background(main_menu)
exten = s,n,WaitExten(5)
exten = s,n,Hangup()
exten = _[1-49],1,AGI(convert_app.phpcli|${EXTEN})
... triggers this message:
-- Executing [EMAIL PROTECTED]:5] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:6] BackGround(Zap/1-1, main_menu) in
new stack
-- Zap/1-1 Playing 'main_menu' (language 'fr')
== CDR updated on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] AGI(Zap/1-1,
convert_app.phpcli|1) in new stack
I don't use CDR. Provided this will not have dire consequences, how
can I disable this?
in cdr.conf:
[general]
enable=no
You may also unload CDR modules. For this do:
ast-dev14*CLI module show like cdr
Module Description
Use Count
cdr_manager.so Asterisk Manager Interface CDR Backend 0
cdr_custom.so Customizable Comma Separated Values CDR 0
app_forkcdr.so Fork The CDR into 2 separate entities0
app_cdr.so Tell Asterisk to not maintain a CDR for 0
app_setcdruserfield.so CDR user field apps 0
func_cdr.soCDR dialplan function0
cdr_addon_mysql.so MySQL CDR Backend0
7 modules loaded
And add in modules.conf:
noload = cdr_csv.so
noload = cdr_odbc.so
noload = cdr_pgsql.so
noload = cdr_sqlite.so
noload = cdr_sqlite3_custom.so
for each module not used.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] realtime queue asterisk 1.6.0-beta5
On Wed, Sep 17, 2008 at 11:57 AM, Ralf Träskman [EMAIL PROTECTED] wrote: Hi I have enabled realtime queue in asterisk, but when i enter a queue i get this and then asterisk crashes. Any clues? -- Executing [EMAIL PROTECTED]:1] Answer(SIP/Ralf-08207de0, ) in new stack -- Executing [EMAIL PROTECTED]:2] Ringing(SIP/Ralf-08207de0, ) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/Ralf-08207de0, 2) in new stack -- Executing [EMAIL PROTECTED]:4] Queue(SIP/Ralf-08207de0, Kundservice) in new stack Segmentation fault Backtrace would tell it. To get backtrace, recompile Asterisk with DONT_OPTIMIZE and then load core file in gdb and launch bt full. For more info see doc/backtrace.txt in asterisk source directory. You can search for existing problems in bugs.digium.com or post here if unsure. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video on Hold?
On Thu, Sep 11, 2008 at 9:15 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Thu, 11 Sep 2008, Russell Bryant wrote: [EMAIL PROTECTED] wrote: Is the idea to switch to another video source or stay with the callers camera? An option for both would be nice. I could see a help desk placing a caller in que and a 1-2 min video coming on showing some simple video of how to hook it up. What I had in mind was to play a video stream that went along with the on hold audio. I was going to make it so if a video file was found with the same name as the audio file being played, it would play it. I rather naively tried this :) Well, Echo() was echoing back video and Record() was recording video and audio line this, and Playback() was playing it back, so ... Sounds fantastic :) Out of curiosity - what formats are supported? Any news on 3G video? I remember some time ago there was some weird application level support. Does chan_mobile supports video too? Would it be possible to have 3G adapter and interact with it? This just brings Asterisk to new level :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Executing dialplan after the call normaly ended
On Fri, Sep 12, 2008 at 2:35 PM, Gergo Csibra [EMAIL PROTECTED] wrote: Hi, The Dial command has the g option, voip-info.org says: If the g option is specified, and the called party hangs up before the calling party, then Dial continues execution at priority n+1. and this works well. But I need to continue the execution if the caller hangs up first too. What do I need to do? Search for h extension Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to disallow the native bridge between the two channel
On Wed, Sep 10, 2008 at 9:48 AM, bala krishnan [EMAIL PROTECTED] wrote:
Hi,
Thanks for your apt response.
I also tried by starting the recording before the dial command through AMI.
But in that scenario also, there are some issue such that both the streams
are not getting updated at the same time. Sometimes one stream takes nearly
2 seconds to getting updated. Here i require both the stream updation needs
to be happening at the same time. And this voice updation issue is not there
when i included the t option in Dial application.
No, what i mean - is to use dialplan command before you dial to
destination peer. For example:
Monitor(ulaw,/tmp/recording-${UNIQUEID},b);
Regards,
Atis
Kindly give your suggestion on this.
Asterisk version - 1.4.21.2
Thanks,
balasam.
On Tue, 09 Sep 2008 Atis Lezdins wrote :
On Tue, Sep 9, 2008 at 3:19 PM, bala krishnan [EMAIL PROTECTED]
wrote:
Hi,
The problem is, when i was starting the recording on zap channels
through AMI by Monitor command, always the out stream recorded as 0
bytes.
So that i did the searching and got the response that t option would
disallow the native bridging between the channels and you will get the
properly recorded out stream files. With that t option, now i am getting
the
properly recorded stream files.
What i would want is, instead of t option in Dial application, is there
any
other way to set this in the configuration file?
Hi,
This should probably belong to list, as I don't have all answers :)
As I understand, you don't want recording unless you start it by AMI.
Theoretically AMI action Monitor should destroy native bridge and
re-bridge call trough Asterisk. It could be a bug, or just
unimplemented feature.
So, i could suggest starting Monitor before Dial command into
temporary dir, and then erase unnecessary recordings.. That way, AMI
Monitor will simply override previous and you will get recording where
you want, and bridge won't be native.
Regards,
Atis
regards,
balasam
On Mon, 08 Sep 2008 Atis Lezdins wrote :
On Mon, Sep 8, 2008 at 11:39 AM, bala krishnan [EMAIL PROTECTED]
wrote:
Hi,
To disallow the native bridge between the zap channels, i enabled
the t
flag in the Dial application. But i dont want to allow the
callee/caller
to
transfer the call.
Why would you need this? It should just take media processing away
from your CPU.
Alternatively you can enable Monitor/MixMonitor, it should keep
Asterisk in media path.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
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Re: [asterisk-users] realtime queue reload
On Tue, Sep 9, 2008 at 1:22 PM, Thomas Winter [EMAIL PROTECTED] wrote: On Monday 08 September 2008 14:44, Atis Lezdins wrote: On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED] wrote: I dont have problem to make a reload by AMI. My questions was if module reload app_queue.so is the right way to do this, because whis reload I reload everything. Its fact that I have to do reload queue otherwise Asterisk did not load realtime database with new settings. Definitely not. Realtime should reload settings on every new call, and this is working for me on periodic_announce and periodic_announce_frequency. However this will work only for new calls, existing calls will have settings as loaded at their enter queue. My Asterisk version is 1.4.19, addons 1.4.6 If you enable debug 1 you should see in your full log: Hi, I have 1.4.21.2 and addons 1.4.7 If I do reload I have this: [Sep 9 12:10:12] DEBUG[4709] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metr ic FROM fileconf WHERE filename='queues.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category , var_name, var_val, id And I have found only this in debug file: [Sep 9 12:12:02] DEBUG[20173] app_queue.c: Queue test has no realtime members defined. No need for update Might be this is the reason, I do add agents with AMI QueueAdd. So changes in realtime queues.conf will not be read, I have to do reload. Oooh, so you have Static realtime. I think it isn't supposed to reload automatically. Go for Real Realtime - http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue or live with module reload :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to disallow the native bridge between the two channel
On Tue, Sep 9, 2008 at 3:19 PM, bala krishnan [EMAIL PROTECTED] wrote: Hi, The problem is, when i was starting the recording on zap channels through AMI by Monitor command, always the out stream recorded as 0 bytes. So that i did the searching and got the response that t option would disallow the native bridging between the channels and you will get the properly recorded out stream files. With that t option, now i am getting the properly recorded stream files. What i would want is, instead of t option in Dial application, is there any other way to set this in the configuration file? Hi, This should probably belong to list, as I don't have all answers :) As I understand, you don't want recording unless you start it by AMI. Theoretically AMI action Monitor should destroy native bridge and re-bridge call trough Asterisk. It could be a bug, or just unimplemented feature. So, i could suggest starting Monitor before Dial command into temporary dir, and then erase unnecessary recordings.. That way, AMI Monitor will simply override previous and you will get recording where you want, and bridge won't be native. Regards, Atis regards, balasam On Mon, 08 Sep 2008 Atis Lezdins wrote : On Mon, Sep 8, 2008 at 11:39 AM, bala krishnan [EMAIL PROTECTED] wrote: Hi, To disallow the native bridge between the zap channels, i enabled the t flag in the Dial application. But i dont want to allow the callee/caller to transfer the call. Why would you need this? It should just take media processing away from your CPU. Alternatively you can enable Monitor/MixMonitor, it should keep Asterisk in media path. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue reload
On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED] wrote: On Sunday 07 September 2008 21:49, Atis Lezdins wrote: On Sun, Sep 7, 2008 at 4:56 PM, Thomas Winter [EMAIL PROTECTED] wrote: is not work for periodic-announce-frequency and periodic-announce. An reload is necessary. Asterisk is 1.4.21.1 It shouldn't be necessary. However you can try queue show queuename from CLI, that would trigger reloading queue's settings. If it doesn't work, enable core set debug 1 and post output when executing reload. As for executing CLI commands, see manager action Command: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command I dont have problem to make a reload by AMI. My questions was if module reload app_queue.so is the right way to do this, because whis reload I reload everything. Its fact that I have to do reload queue otherwise Asterisk did not load realtime database with new settings. Definitely not. Realtime should reload settings on every new call, and this is working for me on periodic_announce and periodic_announce_frequency. However this will work only for new calls, existing calls will have settings as loaded at their enter queue. My Asterisk version is 1.4.19, addons 1.4.6 If you enable debug 1 you should see in your full log: [Sep 8 05:25:58] VERBOSE[27273] logger.c: -- Executing Queue(SIP/90139-c4014774, 22901|t|||300) [Sep 8 05:25:58] DEBUG[27273] app_queue.c: queue: 22901, options: t, url: , announce: , expires: 1220877058, priority: 0 [Sep 8 05:25:58] DEBUG[27273] res_config_mysql.c: MySQL RealTime: Everything is fine. [Sep 8 05:25:58] DEBUG[27273] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM queue_table WHERE name = '22901' [Sep 8 05:25:58] DEBUG[27273] res_config_mysql.c: MySQL RealTime: Everything is fine. [Sep 8 05:25:58] DEBUG[27273] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM queue_members WHERE interface LIKE '%' AND queue_name = '22901' ORDER BY interface [Sep 8 05:25:58] DEBUG[27273] res_config_mysql.c: MySQL RealTime: Everything is fine. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to disallow the native bridge between the two channel
On Mon, Sep 8, 2008 at 11:39 AM, bala krishnan [EMAIL PROTECTED] wrote: Hi, To disallow the native bridge between the zap channels, i enabled the t flag in the Dial application. But i dont want to allow the callee/caller to transfer the call. Why would you need this? It should just take media processing away from your CPU. Alternatively you can enable Monitor/MixMonitor, it should keep Asterisk in media path. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue reload
On Sun, Sep 7, 2008 at 4:56 PM, Thomas Winter [EMAIL PROTECTED] wrote: On Saturday 06 September 2008 21:47, Brian wrote: Hi Thomas, The queue definitions and its member list will be reloaded each time a caller joins the queue. So you don't need to reload it manually. Hi, is not work for periodic-announce-frequency and periodic-announce. An reload is necessary. Asterisk is 1.4.21.1 It shouldn't be necessary. However you can try queue show queuename from CLI, that would trigger reloading queue's settings. If it doesn't work, enable core set debug 1 and post output when executing reload. As for executing CLI commands, see manager action Command: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin AddQueueMember
On Wed, Sep 3, 2008 at 11:09 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: Just out of curiosity, where do you get this AddQueueMember syntax from? Here: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.c om /books/9780596510480.pdf page: 367 Oh so the VOIP Wiki is out of date! It's wiki, anyone can update it. Now, where should we go to for reliable Asterisk info then? asterisk-dev-mc*CLI show application AddQueueMember asterisk-dev-mc*CLI -= Info about application 'AddQueueMember' =- [Synopsis] Dynamically adds queue members [Description] AddQueueMember(queuename[|interface[|penalty[|options[|membername): Dynamically adds interface to an existing queue. If the interface is already in the queue and there exists an n+101 priority then it will then jump to this priority. Otherwise it will return an error The option string may contain zero or more of the following characters: 'j' -- jump to +101 priority when appropriate. This application sets the following channel variable upon completion: AQMSTATUSThe status of the attempt to add a queue member as a text string, one of ADDED | MEMBERALREADY | NOSUCHQUEUE Example: AddQueueMember(techsupport|SIP/3000) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue's
On Wed, Sep 3, 2008 at 9:42 AM, Tobias Ahlander [EMAIL PROTECTED] wrote: Date: Tue, 02 Sep 2008 18:08:52 +1200 From: Paul Crane [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Queue's To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Philipp Kempgen wrote: Tobias Ahlander schrieb: From: Mark Michelson [EMAIL PROTECTED] Tobias Ahlander wrote: Yes, I have autofill set in queues.conf. I suspect that this behaviour is because the Polycom phones I use have 2 lines. Has anyone used this function with polycom phones before? Also, my agents are Dynamic, perhaps this works better with Static agents? Here's my queues.conf (with commented lines deleted for easier reading): [general] autofill = yes monitor-type = MixMonitor [sales] strategy = rrmemory wrapuptime=15 Depending on which Asterisk version you are using, there was a bug in the queue application for some 1.4 releases where the autofill option would only be set properly if it were placed inside a queue. In other words, you may want to try putting autofill=yes inside the [sales] queue in your configuration. Also, if you're using a version of Asterisk 1.2, autofill is not a valid option and you'll be stuck with the behavior you're seeing. Unfortunately this didn't help at all... Anyone else has any tips? Is there a way to limit the polycom phones to only take one call from the Queue at the same time? Asterisk version running is 1.4.13 Maybe the phones have call-waiting enabled? Does it work if you remove the second line? Philipp Kempgen Try setting the call-limit to 1 in sip.conf as well as limitonpeer to yes. - -- Paul Crane Technical Support Officer VentureVoIP Ltd John Wickliffe House 265 Princes Street Dunedin Paul, This option doesn't help me that much. When I have it enabled, I can't put a call on hold and transfer it since Asterisk rejects usage limit to 1. You have to set it to any value, so that device state events are generated, so set it to 10 or 20 to have no actual limit. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play remote file
On Tue, Sep 2, 2008 at 11:44 AM, Pezhman Lali [EMAIL PROTECTED] wrote:
Dear,
do u have any idea to playback a remote file (with url address) ?
for example :
exten = _X.,1,playback(http://www.test.com/test.gsm;);
best
Mani
No direct way, however you can always download and play then (assuming
file is not huge and on other side of world):
exten = _X.,1,System(wget -o /tmp/test-${UNIQUEID}.gsm
http://www.test.com/test.gsm)
exten = _X.,2,Playback(/tmp/test-${UNIQUEID}.gsm)
I placed ${UNIQUEID} so that other call don't start to overwrite while
first one is playing (altough this should probably be well handled by
filesystem).
You may also try downloading in background, but this is a bit tricky,
you have to assume that download will go faster than playback, and i'm
not completely sure that it will work at all - you're welcome to test:
exten = _X.,1,System((wget -o /tmp/test-${UNIQUEID}.gsm
http://www.test.com/test.gsm))
exten = _X.,2,Playback(/tmp/test-${UNIQUEID}.gsm)
This will place wget on background executing and return immediately,
so playback can try handle it.
However more stable would be just mounting remote directory to local
with NFS, or even better push file from web server to Asterisk server.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] AgentCallbackLogin AddQueueMember
On Tue, Sep 2, 2008 at 6:07 PM, Krzysztof Zimnicki [EMAIL PROTECTED] wrote:
Hi
i have problem with AddQueueMember logic.
I need login Agent(Member) in asterisk.
use this option:
for example:
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
and now i want to call to this Agent:
exten = _1XX,1,Dial(Agent/${EXTEN:1})
call to 113 and asterisk should call to Agent = 13 on interface SIP/ekiga.
This doesn't work, How can i do this on Asterisk 1.4(not using
AgentCallbackLogin).
You can't dial Agent/ as it's not channel anymore (not sure that it
was even possible in 1.2), but just queue member name (displayed in
queue show). If you use Local channels for dialing members, you can
enable setinterfacevar=yes in queues.conf and then get MEMBERINTERFACE
variable to dial actual SIP device. The rest of mapping is up to you,
keep it in asterisk db or SQL.
However I must warn you that there is really not much use of Local
channels in queue for 1.4 as there's no state_interface. Queue will
always treat your members as available, and limiting calls with
GROUP_COUNT as in sample docs will create 1 channel for each queue
call per each member, just to find out that it's currently busy.
I would suggest to jump directly to 1.6 if it's new setup, you should
test stability anyway.. otherwise you may try out backport of
state_interface to 1.4 -
http://ftp.iq-labs.net/state_interface-1.4/asterisk-svn-1.4.19-state_interface_101578.patch
Btw, there is currently one very rare deadlock problem in
state_interface, however that shouldn't keep you away, as we have 2000
calls per day and we've seen it only once for half year. I hope it
will be fixed soon.. (putnopvut?)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] is shared_lastcall available in 1.4
On Tue, Aug 26, 2008 at 7:26 PM, Bob Pierce [EMAIL PROTECTED] wrote:
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote:
Are there any plans to back port this feature into upcoming 1.4
releases?
No, new features are added only in trunk, and released in next major
release (1.6).
So what would be involved in back porting this feature for our system?
Do I simply follow the diff from the link you provided and apply the
highlighted changes to the app_queue.c file in my Asterisk source
directory before recompiling?
Generally yes. There's a patch file you can download for automatic
patching, but in this case it doesn't work automatically. So you
manually have to look all pieces that doesn't merge. I already took a
look, and hardest part would be update_status function, because
Asterisk 1.6 uses astobj2 (ao2_lock, ao2_iterator_* and other
functions) to access queue list. You will have to rewrite this part
using old functions - you can see that in update_queues function:
AST_LIST_LOCK(queues);
AST_LIST_TRAVERSE(queues, q, list) {
ast_mutex_lock(q-lock);
If you doubt about some part, you're welcome to ask, i'll try to help
you, but i don't want to provide complete backport to you, as i won't
be able to test it :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Fax issue over cisco gateway
On Wed, Aug 27, 2008 at 6:20 PM, Enrico Pasqualotto [EMAIL PROTECTED] wrote: Hi all, I'm trying to send fax from Hylafax to a remote fax machine through Asterisk and cisco 2801 as E1 gateway. This is my architecture: sendfax - HylaFax - iaxmodem - Asterisk - (SIP) 2801 with E1 card For incoming fax I don't have any problem, but I'm not able to send fax out of 2801. My router conf: dial-peer voice 1 pots destination-pattern .T fax rate disable port 0/2/0:15 ! dial-peer voice 3 pots incoming called-number 53T fax rate disable direct-inward-dial forward-digits all ! ## In asterisk console I see a lot of RTP packets lost: RTP-stats-003*CLI * Our Receiver: SSRC: 642188040 Received packets: 17463 Lost packets: 19686 Jitter:0. Transit: 0. RR-count: 0 * Our Sender: SSRC: 1469234407 Sent packets: 27926 Lost packets: 0 Jitter:0 SR-count: 112 RTT: 0.00 Anyone have idea of this problem? The packet lost quantity is normal? Hi Enrico, In general SIP part is not good at all for fax transmission. However if it's directly connected with no other packets flying by (and having impact on bandwith/latency), you may have high degree of success by using non-compressed codecs (for example G.711). Could you provide SIP debug? I've seen that some switches automatically listen for fax tones, and send T.38 handshake whenever they detect fax on line. Looking into specs, says me that 2801 supports T.38, so perhaps it could be better idea (altough you would have to use Asterisk 1.6 and app_txfax for sending faxes) Also Hylafax log could say something. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is shared_lastcall available in 1.4
On Tue, Aug 26, 2008 at 5:14 PM, Bob Pierce [EMAIL PROTECTED] wrote: On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote: I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, the shared_lastcall option is only in versions 1.6.0 and up. Does anybody have a workaround for this in 1.4? Or maybe a better question: How stable is 1.6 for production use? I'd say - go for backport instead. shared_lastcall is commited in http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985 and it seems that there are no bugfixes for it since. So, backporting should be fairly simple. Also i would suggest subscribing to asterisk-svn and watch for commits to app_queue to not miss any bugfixes to it. Migration to 1.6 could be more time consuming, as there are lot of changes, you will probably have to adjust dialplan, etc. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is shared_lastcall available in 1.4
On Tue, Aug 26, 2008 at 5:39 PM, Bob Pierce [EMAIL PROTECTED] wrote: On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote: I'd say - go for backport instead. shared_lastcall is commited in http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985 and it seems that there are no bugfixes for it since. So, backporting should be fairly simple. Also i would suggest subscribing to asterisk-svn and watch for commits to app_queue to not miss any bugfixes to it. Are there any plans to back port this feature into upcoming 1.4 releases? No, new features are added only in trunk, and released in next major release (1.6). Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing callerID in a context
On Thu, Aug 21, 2008 at 3:11 PM, Andy Dixon [EMAIL PROTECTED] wrote:
Hello,
I am trying to alter the outbound callerID for extensions within a
context I have created.
I wrote the following:
exten = _9.,2,ExecIf($[$[${REALCALLERIDNUM} = 360] | $[$
{REALCALLERIDNUM} = 670]]|Set|CALLERID(num)=581560)
exten = _9.,3,ExecIf($[$[${REALCALLERIDNUM} = 361] | $[$
{REALCALLERIDNUM} = 671]]|Set|CALLERID(num)=581561)
exten = _9.,4,ExecIf($[$[${REALCALLERIDNUM} = 362] | $[$
{REALCALLERIDNUM} = 672]]|Set|CALLERID(num)=581562)
exten = _9.,5,ExecIf($[$[${REALCALLERIDNUM} = 363] | $[$
{REALCALLERIDNUM} = 673]]|Set|CALLERID(num)=581563)
exten = _9.,6,ExecIf($[$[${REALCALLERIDNUM} = 364] | $[$
{REALCALLERIDNUM} = 674]]|Set|CALLERID(num)=581564)
exten = _9.,7,ExecIf($[$[${REALCALLERIDNUM} = 365] | $[$
{REALCALLERIDNUM} = 675]]|Set|CALLERID(num)=581565)
exten = _9.,8,ExecIf($[$[${REALCALLERIDNUM} = 366] | $[$
{REALCALLERIDNUM} = 676]]|Set|CALLERID(num)=581566)
exten = _9.,9,ExecIf($[$[${REALCALLERIDNUM} = 367] | $[$
{REALCALLERIDNUM} = 677]]|Set|CALLERID(num)=581567)
exten = _9.,10,ExecIf($[$[${REALCALLERIDNUM} = 368] | $[$
{REALCALLERIDNUM} = 678]]|Set|CALLERID(num)=581568)
exten = _9.,11,ExecIf($[$[${REALCALLERIDNUM} = 369] | $[$
{REALCALLERIDNUM} = 679]]|Set|CALLERID(num)=581569)
exten = _9.,12,ExecIf($[$[${REALCALLERIDNUM} = 700] | $[$
{REALCALLERIDNUM} = 701]]|Set|CALLERID(num)=581557)
exten = _9.,13,ExecIf($[$[${REALCALLERIDNUM} = 100] | $[$
{REALCALLERIDNUM} = 101]]|Set|CALLERID(num)=581500)
This *should* change the callerID for (for example) 700 and 701 to be
581557, and any extensions not listed above, it should leave them alone.
If I call from extension 666, I get the correct outbound number (as it
does exist), but the rules above are not being followed.
I have tried to use Set(CALLERID(num)=581500) which works okay
slightly further down.
I am aiming for any numbers starting with a 9 to follow the rules
above, and then to follow a further rule (eg if the number starts 901,
or 907)
I'm stuck.. If anyone could help, I would be eternally grateful..
Are you sure ${REALCALLERIDNUM} is set? Alternatively (to AEL) there's
a way how to simplify all this, by using Asterisk extension patterns:
[clid-mangle]
exten = 70[01],1,Set(CALLERID(num)=581557)
exten = 70[01],2,Return()
exten = 10[01],1,Set(CALLERID(num)=581500)
exten = 10[01],2,Return()
; and so on, just better reorganize your extensions so that this can
match patterns better.
[dial-out]
exten = _9.,1,GoSub(clid-mangle,${CALLERID(num)},1)
exten = _9.,2,Dial(SIP/provider)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] US-based echo test servers?
On Mon, Aug 18, 2008 at 8:09 PM, emist [EMAIL PROTECTED] wrote: Hey Nikhil, I have some free time right now and would be willing to set this up for you. Just paypal me $1 for the DID(globalpops fee) and like 50 cents for minutes. Whatever is left over after you're done testing I can refund to you. Oh come on, do you give toll free number or what? Btw i just assigned DID to Echo() app, so Nikhil has what he asked for. And for free :p Regards, Atis Regards, Igor H. Nikhil Nair wrote: Hi, I'm running a small Asterisk server in the UK, just for personal use. I've been experimenting with various VoIP providers for international calls to PSTN numbers, particularly to the US (often California). My results, to date, have been very variable indeed, so much so that I'm considering getting a suitable card and using the PSTN. I have found a VoIP provider with an excellent reputation, and it gives very good quality. However, I seem to get quite a bit of delay at times, enough to make conversation awkward. As the setup at the far end was not completely trivial, I'm not 100% sure the problem was in my connection, but I'd like to test that. Are there any US numbers I can call to get an Asterisk-style echo test? Ideally, a California-based numnber, so I can try to call it from an ordinary PSTN phone here, and compare calling via VoIP, and see if there's an appreciable difference in the delay/quality. I don't anticipate using this for very long, so it doesn't necessarily need to be a free service. Failing that, does anyone have access to a US-based Asterisk server which would allow me to make connections to its echo test? Presumably, if I had this, I could rent a PSTN number from a US-based provider, and point it to the appropriate SIP/IAX address. I expect my total usage would be just a few minutes, though having the facility available for a few weeks would be helpful, to allow me to play around with various options. Again, I'd be willing to pay a modest amount for this. Thanks in advance for any suggestions! Best wishes, Nikhil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stress call test
On Fri, Aug 15, 2008 at 8:31 AM, aby azid [EMAIL PROTECTED] wrote: Hi everyone, I'm required to make a stress call on Asterisk server ( 2000 calls per seconds). Are there tools for me to do this sort of test. I was thinking of sending loads of Asterisk call files simultaneously (starting with 100 call files). Really appreciate if anyone can come up with ideas or tools for me to achieve this. Hi, I've written test framework, you'll need another machine with Asterisk (+php) on it to generate calls. It allows to write scripts in PHP to emulate random customer actions, etc.. You can download it here http://ftp.iq-labs.net/pbx-test/ If you find it useful, or get into some problems, don't hesitate to write me. If you need just bunch of identical calls, you may also try out SIPp. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List?
On Fri, Aug 15, 2008 at 8:56 AM, Karl Fife
[EMAIL PROTECTED] wrote:
Does anyone know enough about the implementation of AstDB to know
whether the data structure is a Hash function, a Balanced-Tree, a
b-Tree, or a Linked List?
I'm trying to estimate the lookup 'cost' of a AstDB with around 160,000
keys? Obviously I already know that it WILL WORK, but the question is
whether the data structure is optimal in the Berkeley DB AS IMPLEMENTED
in Asterisk. AstDB just like CURL is missing some of its features as
implemented, so the generic Berkeley Doc doesn't help much.
The key-space is ideal. It's just npa/nxx lookups so it's UNIQUE and
EVENLY DISTRIBUTED--a perfect for a hash function (or even a balanced
tree). What I do NOT want is a 150k member linked list, or even a
standard b-tree that ends up being 160k entries tall because the values
were inserted in order etc.
In terms of Databases, I know that 160K keys is very small potatoes, but
I want to make sure I understand what's going on under the hood so that
my lookup costs are as low as possible.
Lookups on my Oracle database with tens of millions of records are
instantaneous and inexpensive when implemented properly, and about ten
thousand times slower (actually) when done improperly. If a database
has only 160K keys, it can be tough to tell whether the data structures
are being used efficiently, because even WILDLY INEFFICIENT lookups seem
fast.
Can anyone speak to this?
What is the default data structure?
How many records have you stuck into the AstDB?
Hi,
I can share my experience only. I'm storing call variables in astdb,
daily ~2000 calls, ~20 variables per call. Stored in way like this:
call_variables/${uniqueid}/var=value
Previously i did cleanup of this only nightly, but sometimes i noticed
that lookup in evenings get extremely slow - ~5-10 seconds per
variable. So, i would guess that it's not very optimal, as every
lookup from 2000 keys takes several seconds (sub-keys shouldn't affect
this).
Anyway, my solution was to keep it small by deleting everything upon
hangup of call. A little clutter, but it works more or less.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Asterisk 1.4 T38 UDPTL Pass Through MAX TNT and Linksys 2102
On Thu, Aug 14, 2008 at 4:46 AM, JR Richardson [EMAIL PROTECTED] wrote: Hi All, I finally got the time to test t38 pass through with a TNT, * 1.4.21.1 and Linksys 2102: PRI TNT SIP Asterisk 2102 SharpFax Faxing either direction, the call sets up with ulaw rtp, when fax tones hit the line, both the TNT and the 2102 switch to t38 and udptl packets fly through Asterisk. All looks good, but, once udptl sets up, every few seconds, I get a warning: 'rtp Read too short' on the Asterisk CLI from the TNT side of the session. Faxes never complete, not even a half page, nothing, transmission just ends. There are only a few parameters on the TNT that effect t38 and I've adjusted them all with no change in the results. Pretty much the same results when testing t38 pass through to a Cisco pri gateway as well. So my question is: Does anyone else have this solution working and wouldn't not mind sharing configs? Hi, I have T38 passtrough working in following configuration: Callweaver - Asterisk - SIP proxy (provider) - provider's switch. I never had received such errors, i suspect it must be problem of one of those devices (sending bad packets), so perhaps you can check them by skipping Asterisk, and also try connecting another different device (i.e. Callweaver) to any of endpoints. Alternatively you can try to get Asterisk out of UDPTL path, by enabling re-invites. As for configuration, i had added only t38pt_udptl=yes in [general] section and peer section and everything worked. my sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default-external tos_sip=0x18 tos_audio=0x18 callerid = Unknown dtmfmode=rfc2833 registertimeout=200 ignoreregexpire=no limitonpeer=yes notifyringing=no notifyhold=no allowsubscribe=yes rtcachefriends=yes t38pt_udptl=yes [callweaver] type=friend host=127.0.0.1 permit=127.0.0.1 context=callweaver_out port=7060 allow=all canreinvite=no t38pt_udptl=yes ; note - SIP provider don't have entry, it's dialed by IP. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX t.38 on Asterisk 1.6?
On Fri, Aug 8, 2008 at 1:43 AM, Arturo Ochoa [EMAIL PROTECTED] wrote: Thanks Memo, I've already see that article before, the problem is that this solution is useful when you want asterisk (via t38modem) to terminate the call... Someone send you a Fax using t.28 and this software (t38modem+asterisk+hylafax) will handle the incomming fax. In fact I have a working installation of Iaxmodem+asterisk+hylafax working. The problem seems to be on the implementation of some application to handle the t38 gateway capability of the Asterisk Server. I've read this article http://bugs.digium.com/view.php?id=12931 but it's closed. Ooh, this is cool thing. However i'm afraid because of the mentioned performance issues.I wonder why it was closed, there was a patch for trunk, if some developer would help making it stable - it would be great addition. It would allow simple Hylafax-Iaxmodem-Asterisk-SIP-T38 gateway implementation. Last week i've been trying to get up and running T38modem, and most what i can get out of it is the following setup: Hylafax - T38modem - Asterisk - T38modem - Hylafax. Any attempt to send calls to provider fails. I've been going to other front and tried CallWeaver, it really can send T38 faxes to provider, so i was thinking of implementing virtual modem in Hylafax to send calls trough CallWeaver (CallWeaverModem :D). Regards, Atis Any Ideas? Ing. Arturo Ochoa N Electrosystems S RL Tel. (656)-6230794 -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Guillermo Salas M. Enviado el: Thursday, August 07, 2008 2:00 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] FAX t.38 on Asterisk 1.6? El jue, 07-08-2008 a las 13:31 -0600, Arturo Ochoa escribió: Has anyone have experiencies on this kind of scenario... what version?.. patches?... or any information regarding this goal will be VERY helpful... Hi Arturo, Please ckeck the following URL (on spanish): http://www.sinologic.net/2008-07/como-configurar-un-fax-virtual-t38-en-aster isk/ Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX t.38 on Asterisk 1.6?
On Thu, Aug 7, 2008 at 11:00 PM, Guillermo Salas M. [EMAIL PROTECTED] wrote: El jue, 07-08-2008 a las 13:31 -0600, Arturo Ochoa escribió: Has anyone have experiencies on this kind of scenario... what version?.. patches?... or any information regarding this goal will be VERY helpful... Hi Arturo, Please ckeck the following URL (on spanish): http://www.sinologic.net/2008-07/como-configurar-un-fax-virtual-t38-en-asterisk/ Oh, nice translation. As i don't read spanish - i have to ask about the comment - does it says anything about making this work on real SIP trunk? I've been unsuccessful on that part. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with iaxmodem!
On Wed, Aug 6, 2008 at 4:05 PM, [EMAIL PROTECTED] wrote: Hello, I would like to configure hylafax(4.4.4) + iaxmodem(1.1.1). I use Asterisk and I work on Redhat. I installed the two hylafax and iaxmodem. My configuration of iaxmodem is: (in the file /etc/iaxmodem/ttyIAX0) device /dev/ttyIAX0 owner uucp:uucp mode 660 port 4570 #each line should have it's own port number! refresh 300 server 127.0.0.1 peername IAXmodem #this is the local extension number in FreePBX (create it!) secret 12345 #password for the extension cidname Fax1 cidnumber codec ulaw I added this two lines in /etc/inittab IA:2345:respawn:/usr/local/bin/iaxmodem ttyIAX0 mo:2345:respawn:/usr/sbin/faxgetty -D ttyIAX0 Then I tried to configure hylafax whith the command faxsetup I meet a problem when I want to add a modem with the command faxaddmodem but I can't, I have this response: Serial port that modem is connected to []? ttyIAX0 /dev/ttyIAX0 is not a terminal device. In fact in /dev I don't find ttyIAX0 I added the line /usr/sbin/faxgetty -D /dev/ttyIAX0 in the file /etc/rc.d//rc.local and I tried the command faxgetty -D /dev/ttyIAX0 but nothing!!! I tried the command /usr/local/bin/iaxmodem ttyIAX0 I have the following response: [EMAIL PROTECTED] /usr/local/bin/iaxmodem ttyIAX0 [2008-08-05 17:39:27] Modem started [2008-08-05 17:39:27] Setting device = '/dev/ttyIAX0' [2008-08-05 17:39:27] Setting owner = 'uucp:uucp' [2008-08-05 17:39:27] Setting mode = '660' [2008-08-05 17:39:27] Setting port = 4570 [2008-08-05 17:39:27] Setting refresh = 300 [2008-08-05 17:39:27] Setting server = '127.0.0.1' [2008-08-05 17:39:27] Setting peername = 'IAXmodem #this is the local extension number in FreePBX (create ' [2008-08-05 17:39:27] Setting secret = '12345 #password for the extension' [2008-08-05 17:39:27] Setting cidname = 'Fax1' [2008-08-05 17:39:27] Setting cidnumber = '' [2008-08-05 17:39:27] Setting codec = ulaw [2008-08-05 17:39:27] Opened pty, slave device: /dev/pts/17 [2008-08-05 17:39:27] Removed old /dev/ttyIAX0 [2008-08-05 17:39:27] Created /dev/ttyIAX0 symbolic link [2008-08-05 17:39:27] Registration failed. I don't unerstand why iaxmodem can't register . If someone has an idea, he is welcome!! Thank you What's your iax.conf? For me modem configuration looks like this: [iaxmodem5] type=friend host=dynamic secret=x context=fax permit=127.0.0.1 allow=all P.S. after editing inittab, you also have to execute # kill -HUP 1 So that init process re-reads configuration. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Penalties not working properly
On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? rrmemory isn't ringall, it won't ring all members. But yes - you can use ringall with penalties. My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? What you are seeing is caused by status NOT IN USE. You have to set call-limit in sip.conf for all your phones, to any value, so that device states work correctly, and queue can know that those phones are busy. Now you probably can see in CLI that queue is sending second call to first agent(s). Regards, Atis Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream RS-232 config (slightly off-topic)
On Tue, Aug 5, 2008 at 6:29 PM, Vieri [EMAIL PROTECTED] wrote: Never mind. I set the wrong baud rate. The right values are 115200-8-N-1-No flow control. However, the serial connection is as good or as useless as the telent connection. I have no way to restore factory settings. --- On Tue, 8/5/08, Vieri [EMAIL PROTECTED] wrote: I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone unconfigurable via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the keypad update feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited and I can't change SIP configuration or reset to default values (and see if that helps to bring the HTTP back up). I did try to upgrade and downgrade the firmware with no change at all. The reset button reboots the device but doesn't restore default values! (I tried keeping it pressed for several minutes...) So my last chance before throwing it away is to administer it via serial port. The thing is that Grandstream's official manual says that there is an RS232 serial port for administration but it doesn't say anything else about it (how to connect, how to change config, how to reset the device, etc). There's absolutely nothing regarding RS-232. If someone has this or a similar device and accessed it via serial port then I'd greatly appreciate some quick tips. Thanks, Vieri Have you tried powering it on, while holding reset button? Additionally you can try to leave it for week powered off and hope that there's some old battery keeping up settings. Are you sure that there isn't some enable admin mode command in telnet? It should allow you everything that's available from web. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customized Queuing Strategy
Sorry for previous blank answer :)
On Mon, Aug 4, 2008 at 1:20 PM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Hi Thanks ALL for reply,
If I use cascading queue will it do the trick?? The only problem is (as
mentioned in below example) if a call enters testq and get answered then
after hungup at the agent end only will the call will again enter the
next queue which is testq2 as in this example.??
Check the QUEUESTATUS variable: http://www.voip-info.org/wiki-Asterisk+cmd+Queue
Moreover if I keep penalty 1 for all the first 5 agents and penalty 2 or
higher for all the next 5 agents and implement ringall strategy will it
do the same effect??
Yes
exten = 1589,1,Answer
exten = 1589,2,Ringing
exten = 1589,3,Wait(2)
exten = 1589,4,Queue(testq|t|||45)
if (${QUEUESTATUS=) Hangup();
exten = 1589,5,Queue(testq2|t|||45)
exten = 1589,6,Hangup
Regards,
Atis
thanks in advance.
Syed nasr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Monday, August 04, 2008 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Customized Queuing Strategy
Syed Nasruddin wrote:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5
agents,
firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next 5 Agents.
Set up two queues. Call Queue() on the first queue - corresponding to
#1 - with a rather strict timeout. Fall back on the second queue.
More sophisticated strategies require either the modification of the
source code for app_queue, or custom queue implementation in AGI.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] skype and Asterisk opensource integration
On Mon, Aug 4, 2008 at 10:58 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all except of some commercial hardware / software gateways, is there any opensource or free project to setup a Skype Account on Asterisk? The only one known to the moment is chan_celliax, which is originally for connecting to cell phones by cable, however it supports also skype (just 1 account). It will launch fake X server and original skype, and communicate with it. http://www.celliax.org/ Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customized Queuing Strategy
On Mon, Aug 4, 2008 at 1:20 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Hi Thanks ALL for reply, If I use cascading queue will it do the trick?? The only problem is (as mentioned in below example) if a call enters testq and get answered then after hungup at the agent end only will the call will again enter the next queue which is testq2 as in this example.?? Moreover if I keep penalty 1 for all the first 5 agents and penalty 2 or higher for all the next 5 agents and implement ringall strategy will it do the same effect?? exten = 1589,1,Answer exten = 1589,2,Ringing exten = 1589,3,Wait(2) exten = 1589,4,Queue(testq|t|||45) exten = 1589,5,Queue(testq2|t|||45) exten = 1589,6,Hangup thanks in advance. Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Monday, August 04, 2008 1:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Customized Queuing Strategy Syed Nasruddin wrote: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 4. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Set up two queues. Call Queue() on the first queue - corresponding to #1 - with a rather strict timeout. Fall back on the second queue. More sophisticated strategies require either the modification of the source code for app_queue, or custom queue implementation in AGI. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customized Queuing Strategy
On Mon, Aug 4, 2008 at 2:59 PM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Dear Atis,
I am running in to syntax problem. Sorry only beginner level experience
of conditional checking:
Yes, sorry for that, i just wrote it quickly and didn't checked
expression. Also, i didn't wrote in .conf format, as it's been a long
time since i wrote that.
exten = 1589,1,Answer
exten = 1589,2,Ringing
exten = 1589,3,Wait(2)
exten = 1589,4,Queue(testq|t|||45)
if (${QUEUESTATUS=) Hangup(); since I want to hangup if the caller
has already been catered by an agent and the caller hasnt hanged up, so
what status value should I look for. Moreover syntax of above
conditional statement is complete or something missin:
Exactly, if call has been handled by agent, QUEUESTATUS will be empty.
Otherwise it will be LEAVEUNAVAIL or something like that (not empty)
if (${QUEUESTATUS=) Hangup();
if above condition fails then the control must move to below lines
rather then getting hanged up.
ok, i'll try:
exten = 1589,5,GotoIf($[${QUEUESTATUS}=]?exit)
exten = 1589,5,Queue(testq2|t|||45)
rename to priority 6
exten = 1589,6,Hangup
rename to priority 7 and add label exit:
exten = 1589,7(exit),Hangup
But as said before, you can also use penalties of members. Next
penalty is only chosen if nobody with smallest penalty can't be
dialed. Plus, there will also be advantage that if you dial member for
15 seconds, and at first there is noone with penalty 1 available -
queue will call somebody with penalty 2. Then, if dialed member(s)
don't answer, queue will again try somebody with penalty 1 first.
Regards,
Atis
Thanks
Syed nasr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis
Lezdins
Sent: Monday, August 04, 2008 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Customized Queuing Strategy
Sorry for previous blank answer :)
On Mon, Aug 4, 2008 at 1:20 PM, Syed Nasruddin [EMAIL PROTECTED]
wrote:
Hi Thanks ALL for reply,
If I use cascading queue will it do the trick?? The only problem is
(as
mentioned in below example) if a call enters testq and get answered
then
after hungup at the agent end only will the call will again enter the
next queue which is testq2 as in this example.??
Check the QUEUESTATUS variable:
http://www.voip-info.org/wiki-Asterisk+cmd+Queue
Moreover if I keep penalty 1 for all the first 5 agents and penalty 2
or
higher for all the next 5 agents and implement ringall strategy will
it
do the same effect??
Yes
exten = 1589,1,Answer
exten = 1589,2,Ringing
exten = 1589,3,Wait(2)
exten = 1589,4,Queue(testq|t|||45)
if (${QUEUESTATUS=) Hangup();
exten = 1589,5,Queue(testq2|t|||45)
exten = 1589,6,Hangup
Regards,
Atis
thanks in advance.
Syed nasr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Monday, August 04, 2008 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Customized Queuing Strategy
Syed Nasruddin wrote:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5
agents,
firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next 5 Agents.
Set up two queues. Call Queue() on the first queue - corresponding to
#1 - with a rather strict timeout. Fall back on the second queue.
More sophisticated strategies require either the modification of the
source code for app_queue, or custom queue implementation in AGI.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] finding out on hold channels
On Fri, Jul 25, 2008 at 2:59 AM, Al lists [EMAIL PROTECTED] wrote: I noticed that i' m not getting any manager event for hold and unhold of a channel. is this normal? Also is there any easy way through either CLI or manager to find out which one of the channels are on hold? I checked show channels that did not show a channel being on hold or not, also sip show channels does show that but it has call id instead of channel id. Hi, There was recently a thread regarding this on asterisk-dev (http://lists.digium.com/pipermail/asterisk-dev/2008-June/033466.html). There was message explaining how to do this by adding custom code to Asterisk sources, and I guess it could be already done in trunk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QueueMemberStatus
On Wed, Jul 9, 2008 at 12:00 AM, Jason Dixon [EMAIL PROTECTED] wrote: On Tue, Jul 08, 2008 at 11:00:43AM -0400, Jason Dixon wrote: On Tue, Jul 08, 2008 at 12:10:05PM +1200, Matt Riddell wrote: Action: Command Command: show queue my_queue_name ActionID: my_queue_name_12345 This does not appear to show the correct status of an extension. It appears that ExtensionState also always reports Status of -1. Are there any Actions or Commands that will report the correct status of an extension? So far the only accurate representation I've found of queue members has been the following. $ sudo /usr/sbin/asterisk -r -x show channels | grep '^SIP' SIP/241-b742e010 [EMAIL PROTECTED]:2Ring Dial(Zap/G1/411) $ sudo /usr/sbin/asterisk -r -x show queue support_queue | grep SIP SIP/207 (Ringing) has taken no calls yet SIP/203 (Not in use) has taken no calls yet SIP/202 (In use) has taken no calls yet SIP/201 (Not in use) has taken no calls yet All of the commands I've tried via the AGI have yielded incorrect results. If this sounds wrong, please let me know and I'll resume beating my head against the nearest wall. :) There is QUEUE_MEMBER_COUNT (in 1.4) and QUEUE_MEMBER (in 1.6) dialplan functions which allows to get count of members (in 1.6 also count of free / logged in members). You can use GetVar to evaluate that. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?
On Sun, Jun 29, 2008 at 7:02 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Sherwood McGowan wrote: Gentlemen, I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime. This system works fine with 1.2.28, and everything loads fine with no errors, but when I log an agent in I see the extra message (not in use) by their listing and they are not rang by asterisk when their queue is called. Any ideas? Nobody else? Have you checked call-limit and state information for SIP peers? That was changed between 1.2 and 1.4, and could affect queue state. See the UPGRADE notes. Otherwise You'll have to set core set debug 2 and core set verbose 3, and post full log (debug+verbose) where agents got logged in (if you have also realtime members, just execute queue show on CLI. Then you'll have to give one call to agent, talk for little and disconnect. Then just post that log here. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
On Thu, Jun 26, 2008 at 10:21 PM, Steve Murphy [EMAIL PROTECTED] wrote: On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote: On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote: This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! Hi murf, From some preliminary testing on the CDRfix4 branch it looks like the CDRs for attended transfers are now correct which is fantastic. For blind transfers the CDR for the first call leg is still incorrect with the duration only being recorded up until the point the transfer occurs. I did a blind xfer with my snom360, and got these two cdrs with **TRUNK**: Eventlist: 1. 101 dahdi (used to be zap) phone picked up and 200 is dialed for the snom360 2. 200 (snom360) picks up and answers the call 3. 200 (snom360) hits the Transfer button (101 gets MOH), dials 202 4. 200 (snom360) hits the checkmark button to send off the call (101 starts hearing ringing, 200 starts getting congestion). 5. 202 (eyebeam) answers (101 202 are connected) 6. 101 or 202 hang up. Conversation finished. fxs.01 101,101,200,extension,DAHDI/1-1,SIP/snom360-082c3f68,Dial,SIP/snom360,30,2008-06-26 11:04:08,2008-06-26 11:04:12,2008-06-26 11:05:56,108,104,ANSWERED,DOCUMENTATION,,1214499848.11,, fxs.01 101,101,201,extension,DAHDI/1-1,SIP/murf-eyebeam-082d95d8,Dial,SIP/polycom430SIP/murf-eyebeam,30,2008-06-26 11:06:06,2008-06-26 11:06:12,2008-06-26 11:06:56,50,44,ANSWERED,DOCUMENTATION,,1214499966.13,, Here are the two CDR's with their recorded event times: CDR start answer end 112 3 245 6 above, I called into the snom360, and hit the Transfer button, dialed 201, and got congestion (101 gets moh until I hit the check key), and hung up the snom (200). 201, the eyebeam, rings, I answer. 101 and 201 are connected. 101 hangs up, and the conversation ended. THE SAME PROCEDURE ON THE CDRfix6 branch: fxs.01 101,101,200,extension,DAHDI/1-1,SIP/snom360-0829e2d0,Dial,SIP/snom360,30,Tt,2008-06-26 12:16:37,2008-06-26 12:16:44,2008-06-26 12:17:01,24,17,ANSWERED,DOCUMENTATION,,1214504197.4,, fxs.01 101,101,202,extension,DAHDI/1-1,SIP/murf-eyebeam-082c2b70,Dial,SIP/murf-eyebeam,30,Tt,2008-06-26 12:17:01,2008-06-26 12:17:14,2008-06-26 12:17:49,48,35,ANSWERED,DOCUMENTATION,,1214504197.4,, CDR start answer end 112 4 245 6 Well, time 3 does get lost, but I thought it might be nice to be able to link 1 2 by the coincident times and say, hey, that looks like a blind transfer! One point of dissatisfaction I have with these is the fact that SIP/snom dialed the second CDR, not DAHDI/1. But, if I change it, you won't know that DAHDI/1 was the guy that murf-eyebeam was talking to... tough choices. So, I take it from your above words, that you'd like the 1,2,3; 4,5,6; times on the two CDR's? Can anyone lab this up for 1.2; I don't have enough phones, and I'm not eager to reconfigure the ones I've got for just one test ! I wonder how is this reflected in cdr_addon_mysql. It would show just duration and billsec (at least for 1.4), so i would defineately want this 1 second between 3 and 4 to show up in some record (preferrably in second CDR, as it's not talking time with first user anymore). Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
On 6/26/08, Grey Man [EMAIL PROTECTED] wrote: On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote: This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! Hi, I just wanted to say that we are working on testing our current functionality. We don't use attended transfers, but would like at some point. So, I'll try to report within next week if something else is broken. Hi murf, From some preliminary testing on the CDRfix4 branch it looks like the CDRs for attended transfers are now correct which is fantastic. For blind transfers the CDR for the first call leg is still incorrect with the duration only being recorded up until the point the transfer occurs. What's wrong with that? This fits perfectly for my needs. Is there a way how to exploit this? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI show queues NOT WORKING WELL
On Thu, Jun 19, 2008 at 10:06 PM, Chento Arohuanca [EMAIL PROTECTED] wrote: Just about 30 minutes that I can´t get real information from my Asterisk box. All agents seem to be available but is not true: QUEUE_01 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet [EMAIL PROTECTED] asterisk]# asterisk -rx core show channels Channel Location State Application(Data) SIP/641-08cef808 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/641|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/65-1) Zap/65-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/64-1 [EMAIL PROTECTED]: Up (None) SIP/625-09766788 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/625|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/66-1) Zap/66-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_02|tT|||1800) SIP/620-09358088 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/620|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/63-1) Zap/63-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/94-1 (None) Up Bridged Call(SIP/623-b2b1d070) SIP/623-b2b1d070 [EMAIL PROTECTED]:3 Up Dial(Zap/g3/2714269||tTrRS) SIP/615-08a892c0 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Please help me with this issue! Local channels don't support state information in Asterisk 1.4. For that you either need to use 1.6 or backport of state_interface for 1.4. Then you have to set call-limit for peers, and specify state_interface device when logging in agents. For more information please search for asterisk queue state, as this has been discussed several times. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All I have to say is Murf, SEND ME ONE I'll do anything (within reason) ;-) AEL bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) How about sending those out when certain amount of karma is reached? ;-) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 10:03 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 2:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All I have to say is Murf, SEND ME ONE I'll do anything (within reason) ;-) AEL bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) How about sending those out when certain amount of karma is reached? ;-) Regards, Atis It seems you get these goodies at Astricon events. Unfortuneately it's too far and too expensive for me to get there. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg call recording
On Tue, Jun 17, 2008 at 8:34 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Bikrish Amatya wrote:
Hi all
I am using asterisk as pbx for my company. My company has requirement
that all the incoming and outgoing calls should be recorded for all the
extensions and should be able to play recorded call on extensions basis,
that is , say 123 extension has made what call on the particular date
and should be able to play and listen to it. What is the better way to
achieve this? Any kind of suggestion is truly appreciated.
Bikrish
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A simple web interface, such as asterisk-stats coupled with some basic
modifications to link to a recording that was made with ${UNIQUEID} as
the recording filename (pre extension, use monitor + soxmix to mix the
recordings) will work just fine, I use it on a medium-large installation
that does about 10K calls a day, with no issues in regards to recordings
or ability to access calls/recordings.
I have similar setup, and here are some suggestions from my experience.
Do recording only in native format, that will decrease the load by
transcoding at working time.
Whenever somebody requests to listen, you can mix, transcode and play.
This usually takes few seconds (however depends on call duration). Mix
and transcode (to some lower bandwidth codec) the rest of recordings
at night time.
Personally I record everything in ulaw, and either on listen or at
night transcode to gsm for storage.
Regards,
Atis
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VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Agents getting stuck busy
On Mon, Jun 16, 2008 at 12:30 PM, Kyle Sexton [EMAIL PROTECTED] wrote: Having a weird issue with some agents getting stuck busy on my system. Call will come into the queue and the agent will hit DND, or be DND when the call comes in (DND being the button on eyeBeam softphone, not a star code). After the agent comes back from DND they will be stuck as busy in the queue and I have to reload chan_agent.so in order to get them available. I'm running Asterisk 1.4.17, and the bug sounds a lot like http://bugs.digium.com/view.php?id=9618 but that bug looks to be fixed in 1.4.17. I could suggest you trying on latest version (currently 1.14.21) or at least try this patch http://bugs.digium.com/view.php?id=12127 The description doesn't match your issue, however there was found old code handling dialstatus and translating it to agent state, which could be cause of your problem. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr-custom/Master.csv rotation
On Sun, Jun 15, 2008 at 10:23 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Sat, Jun 14, 2008 at 08:49:04PM -0700, Darryl Dunkin wrote:
It's like asking for directions, and someone tells you to drive,
useless.
Here is what we do here:
Create /etc/logrotate.d/asterisk:
/var/log/asterisk/asterisk-verbose /var/log/asterisk/messages
/var/log/asterisk/debug /var/log/asterisk/queue_log {
daily
rotate 7
compress
missingok
notifempty
sharedscripts
postrotate
/usr/local/bin/log_rot_ast
endscript
}
/usr/local/bin/log_rot_ast contains:
#!/bin/sh
/usr/sbin/asterisk -rx 'logger reload' /dev/null 21
logger reload rotates logs. But not CSV . That's because the CSV CDR
files are not held open.
If they are not held open, you can can just move them away with mv,
next CDR should just write new file.
Regards
A,tis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Idiot's Question
On Sat, Jun 14, 2008 at 8:46 PM, Venefax [EMAIL PROTECTED] wrote:
Believe it or not, I cannot find online a single piece of documentation for
the Asterisk function SPRINTF. This example does not work, for it changes
the caller id.
Set(CALLERID(num)=${SPRINTF(%010lld,0${CALLERID(num)})}),
For instance, if the incoming caller id is 17864335989, I get 0684466805 out
of that function, which is not intended one. To be precise, of the caller
has less than 10 chars, I want to complete it with a string of '0's. If the
caller id is nothing, or empty, I want to replace it with 10 zeroes. I guess
I can figure it out if a link to the documentation of SPRINTF is provided.
Well, 10 chars or 4294967296 to be precise is the limit of integer, so
on 32 bit platform this won't work. Just do the string processing :)
Btw - some kind of str_pad function in dialplan would be nice ;)
Regards,
Atis
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VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 10:51 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis I'm not sure that would be a viable solution, the ResetCDR(w) app+option is only going to write the cdr and then zero it out, but the next time the write occurs wouldn't it just overwrite the existing record? No, next time it will write new record from the point when ResetCDR was called. I use it extensively for call event logging, for example: * Call received to DID A, business hours detected. * Call sent to IVR 1 for 15 seconds * Call waited in queue 2 for 20 seconds etc Regards, Atis Ah thanks Atis! I hadn't played with it before since the documentation gave info that lead me to believe it wouldn't work for me :) Very helpful information :) You're welcome :) Oh, btw, you will definitely need to enable unanswered = yes in cdr.conf as after ResetCDR new entry has disposition NO ANSWER, even if call is answered before. So without this you could loose them. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk calls per second
Hi,
I already gave a hint into right direction, but seems that it got
missed, so basically it would look like this:
exten=_3XX,1,Set(GROUP()=${EPOCH})
exten=_3XX,2,Set(GROUPCOUNT=${GROUP_COUNT(${EPOCH})})
exten=_3XX,3,GotoIf($[${GROUPCOUNT} ${MAX_CALLS}]?120)
exten=_3XX,4,Dial(SIP/${EXTEN})
exten=_3XX,5,Playback(unavailable)
exten=_3XX.,6,Hangup
exten=_3XX,120,Playback(try-later)
exten=_3XX,121,Hangup
Epoch is UNIX timestamp, which changes every second. Probably you
don't even need to use GROUP, but can keep counter for current second
in some database, however that would need database cleanups and locks.
Asterisk builtin DB wouldn't be useful, as it can't increment within
same operation, so some sort of SQL magic should be used. For example
multiple primary keys, one of which is autoincrement, or just
transactions.
However advantage of using GROUP would be that if call disconnects,
it's not counted within GROUP_COUNT anymore, so you can accept one
more call for that second(probably most useful for minute).
Regards,
Atis
On Fri, Jun 13, 2008 at 3:57 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
Hi Edgar,
Thanks for the reply. This setting is good for 10 simultaneous calls.
What i really need is 10 calls being done per second but no limit on
simultaneous calls.
On Fri, Jun 13, 2008 at 2:43 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
Well, as I said, you can tell Asterisk to accept until 10 SIP calls,
for example, at ANY TIME (I don' t understand why per second, I mean,
if the 10 calls are established in the same second, they are acepted,
and so they are if they are established in the same milisecond, while
the max concurrent calls is belowthe limit of 10).
You can do something like this in your dialplan (assuming extensions like
_3XX)
exten=_3XX,1,Set(GROUP()=sip-calls)
exten=_3XX,2,Set(GROUPCOUNT=${GROUP_COUNT(sip-calls)})
exten=_3XX,3,GotoIf($[${GROUPCOUNT} ${MAX_CALLS}]?120)
exten=_3XX,4,Dial(SIP/${EXTEN})
exten=_3XX,5,Playback(unavailable)
exten=_3XX.,6,Hangup
exten=_3XX,120,Playback(try-later)
exten=_3XX,121,Hangup
where ${MAX_CALLS} is a variable defined by you that is the limit of
calls to be accepted
On Thu, Jun 12, 2008 at 12:16 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
yeah something like that. is it possible to set asterisk to make 10
calls per second?
On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I know you can limit the total calls in any given time, for example,
you say I would like to have 10 SIP calls established as maximum.
On 6/11/08, Mark Quitoriano [EMAIL PROTECTED] wrote:
Is there a way to limit or set the calls per second on SIP?
--
Regards,
Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
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APUG! | http://asterisk.org.ph
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Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis I'm not sure that would be a viable solution, the ResetCDR(w) app+option is only going to write the cdr and then zero it out, but the next time the write occurs wouldn't it just overwrite the existing record? No, next time it will write new record from the point when ResetCDR was called. I use it extensively for call event logging, for example: * Call received to DID A, business hours detected. * Call sent to IVR 1 for 15 seconds * Call waited in queue 2 for 20 seconds etc Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk calls per second
On Thu, Jun 12, 2008 at 9:16 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
yeah something like that. is it possible to set asterisk to make 10
calls per second?
On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I know you can limit the total calls in any given time, for example,
you say I would like to have 10 SIP calls established as maximum.
On 6/11/08, Mark Quitoriano [EMAIL PROTECTED] wrote:
Is there a way to limit or set the calls per second on SIP?
Combine GROUP/GROUP_COUNT with category of ${EPOCH}
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group
Calls will still be received by asterisk, however you will be able to
kick them off without proceeding with following dialplan logic.
Regards,
Atis
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Re: [asterisk-users] Camp / Callback feature in 1.4
On Tue, Jun 10, 2008 at 5:34 PM, Phil Knighton [EMAIL PROTECTED] wrote:
Hello
I'm looking for a way to do the following using my Asterisk system and Snom
SIP phones...
Scenario:
Caller on Internal Phone 1 calls internal phone2. Phone 2 is busy (or more
accurately goes straight to voicemail).
Caller on internal phone 1 can press a button / dial a code (explained in
next step) and hangup
When phone 2 is free, phone 1 rings and on answer dials phone 2
I was sure this was called camping - but all the camping stuff I can find,
refers to the caller having to hang on the phone and wait. Am I missing
something?
Anyone have a solution?
Quick solution that comes into mind:
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if (${DIALSTATUS}=BUSY) {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h = {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if (${call_to}!=) {
Set(DB(camp/${exten_copy}/call_to)=);
System(call_to ${exten_copy} ${call_to});
}
}
So, in case if phone2 is busy, store callerid of phone1 in database,
so when phone2 will hangup it will triger a script call_to which
however can originate call trough manager or call-file.
Of course you will need some additional handling in case if multiple
callers decide to camp, or diferent protocols are used, etc.
Regards,
Atis
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Re: [asterisk-users] detecting which party hung up
On Thu, Jun 5, 2008 at 6:57 PM, Lenz [EMAIL PROTECTED] wrote:
Hello list,
I have a problem that looks quite simple but I cannot find a way to fix.
I have a Dial() command and want to detect which party of the call hung up
- if it was the caller or the callee.
In the dialplan, I have the folllowing commands...
exten =
exten = _9XXX.,n,Dial(${MY_TECH}${MY_NUM}||M(call-answer))
; Trapping call termination here
exten = h,1,NoOp( Call exiting: status ${GLOBAL(${GM})} DS:
${DIALSTATUS} HU: ${HANGUPCAUSE} )
I set the ${GLOBAL(${GM})} variable through a macro 'call-answer', and it
works fine for detecting if the call was answered or not (I have other
logic to run at answer time so it fits me okay).
I thought that there would be a way for me to know on the calling channel
if the 'h' was enetered because this channel hung or because the other
bridged party hung, so I tried ${DIALSTATUS} and ${HANGUPCAUSE}, but they
are always the same no matter who hangs up. Am I missing something here?
Thanks
l.
Hi,
add g flag to Dial app, that way Dial will continue to next priority
when ANSWERED but called party hanged up. However if caller will hang
up, channel will jump to h extension.
Regards,
Atis
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[EMAIL PROTECTED]
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Re: [asterisk-users] Dear asterisk-users@lists.digium.com May 87% 0FF
On Fri, May 23, 2008 at 3:19 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, May 23, 2008 at 5:49 AM, VIAGRA (R) Official Site asterisk-users@lists.digium.com wrote: About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the Unsubscribe link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. (c)2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Is this the new DAHDI Viagra? I think, spamfilter should ban every message mentioning Microsoft :p Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Proposed changes for queue timeout
Hello, I've been annoyed quite some time by behavior of queue timeout (specified as argument to Queue app). Basically if I specify timeout for queue 5 minutes, and ring time to agent for 15 seconds, and ring to agent starts at 4:59, agent will receive ring only for 1 second, after which call attempt will terminate. So, the question is - if anybody needs exact queue timing, with possibility that agent calls are terminated without finishing ring timeout? Please see issue http://bugs.digium.com/view.php?id=12690 - there's table of calculations, which explains how values are calculated now, and how I'm proposing. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom XML Files / asterisk
On Thu, May 15, 2008 at 10:08 PM, Robert McNaught [EMAIL PROTECTED] wrote: The way I understood it is that TFTP does not allow you to set a username and password in a URL like tftp://username:[EMAIL PROTECTED] is not possible when setting option 66 Is it not possible to require a username and password with HTTP? I assumed that you could just like if you were protecting the web root directory on a webserver to require authentication credentials, although have never tried this. You can always limit access to HTTP for certain IP range. Isn't that enough? Then add auth in your request string - for example: http://provisioning.mysite.com/secure/234sdfsdf3247sd/- unless you enable directory listing, it should be at same security level as http with authentication or ftp (any of those can be sniffed) Another thing I like in HTTP - you can redirect config read to execute any script, write simple PHP that will generate resulting config, with lookup of correct extension by MAC. Much like DHCP. Regards, Atis Robert On Thu, May 15, 2008 at 10:43 AM, Anthony Francis [EMAIL PROTECTED] wrote: I am confused how TFTP is less secure than HTTP. TFTP does not allow any browsing, ever. Neither technologies will allow the device to authenticate before downloading a configuration file, and both are easily secured by only permitting connections from specific hosts. Robert McNaught wrote: Yes, perhaps a script would always be better than hand-touching these files, and getting an XML editor only really makes it easier on the eyes. On the same subject, I have noticed that Snom and Linksys phones do not support FTP provisioning - only TFTP and HTTP. With TFTP being an insecure option for a hosted architecture, is everyone moving to provision Polycoms with HTTP, so that both can be auto-provisioned via Option 66. One thing I found is that, with option 66 in a LAN router, you cannot specify more than one protocol. Has anyone had any problems provisioning Polycoms with HTTP? On Thu, May 15, 2008 at 1:35 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Robert McNaught schrieb: Does anyone know how to apply a style sheet to the polycom automatic provisioning XML files? Why should applying a stylesheet be different than for any other XML files? Even better, does anyone know of a web-based XML editor where you can just edit the files from a browser directly ie entering in phone number, display name, proxy address etc. From what I gather, most people are just using Notepad to change the files then upload them, or vi from the command line, which is fiddly and time-consuming. Just use your preferred editor. Nobody forces Notepad or vi upon you. Even better: Generate the config files with Perl/PHP/insert favorite language. Grüße, Philipp Kempgen -- Asterisk-Tag.org 2008, 26.-27. Mai - http://www.asterisk-tag.org amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
On Wed, May 14, 2008 at 10:41 AM, Stelios Koroneos [EMAIL PROTECTED] wrote: As people have sugested the ATX power supplies can work without a mobo One thing to watch out for your setup is the actual ampere requirments for your disks i.e Your power supply provides 300W but this is partitioned to different voltages (+5, +12, etc) with different amp charecteristics Disks need 2 voltages. One for the logic (+5V) and one for the motors (+12V) and have different current requirments. Most disk (if not all) mention these ratings on the labels they have What you must do, is to see if by adding the current requirments seperatly for +5V and +12V, does not exceed the power supply's amp rating *for that voltage*, allowing also for a 15% -20% margin, as power consumption will be higher than the nomimal mentioned during disk startup (and you will be starting all your disks at the same time) Also make sure your box has sufficient cooling and there is some short of airflow over the disks, as the number 1 enemy of disks is high temperature and stacking so many disks in a box will create large amounts of heat. I would suggest you to get a good (aka expensive) 500W power supply and use 10-12 disks with it to avoid problems in the long run, Also keep in mind that MTBF specs of SATA disks does not make them an ideal candidate for 24/7/365 operations Another thing is voltage feedback. The Gray wire should be grounded when +5 and +3.3 V is ok for m/b. As +5 is shared also for disk connectors, there could be some problems. Also be advised that you should buy good power supply, as the difference is in voltage stability, and hard disks don't like floating voltages much. I would suggest you to go better for some network oriented setup, use NFS ir CURL for configs, etc. Imagine what will happen if that one PSU will fail. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
On Sun, May 11, 2008 at 8:24 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Sun, May 11, 2008 at 12:24 PM, Robert DeVries [EMAIL PROTECTED] wrote: GrandCentral has a feature where when you call the GrandCentral number it can ring multiple phones. However, it's not the first phone to answer that gets connected, but the first phone to answer AND play a touch-tone after hearing a recording. The advantage of this is that if one of the called phones has voicemail, it won't get connected to the calling party because the VM won't send a touch tone in response to the recording, unlike a live person. I have always resisted implementing a multiple ring scenario with Asterisk that included a cellphone because of the voicemail answering problem, but this seems to be a solution. Anyone know how to implement it with Asterisk? GREAT IDEA! (even if it wasn't yours ;-) I have had so many issues with this and desk phones, cell phones being out of range, turned off, or answering machines set to answer after two rings. If this gets implemented, it would be a great feature and save me tons of complaints and explanations. Maybe a posting on the dev list is appropriate. I would certainly contribute to a bounty. Wouldn't a answer macro do exactly what required. It should be executed before bridge, so ANSWER shouldn't be passed upon it's completed. It can read some tone from keypad, and if that confirms, continue by bridging channels. So, this should work with at least queue in ring-all mode (i feel that it would be correct if Dial would do that too) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
On Sun, May 11, 2008 at 8:49 PM, Matt Watson [EMAIL PROTECTED] wrote: I just took a quick look at the dialplan that freepbx uses for doing call confirmation... the dialplan part of it is actually quite simple... its just a matter of setting the USE_CONFIRMATION varialbe =TRUE. However, the actual magic looks like it happenes through its dialparties.agi... which is a little more complicated than i'd like to try and dissect on a sunday afternoon! but that might be a good place to look at how its done to learn by example. It should be like Dial(SIP/123SIP/456,30,M(confirm)); and macro named confirm that playback the prompt, reads DTMF, and sets value of MACRO_RESULT I know in the freepbx implementation what it does is whenever a handset thats part of the ringgroup answers, they get a recorded message You have an incoming call, press 1 to accept maybe it says something else too... can;t recall at the moment. The first member of the Ring group to hit 1 gets the call... if more than 1 person picks up the handset right away, the first to hit 1 gets it, and the rest hear a sorry, too late, somebody else got it-type message (no idea what it actually says). I suppose just a disconnect, because call was already bridged. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: tricky problem with MOH
On Thu, May 8, 2008 at 11:25 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
I have this simple queue for the reception set up such that the console
queue has only one agent.
I checked the number in the queue and if there is someone there, I play
back a busy please be patient message and then join the call to the
queue.
If there is no one in the queue, the caller will go directly into the
queue and the receptionist phone will ring. This looks fine but while
the call is waiting for the receptionist to pick it up, the caller will
actually hear Music on Hold instead of just ring ring ring. This is
undesirable.
exten =
7100,n(rcl_off_opn),Set(rcv_que_num=${QUEUE_WAITING_COUNT(console)})
exten = 7100,n,GotoIf($[${rcv_que_num} = 0]?rcl_que_jon:)
exten = 7100,n,Playback(rc-busy)
exten = 7100,n(rcl_que_jon),Queue(console)
exten = 7100,n,Wait(2)
exten = 7100,n,HangUp()
Queue(console,r)
would do what you want, but so you would need to have two entry points to queue.
Regards,
Atis
So, the issue is MOH is good for the 2nd and subsequent callers but not
for the first caller who should just hear ring ring ring until the
receptionist picks up the call.
Any thoughts?
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[asterisk-users] Realtime status feature - user feedback needed
Hello users, I had developed several patches that allows to monitor current status of queues/channels in realtime db. For example specifying realtime family channels will make asterisk to keep current channel list in realtime database engine. The same would be for queue callers and queue members (already partially available in 1.4). However I encountered a resistance from Asterisk developers, as they don't wish to accept my patches - because they don't wish to support another interface. As I read in Bug Guidelines, if enough users request this, it should make into asterisk, so I'm asking You to express Your opinion on those features. *** So, realtime status - what's this all about? Basically you get output of show channels, show queues, etc directly in Realtime table (Realtime = database engine system for Asterisk). So, Asterisk will automatically update database upon any changes of channels or queues. *** Why would You need that? In beginning I created this in order to deal with large amount of monitoring software. If there's lot of users monitoring status, some kind of cache should be put into place. With current Asterisk interfaces this would mean either inquiring current status or developing a daemon that follows up all events and collects them to keep current picture always ready. I just decided to move this layer to database engine, which deals really good with this stuff. *** Rapid development of monitoring tools What it takes to create custom monitoring tool now? AMI event listener? AJAX page that gets changes from built-in webserver? All this takes lot of time to learn and start using. Adding just few config lines in extconfig.conf would allow to automatically populate database with current status - so it's accessible easily from any programming language. All the info is just there, no need for processing or analyzing. *** Performance / Scalability Inquerying queue status means that there is lock put on queue list, all queues are traversed, information gathered and then returned. If lot of instances of monitoring software need to have this information, it's obvious that this would mean too much locks. So, as database update is thrown whenever some change is happening, it means that no additional locks are created for monitoring purposes. Transaction is sent to database engine, which keeps relatively small tables of current status. Then any number of clients can access data directly without any locking. Even 200 concurrent calls with 10 new calls per minute would still be a tiny load for MySQL. This can also be scaled by moving database to another machine, thus allowing more raw CPU usage for Asterisk. *** Development maintenance Those changes doesn't introduce any new functions in asterisk code. They utilize currently available Realtime engine which is meant for storage of configuration data. This just extends use of this engine also for status data, so maintenance of this interface should not take lot of work. *** Current patches If You are interested in using those changes right away, here are some backports for 1.4: Channels: http://ftp.iq-labs.net/realtime_channels/ Queue callers: http://ftp.iq-labs.net/realtime_queue_callers-1.4/ Queue members: work in progress, needs some refactoring and optimization to make that effective. Meetme: planned, no patches yet To use any of those patches, you will need to add backport of store/destroy to 1.4: http://ftp.iq-labs.net/realtime_store_destroy-1.4/ *** Supporting this feature If You find that those features would be good for merging into Asterisk, please write a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] update DB on ringing/ catch ringing event
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. For now, not really. You could try Realtime Channels patch I just mentioned here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html This should give you up-to-date list of channels in database, so you can use SELECT * FROM channels WHERE state=Ring; to get currently ringing channels. If You find this patch useful, please add a comment to issue http://bugs.digium.com/view.php?id=12556 that you would like to see Realtime status implemented in future versions of Asterisk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] update DB on ringing/ catch ringing event
On Thu, May 8, 2008 at 12:34 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Atis Lezdins schrieb: On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. For now, not really. You could try Realtime Channels patch I just mentioned here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html Yeah, of course you can do almost anything with a patch. Well, this wasn't specifically written for this requirement. I just want to add some general usage realtime status in Asterisk, and I need user support there :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime status feature - user feedback needed
On Thu, May 8, 2008 at 1:07 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 07 May 2008 16:11:05 Atis Lezdins wrote: However I encountered a resistance from Asterisk developers, as they don't wish to accept my patches - because they don't wish to support another interface. As I read in Bug Guidelines, if enough users request this, it should make into asterisk, so I'm asking You to express Your opinion on those features. That's not quite correct, either. First of all, the correct forum for this is the -dev list, where we discuss development issues. Second, we gave you an alternative way to do this. You could do this with AMI, with the addition of a single query to access current state, then monitor status continuously for updates. And third, it doesn't make a difference how many users request a particular interface -- the development team has to maintain it afterwards, and if you're proposing a new interface, you need to convince the development team that it's worth the extra hassle -- not the users. True, but resistance I encountered gave me impression that there's no way how to convince devs except lot of users asking for this, so i want to see who would find this useful. I hope that this would convince the development team. So we're not opposed to the concept; we are opposed to the particular interface that you chose to use. Modify it, and it will make its way back into Asterisk. Stubbornly stamping your foot and insisting that you have the right way, and the status quo will remain. Unfortunately the concept I'm offering is that There's no need for continuous AMI connection. Current state can be retrieved already (but that needs locking), and incremental updates are available too (but that needs continuous AMI connection). So all together - I'm saying there could be really simple interface for all this - no troubles with locking of lists or keeping persistent connections. Why would user application need to take care of all this, if DB engine can do that. *** Supporting this feature If You find that those features would be good for merging into Asterisk, please write a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Please don't. We've already discussed this to enough detail, and if you choose to modify your code, it will show up in the next major release of Asterisk. I understand that code have to match certain standards, coding guidelines and architecture. I'm willing to do any of this, but so far i see all the discussions are about concept. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime status feature - user feedback needed
On Thu, May 8, 2008 at 3:49 AM, Ex Vito [EMAIL PROTECTED] wrote: On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Tilghman Lesher a écrit : Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. True, that was one of initial options, however I prefer to NOT have yet another layer. I will consider this as an option where appropriate. However this looks quite awkward to me, somehow it reminds me tailing queue_log or CDR and putting result into MySQL database.. just one level more that way. For now, I see only one point against this - having status cleared upon module load/unload makes it easier to follow restarts/module loads. I second that, If there is already a way to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. Agreed. There should be ONE to do it, it should be SIMPLE and as RELIABLE as possible, without interfereing (bad spelling?) with asterisk's operations: the proxy into AMI looks like the way to acheive the required funcionality... After all, that's exactly the purpose of AMI ! Let's keep the codebase as small as possible, let's make asterisk as solid and reliable as possible. Let's not reinvent wheels! Ok, so we're exactly at the point. Yes, I agree that it would act nearly the same way as AMI actions, however there's one great advantage - It would be really easy to set this up for user. AMI proxy would take more effort, need configuration, etc. Then there should be much more development support for proxy than for code within asterisk (if you have noticed, there's no new code, just reusing existing functionality) I think that there should be several ways how to do something, not just one. Having realtime status won't mean that much changes, for now I can see only 4 families for this - queue_members (already existing), queue_callers, channels and meetme. Really nothing more to give full overview of Asterisk Status. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote host can't match request NOTIFY???
On Thu, May 1, 2008 at 1:38 PM, Alan Lord [EMAIL PROTECTED] wrote: Grey Man wrote: On Thu, May 1, 2008 at 7:54 AM, Alan Lord [EMAIL PROTECTED] wrote: Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. snip / Asterisk does not correctly match SIP NOTIFY transactions in at least some cases. Your problem may be related to http://bugs.digium.com/view.php?id=11848. Regards, Greyman. Thanks for that. Not sure I understand it all. I am not actually doing anything when these messages appear. They occur pretty much every minute or so. With or without any calls... There was a post week ago (I was having the same problem). For me it was caused by AudioCodes not understanding voicemail notifications. So, first You can enable SIP debug to see what packets are causing this, and if it's voicemail notifications, turn them off in sip.conf (mailbox= line). Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue callers
On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote: How can I get a list of the callers within a specific queue at any given moment? I need to get the caller IDs of all active calls in a queue then send them out via a udp socket to a listening application on the network (the only data I need to send are two fields: current timestamp and caller id of active queue calls). I have almost all the elements to do this except the best method to retrieve all active caller ids from a given queue. I was wondering if someone already did this. I tried writing a script on the server which connects to the Manager API and receives queue events. I'm basically using the AgentCalled event but it seems clumsy to efficiently detect when the call has ended (connect or abandon) and thus update the remote UDP listening app. I also tried another way by guessing which calls are active via tailing and grepping /var/log/asterisk/queue_log. Finally, a third script method tried parsing the output of show queue right after Callers:. Maybe this is all I really need for my purposes (although less efficient and less real-time than the queue events method because I would need to periodically poll the whole queue statistics) but I only get the originating channel and the wait time. I would require correlating the data to the caller's ID. Has anyone already done something similar? A simple example/script/suggestion would be greatly appreciated. I'm not sure that this is what exactly You need, but I have a patch for app_queue that will store and update queue callers (as well as update lots of fields for queue members) in realtime mysql table. This allows to do many requests for current queue state simultenously, and moves load from asterisk to mysql (which can be on separate machine). So, generally to get active callers with all their callerid/channel info You will have to do just SELECT * FROM queue_callers. It's not very finalized, so I haven't yet posted that to Digium for inclusion in next asterisk versions, but I intend to do that in future. It's been working stable on our production for several months. If You're interested, please reply, and I'll try to separate that patch out from other our patches. Currently I have it updated for 1.4.19, but also have some version for 1.4.14 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue callers
On Tue, Apr 29, 2008 at 1:22 PM, Vieri [EMAIL PROTECTED] wrote: --- Atis Lezdins [EMAIL PROTECTED] wrote: On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote: How can I get a list of the callers within a specific queue at any given moment? I need to get the caller IDs of all active calls in a queue then send them out via a udp socket to a listening application on the network (the only data I need to send are two fields: current timestamp and caller id of active queue calls). I have almost all the elements to do this except the best method to retrieve all active caller ids from a given queue. I was wondering if someone already did this. I tried writing a script on the server which connects to the Manager API and receives queue events. I'm basically using the AgentCalled event but it seems clumsy to efficiently detect when the call has ended (connect or abandon) and thus update the remote UDP listening app. I also tried another way by guessing which calls are active via tailing and grepping /var/log/asterisk/queue_log. Finally, a third script method tried parsing the output of show queue right after Callers:. Maybe this is all I really need for my purposes (although less efficient and less real-time than the queue events method because I would need to periodically poll the whole queue statistics) but I only get the originating channel and the wait time. I would require correlating the data to the caller's ID. Has anyone already done something similar? A simple example/script/suggestion would be greatly appreciated. I'm not sure that this is what exactly You need, but I have a patch for app_queue that will store and update queue callers (as well as update lots of fields for queue members) in realtime mysql table. This allows to do many requests for current queue state simultenously, and moves load from asterisk to mysql (which can be on separate machine). So, generally to get active callers with all their callerid/channel info You will have to do just SELECT * FROM queue_callers. It's not very finalized, so I haven't yet posted that to Digium for inclusion in next asterisk versions, but I intend to do that in future. It's been working stable on our production for several months. If You're interested, please reply, and I'll try to separate that patch out from other our patches. Currently I have it updated for 1.4.19, but also have some version for 1.4.14 Thanks Atis. That patch sounds really neat. Hope it gets into * soon. Just a doubt: suppose the mysql daemon dies for some reason. Will the patched app_queue still handle calls and not hang? It should, as asterisk throws INSERTs, UPDATEs and DELETEs for changing data (queue callers and queue member status), plus it loads existing queue members trough SELECT (as it's now with realtime queue members, just some extra fields). So, I suppose if MySQL dies in middle of operation, SELECT should fail and Asterisk should just continue with what it has in memory. Btw, You should be able to also use static or dynamic queue members (not realtime) in combination with realtime queue calls. Btw, I never experienced that MySQL dies, it's more often that Asterisk dies. So, are You interested in applying this patch yourself? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue
Atis Lezdins wrote: Queue will continue if called person hangs up (and there's no option). If caller hangs up, call goes to h extension in same context. Just the same way as Dial with 'g'. There's a change in 1.6 that allows called channel to continue if caller hangs up, so probably something like this could be applied also to Queue (or was that actually working with using Local channels?). On Wed, Apr 23, 2008 at 8:18 PM, Al Baker [EMAIL PROTECTED] wrote: Why would you want a channel to continue after the caller has hung up. I clearly am missing something here because I can't see what good that would be. What do people do with this Continued Channel ? What is is used for ? How Does having it help you ? ??? To play something to called party. I'm not familiar with that feature too deep, but I guess it's not caller channel but called channel that's continued. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6 that allows called
channel to continue if caller hangs up, so probably something like
this could be applied also to Queue (or was that actually working with
using Local channels?).
Regards,
Atis
On Wed, Apr 23, 2008 at 7:13 PM, AnDY [EMAIL PROTECTED] wrote:
Thank you for your answer.
But the Dial command has a option 'g' which means that after succes will
proceed next priorities in the dialplan. Is there something also for
Queue() because according to manual there is no option for it. So I am
looking for some other solution.
Andy
Tony Mountifield napsal(a):
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
Hello everybody.
I was looking for the solution but nothing found. I have this in my
extensions.conf:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
But when the call is placed in the queue and somebody answer it, it will
throw an error:
== Spawn extension (default, 211, 4) exited non-zero on
'Local/[EMAIL PROTECTED],2'
And no other command in extensions is executed.
Any suggestions?
Queue() is like Dial(), in that if it succeeds in connecting to someone,
it will not return to the next priority in the dialplan. However, if you
define an 'h' extension, that will get executed when the call is complete:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
exten = h,1,NoOp(${QUEUESTATUS})
exten = h,2,NoOp(${DIALSTATUS})
Cheers
Tony
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VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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[asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS
Hi, I experience my log flooded with warning messages like this: [Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up I traced this down to point when we added to sip.conf status notifications: allowsubscribe=yes rtcachefriends=yes So, those notifications allow for queue to display (In Use) etc, and creates no warnings for other devices except Audiocodes gateway. I wonder is there any way how to disable this message in Asterisk, or make Audiocodes act correctly? Below is the sip debug for this (xx.xx.xx.xx is Audiocodes, yy.yy.yy.yy is Asterisk). Regards, Atis - [Apr 14 01:30:24] VERBOSE[19514] logger.c: Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: NOTIFY) [Apr 14 01:30:24] VERBOSE[19514] logger.c: Reliably Transmitting (NAT) to xx.xx.xx.xx:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport From: Unknown sip:[EMAIL PROTECTED];tag=as436bf308 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) --- [Apr 14 01:30:24] VERBOSE[19514] logger.c: --- SIP read from xx.xx.xx.xx:5060 --- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport From: Unknown sip:[EMAIL PROTECTED];tag=as436bf308 To: sip:[EMAIL PROTECTED];tag=1c73477527 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Contact: sip:xx.xx.xx.xx Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Content-Length: 0 - [Apr 14 01:30:24] VERBOSE[19514] logger.c: --- (10 headers 0 lines) --- [Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS
On Tue, Apr 22, 2008 at 3:15 PM, Grey Man [EMAIL PROTECTED] wrote:
For blind transfers Asterisk will send the call back to the dial plan
and into the TRANSFER (I think, could be a different name) context if
it exists. Within that context you can access the channel that was
answered on the original call using ${DIALEDPEERNUMBER}.
Note that this mechanism cannot be use for attended transfers as they
are not sent back to the dial plan for processing.
I apologize, but I don't have any problems with transfers. The
warnings I get in log appears there even without any calls going on.
Maybe You replied to wrong topic?
Regards,
Atos
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VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] G729 license count...
The codec in use for a specific channel doesn't even care if that channel exists over zapata analog or digital cards, sip channels, iax[2] channels, smoke signals, etc. If you care to use ping pong balls and the atlantic ocean as your medium, you should be able to interface with the g729 codec if you still needed to :D Although I wouldn't expect there to be much error correction inherent in the Atlantic. I would not risk sending my data trough new cutting edge transports You mentioned. Instead I prefer to use proven technologies, and preferably documented in RFC - for example RFC 2549 IP over Avian Carriers with Quality of Service. There are even some modifications to this by using flash cards instead of paper, and that beats speed of ADSL. However that still doesn't seems best for my VoIP traffic because of latency. The codecs are modules for *asterisk* and not for the cards themselves. That's true. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about queue
Hey, I just found out today that it doesn't work on Asterisk 1.4.19 (at least for realtime queues) if you have autofill=yes in queues.conf. However it works if you add it in queue settings for each queue, for realtime that would be ALTER TABLE queue_table ADD COLUMN autofill TINYINT(1) UNSIGNED DEFAULT 1; For following this issue, see http://bugs.digium.com/view.php?id=12445 Regards, Atis On Sat, Apr 12, 2008 at 4:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global call limit
On Wed, Apr 16, 2008 at 12:46 AM, Vinz486 [EMAIL PROTECTED] wrote:
Hi,
i'm new in asterisk programming.
Maybe my question was posted thousand times but i found nothing using google.
I'm looking for a method to limit the total simultaneous calls
(inbound and outbound) that pass from internal phones to 2 SIP
providers.
I found the calllimit option but it works only on a per-channel basis.
Instead i want limit the total amount of calls, abstracting from which
SIP provider us used.
This to keep good audio quality for active calls and rejecting new
arriving: this is needed for a PBX connectect with a poor ADSL having
only 256kbit in upload: so i want permit only 3 calls (256 / 80 kbit)
and rejecting the 4th.
Any solutions?
if (${GROUP_COUNT([EMAIL PROTECTED])}) function in combination with
Set(GROUP(a)=x)
or Set([EMAIL PROTECTED])
Regards,
Atis
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Skype: atis.lezdins
Cell Phone: +371 28806004
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Re: [asterisk-users] Asterisk 1.4.19 crash with Realtime using SIP peers
On Wed, Apr 9, 2008 at 5:29 PM, Trevor Peirce [EMAIL PROTECTED] wrote: Mindaugas Kezys wrote: Hello, Asterisk 1.4.19 crashes everytime using Realtime and SIP peers Yes I also saw this and had to revert. Calls to the IVR seemed to be fine, but as soon as two peers call each other it crashes as the call progresses (never connects). I haven't had a chance to explore any further and therefore haven't posted a bug either. Perhaps this weekend if nobody does first. So far works fine for me. Sample peer setup below. Had one issue with peers where ipaddr was 0 (and hostname used instead), but adding this patch ( http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?r1=113012r2=113240 ) seems to solve everything. Regards, Atis *** 1. row *** id: 2 name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: Atis 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: sip:[EMAIL PROTECTED]:5061 host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: 21168 type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1207763735 ipaddr: 192.168.1.123 regexten: cancallforward: yes setvar: call-limit: 4 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interrupting MOH
Sorry for top-posting, but seems everyone on this thread did so. Also that would be my suggestion for now - call queue with periodic-announce. However i see that this would make nice architectural improvement - allow inject sound files into MoH stream. This would be useful for example in call queues - to inject all the queue announcements into MoH directly, rather than play them while blocking further queue actions. Regards, Atis On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge [EMAIL PROTECTED] wrote: I think that's still a better idea than using a dump the caller into meetme hack and is actually what I was going to suggest. If you want something simpler than a queue then inject the sounds into the moh already. On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis [EMAIL PROTECTED] wrote: You may be able to achieve the desired result using queues rather than Dial statements. Overkill perhaps, but it's the only way I can think to implement it at the moment. John Millican wrote: Tilghman Lesher wrote: On Tuesday 01 April 2008 05:14:25 Pete Kay wrote: I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like please wait while we transfer your call. How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can be accomplished? The only way to do that currently is to implement the prompt within the MOH stream itself. Just off the top-o-my head(YMMV), couldn't you create a meetme and play hold music into the meetme and then also play the prompt into the meetme at the same time without interrupting the hold music? This would obviously not work for high load but... JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interrupting MOH
On Wed, Apr 2, 2008 at 11:05 PM, Brent Davidson [EMAIL PROTECTED] wrote: You could also, conceivably, handle this outside of asterisk by using a more complex MOH stream source. For instance, use a shoutcast client as the MOH source, run your own shoutcast server streaming your music and have a script set up to periodically interrupt the stream being served to the shoutcast server and inject an announcement. (Keep in mind that this is an off the top of my head suggestion so I don't have exact details for implementation, but I'm sure it can be done.) That would need one shoutcast stream per call.. not very reasonable.. Regards, Atis Good luck, Brent Matt Florell wrote: Hello, We achieve this using an AGI script in the VICIDIAL project for our version of inbound queues. You start MoH then when you stream a sound to the channel it will stop MoH then after the sound is done you start MoH back up again. Probably a bit more involved than what you want, but it dose work well for us. MATT--- On 4/2/08, Atis Lezdins [EMAIL PROTECTED] wrote: Sorry for top-posting, but seems everyone on this thread did so. Also that would be my suggestion for now - call queue with periodic-announce. However i see that this would make nice architectural improvement - allow inject sound files into MoH stream. This would be useful for example in call queues - to inject all the queue announcements into MoH directly, rather than play them while blocking further queue actions. Regards, Atis On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge [EMAIL PROTECTED] wrote: I think that's still a better idea than using a dump the caller into meetme hack and is actually what I was going to suggest. If you want something simpler than a queue then inject the sounds into the moh already. On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis [EMAIL PROTECTED] wrote: You may be able to achieve the desired result using queues rather than Dial statements. Overkill perhaps, but it's the only way I can think to implement it at the moment. John Millican wrote: Tilghman Lesher wrote: On Tuesday 01 April 2008 05:14:25 Pete Kay wrote: I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like please wait while we transfer your call. How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can be accomplished? The only way to do that currently is to implement the prompt within the MOH stream itself. Just off the top-o-my head(YMMV), couldn't you create a meetme and play hold music into the meetme and then also play the prompt into the meetme at the same time without interrupting the hold music? This would obviously not work for high load but... JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689
Re: [asterisk-users] callers in queue passed to agents who accept only one call at a time
On Thu, Mar 27, 2008 at 6:32 PM, Vieri [EMAIL PROTECTED] wrote: I have a queue I configured as strict and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy). Instead, it's as if they had call waiting and incoming calls keep popping up while they're conversating with the first caller and they would like to avoid this. I guess the easiest solution would be to find an open-source or free softphone that can be configured to accept only one call at a time (currently using SJphone). Another solution would be if I could tell the Queue() application that if an agent is InUse then don't pass the call. Still another yet more delicate solution would be to have a custom script receive manager events related to the queue which in turn replies with an agi command. For example, whenever an agent answers a call I think that an event such as QueueMemberStatus can be triggered (although I don't know how). If the custom script could receive this event in realtime then it would run an agi command such as QueueRemove(busyagent...). When the agent is free again I suppose the same event is triggered and the custom script can QueueAdd(freeagent...). Could anyone please give me some pointers on this? In queues.conf set ringinuse=no Also make sure that you don't use realtime sip peers (or use rtcachefriends with that). Probably you also need call-limit set to any value in sip.conf For more info see http://www.voip-info.org/wiki-Asterisk+config+sip.conf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling 3 different call ending causes
On 3/20/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Date: Wed, 19 Mar 2008 11:31:57 +0200
From: Atis Lezdins [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Handling 3 different call ending causes
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
On 3/17/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
You can handle the 'h' extension in the dial plan, which will supply
the
${CHANNEL} that was hung up, and possibly some additional dial plan
variables as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
Using these, you can piece together who hung up on whom, etc.
#2 is handled by fallthrough in the dial plan that causes the
instructions
to continue executing to the next priority for that extension, whereas
if the call completes (Dial() is successfully connected), this does not
happen.
I''ve tried to use the h extension in combination with the ${CHANNEL}
in the dialplan as suggested on the wiki page, but I haven't had any luck
with it.
For this test I have a Sipura phone with number 1003 and a X-lite with
1203.
If I let the time go by (the 20 seconds defined in the Dial Command) I
get the following:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
If I let the Sipura hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
Lastly if I let the X-lite hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
Yes they are all the same :(
Perhaps there's something wrong with my code? Its just a small
context with the following for this test:
[hangupcause]
exten = s,1,Dial(SIP/1203,30,gL(1[:5000][:5000]))
exten = s,2,NoOp(Callee hangup)
exten = h,1,NoOp(Channel hungup is ${CHANNEL})
Have I missed something basic here or what?
This should allow you to distinguish caller and callee hangups. I suppose
dial time limit will match Callee hangup, but you can check that by
${ANSWEREDTIME} or some sort of timestamp checking before and after Dial
(altough that would include ringing time)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
Hello List,
Ok, I solved it by using this code. This will work for me since the variable
${timeleft} is always in complete seconds. Thank you all for the ideas and
pointers :)
context hangupcause {
s = {
Set(timeleft=7000);
Dial(SIP/1203,30,gL(${timeleft}[:4000][:4000]));
if(${timeleft} = (${ANSWEREDTIME}*1000)) {
jump [EMAIL PROTECTED];
} else {
jump [EMAIL PROTECTED];
}
}
h = {
NoOp(Caller Hangup);
}
}
context hangupcause2 {
s = {
NoOp(Callee Hangup);
}
}
context notimeleft {
s = {
NoOp(Time's up!);
}
}
I would change that to = just for reliability - you never know :)
Regards,
Atis
--
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VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] hint status unavailable
On 3/20/08, Stefan Schmidt [EMAIL PROTECTED] wrote: hello, i am trying to set up a asterisk server (version 1.2.26 by now) with realtime configuration but the user shouldnt register directly to the server, instead i have set up a ser registration proxy. Everything works fine so far, but i can´t use the hint feature. Its possible to subscribe to a given hint, but the status is allways unavailable and also i dont get a notify. Could someone help me finding a solution for this problem? I want to get notifies for hints where the user isnt registered on the asterisk itself. Thanks best regards Steve Smith ps: allready posted on Dev lists with the result this isnt a dev- related topic. What did you mean by realtime config? Realtime SIP users, realtime dialplan? If it's just SIP users, you should have some success with rtcachefriends=yes in sip.conf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint status unavailable
On 3/20/08, Steve Davies [EMAIL PROTECTED] wrote:
On 20/03/2008, Johansson Olle E [EMAIL PROTECTED] wrote:
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything
works
fine so far, but i can´t use the hint feature. Its possible to
subscribe
to a given hint, but the status is allways unavailable and also i dont
get a notify.
Could someone help me finding a solution for this problem? I want to
get
notifies for hints where the user isnt registered on the asterisk
itself.
That is something we all want, but it doesn't work now unless you add
a third party software. I haven't seen anything that solves the issue,
but have
a few ideas.
The question here is how should one asterisk be able to know anything
about
devices it doesn't control? It's a pbx, not an artificial intelligence
software.
There is work going on in the development group to make it possible
to apply a message bus between Asterisk servers so that Asterisk
servers can share call states. When that is up and running and tested,
it will be part of a future Asterisk release.
So the answer in short is not possible today, maybe tomorrow
Regards,
/olle
Perhaps in a similar thread, is it possible to somehow SET the state
of a hint from the dialplan? Perhaps a bit like:
Set(${ChanIsAvail(hint,234)}=Busy)
or perhaps have a pseudo-device facility where you can add it to the
end of the hint list to hint-the-hint. Something like:
exten = 234,hint,SIP/myphonePSEUDO/234
exten = *78,1,ChanAvailIs(PSEUDO/234,Busy)
exten = *791,ChanAvailIs(PSEUDO/234,Unknown)
This could be very useful for presence indication.
Huh, this hint hint would be useful for queues with local channel
state_interface too.. i think some general usage way could be added to
allow combining of device states.
Regards,
Atis
--
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VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Handling 3 different call ending causes
On 3/17/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
You can handle the 'h' extension in the dial plan, which will supply the
${CHANNEL} that was hung up, and possibly some additional dial plan
variables as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
Using these, you can piece together who hung up on whom, etc.
#2 is handled by fallthrough in the dial plan that causes the instructions
to continue executing to the next priority for that extension, whereas if
the call completes (Dial() is successfully connected), this does not happen.
I''ve tried to use the h extension in combination with the ${CHANNEL} in the
dialplan as suggested on the wiki page, but I haven't had any luck with it.
For this test I have a Sipura phone with number 1003 and a X-lite with 1203.
If I let the time go by (the 20 seconds defined in the Dial Command) I get
the following:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup
is
SIP/1003-08a491b8) in new stack
If I let the Sipura hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup is
SIP/1003-08a491b8) in new stack
Lastly if I let the X-lite hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup is
SIP/1003-08a491b8) in new stack
Yes they are all the same :(
Perhaps there's something wrong with my code? Its just a small context with
the following for this test:
[hangupcause]
exten = s,1,Dial(SIP/1203,30,gL(1[:5000][:5000]))
exten = s,2,NoOp(Callee hangup)
exten = h,1,NoOp(Channel hungup is ${CHANNEL})
Have I missed something basic here or what?
This should allow you to distinguish caller and callee hangups. I
suppose dial time limit will match Callee hangup, but you can check
that by ${ANSWEREDTIME} or some sort of timestamp checking before and
after Dial (altough that would include ringing time)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Asterisk 1.4 reliability problems
I would suggest taking latest 1.4 branch from SVN (or 1.4.19-rc3 when it's out). There has been few deadlocks fixed since rc2. Recompile asterisk with DEBUG_THREADS enabled (in make menuselect), If you're not using safe_asterisk script to start it, you should execute also ulimit -c unlimited before launching asterisk.. When your asterisk is deadlocked, open CLI and execute core show locks. Copy that output, and submit to bugs.digium.com - it will tell developers where exactly is problem. Then, do killall -11 asterisk. It will dump asterisk to core file, and that might provide helpful information later. If your have been requested backtraces, look in /tmp (or in directory you launched asterisk from) for core file. Open that core file with gdb /usr/sbin/asterisk core. and take a dump of thread apply all bt full (make sure you set set pagination off in gdb before this) Regards, Atis On 3/18/08, Norman Franke [EMAIL PROTECTED] wrote: Check around on bugs.digium.com. You'll find a number of issues reported that sound similar. I'm hoping that 1.4.19 will fix a lot of stuff, since the release candidates seem much more stable to me. I couldn't keep Asterisk up for more than a few days before on 1.4.18. I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. 1.4.17, which you mentioned, is also very buggy. 1.4.18 fixed many issues. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 7:40 AM, [EMAIL PROTECTED] wrote: We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy problem causing playback () to fail
On 3/18/08, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am having problem with my Asterisk installation and find out it has to do with ztdummy. if the ztdummy module is loaded, the asterisk playback() command will not play files. DTMF is still properly received. If the ztdummy module is unloaded, sound playback works again. Here is my version zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I just have ztdummy removed forever and the system will work? I saw from manual that ztdummy is required. ztdummy is required by meetme application. If you have no intention to use it, you might very well remove. I've seen this problem once, however recompiling everything and restarting helped me. I would suggest you just doing make clean on zaptel and asterisk, then compile first zaptel, then asterisk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk.conf uniquename or sysname for uniqueid field in CDR
On 3/18/08, Vieri [EMAIL PROTECTED] wrote: --- Vieri [EMAIL PROTECTED] wrote: I set uniquename = MYHOST in asterisk.conf (under [options]) so that my uniqueid data shows up as MYHOST.time.seq. First of all, I would like to know if uniquename (or sysname?) will still be valid across future * versions (mainly 1.6). Secondly, is there a way to specify uniquename as an asterisk option at the command line? (asterisk -h doesn't show me anything regarding this feature) Finally, how can I set uniquename to a system value (say, dynamically set to whatever `hostname` yields)? Something like uniquename = `hostname` so that I don't have to statically set it on each asterisk server? I just realized that uniquename is only available after applying the BRISTUFF patches. So let me rephrase my question: will Asterisk ever include the uniquename feature in its base code? If not, why? (I would prefer not to apply BRIstuff since I don't have Junghanns hardware). Look into doc/asterisk-conf.txt - probably you can use systemname. Asterisk config files also support #exec directive, so you can create your regular asterisk.conf without sysname and create shell script: #!/bin/bash cat asterisk.conf.template echo sysname=`hostname`. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
On 3/17/08, Rajkumar S [EMAIL PROTECTED] wrote: On Mon, Mar 17, 2008 at 6:30 PM, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: Forgot to add: Multiple queues fo sip phone, it is normal that sometimes it is ringed, as reported busy for 1 queue and free for another. you limitited incoming call to max 1 ' incominglimit=1' so ;) My understanding was that if a SIP phone is busy, either due to a call from queue or a call from another sip phone or even making an out bound call, the queue application would detect that and skip trying that channel. Is this assumption wrong ? If that would be queue, it would have different log entry. This seems, a result from Dial(SIP/2505,,). There are two different settings. You can increase call-limit (or incominglimit) in sip.conf - so devices will be able to take several simultenous calls. So, even if SIP device has one call (and call-limit is more than one), device state of SIP device will be In Use, and that's where ringinuse parameter of Queue application comes in - if set to 0, Queue won't ring and you will see a bit different message. Hope that this explains architecture. As for current problem - i suspect that device state don't get updated correctly for Queue application, so Queue tries to dial device, and call-limit blocks it from doing so. There's a patch, currently in testing (issue 12127), it should fix this, however if you intend to keep incominglimit to 1, and don't use local channels - there's nothing to worry about. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Order of queue member list
On 3/17/08, C. Chad Wallace [EMAIL PROTECTED] wrote: We just recently upgraded from Asterisk 1.2 to 1.4, and quickly noticed a change in the behaviour of the queues--a change that we cannot live with. We've used AddQueueMember/RemoveQueueMember to manage logging into and out of our queues for over a year now with Asterisk 1.2, and in that version the queue members were sorted in such a way that the person who had been logged in the longest would be the first one to get a call. But when we deployed 1.4 last week, we noticed that the member list was no longer sorted based on login time. It seemed to have a pre-set order that members were always placed into. After looking at the code (apps/app_queue.c), I found the cause of this. In 1.2, the members were stored in a linked list, so when someone logged in, they were placed at the end, and when calls were handed out, it was done starting at the front of the member list (or vice-versa, but either way, it has the same effect). In 1.4, the member list was changed to an ao2_container, which apparently uses a hash table, and iterates through the list in a fixed order, meaning one of our agents is always the favourite for a call, and it is quite unfair. Now, I know that the ordering of members in the queue in 1.2 was not documented, and it may not have even been intentional, but it was very appropriate for our business model, and we'd like to find a way to get it back. Is there a way to control the order in which the ao2_iterator returns the items? Even a random distribution would be better than the current--which always favours some agents over others. And before anyone mentions the strategy setting in queues.conf, I should say that we use leastrecent, but because of the ratio of agents to queues in our business, the strategy doesn't come into effect immediately. With many agents answering each queue, it takes a while for each of them to get a call. Until then (which usually takes about half of each day), the calls are distributed based on the ordering of the member list. We have switched to the rrmemory strategy for now, but we've yet to notice what effect that has--and our ideal would be to use leastrecent along with the behaviour that Asterisk 1.2 exhibited. I would suggest adding: cur-lastcall = time(NULL); within create_queue_member() function. This will allow you speed bonus from hashtable in some places, and will make sure the login time gets registred. You can also consider updating lastcall in set_member_paused() - i'm having both of those. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue log vs. cdr
On 3/13/08, Vieri [EMAIL PROTECTED] wrote: Hi, Surely, I must be overlooking something. If I run the following SQL queries I don't get the same number of rows. Is this coherent? mysql select * from queue_log where queuename = '4010' and FROM_UNIXTIME(time) between 2008030800 and 20080313145900 group by callid; 357 rows in set (0.01 sec) mysql select * from cdr where dst = 4010 and calldate between 2008030800 and 20080313145900 group by uniqueid; 219 rows in set (0.19 sec) Thanks! Hmm, didn't knew that queue_log can be written into MySQL.. that's something useful for me :) Is callid in queue_log the same uniqueid? You can do something like this: CREATE TEMPORARY TABLE a TYPE=MEMORY select * from queue_log where queuename = '4010' and FROM_UNIXTIME(time) between 2008030800 and 20080313145900 group by callid; CREATE TEMPORARY TABLE b TYPE=MEMORY select * from cdr where dst = 4010 and calldate between 2008030800 and 20080313145900 group by uniqueid; and then compare: SELECT * FROM a WHERE callid NOT IN (SELECT uniqueid FROM b) SELECT * FROM b WHERE uniqueid NOT IN (SELECT callid FROM a) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call tracing - Asterisk 1.4
On 3/12/08, Louwrens Benadé [EMAIL PROTECTED] wrote:
That's the way how i have it workin. Of course, this wasn't done in
one day, i've been working on details for weeks. Generally i use
CDR,
and manipulate it with ResetCDR, NoCDR, and link them together by
first uniqueid. This works great for IVRs, extension2extension
calls,
outgoing calls, blind transfers, queues. So i can take any call and
see what was done to it, where it was transferred, duration of each
step and so on, so on. However it won't work for conferences (you
don't know that person will join conference unless it joins, and
then
it's too late to change uniqueid, first cdr may be already writted),
and i haven't implemented that for blind transfers. But generally if
you want all that in DB, manipulating CDR is the way to go.
When you will have more specific questions, please ask, i'm sure
somebody will answer :)
So I'm not the only one :)
Ok, because of my lack of knowledge about using the dial-plan, I've resorted
to using Trixbox (don't laugh). I've managed to find where the initial
uniqueid is inserted which I then pump into a variable, and from there into
the 'userfield' in the CDR.
The problem I'm having at the moment is that I can't figure out when the
next hit in the CDR takes place. I've found the macro that (I think)
generates it, but no matter what I try, I can't populate the 'userfield' for
the next event.
So here are my questions:
1. Is the next event in the CDR inserted by ResetCDR or NoCDR?
NoCDR wouldn't cause that, as that's supposed to skip posting current
CDR. Next entry would be caused by either ResetCDR(w) or some
application that creates new channel (i.e. Dial or Queue). You can
enable full log and set verbosity and debug to higher values, to see
all what's going on.
2. Can I use a locally defined variable ( exten =
s,n,Set(v_identme=${CDR(UNIQUEID)})) ) or do I have to use a global
variable?
I'm not sure about value of ${CDR(UNIQUEID)}, but you can use just
${UNIQUEID}. If you want to pass variable to child channels, you
should make it inheritable. I'm using:
Set(__call_id=${UNIQUEID})
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Call tracing - Asterisk 1.4
On 3/11/08, Louwrens Benadé [EMAIL PROTECTED] wrote: Hi guys I've just read this about the upcoming release of * 1.6: Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing. That sounds brilliant! But I'm in desperate need of something to handle call tracing in 1.4... Does anyone know how I can accomplish this? I thought about using the originating uniqueid and populate for every event related to the call (transfers, etc), but I'm having trouble reading the dialplan to see what executes where :( That's the way how i have it workin. Of course, this wasn't done in one day, i've been working on details for weeks. Generally i use CDR, and manipulate it with ResetCDR, NoCDR, and link them together by first uniqueid. This works great for IVRs, extension2extension calls, outgoing calls, blind transfers, queues. So i can take any call and see what was done to it, where it was transferred, duration of each step and so on, so on. However it won't work for conferences (you don't know that person will join conference unless it joins, and then it's too late to change uniqueid, first cdr may be already writted), and i haven't implemented that for blind transfers. But generally if you want all that in DB, manipulating CDR is the way to go. When you will have more specific questions, please ask, i'm sure somebody will answer :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audiocodes MP124-FXS replying BUSY when line is not.
:14] VERBOSE[31897] logger.c: Transmitting (NAT) to ee.ff.gg.hh:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport From: 28901-2067217913 sip:[EMAIL PROTECTED];tag=as18481a04 To: sip:[EMAIL PROTECTED];tag=1c1673732975 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 11:13:14] VERBOSE[30165] logger.c: -- SIP/90166-c45079a0 is busy [Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime: Everything is fine. [Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime: Delete SQL: DELETE FROM channels WHERE uniqueid = '1205172794.6453' [Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime: Deleted 1 rows on table: channels [Mar 10 11:13:14] DEBUG[30165] chan_sip.c: Call to peer '90166' removed from call limit 8 [Mar 10 11:13:14] VERBOSE[30165] logger.c: == Everyone is busy/congested at this time (1:1/0/0) - end of log --- -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
