Re: [asterisk-users] show queue's name and other info in incoming call to queue member
On Mon, Dec 7, 2009 at 10:00 AM, Giedrius Augys voi...@gmail.com wrote:
hello,
I've callcenter and our queue members want to see on their IP phone's
display queue's name , from which incoming call was originated, for
example client's_number - Sales. This problem appears when one member
can belong to couple queues. Work around would be setting calling name with
such information.
If Your phone supports text CLID:
Set(CALLERID(name)=${CALLERID(num) - Sales);
Queue(sales);
If not, You can just add some digit in front/end of CALLERID(num).
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] show queue's name and other info in incoming call to queue member
On Thu, Dec 10, 2009 at 2:54 AM, Atis Lezdins a...@iq-labs.net wrote:
On Mon, Dec 7, 2009 at 10:00 AM, Giedrius Augys voi...@gmail.com wrote:
hello,
I've callcenter and our queue members want to see on their IP phone's
display queue's name , from which incoming call was originated, for
example client's_number - Sales. This problem appears when one member
can belong to couple queues. Work around would be setting calling name with
such information.
If Your phone supports text CLID:
Set(CALLERID(name)=${CALLERID(num) - Sales);
Ooops, syntax validation was off:
Set(CALLERID(name)=${CALLERID(num)} - Sales);
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] queue issue
On Tue, Sep 1, 2009 at 4:35 AM, Paul Halespdha...@optusnet.com.au wrote: Miguel Molina wrote: Paul Hales escribió: I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH Hi, Maybe maxlen = 1? Cheers, Hmmm - almost. Maxlen limits the amounts of calls waiting for the queue, not the amount of callers talking to queue members. You can do any limitations i can imagine with Set(GROUP()=...) and GROUP_COUNT. Do You actually need rest of callers to wait in queue while one is speaking, or disconnect them before they enter queue? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 2009 - October 13 - 15 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 with rtcachefriends=no problems...
On Wed, Aug 26, 2009 at 12:11 AM, Mauro Sergio Ferreira Brasilmauro.bra...@tqi.com.br wrote: Hello there! Problem found. For some reason, the update statement below is generated with an invalid atribution of empty value '' to field port that is an integer. Because of that, this record keeps with prior fullcontact information that was updated by another client (which uses a different port) what leads to wrong client rtp packets routing... wow... that was weird... :-) [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query: UPDATE sip_buddies SET fullcontact = '', ipaddr = '', port = '', regseconds = '0', username = '', regserver = '' WHERE name = '101' [Aug 25 17:57:43] DEBUG[20801] res_config_mysql.c: MySQL RealTime: Query Failed because: Incorrect integer value: '' for column 'port' at row 1 First of all... my appologies by the false alarm. But now I need your help to identify why is this update statement being generated wrongly. Does someone have any idea ? Asterisk Realtime Architecutre currently treats all fields as strings. I wish too that it would take into account actual field type retrieved from DESCRIBE statement and add the quotes only if it's string. You can safely do ALTER TABLE sip_buddies CHANGE COLUMN port port VARCHAR(5); Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 2009 - October 13 - 15 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] application missed in asterisk 1.6.1 - SetCallerID()
On Wed, Aug 26, 2009 at 5:03 PM, harry Rrhm.noa...@gmail.com wrote: Hi A few day ago, I notice that some applications missed in asterisk 1.6.1 release even if *.so file which normally create them were compiled during Asterisk install. SetCallerID(), SetCIDNum(), SetCIDName(), SetLanguage() ... and maybe so more. anyone already notice that to ? If it's not normal, anyone have an solution to it ? Read the UPGRADE.txt Solution is to use functions instead: Set(CALLERID(name)); Set(CALLERID(num)); Set(CHANNEL(language)); etc Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 2009 - October 13 - 15 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] T.38 pass-through 488 handling problem
On Mon, Jun 8, 2009 at 2:06 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 INVITE INVITE ---200OK-- ---200OK-- ACK--- ACK--- INVITE w/T.38- --INVITE w/ T.38-- -488-- --ACK- --BYE- -200-- Asterisk does not forward the 488 back to the caller, but hangs up the callee's call leg. Further, the caller's call leg will not be hung up. Is somebody aware of this problem and a fix? T.38 passthrough is possible if BOTH devices support T.38, so Asterisk don't have to transcode anything. You could try 1.6 with some gateway app (don't remember if there exists any and in what state), or just write a RxFax which would then generate call with TxFax. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] T.38 pass-through 488 handling problem
On Mon, Jun 8, 2009 at 7:00 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Atis Lezdins schrieb: On Mon, Jun 8, 2009 at 2:06 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 INVITE INVITE ---200OK-- ---200OK-- ACK--- ACK--- INVITE w/T.38- --INVITE w/ T.38-- -488-- --ACK- --BYE- -200-- Asterisk does not forward the 488 back to the caller, but hangs up the callee's call leg. Further, the caller's call leg will not be hung up. Is somebody aware of this problem and a fix? T.38 passthrough is possible if BOTH devices support T.38, so Asterisk don't have to transcode anything. You could try 1.6 with some gateway app (don't remember if there exists any and in what state), or just write a RxFax which would then generate call with TxFax. That's not the problem. Asterisk should just relay back the 488 so that Faxing happens with g.711. Ok, then You have to look into headers and log, Asterisk should say something.. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 core CDR system for multilpe servers
On Thu, Jun 4, 2009 at 6:42 PM, Jeff LaCoursiere j...@jeff.net wrote:
On Thu, 4 Jun 2009, Danny Nicholas wrote:
Do you want a live repository or just a common gathering of the data? If
LR then you should set up a deamon on each box to transfer records as they
occur using something like the DBI functionality of PERL. If not, then just
do a mysql dump periodically and ssh the files to the common server.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo A
Gonzalez
Sent: Thursday, June 04, 2009 10:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about core CDR system for multilpe
servers
Hi all! I?m not sure if it is the correct place but, I?ve five boxes running
asterisk and each one with his own cdr mysql database. What Im looking for
is to get a core CDR system that holds information stored on each asterisk
server. Have you any suggestion/process to accomplish that?. Thanks!!!
Gustavo A. Gonz?lez
How about just configuring cdr_mysql.conf to connect to the one machine
you want to collect the records? No need to keep them on all the machines
and have some complex copying setup...
Exactly my point.
There's a system name (or something similar) option in asterisk.conf
which would prepend system name to ${UNIQUEID}, so You just have to
make sure that uniqueid is enabled in cdr_addon_mysql, so each CDR in
database will be marked from specific system.
However I would suggest not doing heavy SELECT's on this database, set
up another slave for reports, as each table lock will cause asterisk
posting a CDR to wait (and current call posting a CDR will wait in
silence)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Queue - Multiple Transfer
On Sat, May 30, 2009 at 3:50 AM, Kurian Thayil kurianmtha...@gmail.com wrote: Hi all, I ve setup a queue with 2+ agents for managing our inbound calls from customer. Using Asterisk 1.2.18 in a CentOS box. Agents login using AgentCallbackLogin application and I use a BASH AGI to accomplish this as there are some validations done with MySQL DB. Im aware that transfer could be done with option 't' in the queue() application and I was able to successfully transfer calling party (client) to another agent. But is it possible for the new agent to transfer this calling party to another agent? ie Does a second transfer is supported in a Queue? Second transfer wouldn't occur in Queue anymore, as first transfer makes call to bridge outside of Queue. If Queue doesn't pass t flag to the subsequent dial of transfer, You should create transfer context, and set TRANSFER_CONTEXT variable, and put a Dial with t flag there. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] New tutorial: storing audio recordings per day
On Mon, May 25, 2009 at 7:42 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:
Hi everyone,
after doing the same thing multiple times and struggling to remember how it
was done, I have prepared a small tutorial that explains how to save
monitored files in different folders per day. This is quite useful
becausethe resultingfile system is way more manageable than having maybe
100,000 files all saved in the same folder.
You can find the tutorial here:
http://astrecipes.net/index.php?n=387
As always, comments and suggestions are welcome.
l.
PS. I am also working on some scripts to normalize existing recordings
all-in-one-directory... if anybody is interested, please contact me.
Actually You don't have to create folders in advance, as Asterisk will
automatically create them when needed. Just make sure that Asterisk
process is owner of parent directory.
Set(__call_day=${STRFTIME(|${TIMEZONE}|%Y/%m/%d)});
Set(MONITOR_FILENAME=${MONITOR_DIR}/${call_day}/call-${UNIQUEID});
Monitor(ulaw,${MONITOR_FILENAME},b);
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Queue Load, Asterisk Disconnected
|20090513-092731|1242196051.186: CALLFILENAME=1242196051.186 -- AGI Script recordingcheck completed, returning 0 -- Executing Monitor(Local/2...@from-internal-e5d7,2, wav49|1242196051.186| mb) in new stack -- Executing Macro(Local/2...@from-internal-e5d7,2, dial|30|Ttr|221) in new stack -- Executing AGI(Local/2...@from-internal-e5d7,2, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is '0227559600' number is '0227559600' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 221 to extension map -- dialparties.agi: Extension 221 cf is disabled -- dialparties.agi: Extension 221 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 -- dialparties.agi: Checking CW and CFB status for extension 221 dialparties.agi: Extension 221 is not available to be called dialparties.agi: Extension 221 has call waiting disabled -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp(Local/2...@from-internal-e5d7,2, Returned from dialparties with no extensions to call) in new stack -- Executing Set(Local/2...@from-internal-e5d7,2, DIALSTATUS=BUSY) in new stack -- Executing GotoIf(Local/2...@from-internal-e5d7,2, 1?s-BUSY|1) in new stack -- Goto (macro-exten-vm,s-BUSY,1) -- Executing NoOp(Local/2...@from-internal-e5d7,2, Extension is reporting BUSY and has no Voicemail) in new stack -- Executing Busy(Local/2...@from-internal-e5d7,2, ) in new stack -- Local/2...@from-internal-e5d7,1 is busy -- Called Local/2...@from-internal/n == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Local/2...@from-internal-e5d7,2' in macro 'exten-vm' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Local/2...@from-internal-e5d7,2' -- Executing Macro(Local/2...@from-internal-a118,2, exten-vm|novm|225) in new stack -- Executing Macro(Local/2...@from-internal-a118,2, user-callerid) in new stack -- Executing Set(Local/2...@from-internal-a118,2, AMPUSER=) in new stack -- Executing GotoIf(Local/2...@from-internal-a118,2, 0?109) in new stack -- Executing Set(Local/2...@from-internal-a118,2, EMERGENCYCID=) in new stack -- Executing Set(Local/2...@from-internal-a118,2, AMPUSERCIDNAME=) in new stack -- Executing GotoIf(Local/2...@from-internal-a118,2, 1?7) in new stack -- Goto (macro-user-callerid,s,7) -- Executing NoOp(Local/2...@from-internal-a118,2, Using CallerID 0227559600 0227559600) in new stack -- Executing Set(Local/2...@from-internal-a118,2, FROMCONTEXT=exten-vm) in new stack -- Executing Macro(Local/2...@from-internal-a118,2, record-enable|225|IN) in new stack -- Executing GotoIf(Local/2...@from-internal-a118,2, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Local/2...@from-internal-a118,2, recordingcheck|20090513-092731|1242196051.188) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20090513-092731|1242196051.188: Inbound recording enabled. recordingcheck|20090513-092731|1242196051.188: CALLFILENAME=1242196051.188 -- AGI Script recordingcheck completed, returning 0 -- Executing Monitor(Local/2...@from-internal-a118,2, wav49|1242196051.188| mb) in new stack -- Executing Macro(Local/2...@from-internal-a118,2, dial|30|Ttr|225) in new stack -- Executing AGI(Local/2...@from-internal-a118,2, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi -- dialparties.agi: priority is 1 dialparties.agi: Caller ID name is '0227559600' number is '0227559600' dialparties.agi: Methodology of ring is 'none' asterisk*CLI Disconnected from Asterisk server Executing last minute cleanups [r...@asterisk ~]# Any Help will be highly appreciated Hello, First, 1.2 branch is quite old, and bugs are not going to be fixed anymore (except security releases). So in this case You're on Your own - if you find problem in code, You can fix it by Yourself, and keep the patch/share with others.. but it won't be accepted anywhere. Asterisk has had a lot of changes in Queue between 1.2 and 1.4, so You should consider upgrade benefits. If You want to debug - You should read on Asterisk Debugging - http://www.voip-info.org/wiki/view/Asterisk+debugging - and compile asterisk without optimizations, make it crash and then look into core file with gdb. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835
Re: [asterisk-users] Support of /* */ comments in ael.vim
On Mon, May 11, 2009 at 1:55 PM, Philipp Kempgen philipp.kemp...@amooma.de wrote: Olivier schrieb: It seems /* */ comments are not supported in ael.vim (which brings AEL syntax-highlighting to vim). Are C-style comments supported in AEL? I don't think so. They are. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Preferred language for Asterisk AGIs development ?
On Tue, May 5, 2009 at 9:52 AM, Kashif Naeem kas...@haditelecom.com wrote: Hello, We are going to start development for a product based over Asterisk. According to you, which is the preferred language for AGIs / IVRs development in Asterisk. I got opinions that Perl is going to be replaced by PHP for all future developments. Just use the language You write the rest of system. If it's web application, and You use PHP, You can use existing codebase of Your project :) Just be careful with this: asterisk and web server are usually run in separate processes, so having common cache is troublesome unless You check internally for effective uid and call sudo internally. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Outgoing Queues
Shouldn’t the member has the statics per queue? I mean, I have 2 queues test1 and test2, with member 1001 for example for both queues, if I make a call to queue test1 and the member 1001 answers the call, the statics for the member is up in both queues, (has taken 1 call….), this should be per queue basis don’t you think? Yes it's so, unless You have enabled shared_lastcall, in which case lastcall and call counter is shared across queues in order to acquire fair call distribution strategy. You shouldn't use queue data for statistics, there's queue_log for that. This is purely monitoring info which can get lost during restarts/reloads. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Record in mp3
Secondarily, MPEG audio compression takes a lot of CPU. Until the last few years, desktop CPUs weren't even capable of doing realtime MPEG audio compression, which is necessary if you're going to have the recording ready by the time the audio input is terminated. Above and beyond that, even modern CPUs are limited in how many concurrent streams can be MPEG-compressed, which may cause problems if you're encoding multiple channels to MP3 at the same time. Well, actually it's lot of CPU for encoding 44kHz stream. I wonder how it would scale to encode 8kHz.. We currently do a daily routine to compress all ulaw files to mp3 at night time, and it takes ~6 hours of processing on 1 CPU (no parallel processing). Regarding legal reasons, can't it be linked with lame within asterisk-addons? Regards, Atis ___ -- 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-dev] How to get to 10.000 open calls
# moving to -users as this belongs there. It is a nice idea to run several Asterisk processes simultenously, it will defineately help with multithreading. However I would suggest trying less instances - that would perhaps give greater benefit, as Asterisk has it's own threading. For example 8 instances of Asterisk / 4 instances.. However, in this case - if You go for splitting everything up, You could just simply drop in more machines. I think it would be more cost-effective to have 8 machines with 2 cores each. and that would additionally provide better I/O performance. Anyway, You can try throwing those calls and see how much can You get. As for directrtp=yes - i'm not sure what it does, but perhaps it's meant to be canreinvite=yes? Set it for each peer, and make sure You dial to peer, not to IP (as I recall - this didn't work globally) Regards, Atis On Wed, Apr 22, 2009 at 10:31 AM, Venefax vene...@gmail.com wrote: Yes, I have the box. And I will get the calls next week. I was thinking to use the Asterisk feature where you can start different Asterisk using -C \path_to\config\file, and start 15 instances. But to be able to load balance it is a nightmare, since many clients do not accept or follow redirects (SIP 302 Moved). I am out of tricks, unless I setup another technology for load balancing but then why not use the same (x) technology for everything? What technology would that be that can handle 10.000 sip connections, not touching the media? My Cisco 7301 would not scale so far out. -Original Message- From: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Wednesday, April 22, 2009 3:19 AM To: asterisk-...@lists.digium.com Subject: Re: [asterisk-dev] How to get to 10.000 open calls On Wed, Apr 22, 2009 at 02:48:11AM -0400, Venefax wrote: I am using 1.6.2 and directrtp=yes. I need to scale to 10.000 open calls on a box with 1288 GB or RAM and 16 Cores. Is there any modification to the source code that would be obvious, any bottlenecks? I will never to transcoding and the media should, theoretically, flow outside. I have 15 IP addresses already configured in the same box, on two different nics, to spread the interrupts. Is this a dream or will this work with some tweaking? Do you have the system now? While it's most likely be a dream, identifying the current bottlenecks might be useful :-) Just a few uneducated guesses of my own: * More than one IP per NIC won't help and only cause some administrative issues * I'm not sure how much the extra memory can help. I suspect htat if you boot the system with mem=whatever_needed_for_16GB the results won't differ greatly * It would also be interesting to see how the results scale with various values numbers of cores. This is again something you can set at boot (numcpus=N). I wonder just how far from linear it will be. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Fri, Apr 17, 2009 at 1:35 PM, Florian Hackenberger f.hackenber...@chello.at wrote: On Friday 17 April 2009, Michael wrote: Trying to link Hylafax - Callweaver or Asterisk is unfortunately a waste of time. You need to use the built in fax support and write from scratch the necessary scripts and dial plan to deal with faxes. Ok, that's fine. Well, not really a waste of time. As I mentioned - Hylafax has many desktop clients, it's better to just write few scripts than to design desktop software for your own setup. If You have a need of sending faxes, You'll probably need a desktop client too. I have successfully got a T.38 set up working, but it wasn't devoid of a lot of bother along the way. And how do the T.38 calls get to callweaver? Directly from the SIP provider, or does asterisk forward them to callweaver? Which version of asterisk (on which distribution) and which version of callweaver are you using? Ok, our setup is the following: Inbound call arrives from SIP provider to Asterisk 1.4.19 Asterisk Dials Callweaver (1.2.0 as I recall) on localhost CallWeaver uses RxFax, which causes call to be switched to T.38, Asterisk does T.38 passtrough. CallWeaver executes shell script at the end which emails the .tiff file to recipient. User prints document to Hylafax Desktop client. Document is sent to Hylafax server Hylafax executes shell script specified in SendFaxCmd. Shell script creates callfile for CallWeaver CallWeaver dials destination number to Asterisk Asterisk forwards call to SIP operator CallWeaver uses TxFax to send .tiff file already generated by CallWeaver. As we are currently on stable 1.4 version, we chose to use CallWeaver for this, but we plan to simplify whole setup when migrating to Asterisk 1.6, which would take over CallWeaver functions. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Fri, Apr 17, 2009 at 4:03 PM, Michael mich...@networkstuff.co.nz wrote: On Sat, 18 Apr 2009 00:54:07 you wrote: Well, not really a waste of time. As I mentioned - Hylafax has many desktop clients, it's better to just write few scripts than to design desktop software for your own setup. If You have a need of sending faxes, You'll probably need a desktop client too. I wish it wasn't a waste of time because Hylafax would make a great front end to a T.38 SIP system. I use Hylafax myself, but not with SIP. The problem is that there is no reliable, or really any viable way to achieve this when using T.38 as the carrier uplink. Could You explain this? I really don't understand Your point. The setup you describe does not have a audio data path connection to Hylafax and I wonder why the convoluted method when the same could be achieved using Callweaver alone and some custom scripting. Why would the audio data path would be necessary? In our setup CallWeaver effectively acts as modem, and talks T.38 with provider. Please see my previous statement about desktop client software. I doubt that this can be simply achieved with custom scripting. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Fri, Apr 17, 2009 at 4:31 PM, Michael mich...@networkstuff.co.nz wrote: The problem is that there is no reliable, or really any viable way to achieve this when using T.38 as the carrier uplink. Could You explain this? I really don't understand Your point. On voip-info there is a how to using T38modem. Congrats to anyone who can get it working. Well, i initially wrote that howto, after numerous hours of unsuccessful compilations and wrong versions, but T38modem didn't prove to work with our provider. Later, one Russian guy managet to get this working with he's provider. That's why i put CallWeaver (which basically has the same T.38 stack as Asterisk 1.6) in it's place. The setup you describe does not have a audio data path connection to Hylafax and I wonder why the convoluted method when the same could be achieved using Callweaver alone and some custom scripting. Why would the audio data path would be necessary? In our setup CallWeaver effectively acts as modem, and talks T.38 with provider. Fax information data path to be pedantic. Data from Hylafax to CallWeaver is passed as TIFF image - thus no data/quality loss. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Exit Dial Application
On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller
fuch_li...@kurtkrenn.com wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Danny Nicholas schrieb:
Here's how core show application dial says you should do it:
Change your dial to
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
I'm not sure if this is correct. core show application dial says:
Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL])
If I configure what you wrote, then callback is passed as URL to the called
party.
The optional URL will be sent to the called party if the channel supports
it.
I don't think that's what I want.
What I want is: If A dials B and B doesn't answer, A can press 5 and place an
automatic
callback. If B is back and places or takes a call, the automatic callback to
A should be
started.
I've found a possibility to do this via answering the call before the dial.
But ... that's
not an ideal solution. I would prefer not to answer the call in the dialplan.
Does the
option 'd' implies an answered channel? Or is this a Bug?
I think the limitation could be by analogous Zap phones, as they
probably don't support sending DTMF on unanswered channel. You could
try it opposite way - Dial from SIP phone to Zap.
Regards,
Atis
--
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a...@iq-labs.net
Skype: atis.lezdins
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Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
On Tue, Apr 14, 2009 at 4:52 PM, Florian Hackenberger f.hackenber...@chello.at wrote: On Tuesday 14 April 2009, Michael wrote: asterisk-1.6 with app_fax built-in Try 1.6. You'll be glad you did. While I have not tried Asterisk 1.6 because I settled on Callweaver at the time (which has native T38 support), I *strongly* recommend going with software that has native T38 support. This could be Asterisk 1.6 or Callweaver. So +1 for the above. Thank you all for your comments. I'm unfortunately stuck with asterisk 1.4, because it took a considerable amount of time to patch it to be stable (and feature complete) enough for my requirements. Integrating Callweaver might be an option, however I would loose the advantage of easily controlling the fax call before handing it off to hylafax. With asterisk 1.6, is it possible to use hylafax, or would asterisk terminate the fax calls itself? Are there any success stories with t38modem, asterisk and hylafax? I tried T38modem, it works nicely in local setups, but I never got it working with our SIP provider. As for stable Asterisk 1.4, we also do have it, but there's CallWeaver on the same machine sending all fax calls to Asterisk 1.4 in T.38 passthrough (for unified billing and logging). Hylafax has one big advantage - many desktop clients that allow easy fax sending. It can be configured to execute custom scripts that grab generated .tiff files and feed them to CallWeaver. Just search list archives, I've writen detailed descriptions of this mechanism. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Ignoring time spent waiting in queue in CDR
On Tue, Apr 14, 2009 at 4:15 PM, Jared Smith jsm...@digium.com wrote: - Scott Gifford sgiff...@suspectclass.com wrote: The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent. You're absolutely right -- the CDR information is for the entire call. Instead, look at the queue log (typically written to /var/log/asterisk/queue_log). It will tell you most (if not all) of the information you need for creating call queue reports. Most, but not all.. Short answer - do an ResetCDR() before entering Queue. This will set CDR Answer status to NO ANSWER, and next answer by agent will answer the CDR, so You will have two distinct values - duration and billsec. Duration will be total length, but billsec will be conversation time. On the other hand, You can easily link queue_log with CDR, by enabling storing of UNIQUEID within CDR record. The same UNIQUEID will be in queue_log for CONNECT and HANGUP events. We do have purely CDR based billing implemented, but it requires some attention upon upgrading Asterisk, as some tiny details might change, so careful testing is a must. We are happy, as it allows to see complete call flow for every call, group them easily etc. There's a sample screenshot: http://ftp.iq-labs.net/screenshots/cdr_view.jpg However You should really have a think about what are Your requirements, and how they could change in future. Perhaps using the queue_log would allow rapid implementation and changes. Also, make sure to take a look at queue_log on Asterisk 1.6.0/1.6.1, they have some nice features added. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Exit Dial Application
CLI core show application Dial
d- Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
if it exists.
Regards,
Atis
On Tue, Apr 14, 2009 at 7:49 PM, Christoph Fürstaller
fuch_li...@kurtkrenn.com wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial,
but I wanna do it during the dial, when user A is waiting for user B,
answering the phone. This should be possible, right?
I hope anyone knows if this is possible.
Chris...
Danny Nicholas schrieb:
I'd change callback to this
[callback]
Exten = s,1,Playback(press5msg)
Exten = s,n,Waitexten(5)
Exten = s,n,Hangup
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup if
they don't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fuerstaller
Sent: Tuesday, April 14, 2009 5:04 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Exit Dial Application
Hi,
I' try to implement an automatic callback mechanism, just for local SIP
calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be
possible to press
5 to place an callback.
Here is my dial:
exten = _X.,1,Set(EXITCONTEXT=callback)
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for callback.
[callback]
exten = 5,1,agi(str_concat.sh)
exten = 5,n,Hangup
If I call someone and press 5, nothing happens. What could be a problem?
DTMFmode is RFC2833 for all SIP Accounts. DTMF's are transmitted correctly,
I can enter
the voicmail menue.
I'm using Asterisk 1.4.21.1.
Any successions are very appreciated.
Chris...
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- --
commpany dialog solutions gmbh
Dipl.-Ing.(FH) Christoph Fürstaller
IP-Communications
Ischlerbahnstraße 14, 5301 Eugendorf
Tel: +43 662 879512 Fax: +43 662 875960
IP-Tel: +43 780 commpany (26667269)
Email: c.fuerstal...@commpany.at
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (MingW32)
iEYEARECAAYFAknkvqoACgkQR0exH8dhr/ZqRACfV7KLoTMl9RgH0QNIPiJ/Gq9G
5dcAoIVK3L7pxTBLZrDi+kJGpOCPVa47
=hEGE
-END PGP SIGNATURE-
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--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Ring All Queue
On Tue, Apr 14, 2009 at 9:09 PM, Jim Dickenson dicken...@cfmc.com wrote:
At least in version 1.6.0.x you can specify a macro to be executed when the
agent answers the queued call. This is an argument to the queue application.
Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,g
osub][,rule])
The optional macro parameter will run a macro on the calling party's channel
once they are connected to a queue member.
Here is what my Macro does:
exten = s,1,UserEvent(DidQueue,ActionID:${CfMC_ActionID} ${UNIQUEID}
${CHANNEL} ${CfMC_AgentToUse} ${CfMC_DialInfo}
${CfMC_QueueToUse} ${MEMBERINTERFACE} ${MEMBERNAME})
${MEMBERINTERFACE} and ${MEMBERNAME} have info about the agent that answered
the call.
Just test this with multiple simultenous answers, so You don't get any
surprises. I'd recommend putting Wait(10) into that macro (actually
GoSub in 1.6) and trying to pick up second ringing phone while first
is in Wait().
I haven't gotten into 1.6 yet, but here are some related problems on
1.4 with some backports:
http://bugs.digium.com/view.php?id=13335
http://bugs.digium.com/view.php?id=14859
Once You'll get the agent in some variable within answer part of
dialplan, it's just a matter of storing this into per-call database
entry and reading from parrent channel. See function DB and variable
UNIQUEID for that.
Of course, if You need it only on hangup, Luis suggestion will work
just fine, use Asterisk Realtime engine to read value from realtime
queue log.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Exit Dial Application
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
doesn't : /
Oh, sorry, missed that part :)
Try enabling full log in logger.conf, set verbosity to 3 and debug
to 1, and see what goes in it.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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Re: [asterisk-users] Exit Dial Application
On Tue, Apr 14, 2009 at 9:14 PM, Christoph Fürstaller fuch_li...@kurtkrenn.com wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Atis, No problem : ) I tried it again, here is the log output: -- Executing [...@from-pbx:1] Set(Zap/31-1, EXITCONTEXT=callback) in new stack -- Executing [...@from-pbx:2] Dial(Zap/31-1, SIP/236||d) in new stack -- Called 236 -- SIP/236-0825f928 is ringing -- SIP/236-0825f928 is ringing -- SIP/236-0825f928 is ringing -- SIP/236-0825f928 is ringing That's CLI interface output, log should have timestamps and much more detail in it. Check /var/log/asterisk/full (assuming default install location). You'll need to enable full line in logger.conf, restart Asterisk and issue core set verbose 3 and core set debug 1 in CLI. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Exit Dial Application
On Tue, Apr 14, 2009 at 11:11 PM, Christoph Fürstaller fuch_li...@kurtkrenn.com wrote: Thanks for the hint. I've looked aht the full log. I've attached a snipplet from the file. But I can't see anythin which can help me. Very interesting, but not helpful for me : / Is it possible to deactivate the 'd' option? Or what else could cause my problem? Ok, at first glance the app_macro looks suspicious, can You try calling dial without Macro? If unsuccessful, You could enable debug level 2, it will tell way much more of everything, including DTMF events etc. Btw, does DTMF work at all for this Zap/ line? You could verify that by using Read before Dial. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 is not designed for University with largeuser base?
About the database polling - i think for such a installation you could create something like a database to config files script - so not to use realtime. This should solve this problem. No need for that. There's rtcachefriends setting in sip.conf, and if you have to update user credentials from some interface, just issue sip prune realtime peer xxx trough manager. Also, in Asterisk 1.6 res_mysql driver can take advantage of MySQL master/slave setups, so You can distribute Your database load to separate read/write hosts. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] colorized logfiles in asterisk 1.6.0.6
On Fri, Mar 6, 2009 at 11:59 PM, Tiago Durante tiagodura...@gmail.com wrote:
On Fri, Mar 6, 2009 at 10:39 AM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
Danny Nicholas wrote:
The log files themselves are not in color. It would be a style sheet change
on the GUI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johann
Steinwendtner
Sent: Friday, March 06, 2009 2:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] colorized logfiles in asterisk 1.6.0.6
Hello !
I've upgraded our testsystem from asterisk 1.4.21 to asterisk 1.6.0.6.
We 've noticed that the log files are now in colour.
I could not find a note in the upgrade section about this.
Is this a feature or a bug ?
It might be usefull to have them not in colour.
best regards
Hans
Sorry, that I wasn't clear enough. The logfiles contains escape codes +
the colour codes.
e.g.:
[Feb 12 13:38:30] VERBOSE[19816] logger.c: == Registered custom function
'ESC[1;36;40mSQL_ESCESC[0;37;40m'
[Feb 12 13:38:30] VERBOSE[19816] logger.c: func_odbc.so = (ESC[33;40mODBC
lookupsESC[0;37;40m)
[Feb 12 13:38:30] VERBOSE[19816] logger.c: == Registered application
'ESC[1;36;40mReadFileESC[0;37;40m'
I do not use a GUI.
same thing happens to me, as far as I noticed only in one server...
asterisk 1.6.0.5...
when you do a, lets say, tail -f /var/log/asterisk/full its kinda of
cool, because you can check the log with colors... but the log itself
become a mess...
regards,
Well, it's nice in console, but not that for analyzing. I would prefer
disable option for that.
Also i wrote simple wrapper, to allow browsing logs from web. It might
need some tuning (see if ANSI: string comes out and add some colors
- only 2 colors are converted for now), but it works nice for my
asterisk logs :)
Regards,
Atis
function logfile_ansi_to_html($str) {
$tokens = explode(chr(27).'[',$str);
$result = array_shift($tokens);
foreach ($tokens as $k=$v) {
$end = 8;
$code = substr($v,0,$end);
if ($code=='0;37;40m') $result .= '/b';
else if ($code=='1;36;40m') $result .= 'b style=color: navy';
else if ($code=='1;35;40m') $result .= 'b style=color: #cc3366';
else $result .= 'ANSI:'.$code;
$result .= substr($v,$end);
}
return $result;
}
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Bounty- CDR Bug Fix
Well, i can share mine backports of queue_log into mysql for 1.4.
Basically you need two backports (that's why there are numerous
files). Realtime store/destroy allows Asterisk Realtime engine to use
INSERT's on MySQL. It needs two patches - one for Asterisk, one for
Asterisk-addons (mysql part). And then there's itself queue_log
realtime patch.
I'm using it on Asterisk 1.4.19 for some half year already, so it
could be considered stable. I just tested and it does apply cleanly to
Asterisk 1.4.23 (and was previously working with latest Addons-1.4.7).
Also an advantage of this is - that it's already merged into 1.6.0, so
upgrade shouldn't be a problem.
So, some brief instructions:
1) apply
http://ftp.iq-labs.net/realtime_store_destroy-1.4/asterisk_realtime_store_destroy_1.4.19.patch
to Asterisk
2) apply
http://ftp.iq-labs.net/queue_log-1.4/asterisk_queue_log_realtime_1.4.19.patch
to Asterisk
make make install
3) make dist-clean on Asterisk-addons
4) Apply
http://ftp.iq-labs.net/realtime_store_destroy-1.4/asterisk_addons_realtime_store_destroy_1.4.6.patch
to Asterisk-addons
make make install
* ensure that res_mysql.conf has working connection:
[general]
dbhost = localhost
dbname = asterisk
dbuser = asterisk
dbpass = pass
dbport = 3306
dbsock = /tmp/mysql.sock
* add to extconfig.conf:
queue_log = mysql,asteriskcdrdb,queue_log
* create mysql table:
CREATE TABLE queue_log (
id int(10) unsigned NOT NULL PRIMARY KEY AUTO_INCREMENT
time int(10) unsigned,
callid varchar(20),
queuename int(10) unsigned,
agent varchar(40),
event
enum('ABANDON','ADDMEMBER','AGENTCALLBACKLOGIN','AGENTCALLBACKLOGOFF','AGENTDUMP','AGENTLOGIN','AGENTLOGOFF','COMPLETEAGENT','COMPLETECALLER','CONFIGRELOAD','CONNECT','EDITMEMBER','ENTERQUEUE','EXITEMPTY','EXITWITHKEY','EXITWITHTIMEOUT','PAUSEALL','PAUSE','QUEUESTART','REMOVEMEMBER','RINGNOANSWER','SYSCOMPAT','TRANSFER','TRANSFERATTENDED','UNPAUSE','UNPAUSEALL'),
data varchar(255)
)
Regards,
Atis
On Thu, Mar 5, 2009 at 8:54 PM, Robert Broyles rob...@poornam.com wrote:
The patch I was referring to is:
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queue_logging
It doesn't work for the current SVN 1.4
--
Regards,
Robert Broyles
Anthony Francis wrote:
Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a
patch way back for 1.2 that allowed all queue log events to sh,ow up in
the AMI, just haven't had time to make a new version for 1.6.
Maybe this time I can get the patch in trunk and it will always be there.
Robert Broyles wrote:
Problem is, without going to 1.6, I can't get the queue log or events
posted to MySQL in realtime.
There used to be a patch out there for queue_log, but it doesn't work
with versions 1.4.21 or higher.
--
Regards,
Robert Broyles
Anthony Francis wrote:
Robert Broyles wrote:
I saw some of the heat about the $20 bounty earlier. So I don't want to
put a low bounty out.
Quote me a bounty, and I'll see if I can get it approved by management. :-)
I'm in need of getting this bug fixed. Bug has all of the details, but
basically 1.4.22 broke it all.
I've waited as long as I can - hoping the bug would 'resolve itself' -
but now I'm putting a bounty out on it.
http://bugs.digium.com/view.php?id=13691
I would not recommend using CDR's for queue data, instead I use the
queue events, or at a minimum the queue log.
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a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Bounty- CDR Bug Fix
On Wed, Mar 4, 2009 at 6:24 PM, Robert Broyles rob...@poornam.com wrote: By the way, I'm more than happy to send murf a case of rootbeer (or real beer assuming he's legal :-P ) if this bug and/or related bugs can be resolved soon. :-) Bottle of Riga Black Balsam (45%), just have to figure out a way to send it :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Meetme - play the name
On Mon, Dec 29, 2008 at 5:51 AM, sasikala kala sasi_jeyalaks...@yahoo.com wrote: Hi, Thanks for your prompt reply. Let me clarify some thing on my requirement. That's not a realistic expectation. How can you presume that because callerid is xyz that it's always the same person calling? You can not. Office's routinely have one main number with callerid being the same for all office users. I would not be surprised to find two users calling in separately from the same office having the same callerid, where you can not tell them apart based on callerid. In my case, every person is having DID (individual, unique across whole office), so this feature is called for. This is good reasoning for local users. The name prompt from voicemail could be used and made more generic. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Using Asterisk to measure call quality: Introducing Recqual
On Mon, Dec 22, 2008 at 5:37 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: Recqual is collection of scripts using Asterisk and other Linux utilities to measure call quality on an automated basis. -How it Works- Recqual was designed to detect audio quality problems in a call path that may not be visible from the technology being used locally. Whether it's SIP, ZAP, or IAX the fact is there are many potential sources of call quality problems in just about any call being made. Often times a SIP provider may resell services, for example. While the delivery of IP packets to/from this provider may look excellent there may be other problems upstream that an analysis of the IP packets, path, etc may not be able to detect. In scenarios such as this the only way to identify call quality problems is to analyze the audio itself. Regardless of method or transport being used, the goal of any telephony system is to deliver reliable, consistent call quality. Recqual is designed to allow you to place a large number of automated calls (using Asterisk) using different call scenario files. The key here is consistency. When Asterisk places the outbound call (and answers the inbound call) it will generate a set of tones while recording the return audio path. Once the run has finished Ecasound will run with various filters and noise gates to detect certain amounts of distortion, signal loss, etc. Calls either pass or fail based on how much variation there is in the audio once it has been returned. Of course you can pass audio through any combination of networks - including the PSTN. Almost any call quality problem(s) can be detected with this method. Whether it's one way calls, echo, dropped packets, distortion, etc ecasound should be able to isolate the problem calls. If not you can just tweak the script ;). Only calls that fail are saved. These files can be imported into your favorite audio processing utility and/or run through Ecasound again if you'd like to tweak the process script to detect them automatically. Recqual has been designed (and optimized) to work with SIP channels. For example, it has the ability to correlate problem calls with specific RTP endpoint IP addresses. However, due to the protocol independent nature of Asterisk you can use just about any channel type with a few simple changes. --- So there you have it. I've used this with a great deal of success but I think there is still a lot to be done. More on my blog here: http://blog.krisk.org Thoughts? Hi, This is good idea, and i will probably try it out someday next year (too busy completing my business requirements :) I took a look at asterisk patch, and it seems quite simple. I just don't see the point of removing if(debug). You could easily get this additional logging into Asterisk trunk (if preserving RTP info in debug level), and starting asterisk with debug 1. So, then it would be easier to install recqual. Also, being able to run on unmodified version of Asterisk, it would be good to allow keeping current dialplan and just route test calls trough it. So, people would be able to keep track of their billing, etc for those test calls. Also, thanks for showing us magics of ecasound. I have similar project (pbx-test-framework) that allows IVR/Queue/etc testing in automated mode. Recording everything and checking voice quuailty would be great addition :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] Using Asterisk to measure call quality: Introducing Recqual
On Tue, Dec 23, 2008 at 11:17 PM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: This is true, however, I wasn't very excited about any other debug messages that might get printed with debug 1. I knew I only needed the endpoint RTP address, so I just removed the if. Of course you could always just run with debug 1 instead of the patch too. Again, this modification isn't strictly required. I just did if for SIP providers that give unpredictable media endpoint IP addresses... :) Debug 1 isn't that much. Just grep for line you're using and everything should work fast and fine. Sometimes i even log our production servers for weeks with debug 1. So i would suggest submiting this modification to digium bugtracker, if it really helps tracking ip's. Thanks again, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 1.4.22 Queues problems (Fifo or not ?)
On Thu, Dec 18, 2008 at 8:21 PM, Benoit maver...@maverick.eu.org wrote:
I'm having a question with asterisk queue system, is it a fifo or a lifo
or random ?
Sometimes when we have people waiting in the queue and new agents are
connected to handle the load the first call that is handled is not the
one which
is already waiting for 4min, but the new one which has just arrived.
However this doesn't happens everytimes
Is it normal ?
Calls are distributed in Priority+FIFO. Do you set ${QUEUE_PRIO}
before sending call to queue? Perhaps you're forgetting it in some
part of dialplan.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
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 1.4.22 Queues problems (Fifo or not ?)
On Thu, Dec 18, 2008 at 8:39 PM, Philipp Kempgen philipp.kemp...@amooma.de wrote: Benoit schrieb: I'm having a question with asterisk queue system, is it a fifo or a lifo or random ? Depends on the strategy. http://www.voip-info.org/wiki-Asterisk+call+queues Strategy affects which agent will be next to get call, but not which call will be sent to next agent (if i understood OP correctly) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 1.4.22 Queues problems (Fifo or not ?)
On Thu, Dec 18, 2008 at 8:50 PM, Darrin Henshaw dhens...@ignition.bm wrote: I believe you are correct Atis. Philipp within your queue setup do you have any announcements? If so read the posting on queues.conf(http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf), announcements will have an effect on the order that calls are picked up. Yes, announcments could also affect this. If announcement is being played to caller, he won't get connected at that point, and other call could jump in front of him. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 1.4.22 Queues problems (Fifo or not ?)
On Thu, Dec 18, 2008 at 9:44 PM, Benoit maver...@maverick.eu.org wrote: Atis Lezdins a écrit : On Thu, Dec 18, 2008 at 8:50 PM, Darrin Henshaw dhens...@ignition.bm wrote: I believe you are correct Atis. Philipp within your queue setup do you have any announcements? If so read the posting on queues.conf(http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf), announcements will have an effect on the order that calls are picked up. Yes, announcments could also affect this. If announcement is being played to caller, he won't get connected at that point, and other call could jump in front of him. Regards, Atis No, no announcments whatsoever, only a music on hold of a directory type with two files. You could enable core set verbose 3 and core set debug 1, and then post corresponding log when you see this happens. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 / Hylafax
On Tue, Dec 16, 2008 at 2:43 PM, Michael mich...@networkstuff.co.nz wrote: Recently i also posted some rough configuration sample of my setup on http://lists.digium.com/pipermail/asterisk-users/2008-November/222531.html Please mind, that if you're trying T38modem, you should get versions exactly as specified in voip-info.org, otherwise they might not work with Opal (which adds SIP protocol, as T38modem was originally for H.323) I used the SVN versions as recommended. Do you have any idea why T38 modem complains about libavcodec on start up? I'm really not sure. You can try installing ffmpeg of course. Local copies of opal i have mentions libavcodec/ffmpeg only in plugins dir. Did you compiled plugins? Perhaps you can try deleting everything there. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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 / Hylafax
On Sun, Dec 14, 2008 at 1:01 PM, Michael mich...@networkstuff.co.nz wrote: This path will not work. As You mentioned, * supports T38 path through only. In Your setup there will be a conversion on the * box between T38 via SIP provider and IAX (which uses G711 codec in this case). To make it work, use newer versions of t38modem and replace the iaxmodem with it. Newer versions of t38modem supports SIP, so that Your path will be PSTN = T.38 aware SIP provider = Internet = My machine (Asterisk) = T38-modem on localhost = Hylafax. I am using t38 modem 1.0.0, which AFAIK is the latest. I have also tried the t38 method (as documented on voip-info using opal and pwlib) and have not been able to get this to work. As a side when I start the t38 modem it complains it can find libavcodec, which is on my machine. I don't know if this is material or not? Michael Hi Michael, I just saw your topic on -dev (which is not appropriate there) and it reminded me to write a reply to this post (i've marked it for reply anyway :). The directions at voip-info.org are originally written by me, i spent lot of hours searching mailing lists etc to get at least something working. So, the question is - how far did you get with this. With this setup you should be able to do: Hylafax = T38modem = Asterisk = T38modem = Hylafax. That would be a verification for setup, that you should have everything working. Then you would want to test it with switch of SIP provider to see if it understands T38modem (as in quoted diagram). Unfortunately i didn't manage to get T38modem to understand switch of our provider, so i quickly tested with callweaver, and ended up with: PSTN = T.38 SIP provider = Internet = Asterisk = Callweaver = custom scripts = Hylafax. In my setup everything is located on the same machine and Callweaver runs smoothly together with asterisk on different ports. You should be able to do the same as Callweaver does with Asterisk 1.6 (if you're not bound to 1.4 setup) Recently i also posted some rough configuration sample of my setup on http://lists.digium.com/pipermail/asterisk-users/2008-November/222531.html Please mind, that if you're trying T38modem, you should get versions exactly as specified in voip-info.org, otherwise they might not work with Opal (which adds SIP protocol, as T38modem was originally for H.323) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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] 1.6 upgrade issues
On Tue, Dec 16, 2008 at 8:36 PM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
On Monday 15 December 2008 22:03:37 Chris Bagnall wrote:
Greetings list,
Over the last few days I've been gearing up to replace a couple of our
servers with 1.6 as something of a testbed, but I'm encountering a few
problems, and wondering if anyone can help...
In extensions.conf, there are a number of contexts defined for each group
of users, along the lines of: [groupa] [groupb] etc.
In each of those, there's a command include = outbound
[outbound] has entries similar to the following:
exten = _0[123],1,Macro(outbound,${EXTEN}, provider1, provider2)
the macro outbound is defined in extensions.ael as follows:
macro outbound (number, route1, route2) {
dosomestuff;
}
This has worked fine in 1.2 and 1.4, but seems to be choking on 1.6. I've
looked through the various changes.txt files, and have read mention of
replacing macro calls with Gosub(), but I'm not sure that's relevant to
this issue.
It is precisely relevant to this issue. All subroutines, whether they're
called macros or not, in AEL (in 1.6) are Gosub routines. So to invoke that
subroutine, you need to call out with Gosub, not with Macro. So it probably
should be along the lines of: Gosub(outbound,s,1
(${EXTEN},provider1,provider2)).
Actually there's ampersand operator prefixing macro name, so AEL
parser will automatically check dependencies etc:
outbound(${EXTEN},provider1,provider2);
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
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 spoken digits
On Thu, Dec 11, 2008 at 4:25 PM, Michael [EMAIL PROTECTED] wrote: How do I customize the digits 0 to 9? I have tried changing the paths in say.conf and nothing changes. I would like to do this without over writing the existing files, so I can have all my custom files in one location. http://www.voip-info.org/wiki/view/Asterisk+cmd+SetLanguage Set(CHANNEL(language)=my) and put your digits in /var/lib/asterisk/sounds/my/digits Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] config from DB
On Sun, Dec 7, 2008 at 9:19 AM, [EMAIL PROTECTED] wrote:
Hi Everyone,
Sorry, if this has been already discussed, but maybe someone encountered
interesting issue.
I have an * Dialplan configured with MySQL db. Everything works excellent,
except, I can't specify the ex-girlfriend logic. For example
Context exten Priorityapp
appdata
Default 400 1 Wait20
Default 400/100 1 Wait10
So, it does not matter what is my callerid, it will always go in wait(20) If
user with callerID 100 will try to dial x400, it will go to wait(20) as
well, and never wait(10). In another words Asterisk will disregard this
logic. If I place this logic in the extensions.conf file it will work as a
charm - no problem.
Thank you for your help.
Hi,
I believe it's a technological limitation. If it's in extensions.conf,
Asterisk can easilly draw a map of all possible matches in memory,
however for db it has to do query for each possible match. Perhaps
matching specific CallerID's was never thought of in realtime. If it
would work the same way as extension matching, probably a separate
column would be better (but that's just thought of how it should be).
Anyway, you can try creating separate context with callerid in exten
and then GoSub(${CALLERID(num)}) to it. Remember that ${EXTEN} is just
any number in your dialplan, and you can set it to CallerID when
jumping to other context. Upon returning from gosub it would be back
the same.
Regards,
Atis
--
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VoIP Project Manager / Developer,
IQ Labs Inc,
[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] CDR Design
On Fri, Dec 5, 2008 at 2:35 PM, Andrew Thomas [EMAIL PROTECTED] wrote: I'd disagree. In some cases a event based system would be the best solution, but in systems with high call volumes, scanning through events looking for the proper billing information and parsing them would be a hard job compared to CDRs. That's just it - you wouldn't be 'scanning' any CDR's - you'd be given Events. Your 3rd party app could then do anything it wanted to with them. Events are real time - not historic (like CDR's). Events are presented as they happen (hold, ring, etc) - CDR's are usually presented AFTER the call has finished so you miss things like hold-times etc. Remember, I am not saying that everyone should stop using the CDR's if they feel comfortable with them - but I, for one, don't trust them for building a stable billing platform or a real time stats package (which more and more customers seem to want these days). Pardon me, I have created realtime stats package that's based on CDR, you see new info immediately after call leg/event is over http://ftp.iq-labs.net/screenshots/cdr_view.jpg If you start to change the CDR's to account for extra bits (using a unique ID) then your 'scanning' actually increases as you will need to tie up all the unique ID's to get one full call progress path. This is exactly how real-time billing works. If you somebody wants it, they put in custom ResetCDR(w) in their dialplan and have all kinds of events logged. Having Asterisk write all the timestamps/durations into database is just much simpler. Please note, I am not trying to cause flame wars here - just stating that I'd love an event based stream, that I can parse any way I see fit. I know there's the AMI - but that is a 2-way, give-you-everything solution. All I want is to know when a handset and/or trunk does something (I don't care about SIP registrations etc). I guess we'll just have to wait and see what santa murf gives us all for Christmas :). I really want to contribute this discussion (and RFC), i'm reading it and i have lot of to say, but it's hard to find time for reading RFC (i'm in middle yet). So, i hope this will go on and allow me to respond with some objective comments. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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 Design
On Fri, Dec 5, 2008 at 3:41 PM, Andrew Thomas [EMAIL PROTECTED] wrote: Pardon me, Granted ;). I have created realtime stats package that's based on CDR, you see new info immediately after call leg/event is over I see what you are saying but can you show hold-times etc? For example, call comes in to A, A puts call on hold, A dials B, B answers A, A transfers call to B, B speaks to caller. Basic PBX functionality - but how long did it take B to answer A? What if B is an external number (trunk to trunk)? To illustrate - dial an external number and, while on that call, check your CDR's - there isn't any. Now put that call on hold, still none, now call another internal extension - still none. Now hang up and transfer the call. Now there is one CDR for your call. That isn't real-time - that's historic (ie. it happens AFTER the call is finished). Well, by real-time i meant that you don't have to run CDR processing at the end of month to see some reports/logs, etc. When i started to write this implementation, luckily i didn't had much expertise in telephony, so i did it from programmers point of view. There's even funny story about this in our company - we had some Project managers and Development managers hired later who had lots of experience in telephony, and at some point when discussing some minor problems with my implementation, they told me that this is not the way how to do it. Telco's do all processing at end of month, so this system won't last for long. Currenty everybody in our company probably would be very disappointed if they wouldn't be able to see fresh data in reports immediately. Regarding hold and transfers, yes - you can't achieve that in real-real-time, but i think it's not important. Calls don't last for several days, and even if they do - you have different view where you can see active calls. It's completely logical that CDR's get posted after finishing certain action - in order to account correct timing. Otherwise Asterisk would have to post a record, and later modify it (yet another minefield). The CDR that's produced here will show your call to the outside world - and its duration etc. So far, so good (for historic reporting). Now get the person you transferred the call to to hang up. Another CDR record - but this show as you talking to the internal extension - not the external extension talking to the outside world. Therefore, if the 2nd extension stays on that call for a long time - who's picking up the bill? Current CDR's are lacking in this respect - and I think this is what murf is trying to sort out (please jump in here murf). I would like to comment really much of this, but I'll refrain until i complete reading Murf's RFC. I just don't feel competent enough to speak about this without reading he's ideas first. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] top posting again [was: Re: CDR Design]
On Fri, Dec 5, 2008 at 3:47 PM, Apostolos Pantsiopoulos [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: Top posting strikes again: On Fri, Dec 05, 2008 at 01:39:59PM +0200, [EMAIL PROTECTED] wrote: Quote : Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Who wrote that? [snip the rest of the reply] Andrew Thomas wrote: [snip] Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Why didn't you place your reply here? We have archives of the list. We can spot the original message. [snip more useless quoting resulted from top-posting] Sorry I did not know you have a non-top-posting policy It's not official policy, however it's pleasant in long discussions. It's good to make it a personal habit :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] top posting again [was: Re: CDR Design]
makes sense On Fri, Dec 5, 2008 at 7:59 PM, David fire [EMAIL PROTECTED] wrote: i have the solution so every one is happy i will write over and below :- ) 2008/12/5 Tilghman Lesher [EMAIL PROTECTED] On Friday 05 December 2008 11:11:33 Wilton Helm wrote: I guess there is a variety of opinions on this, some of which relates to the tools a person is using. The absolutely most offensive thing to me in a post is to have to scroll through a bunch of copied original material that I've already read six times to get to the new part. My own preference is not to quote anything other than a short phrase snippet that is directly being replied to or failing that, at least put the original after the new material for those who might want it. I realize that not everyone sees it that way, but maybe it throws a different perspective in the mix. If you are asking a question in a public forum, will you jump up and down and shout your question? Or will you ask your question in accordance with the rules that were set out in advance and is most likely to get you the answer you require? While email is far more anonymous than a town meeting, your adherence to established standards is no less required. -- Tilghman ___ -- 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 i have the solution so every one is happy i will write over and below :- ) əsuəs səʞɐɯ -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] CLI and choice of messages
On Fri, Dec 5, 2008 at 6:00 PM, Mike [EMAIL PROTECTED] wrote: Is there a way, for debugging purpose, to have a level where only Noop() cmds are shown in the CLI but nothing else in the dialplan appears (except for errors and warnings or course)? Replace NoOp(something) with Verbose(something) and it will be printed out with Verbosity of 0. That's default verbosity you see in CLI. NoOp really does nothing as opposed to Verbose(), so you will see it only in -- Executing message which has verbosity 2. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] top posting again [was: Re: CDR Design]
On Sat, Dec 6, 2008 at 4:59 AM, Bob Gustafson [EMAIL PROTECTED] wrote: If I notice that someone has started a bottom post, I will follow. But, if I am the first, I will top post. When I look at a new email, I don't like to scroll to the bottom to find out what is new. If you know of a mail reader which will automatically scroll to the top of the latest info, let me know. If there is a technological fix, perhaps these threads will die down. GMail webinterface does automatically hides quotations. I expect that other mail clients are following. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Parking calls
On Wed, Dec 3, 2008 at 1:47 AM, Sebastian [EMAIL PROTECTED] wrote: I found other solution, I can use cannel local to dial to an extension with m parameter, then I can put Ringing as the first thing to do that will follow processing the next lines of the dialplan, with the m option MOH will sound instead of ringing, and I can do the heavy work there till I finish and do other things with the call. Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas Sent: martes, 02 de diciembre de 2008 07:37 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Parking calls This seems to be an AGI/Music on Hold solution to me. For parking to work, you would have to know which lot you parked the call in and pick it back up when done, assuming that another user did not pick it up and that the caller did not hang up. From the dialplan, you would call an AGI. The AGI would do something like this: print STDOUT EXEC background /var/lib/asterisk/sounds/wait-moment \n system(program2.agi ) exit; program 2 would run while the sound played. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Tuesday, December 02, 2008 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parking calls It is not a parking solution. Sebastian wrote: Any idea? Please I need advice. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Sent: lunes, 01 de diciembre de 2008 11:58 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Parking calls Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing à follow = true 5. Endwhile 6. UnParkCall 7. Go on….. The idea is let the call waiting while I do some things on the dialplan, is it possible?? Maybe is not parking the solution?? -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- 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 __ Information from ESET Smart Security, version of virus signature database 3659 (20081202) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 3659 (20081202) __ The message was checked by ESET Smart Security. http://www.eset.com Hi, You can try to use MusicOnHold() application to do this. However docs don't say how to terminate it. I presume that Playback(silence/1) would stop it, you're welcome to try it out :) That should be better solution than creating child channel etc. Also, i'm curious what kind of massive processing you need in dialplan? It's best practice to don't do anything that may delay call for long, as caller can get bored/angry etc. If you just have to do something heavy for each call and you don't use result of that operation to determine next step of call, you can do: System((/usr/bin/do-something.sh)) note, the ampersand after first brackets will make to run shell command in background. If you need the result of some operation to send call further, you should optimize that as much as possible by creating some kind of cache. Also, there's a trick - you can launch background shell command at beginning of call, then send customer to IVR or even Dial() and at later point check results. For example you can add G or M argument to Dial() to execute part of dialplan macro/gosub upon answer. Hope that my explanation helps :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Parking calls
On Wed, Dec 3, 2008 at 7:27 PM, Sebastian [EMAIL PROTECTED] wrote:
The thing is I have to wait checking a database value to change the state,
that duration is not long, but on any case I don't know when will be ready
to go on.
If I use MusicOnHold app the dialplan get stuck there and there's no further
movement on my dialplan lines.
I will have a while loop checking for a database value to change, if it
changes the call will go on through the dialplan depending on the result,
but I can't make the call wait without any sound (I thought PlayTones could
be a possibility but I prefere MOH).
For these reasons I can't use a shell script launched in background.
Is there any way to launch in background some app like Background but follow
with the next dialplan line while it plays the sound??
(Just like Ringing does on my solution), I know making a local channel is
not the best solution, but at this moment I can't think on a different one
that not involves agi.
Any idea??
AMI action Redirect -
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect
Of course you would need some script to send this action, but as long
as you control writes to database it shouldn't be a problem. All you
need is to store ${CHANNEL} name of current channel before entering
MusicOnHold().
Also you could take a look at GROUP_COUNT function, perhaps it in some
way can help you :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[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] Parking calls
On Thu, Dec 4, 2008 at 1:25 AM, Sebastian [EMAIL PROTECTED] wrote:
I don't understand how can I solve my situation with this
Ok, a simplified sample (i used PHP because i use it daily, but any
language is good):
context incoming {
_X. = {
Answer();
System(channel-waiting.php ${CHANNEL});
MusicOnHold();
}
}
context continue {
_X. = {
// you reached your condition
Playback(tt-monkeys);
Dial(SIP/something);
}
}
then a channel-waiting.php would store ${CHANNEL} name somewhere in database.
Then, assuming you can execute some code WHEN you change the database
value you wanted to monitor in loop, you launch a script that sends
AMI Redirect action.
For example:
?
$channel = 'SIP/123-abc'; // retrieved from DB where set by channel-waiting.php
require phpagi-asmanager.php;
$as = new AGI_AsteriskManager();
$res=$as-connect(localhost,username,password);
if($res==FALSE) {
echo Connection failed.\n;
}
$res=$as-send_request(Redirect,
array(
Channel=$channel,
Context=continue,
Exten=123,
Priority=1
)
);
if($as-resp_is_success($res)){
echo it worked!;
}
?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: miércoles, 03 de diciembre de 2008 03:48 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parking calls
On Wed, Dec 3, 2008 at 7:27 PM, Sebastian [EMAIL PROTECTED] wrote:
The thing is I have to wait checking a database value to change the state,
that duration is not long, but on any case I don't know when will be ready
to go on.
If I use MusicOnHold app the dialplan get stuck there and there's no
further
movement on my dialplan lines.
I will have a while loop checking for a database value to change, if it
changes the call will go on through the dialplan depending on the result,
but I can't make the call wait without any sound (I thought PlayTones
could
be a possibility but I prefere MOH).
For these reasons I can't use a shell script launched in background.
Is there any way to launch in background some app like Background but
follow
with the next dialplan line while it plays the sound??
(Just like Ringing does on my solution), I know making a local channel is
not the best solution, but at this moment I can't think on a different one
that not involves agi.
Any idea??
AMI action Redirect -
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect
Of course you would need some script to send this action, but as long
as you control writes to database it shouldn't be a problem. All you
need is to store ${CHANNEL} name of current channel before entering
MusicOnHold().
Also you could take a look at GROUP_COUNT function, perhaps it in some
way can help you :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
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__ Information from ESET Smart Security, version of virus signature
database 3660 (20081203) __
The message was checked by ESET Smart Security.
http://www.eset.com
__ Information from ESET Smart Security, version of virus signature
database 3662 (20081203) __
The message was checked by ESET Smart Security.
http://www.eset.com
___
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--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[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] Parking calls
On Thu, Dec 4, 2008 at 3:30 AM, Sebastian [EMAIL PROTECTED] wrote:
Do you thik theres no chance to do it directly from dialplan like if I use
PlayTones, and the call will follow to the next line on the dialplan?
I think that the best solution would be make a play musiconhold but not wait
indefinitely, something like StartMOHAsync and StopMOHAsync.
What do you think?
Yeah, it would be best, but i'm really not aware of anything like
that. Perhaps app_jack in 1.6 could do something (but don't ask me
how, i've just heard rumors of it).
The hold/background stuff is not my field, i just spitted out ideas of
how i would solve it. I looked at available commands, and if you say
MusicOnHold doesn't stop, you have to terminate it somehow.
Regards,
Atis
Thanks for your solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: miércoles, 03 de diciembre de 2008 10:31 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parking calls
On Thu, Dec 4, 2008 at 1:25 AM, Sebastian [EMAIL PROTECTED] wrote:
I don't understand how can I solve my situation with this
Ok, a simplified sample (i used PHP because i use it daily, but any
language is good):
context incoming {
_X. = {
Answer();
System(channel-waiting.php ${CHANNEL});
MusicOnHold();
}
}
context continue {
_X. = {
// you reached your condition
Playback(tt-monkeys);
Dial(SIP/something);
}
}
then a channel-waiting.php would store ${CHANNEL} name somewhere in
database.
Then, assuming you can execute some code WHEN you change the database
value you wanted to monitor in loop, you launch a script that sends
AMI Redirect action.
For example:
?
$channel = 'SIP/123-abc'; // retrieved from DB where set by
channel-waiting.php
require phpagi-asmanager.php;
$as = new AGI_AsteriskManager();
$res=$as-connect(localhost,username,password);
if($res==FALSE) {
echo Connection failed.\n;
}
$res=$as-send_request(Redirect,
array(
Channel=$channel,
Context=continue,
Exten=123,
Priority=1
)
);
if($as-resp_is_success($res)){
echo it worked!;
}
?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: miércoles, 03 de diciembre de 2008 03:48 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parking calls
On Wed, Dec 3, 2008 at 7:27 PM, Sebastian [EMAIL PROTECTED] wrote:
The thing is I have to wait checking a database value to change the
state,
that duration is not long, but on any case I don't know when will be
ready
to go on.
If I use MusicOnHold app the dialplan get stuck there and there's no
further
movement on my dialplan lines.
I will have a while loop checking for a database value to change, if it
changes the call will go on through the dialplan depending on the result,
but I can't make the call wait without any sound (I thought PlayTones
could
be a possibility but I prefere MOH).
For these reasons I can't use a shell script launched in background.
Is there any way to launch in background some app like Background but
follow
with the next dialplan line while it plays the sound??
(Just like Ringing does on my solution), I know making a local channel is
not the best solution, but at this moment I can't think on a different
one
that not involves agi.
Any idea??
AMI action Redirect -
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect
Of course you would need some script to send this action, but as long
as you control writes to database it shouldn't be a problem. All you
need is to store ${CHANNEL} name of current channel before entering
MusicOnHold().
Also you could take a look at GROUP_COUNT function, perhaps it in some
way can help you :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[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
__ Information from ESET Smart Security, version of virus
signature
database 3660 (20081203) __
The message was checked by ESET Smart Security.
http://www.eset.com
__ Information from ESET Smart Security, version of virus
signature
database 3662 (20081203) __
The message was checked by ESET Smart Security.
http://www.eset.com
___
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Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
On Tue, Dec 2, 2008 at 8:22 PM, Dave Fullerton [EMAIL PROTECTED] wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 I see a reference in the 1.6 changelog that refers to SENTINEL not existing in 1.6.0 2008-06-27 01:09 + [r125648-125684] Mark Michelson [EMAIL PROTECTED] * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 -Dave ACK [CC] manager.c - manager.o manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released
On Tue, Dec 2, 2008 at 9:56 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Dec 02, 2008 at 01:22:16PM -0500, Dave Fullerton wrote: Is anyone else having difficulty compiling 1.6.0.2? It bombs out when compiling manager.c manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Error 1 make: *** [main] Error 2 On what platform is it? Fedora Core release 6 (Zod) - Linux ast-dev14 2.6.21.1skvt #1 Fri May 18 10:14:35 EEST 2007 i686 i686 i386 GNU/Linux Fedora release 8 (Werewolf) - Linux asterisk-dev-mc 2.6.24.7-92.fc8 #1 SMP Wed May 7 16:26:02 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Debian Etch (4.0) - Linux saule 2.6.18-6-xen-686 #1 SMP Thu May 8 11:28:36 UTC 2008 i686 GNU/Linux Debian Sid - Linux debian 2.6.26-1-686 #1 SMP Thu Oct 9 15:18:09 UTC 2008 i686 GNU/Linux 1.6.0.1 compiled fine on at least two Fedoras. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Priority between calls from different queues
On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Priority between calls from different queues
On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED] wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. With queues there are much more such situation than just this one ;) Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. For in-between queues you can use weight. So, if queue1 has more weight than queue2, and agent1 is available (and is in both queues), he will receive call from queue1 (no matter how long other caller waits in queue2). Also, there's wrapuptime. It means - how many seconds agent should not receive call after completing previous queue call. So, if agent receives call from queue1 and it has wrapuptime 10 seconds, then he ends call, he might immediately receive call from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls are ready to go to agent in different queues. Also, you can give priority to different callers within queue by setting QUEUE_PRIO variable before sending call to queue. You could try to describe why you need two queues and what should be rules to distribute calls - so we can help you with overall architecture. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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 This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone
Re: [asterisk-users] Priority between calls from different queues
On Fri, Nov 28, 2008 at 4:51 PM, equis software [EMAIL PROTECTED] wrote: In both queues have the same wrapuptime, there´s not a problem... With weight property I can´t resolve my problem...I want to answer calls of both queues sorted by time, like a big FIFO or like if I had only one queue I'm afraid that it's not possible. There will be too much cases when one queue can choose to call agent ignoring another queue. What i meant with wrapuptime - even if it's the same (and you don't use shared_lastcall), second queue won't know that agent has just ended conversation - so it will send call to agent. I guess that there would be some more such race conditions for having free agent. If you really need FIFO, you would have much better luck with having one queue and then thinking how to customize it for different callers. Single instance of Queue is built like FIFO for calls (with bucket of agents). For example - wait time you can specify as argument to Queue(). As for different caller amount, you can assign them to groups and use GROUP_COUNT to determine how many they are in each group. If you need some more differentiation, just ask, and we'll try to give ideas. Oh, btw - you could also try to create one fake agent in queue1 and queue2 (with ringinuse=yes) and use Local channel to send those calls to queue-real where your agents reside. However, i'm not sure that this will work, as queue-real might answer channel, even if you set r option.. not sure is this a problem, but it could be complex :) Regards, Atis regards On Fri, Nov 28, 2008 at 12:31 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED] wrote: One thing you also will run into is listed here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf. Here is the interesting part: Note that calls are not offered to queue members whilst the announcement is playing and it is possible for callers to slip ahead in the queue as a result. For example, call 1 arrives and is queued. Call 2 arrives ten seconds later and is queued. After twenty seconds, call 1 is played the periodic announce message. Exactly one second after call 1 starts hearing the message an agent becomes free. Since call 1 is tied up with announcements, call 2 is successfully offered to the agent. Call 1 remains on hold and yet a call which arrived later has been serviced. Basically you can see that if you have announcements played, that could cause your order of answered calls to be not what you expect. With queues there are much more such situation than just this one ;) Cheers, Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC Ignition Support Center | www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of equis software Sent: Friday, November 28, 2008 10:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Priority between calls from different queues I saw QUEUE_PRIO but it works inside a queue not between queues. I need to use two queues because their have different settings like max time waiting, max amount of calls in queue and others. For in-between queues you can use weight. So, if queue1 has more weight than queue2, and agent1 is available (and is in both queues), he will receive call from queue1 (no matter how long other caller waits in queue2). Also, there's wrapuptime. It means - how many seconds agent should not receive call after completing previous queue call. So, if agent receives call from queue1 and it has wrapuptime 10 seconds, then he ends call, he might immediately receive call from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis Regards On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED] wrote: Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 - call1 queue1 - call2 queue2 - call3 queue2 - call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could I set priority between queues? Hello, Queue has lot of different settings, like wrapuptime, strategy, etc. Also two queues usually don't know about each other, with few exceptions. One of them is shared_lastcall (introduced in Asterisk 1.6.0). There's also weight - it will help to give priority to specific queue if multiple calls
Re: [asterisk-users] Any 1.6 SendFAX example ?
On Thu, Nov 27, 2008 at 1:03 PM, Olivier [EMAIL PROTECTED] wrote:
Hi,
Do you have any example showing how to use SendFAX ?
I can see several examples of ReceiveFAX but not a single one showing
SendFAX.
This is not from 1.6, but rather from callweaver attached to Asterisk 1.4.
When i'll finally switch to 1.6, i intend to just move those contexts
to Asterisk dialplan.
extensions.conf:
[fax_out]
exten = _X.,1,NoOp(--- sending fax to ${EXTEN} ---)
exten = _X.,n,SipDTMFMode(inband)
exten = _X.,n,TxFAX(${TIFF},caller,debug,ecm)
exten = _X.,n,Hangup()
exten = h,1,NoOp(--- done sending fax ---)
exten = h,n,NoOp(TX: REMOTESTATIONID is ${REMOTESTATIONID})
exten = h,n,NoOp(TX: FAXPAGES is ${FAXPAGES})
exten = h,n,NoOp(TX: FAXRESOLUTION is ${FAXRESOLUTION})
exten = h,n,NoOp(TX: FAXBITRATE is ${FAXBITRATE})
exten = h,n,NoOp(TX: PHASEESTATUS is ${PHASEESTATUS})
exten = h,n,NoOp(TX: PHASEESTRING is ${PHASEESTRING})
exten = h,n,NoOp(TX: DIALSTATUS is ${DIALSTATUS})
exten = h,n,System(${SCRIPT}/fax_out_end.php --status
${uniqueid_storage} --pages ${FAXPAGES} --resolution ${FAXRESOLUTION}
--bitrate ${FAXBITRATE} --phase
exten = failed,1,NoOp(--- failed sending fax ---)
Then, to send a fax, generate tiff file and call-file.
Snapshot of my PHP generating call-file from hylafax job:
$channel = 'SIP/'.$job['number'].'@asterisk-t38';
$destination =
array('context'='fax_out','extension'=$job['number'],'priority'='1');
$vars = array(
'LOCALSTATIONID' = 'CallWeaver-T38-TxFax',
'T38CALL'='1',
'TIFF'=$job['private']['tiff_file'],
);
$callerid = 'CallWeaver T38 TxFax';
$waittime = 180;
$deliver_time = NULL;
$filename = NULL;
$retries = array();
$callfile_dir = T38_CALLFILE_DIR.'/';
$result =
ast_originate_callfile($channel,$destination,$vars,$callerid,$waittime,$deliver_time,$filename,$retries,$callfile_dir);
Of course you'll need ast_originate_callfile which writes data to file
and then moves to correct dir. I would publish that, but it's full of
my constants and realted to much other libs..
Basically, you dial destination number (SIP/[EMAIL PROTECTED]) and send
local side of channel to fax_out,${NUMBER},1 which does SendFax.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[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] Any 1.6 SendFAX example ?
On Thu, Nov 27, 2008 at 4:39 PM, Olivier [EMAIL PROTECTED] wrote: Thanks for this detailed reply. I was trying to test SendFAX, ReceiveFAX as first on my way to Hylafax with either iaxmodem or t38modem. Have you tried any of those 2 (iaxmodem or t38modem) ? Which one would you pick ? We had IAXmodem with Hylafax installation base, and we sent our faxes out trough PRI. Then we switched to pure SIP, but were unable to get T38modem to work with our provider. So, we wrote a wrapper for Hylafax that grabs processed tiff file from outgoing spool and generates call file for Callweaver (which sends trough Asterisk with T38 passtrough). So, if you have PRI ir analogue lines, use IAXmodem, otherwise you have to do either T38modem or SendFax. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] SVN
On Wed, Nov 26, 2008 at 1:32 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 09:06, Wed 26 Nov 08, Alex Montoanelli wrote: Hello, everyone. Anybody know when that svn will be available again? Regards Hey, I can checkout stuff fine from svn.digium.com. Maybe you can provide some more info about how it's not working for you. Probably it's that http://svn.digium.com/ gives 403 error. As i recall, it showed up when some search engine tried to indexing whole SVN ignoring robots.txt, so Digium disabled root page. Now you can access it by adding /view/ to URL. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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 1.6 mysql cdr log problem
On Tue, Nov 25, 2008 at 2:19 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote: On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED] wrote: I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized there is some differences between 1.2 (my previous system) and 1.6 log system. I suspect that you have some unique index on the table which is conflicting with the inserted fields. Once you figure out which field is causing the conflict, it should be easier to figure out where the problem actually lies. You should also check Asterisk log for warnings. 1.6 should detect table structure and warn about missing fields. If it's so, perhaps you can change asterisk - mysql (res_cdr_addon_mysql if i remember correctly) to do an alter on your table - then it will automagically create missing fields. You remember incorrectly. None of the CDR drivers currently have the capability to alter tables. What they will do is to adapt to the table structure and insert only the required fields. Only realtime table drivers have the capability of altering tables and then, only if you turn that behavior on. By default, Asterisk does not alter table structures. Oh, my mistake :) BTW, if you 'core set debug 2 cdr_addon_mysql.c' (and make sure debug is enabled to the console, via /etc/asterisk/logger.conf), then the SQL will be printed to the console. That should help you find where the problem lies. Actually SQL's are logged with debug 1, debug 2 is too much of everything. I didn't knew that you could use filename.c when setting debug level, seems quite useful :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] SendImage()
On Sun, Nov 23, 2008 at 10:00 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Saturday 22 November 2008 22:18:05 Rob Hillis wrote: Philipp Kempgen wrote: SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the following channel variable upon completion: SENDIMAGESTATUSThe status is the result of the attempt as a text string, one of OK | NOSUPPORT ---cut--- in 1.6: ---cut--- SendImage(filename): Sends an image on a channel. Result of transmission will be stored in SENDIMAGESTATUS channel variable: SUCCESS Transmission succeeded FAILURE Transmission failed UNSUPPORTED Image transmission not supported by channel ---cut--- Is there any reason to break backwards compatibility? Why is SUCCESS better than OK and UNSUPPORTED better than NOSUPPORT? IMHO there was no need to change anything except for adding the FAILURE return status. This is a case of damned if you do, damned if you don't. That is a perfect complaint, and I understand it completely. On the other side, we are criticized for inconsistent behavior, inconsistent status names, etc. So we've chosen to make Asterisk more consistent going forward, with the one-time problem of a slight change in behavior. Current users see an issue either way, and future users won't see a problem at all. Perhaps somebody from -dev team can be delegated to check naming consistency of new features? So, whenever a feature is added (perhaps at code review), he checks naming to match best of he's opinion. I know that original developers might be stubborn to keep their own names, however that leads to inconsistencies and such changes later. So, if one person is responsible of that, even if change is insignificant, nobody should be offended.. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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 1.6 mysql cdr log problem
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED] wrote: Hi all! I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized there is some differences between 1.2 (my previous system) and 1.6 log system. You should also check Asterisk log for warnings. 1.6 should detect table structure and warn about missing fields. If it's so, perhaps you can change asterisk - mysql (res_cdr_addon_mysql if i remember correctly) to do an alter on your table - then it will automagically create missing fields. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Log level of 500 Server Internal Error.
On Fri, Nov 21, 2008 at 7:48 PM, Alex Balashov [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be great indication that something is not ok - either outgoing trunk or local phone is bad. That would generate a lot of debate about what sorts of signaling error classes are useful to include in the fixed logs and which aren't. Best thing to do is just to run your own packet capture and grep for things of interest to you. Yes, that's what i would like to start. If a call fails, i think it's reasonable enough to log a warning message. If i haven't seen this before, how would i know that it's bad and search for it? IMHO it's a good indication for network problem (as was midget packet warning recently) Define fails. There are many different scenarios applicable to many different people's situations, and I doubt Asterisk can be set up to log them all. SIP also has a complicated state machine; sometimes call failures can occur further up the setup flow and not as an immediate failure response. That, I think, is what I was trying to put forth as a possible reason why Asterisk doesn't do what you're asking, which is otherwise a fairly obvious thing to do. Well, of course there are different scenarios. Asterisk shouldn't warn if device sends REGISTER and it replies with UNAUTHORIZED, however it shold warn when device sends wrong authorization. There are lot of cases, perhaps not all can be implemented right now, as some would need complete state information to determine correct/wrong behavior. Of course there are people who do handling of unsuccessful Dial() and send outgoing call trough other provider or incoming - to voicemail. However if SIP device is registered or set as peer, and replays with 500 Internal server error or something similar - that would give pretty much useful info to newbies about what's going wrong. As there's currently no complete way how to react to SIP responses, and DIALSTATUS=CONGESTION isn't much useful. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] database queries from extensions.conf
On Mon, Nov 24, 2008 at 8:01 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: For me, the best is the curl function, along with res_config_curl. Best of all worlds - pass a web query to *whatever* backend system you want to implement. No messy ODBC drivers. It's really, really good stuff ;) However you probably can't use it for transactions within call workflow. For example: Customer calls in Start transaction Do query 1 Play prompt A Do query 2 Play prompt B Do query 3 End transaction So, if customer hangs up in middle, you don't execute transaction. That's the thing how it should be done with ODBC or whatever :) Regards, Atis Julian. Jared Smith wrote: On Sun, 2008-11-23 at 00:47 -0500, Al Baker wrote: Quote The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. How can you be VERY HappY with something that allows ONLY single statemts of SQL My intention here is not to start a flamewar over which one is *best*, or worse to start arguing about who is right instead of what is right. You're absolutely correct in your assertion that func_odbc doesn't currently support multi-statement or transactional statements, which is obviously a limitation to some people. As I pointed out in my other response to this thread this morning, Tilghman Lesher is working on that. Feel free to look at his odbc_tx_support branch on the web at http://svn.digium.com/view/asterisk/team/tilghman/odbc_tx_support/, or to check it out via Subversion at http://svn.digium.com/svn/asterisk/team/tilghman/odbc_tx_support/ One other way of working around the problem is to use stored procedures in the database. That being said, I guess I'll articulate my own personal reasons for preferring func_odbc, and leave it at that. 1) I like that my dialplan isn't tied to one particular database. I've done a *lot* of database work in my short career, including being a sysadmin for one of the largest MySQL database installations in the world. I *love* the fact that the ODBC abstraction layer means I can easily change my backend database from MySQL to PostgreSQL (or Oracle or SQL Server, heaven forbid!) at the drop of a hat. I realize that might not be a big attraction for some, but for me it's a big plus. 2) I don't like the licensing mess associated with linking MySQL directly to Asterisk. I'm sure there are a few people on the list that really enjoy the convoluted logic of tip-toeing the licensing minefield of linking (dual-licensed) Asterisk with (dial-licensed) MySQL, but I prefer to avoid the minefield altogether and use ODBC. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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, IQ Labs Inc, [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] Ping
On Fri, Nov 21, 2008 at 4:59 PM, Sebastian Milioto [EMAIL PROTECTED] wrote: Ping ___ -- 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 Pong GMail's preview looks fun - Ping -- Bandwidth and Colocation Provided by http://www.api-digital.com; Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Log level of 500 Server Internal Error.
Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be great indication that something is not ok - either outgoing trunk or local phone is bad. Any opinions? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Log level of 500 Server Internal Error.
On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov [EMAIL PROTECTED] wrote: Atis Lezdins wrote: Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be great indication that something is not ok - either outgoing trunk or local phone is bad. That would generate a lot of debate about what sorts of signaling error classes are useful to include in the fixed logs and which aren't. Best thing to do is just to run your own packet capture and grep for things of interest to you. Yes, that's what i would like to start. If a call fails, i think it's reasonable enough to log a warning message. If i haven't seen this before, how would i know that it's bad and search for it? IMHO it's a good indication for network problem (as was midget packet warning recently) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Limit the number of users in a meetmeconference?
On Fri, Nov 21, 2008 at 5:46 PM, Danny Nicholas [EMAIL PROTECTED] wrote: Armed with a little more information, here is a more realistic reply. In the 1.6.0.1 code, app_meetme.c defines maxusers in line 369 and sets the max value in line 870 to 0x7fff. Therefore changing line 870 would allow you to limit the maxusers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Friday, November 21, 2008 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Limit the number of users in a meetmeconference? Hi Dan - I found the maxusers defined in meetme.c, but I'm not sure how this value is set. Does anybody know if one can limit the number of users permitted in a meetme conference? I know there's MeetmeCount(), but I'd rather avoid the dialplan logic and just set maxusers instead. That feature is primarily used with RealTime conferences. The maxusers value is read from a database and enforced on RealTime enable conferences. This presumes you are looking at 1.6.X or Trunk code... Ah. No realtime for me, so I guess I'll just stick with using MeetmeCount() in the dialplan. Thanks for the info! - Noah If it's in realtime, then it should also work from config file. If it's not, then file a bug. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [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] Any other free toll free SIP providers out there?
On Thu, Nov 20, 2008 at 2:50 PM, SIP [EMAIL PROTECTED] wrote:
Tom Browning wrote:
FWD (Free World Dialup) allows any SIP call to US toll free numbers
via [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
This works WITHOUT the need to be registered at FWD so in my dialplan
I have something like:
exten = _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r
http://fwd.pulver.com/*$%7BEXTEN:1%7D,60,r)
exten = _8.,2,Hangup
And I just dial 8-1-8xxyyy and presto ... calls go through just
fine 99% of the time.
I'm wondering if there are any other providers out there that allow
calls to toll free numbers without the need of being registered? I'd
like to have a backup or two.
Tom
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IdeaSIP doesn't require registration for free Toll-Free.
[EMAIL PROTECTED]
Wow, that's helpful.
I googled a bit, and found this lost page:
http://www.voip-info.org/wiki/view/Toll+Free+Termination+Providers
So, now it's updated with FWD and IdeaSIP, and linked from VoIP
Service Providers
Perhaps anyone who uses them can check examples - the ${EXTEN:1} part
seems wrong.
I wonder are there any legal issues if they were included in Asterisk
sample config? Or perhaps they could even pay for advertising to get
included there ;-)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Macro conversion in 1.6
On Thu, Nov 20, 2008 at 5:57 PM, Joseph L. Casale
[EMAIL PROTECTED] wrote:
I create my sip users using a common macro in 1.4:
[internal]
exten = 200,1,Macro(phones|200|SIP/200)
[macro-phones]
exten = s,1,Dial(${ARG2}|45|Tt)
etc...
But now in 1.6 this fails:
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/201-0942b530,
phones|200|SIP/200) in new stack
[Nov 20 08:55:55] WARNING[5958]: app_macro.c:201 _macro_exec: No such context
'macro-phones|200|SIP/200' for macro 'phones|200|SIP/200'
-- Executing [EMAIL PROTECTED]:2] Wait(SIP/201-0942b530, 1) in new
stack
-- Executing [EMAIL PROTECTED]:3] Playback(SIP/201-0942b530, invalid)
in new stack
-- SIP/201-0942b530 Playing 'invalid.gsm' (language 'en')
Why does the user's extension get created (all the phones work) but I can't
dial to it?
AFAIR it was mentioned in UPGRADE.txt that argument separator was
changed from pipe to comma. Unless you read it, you might also
experience lot of other problems.
It should be Macro(phones,200,SIP/200)
However it's not recommended to use macro's, you are encouraged to
convert them to GoSub's, as they now support arguments.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[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] IF else
On Wed, Nov 19, 2008 at 4:05 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Wed, 19 Nov 2008, michel freiha wrote:
Hi all,
I have the following context in extensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup
If the customer dial 111, it'll be router to the entry with priority 21,
else it'll go to priority 2...I would like to add a third condition that if
the user dial let's say 112 it'll go to the priority 28 let's say
1. Stop using numbers.
2. Start using labels.
3. Add comments.
exten = _X.,1,Gotoif($[${EXTEN} = 111]?exten111)
exten = _X.,n,Gotoif($[${EXTEN} = 112]?exten112)
exten = _X.,n,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
exten = _X.,n(exten111),Noop(Dialled 111)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
exten = _X.,n(exten112),Noop(Dialed 112)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else if(${EXTEN}=112) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else {
DeadAGI(a2billing.php);
Wait(2)
Hangup();
}
}
2) Start using extension masks (also works with AEL):
[a2billing]
exten = _111,1,Noop(Dialled 111)
exten = _111,n,Playback(AR_GetGiveToID)
exten = _111,n,Wait(2)
exten = _111,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _111,n,Wait(2)
exten = _111,n,Playback(/tmp/asterisk-recording)
exten = _111,n,Wait(2)
exten = _111,n,Hangup
exten = _112,1,Noop(Dialed 112)
exten = _112,n,Playback(AR_GetGiveToID)
exten = _112,n,Wait(2)
exten = _112,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _112,n,Wait(2)
exten = _112,n,Playback(/tmp/asterisk-recording)
exten = _112,n,Wait(2)
exten = _112,n,Hangup
exten = _X.,1,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[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] IF else
On Wed, Nov 19, 2008 at 6:51 PM, Steve Edwards
[EMAIL PROTECTED] wrote:
On Wed, 19 Nov 2008, Atis Lezdins wrote:
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else if(${EXTEN}=112) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else {
DeadAGI(a2billing.php);
Wait(2)
Hangup();
}
}
You're missing a couple of semi-colons.
Sorry, that was untested proof of options :)
2) Start using extension masks (also works with AEL):
[a2billing]
exten = _111,1,Noop(Dialled 111)
exten = _111,n,Playback(AR_GetGiveToID)
exten = _111,n,Wait(2)
exten = _111,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _111,n,Wait(2)
exten = _111,n,Playback(/tmp/asterisk-recording)
exten = _111,n,Wait(2)
exten = _111,n,Hangup
exten = _112,1,Noop(Dialed 112)
exten = _112,n,Playback(AR_GetGiveToID)
exten = _112,n,Wait(2)
exten = _112,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _112,n,Wait(2)
exten = _112,n,Playback(/tmp/asterisk-recording)
exten = _112,n,Wait(2)
exten = _112,n,Hangup
exten = _X.,1,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
And, just in case the 2 extensions really are supposed to do the exact
same thing, use extension pattern matching:
context a2billing
{
_11[12] =
{
playback(AR_GetGiveToID);
wait(2);
record(/tmp/asterisk-recording:ulaw,,5);
wait(2);
playback(/tmp/asterisk-recording);
wait(2);
hangup();
};
_x. =
{
deadagi(a2billing.php);
wait(2);
hangup();
};
};
(The above is my first attempt at AEL. It parses, but it hasn't actually
been tested.)
I would question the use of deadagi() in a non-h extension. Are signals
not being trapped correctly in a2billing.php?
AFAIK that's how a2billing is built, it's intentionally DeadAGI on
live channel. Ugly hack that gives warnings all the time in logs, but
it works and seems to provide correct billing info :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[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] Debugging Asterisk
On Mon, Nov 17, 2008 at 10:26 AM, Mikel Lindsaar [EMAIL PROTECTED] wrote:
Hello all,
Two questions:
1) What do people on the list do to debug phone quality issues. Phone
quality seems to be a very subjective thing. But are there metrics that you
can work against? Like maybe generating a tone and measuring the return
quality etc? It looks like all trial and error right now. If that is the
way it is, then fine. But anything more accurate / scientific?
For this you should search RTCP, recently there was a topic here. It's
not perfect, but at least gives you some data.
2) Also wondering what people do when parsing asterisk verbose output in the
log. Specifically, following a certain call. Asterisk's verbose output
logs in sequence of action, which is good, but if you have 40-50
workstations going at once, tracking the progress of one call you are trying
to make can be difficult. Obviously you can follow the channel as it goes
through. But I am wondering if there is a smarter way, like telling
asterisk to only log on certain numbers etc.
Any hints or tricks on this would be appreciated.
For me, the trick is to pass uniqueid from first channel down to child
channels. So, in child channels all you have to do, is print inherited
id. For example, this should do:
context incoming {
_X. = {
Set(__call_id=${UNIQUEID});
// do whatever
}
}
context queue_ring {
_X. = {
Verbose(${call_id});
// ring agent
}
}
So, for example queue creates child channel, which lives in separate
thread, but it will at beginning display parent's call_id.
As i do also register that call_id with CDR, my call log viewer allows
also to click on call_id and open new window with full log of this
call.
But anyway, as soon as you've found out call_id, you can do:
1) grep on log for this call_id
# cat /var/log/asterisk/full | grep 1234567890.123 /tmp/callid.txt
2) sort results, and get ony process id's of threads:
# cat /tmp/callid.txt | grep -o -P
(?:ERROR|WARNING|VERBOSE|DEBUG)\[([0-9]*)\] | grep -o \[[0-9]*\] |
uniq | sort | uniq /tmp/callid_pids.txt
So, now you have all unique process id's involved in this call.
Now you can do:
# cat /var/log/asterisk/full | grep -F -f /tmp/callid_pids.txt
callid_$1.log
Well, that's the genera, in result you get file containing log only
for one call. Then this can be improved by adding date filters (one
full file might contain repeating call id's). Also if using log
rotating, you'll have to find out in which log this call is, and then
do a zcat on compressed logs.
Also, i've heard that this approach of one uniqeid for all child
channels has been committed in trunk, it's called linked_id 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
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Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained
On Tue, Nov 18, 2008 at 12:59 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Monday 17 November 2008 04:50:43 pm Anthony Francis wrote: How do you go about determining this has happened? Tilghman Lesher wrote: Similarly, we will probably end-of-life 1.4 when a majority of users make the jump to 1.6. There are various measures, such as the questions people ask. Also, the comparative numbers of unique IPs downloading 1.4 releases, as opposed to 1.6 releases. Do you also count SVN checkouts, because that's what i usually do. And then there's also SVN switch, to update to other tag (for example 1.4.19 to 1.4.22) 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] Looking for a good lightweight Linux softPhone
On Fri, Nov 14, 2008 at 7:07 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: On Fri, 14 Nov 2008, Gordon Henderson wrote: On Fri, 14 Nov 2008, Tilghman Lesher wrote: On Friday 14 November 2008 09:19:22 Gordon Henderson wrote: On Fri, 14 Nov 2008, Tzafrir Cohen wrote: On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote: I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. Ekiga is a bit nicer for an end-user. Twinkle is probably what you'd want for testing and such. What I really want is a command-like dialer. Call me a boring old fart, but I'm utterly fed-up with the current bloatware out there. So with the IAX library and some time I might just come up with something. Technically, you can use Asterisk as a softphone, by using chan_alsa or another console channel driver. The interface is nothing if not command-line oriented. Maybe the word bloatware got lost in the trans-atlantic translation... If I had a command-line program where I could type: call 07712191046 then I'd be happy. Being able to use the arrow keys to adjust volume would be a bonus, but I already have a command-line mixer program. Gordon (the luddite) Hmm, reading the thread I assumed you were already running asterisk, since you did actually post on the asterisk user mailing list. As was pointed out you can use the asterisk command line interface to do just what you are suggesting (assuming you have a compatible sound card), and you could even wrap it in a simple shell script so you have what you say you want: call number Regardless, if you are planning to write an IAX lib based command line tool you will need an asterisk server to connect to to place your calls. I am not understanding where you think the bloatware is coming into play. So are you sitting at the console of the machine running asterisk or is this something that you would use from a standalone *nix workstation that would use the net to route your call? A small shell/perl/whatever script that takes care of minimized asterisk config could really kick ass. I wonder does anybody feels up to the challenge to create/maintain it, and push to common distros. 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] RTP LOG
On Sat, Nov 15, 2008 at 6:47 AM, Max Alex [EMAIL PROTECTED] wrote:
Hi All,
Thanks for reply
i have tried for this, it looks fine for me,
but is there any way to check rtp log while call is connected or any way to
enable it to write in log file.
Please give me some guide lines!
thanks in advance.
CLI rtcp stats
CLI rtcp debug
and as i recall you might also need sip set debug on in order to
link this to calls/ip's, as rtcp stats are reporting only SIP call id.
Regards,
Atis
Thanks,
Max Alex
Voip Developer
On Sat, Nov 15, 2008 at 3:21 AM, Benny Amorsen [EMAIL PROTECTED]
wrote:
Positively Optimistic [EMAIL PROTECTED] writes:
exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
exten = h,2,Hangup()
results in
Set(SIP/rpx2399a-b61fc5e0,
CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00)
Does it still only report what was in the last incoming RTCP packet?
/Benny
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Re: [asterisk-users] Looking for a good lightweight Linux softPhone
On Fri, Nov 14, 2008 at 10:27 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Nov 14, 2008 at 08:34:48PM +0200, Atis Lezdins wrote: On Fri, Nov 14, 2008 at 7:07 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote: On Fri, 14 Nov 2008, Gordon Henderson wrote: On Fri, 14 Nov 2008, Tilghman Lesher wrote: On Friday 14 November 2008 09:19:22 Gordon Henderson wrote: On Fri, 14 Nov 2008, Tzafrir Cohen wrote: On Fri, Nov 14, 2008 at 02:02:14PM +, Gordon Henderson wrote: I used to use IDEFISK, but since it was taken over/renamed into Zoiper it's been really hard work - now I'm told that they won't support my chosen distribution - Debian Etch - the current stable version of Debian I prefer. Ekiga is a bit nicer for an end-user. Twinkle is probably what you'd want for testing and such. What I really want is a command-like dialer. Call me a boring old fart, but I'm utterly fed-up with the current bloatware out there. So with the IAX library and some time I might just come up with something. Technically, you can use Asterisk as a softphone, by using chan_alsa or another console channel driver. The interface is nothing if not command-line oriented. Maybe the word bloatware got lost in the trans-atlantic translation... If I had a command-line program where I could type: call 07712191046 then I'd be happy. Being able to use the arrow keys to adjust volume would be a bonus, but I already have a command-line mixer program. Gordon (the luddite) Hmm, reading the thread I assumed you were already running asterisk, since you did actually post on the asterisk user mailing list. As was pointed out you can use the asterisk command line interface to do just what you are suggesting (assuming you have a compatible sound card), and you could even wrap it in a simple shell script so you have what you say you want: call number Regardless, if you are planning to write an IAX lib based command line tool you will need an asterisk server to connect to to place your calls. I am not understanding where you think the bloatware is coming into play. So are you sitting at the console of the machine running asterisk or is this something that you would use from a standalone *nix workstation that would use the net to route your call? A small shell/perl/whatever script that takes care of minimized asterisk config could really kick ass. What asterisk configuration exactly? Well, take minimal list of modules, write dialplan of few lines, make all incoming calls to ring speaker/other configured device. Probably somebody has done this already, i just haven't seen anything public. I wonder does anybody feels up to the challenge to create/maintain it, and push to common distros. Both chan_oss and chan_also tend to be disabled by default. This is because both can't be loaded (which would generate an error message). Asterisk also thinks that it has all the sound channel for itself. Even with chan_alsa, IIRC, it is not possible to play any other sound while asterisk uses the speaker/microphone. Script could take care of it, load corresponding module (specified by command-line arg) etc etc.. Generally i'm thinking to create something like Asterisk Softphone which is full featured softphone, usable from shell. # asph start alsa [EMAIL PROTECTED] # asph add [EMAIL PROTECTED] # asph dial [EMAIL PROTECTED] # asph answer # asph hangup 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] QueueLog from AMI
On Wed, Nov 12, 2008 at 6:44 PM, Sebastian Gutierrez [EMAIL PROTECTED] wrote: Hi, How can I pass the following data to te queuelog via ami?? Agent,data. ?? I'm doing this: Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n And thath works fine getting the log with the event but I cant find how to pass the agent and data parameters Any idea? From app_queue.c (1.6.0): queuename = astman_get_header(m, Queue); uniqueid = astman_get_header(m, UniqueId); interface = astman_get_header(m, Interface); event = astman_get_header(m, Event); message = astman_get_header(m, Message); ast_queue_log(queuename, S_OR(uniqueid, NONE), interface, event, %s, message); So, agent would be Interface and data would be Message. However, i wonder why do you need to pass Login event, as any kind of Queue Login (dialplan or AMI) would do that automatically. Regards, Atisw -- 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] QueueLog from AMI
On Wed, Nov 12, 2008 at 7:31 PM, Sebastian Gutierrez [EMAIL PROTECTED] wrote: Not if I have realtime, I'm inserting and deleting from queue_members table, so I don't have that info. As am I. I posted a patch that fixes this, so you could be interested in keeping it in mind (if not even backporting 3 added lines) when upgrading to 1.6.1. http://svn.digium.com/view/asterisk?view=revrevision=120166 Regards, Atis -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins Enviado el: Wednesday, November 12, 2008 3:16 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] QueueLog from AMI On Wed, Nov 12, 2008 at 6:44 PM, Sebastian Gutierrez [EMAIL PROTECTED] wrote: Hi, How can I pass the following data to te queuelog via ami?? Agent,data. ?? I'm doing this: Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n And thath works fine getting the log with the event but I cant find how to pass the agent and data parameters Any idea? From app_queue.c (1.6.0): queuename = astman_get_header(m, Queue); uniqueid = astman_get_header(m, UniqueId); interface = astman_get_header(m, Interface); event = astman_get_header(m, Event); message = astman_get_header(m, Message); ast_queue_log(queuename, S_OR(uniqueid, NONE), interface, event, %s, message); So, agent would be Interface and data would be Message. However, i wonder why do you need to pass Login event, as any kind of Queue Login (dialplan or AMI) would do that automatically. Regards, Atisw -- 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 -- 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] QueueLog from AMI
On Wed, Nov 12, 2008 at 9:40 PM, Sebastian Gutierrez [EMAIL PROTECTED] wrote: Would this be part of 1.6.1 release??? AFAIK yes. It's already in branch. However you might be confused about when it will be. Digium has changed numbering for releases, 1.6.1 is next major release, so it won't be out in month or two. Next release in 1.6.0 branch will be 1.6.0.2. Regards, Atis Regards -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins Enviado el: Wednesday, November 12, 2008 5:12 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] QueueLog from AMI On Wed, Nov 12, 2008 at 7:31 PM, Sebastian Gutierrez [EMAIL PROTECTED] wrote: Not if I have realtime, I'm inserting and deleting from queue_members table, so I don't have that info. As am I. I posted a patch that fixes this, so you could be interested in keeping it in mind (if not even backporting 3 added lines) when upgrading to 1.6.1. http://svn.digium.com/view/asterisk?view=revrevision=120166 Regards, Atis -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins Enviado el: Wednesday, November 12, 2008 3:16 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] QueueLog from AMI On Wed, Nov 12, 2008 at 6:44 PM, Sebastian Gutierrez [EMAIL PROTECTED] wrote: Hi, How can I pass the following data to te queuelog via ami?? Agent,data. ?? I'm doing this: Action: QueueLog\r\nQueue: queueprueba\r\nEvent: Login\r\n\r\n And thath works fine getting the log with the event but I cant find how to pass the agent and data parameters Any idea? From app_queue.c (1.6.0): queuename = astman_get_header(m, Queue); uniqueid = astman_get_header(m, UniqueId); interface = astman_get_header(m, Interface); event = astman_get_header(m, Event); message = astman_get_header(m, Message); ast_queue_log(queuename, S_OR(uniqueid, NONE), interface, event, %s, message); So, agent would be Interface and data would be Message. However, i wonder why do you need to pass Login event, as any kind of Queue Login (dialplan or AMI) would do that automatically. Regards, Atisw -- 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 -- 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 -- 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] List eating mail again?
On Thu, Nov 13, 2008 at 1:23 AM, Doug Lytle [EMAIL PROTECTED] wrote: Doug Lytle wrote: I've replied to two emails in the last two days and haven't seen them yet. Please ignore, I must be getting blind. I seem to have missed them. Settinu up bounce upon your post in list settings and then filtering them to separate folder helps a lot. Regards, Atis Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] tired of midget packet received warnings
On Sat, Nov 8, 2008 at 9:20 AM, Louis-David Mitterrand [EMAIL PROTECTED] wrote: On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote: Tzafrir Cohen wrote: On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. Maybe it's me, but I think that warning should be regarding a problem I can fix. Malformed network content does not neceserily fall under that definition. notice? Absolutely it does. Warnings of malformed packets are often (as mentioned above) symptomatic of network problems. Fix the network problem, fix the warning. C'mon, even firewalls give you the option of _not_ logging malformed packets! fiaif does. Else your logfile would be the weak point of your system. And what if you can't fix the source of these packets? And what if friendly peers outside of your realm (likely to iax-call you, so can't block them) sends these packets? There are holes in your logic. So asterisk has to be puritan of the lot? Holier than thou? Pro-life with malformed packets? I see where this is going and I don't like it one bit. Asterisk offers very much the same flexibility. You can disable specific log levels (for example warnings) in logger.conf or you can log everything to syslog, where filter out this specific message. 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] Variable Scope Question
On Thu, Nov 6, 2008 at 6:12 PM, Brent Davidson
[EMAIL PROTECTED] wrote:
If I have a global variable in my dialplan and I change it, does that
change immediately take affect for all calls that are active?
Here is my situation. The company I work for has two office groups that
share a building. The two offices are separate companies but support
one another and want to be able to transfer calls as if they were all on
the same phone system. Each company has 4 incoming voice lines and
calls on those lines should be sent to the appropriate main menu.
As it stands I have a context called internal that defines all of the
internal extensions for both offices then I have two virtually operator
contexts, two virtually identical mainmenu contexts and two virtually
identical admin contexts that allow them to record the appropriate
mainmenu greetings.
What I'd like to do would be to consolidate the mainmenu and operator
contexts and create a CompanyA and CompanyB context that sets a variable
for the appropriate company then jumps into the mainmenu or operator
context carrying that value so the correct greetings are played and the
correct operator extension is used. The variable would need to be one
that only affects the current call and no others since there is the
potential to have 4 calls coming in to each office at the same time.
Any ideas on the best way to handle this?
Hello,
there is not a definite best way, however variable approach sounds ok.
All you need is channel variable as opposed to global variable.
Whenever call starts (internal or external), you can match mask of DID
or CallerID (for internal extensions) and just execute
Set(__company=A). Two underscores means that this variable will be
inherited in every child channel, so wherever the call will go (within
Asterisk of course) you will have variable ${company}
For more information please see http://www.voip-info.org/wiki-Asterisk+variables
Regards,
Atis
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Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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Re: [asterisk-users] [OT] Capitalism (was: Spam from DIDForSale [EMAIL PROTECTED])
On Thu, Nov 6, 2008 at 7:50 PM, Anthony Francis [EMAIL PROTECTED] wrote: http://en.wikipedia.org/wiki/Jacque_Fresco A resource based economy. Greg Woods wrote: On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. Socialism has already completely failed. What should we do, go back to a barter economy? :-) Thanks for interesting link :) Didn't knew any such projects exist. I recently submitted idea for Google Project 10^100 which would help implementing Resource Basec Economy (i just didn't knew that such term exists). Can't wait January 27th.. :) 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] AEL NoOp not working
On Wed, Nov 5, 2008 at 12:39 PM, Olivier [EMAIL PROTECTED] wrote: Hi, I've new to http://www.voip-info.org/wiki/view/Asterisk+AEL2 I'm using NoOp and Verbose functions inside extensions.ael. Strangely, NoOp is not printing anything in Asterisk console while Verbose is working. Am I missing something obvious ? Hi, NoOp is not outputting anything, it's just does nothing, however you should still be able to see Executing NoOp(blablabla) in console, as it's a 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phishing attempt
On Wed, Nov 5, 2008 at 5:46 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Wed, Nov 5, 2008 at 10:26 AM, Roderick A. Anderson [EMAIL PROTECTED] wrote: FYI/Heads up, I /just/ received what looks like a phishing attempt for information about Open Source PBX usage. It says it comes from Digium but all the links (including the one for digium.com) point elsewhere. Rod -- I doubt it is a Phishing Attempt, there are many free (or paid) survey websites that make creating surveys extremely easy, rather than coding and hosting the survey yourself. I see it all the time. Anyways, it asks for nothing but usage statistics, so I am not sure what they are Phishing for? Obviously, they already have your email, I guess they could get your IP and usage stats.. Harmless in my opinion, I am 99.9% sure it is from Diigium but even if not, I would answer the same questions if it were a survey from anyone in the community. I just would not like the direct delivery method rather than the list. Yes, but first impression was somehow weird - i received email from Digium with links pointing elsewhere. I doubted for few seconds before clicking anything (old habit from windows systems :) Source of email seems legitimate, i just wonder why Digium didn't survey on it's own page. It could have been done in simple way with Google Documents, putting survey HTML on Digium site. Or just code a few lines with PHP and you have exactly the same survey. Btw, i somehow recall filling this out already a long time ago. Regards, Atis Received: by 10.210.124.9 with SMTP id w9cs37934ebc; Wed, 5 Nov 2008 07:25:22 -0800 (PST) Received: by 10.65.189.20 with SMTP id r20mr825665qbp.71.1225898713168; Wed, 05 Nov 2008 07:25:13 -0800 (PST) Return-Path: [EMAIL PROTECTED] Received: from mail.email.digium.com (mail.email.digium.com [66.48.80.65]) by mx.google.com with ESMTP id k30si23651859qba.4.2008.11.05.07.25.12; Wed, 05 Nov 2008 07:25:13 -0800 (PST) Received-SPF: pass (google.com: domain of [EMAIL PROTECTED] designates 66.48.80.65 as permitted sender) client-ip=66.48.80.65; Authentication-Results: mx.google.com; spf=pass (google.com: domain of [EMAIL PROTECTED] designates 66.48.80.65 as permitted sender) [EMAIL PROTECTED] ReturnPath: Digium[EMAIL PROTECTED] X-Mailer: SMTP Message-ID: [EMAIL PROTECTED] MIME-Version: 1.0 From: =?utf-8?B?RGlnaXVt?= [EMAIL PROTECTED] -- 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] AEL NoOp not working
On Wed, Nov 5, 2008 at 5:28 PM, Olivier [EMAIL PROTECTED] wrote: 2008/11/5 Atis Lezdins [EMAIL PROTECTED] On Wed, Nov 5, 2008 at 12:39 PM, Olivier [EMAIL PROTECTED] wrote: Hi, I've new to http://www.voip-info.org/wiki/view/Asterisk+AEL2 I'm using NoOp and Verbose functions inside extensions.ael. Strangely, NoOp is not printing anything in Asterisk console while Verbose is working. Am I missing something obvious ? Hi, NoOp is not outputting anything, it's just does nothing, however you should still be able to see Executing NoOp(blablabla) in console, as it's a command. Yes, that's the point : I don't see anything in console (I wasn't expecting anything else to happen). Strange ... Ok, i played with this, and seems that Executing ... lines are shown in CLI only on verbose level 3 or higher. So, either start asterisk with asterisk -vvv or issue core set verbose 3 in CLI. 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] [OT] Flash player for call recordings - 8khz
Hello, I'm trying to find simple MP3 player in flash, to integrate it with call recordings. My requirements would be: * simple UI * buffering (would be nice) * slider * volume control * support of 8kHz stereo mp3 * javascript access to seek/position * free for any use (GPL, MPL, MIT, BSD) So far I've found that JWplayer[1] does great with my recordings. However it's not small in size, as there's video player and playlists - none of which i need. Also it should be paid, even if i use it internally in company, which i don't like. Also I found niftyPlayer[2], which would be perfect, however it creates chipmunk sound out of 8kHz recordings, so i wonder what's the difference. Could anybody share their experience with call recordings? What i have found to be useful: * Record everything in G.711 (as it's native codec, thus less transcoding and more quality) * Do a nightly (or per request) conversion to stereo MP3's preserving 8kHz. sox -M is great for this. Resulting files have smaller size than in GSM or WAV format, so you can keep more recordings. References: [1] http://www.jeroenwijering.com/?item=JW_FLV_Player [2] http://www.varal.org/media/niftyplayer/ 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] Fring: Open VPN client to be installed on the mobile, which mobile?
As i recall, iphone runs on Mach kernel, which should be UNIX compatible. Of course you would have to jailbreak it. Or you can try Openmoko, which is pure Linux. Regards, Atis On Mon, Oct 27, 2008 at 3:08 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Which mobile phone work with UNIX or support UNIX? Regards Bilal --- On Mon, 10/27/08, Alex Balashov [EMAIL PROTECTED] wrote: From: Alex Balashov [EMAIL PROTECTED] Subject: Re: [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile? To: [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, October 27, 2008, 7:34 AM Yes, but Windows is ghetto. bilal ghayyad wrote: Why u r assuming UNIX? There are open vpn that works with windows. OK, let us go with UNIX idea, the question is which mobile that can run UNIX or embeded version of UNIX? As I know that mobiles have special OS which is not UNIX and not normal windows (like xp, vista, 98, NT, 2000, 2003). I might need to ask the question in another way: Did anyone try Open VPN client on the Nokia Communicator or any mobile phone? What was the used OS with that mobile? Regards Bilal --- On Mon, 10/27/08, Alex Balashov [EMAIL PROTECTED] wrote: From: Alex Balashov [EMAIL PROTECTED] Subject: Re: [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile? To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, October 27, 2008, 7:17 AM Something that runs some embedded version of a UNIX derived OS. bilal ghayyad wrote: Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- 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 -- 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] Agents log in afterhours
On Fri, Oct 24, 2008 at 9:33 PM, Ing. Jorge S Alanís Garza [EMAIL PROTECTED] wrote: Hi all, I received a report of a client which stated that two of its agents are logging in to the queues when they actually aren't there working. They appeared to be logged on all night. They thought they weren't logging off correctly, but they checked one of them and he was following the procedure. Any ideas of what can be happening? Is there a way to prevent logins to queues afterhours? The question is actually - what impact does it have. Agents could login in working time and just forget to log out. So any deny would be ineffective. We have DID routing determined by free member count in queues + working hours. Some configurations allow calls to go to queue within working hours if there are no agents, and some don't allow to accept calls in after hours. So, in order to not have callers wait in queue within afterhours we have several methods used together: 1) If agent don't answer a call, he gets either paused or logged off. Paused agents don't count as free, but they are still around, so routing might send call to IVR first to welcome caller and give agents some time. 2) Within after hours all agents are logged out every 15 minutes. So, they are allowed to work after official working hours, but they just have to relogin every 15 minutes. Realtime queue members in MySQL and cron script makes this quite straightforward :) 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] [help] Realtime Swich any context dinamically
On Tue, Oct 21, 2008 at 1:46 PM, morteza kashani [EMAIL PROTECTED] wrote: when i wnat to working with realtime and mysql for any context i have to insert (switch = Realtiem/[EMAIL PROTECTED]) statment into extensions.conf for example if i want to have 10 context, i have to insert these lines into extension.conf : [context1] switch = Realtiem/[EMAIL PROTECTED] [context2] switch = Realtiem/[EMAIL PROTECTED] [context3] switch = Realtiem/[EMAIL PROTECTED] i want to switch these 10 context whithout insert these 10 lines how can i switch any context to real time dinamically ? Hello, short answer is - you can't. If you really need lot of contexts added, you can place a #exec line in your dialplan, write a script that would do SELECT DISTINCT context FROM extensions_table and for each of results print out switch= lines. However you would need to issue dialplan reload or AEL reload whenever you add a context. Regards, Atis P.S. try to not post twice :) -- 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] a little regex help needed
On Mon, Oct 20, 2008 at 9:20 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid(num) of 2124571123 to generate an if test
of [02124571123 == 021245711*] or TRUE.
This is not the correct way to use regular expressions. Regular
expression is matched to data withing REGEX function, and it just
returns match/don't match.
Here's description
REGEX(regular expression data)
[Synopsis]
Regular Expression
[Description]
Returns 1 if data matches regular expression, or 0 otherwise.
Please note that the space following the double quotes separating the
regex from the data
is optional and if present, is skipped. If a space is desired at the
beginning of the data,
then put two spaces there; the second will not be skipped.
So, it would be something like:
${REGEX(21245711.* ${CALLERID(num)})}
But I've messed up the regex statement somehow.
In regular expressions, the * means zero or more of the preceding
character, so the way you have that written means 021245711 and zero or
more 1s. What you want instead is 021245711.*, which means
021245711 followed by at least on other character.
correction - 021245711.* would match also 021245711 as * allows zero
or more and dot means any character.
Hopefully that sets you on the right path. Don't forget that Asterisk
has two regex operators that can be used in expressions as well...
they're the ':' and '~' operators.
I wonder what are those used for? Never heard of that.
Are you really sure you need regular expressions there? Asterisk has
it's own number pattern matching, as it's much easier to read, and
would allow easy adding/removing some specific masks. Here's one
sample:
[main]
..
exten = s,n,GoSub(callerid-update,${CALLERID(num)},1)
..
[callerid-update]
exten = 021245711XX,1,Set(CALLERID(name)=Office: ${CALLERID(num)});
exten = 021245711XX,2,Return();
exten = _X.,1,Return();
exten = i,1,Return(); // just for safety :)
Of course there's also direct callerid matching, so you can match
dialed extension and callerid in same rule, but this looks simpler to
me in this case :)
For more info see
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf and
search for ex-girlfriend :)
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_log to mySQL backport to 1.4
Hi John, this is getting quite strange, and i'm becoming quite curios why it's not working :) Could you try first setting up realtime for SIP or queues? This should work out-of-the-box on 1.4. http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue For beginning you can just add queues=.. in extconfig.conf to see that SQLs go trough and return errors. You should see SELECT's in your log whenever accessing. For example, entering into CLI: ast-dev14*CLI queue show myqueue would write into log: [Oct 15 04:04:09] DEBUG[9935] res_config_mysql.c: MySQL RealTime: Everything is fine. [Oct 15 04:04:09] DEBUG[9935] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM queue_table WHERE name = 'myqueue' Also i would suggest enabling full log, as it's one place you can see everything. Then use grep to search for realtime messages. Your logger.conf should already have commented line: full = notice,warning,error,debug,verbose Then you can do: # tail -fn0 /var/log/asterisk/full | grep -F res_config_mysql to see every message about realtime driver. After this you can unload and load module or restart asterisk completely (if restarting, make sure it's started with -vvvd). To reload module, use: ast-dev14*CLI module unload res_config_mysql.so MySQL RealTime unloaded. ast-dev14*CLI module load res_config_mysql.so == Parsing '/etc/asterisk/res_mysql.conf': Found MySQL RealTime driver loaded. Loaded res_config_mysql.so = (MySQL RealTime Configuration Driver) On Wed, Oct 15, 2008 at 9:05 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: Hi Atis, queue_log = mysql,asteriskcdrdb,queue_log that is engine,database,table If it's wrong, you should see some warnings when asterisk is starting up. Thanks for the suggestion. I did not put in queue_log for table and it has just taken the default which is queue_log. In the console startup, you can see below that it has successfully bound queue_log to /mysql/db1/queue_log. # asterisk -rvvv Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. [...] == Parsing '/etc/asterisk/extconfig.conf': Found == Binding queue_log to mysql/db1/queue_log Connected to Asterisk 1.4.21.2 currently running on machine Verbosity is at least 3 This seems somehow strange. If you connect to running asterisk with -r, you shouldn't see parsing extconfig.conf, as it should be parsed on startup time. You could also add -d to enable debug 1. In /var/log/asterisk/messages, I saw: [Oct 15 15:31:48] NOTICE[20941] config.c: Registered Config Engine mysql Another idea that came into my mind is, that (if this config doesn't still work) you might have to do make dist-clean within asterisk-addons after reinstalling asterisk, and then configure, make, make install. It's because addons do use headers from installed version of asterisk, and they might not have correct declarations. Basically, I did: - Asterisk-1.4.21.2 make clean ./configure make make install - Asterisk-addons-1.4.7 make dist-clean ./configure make make install Yes, this is completely correct (assuming you restarted asterisk after :) Also, you mentioned that you checked /var/log/asterisk/messages, however i think debug is written into file called debug. Anyway you can enable full in logger.conf and get everything there. To debug this you shouldn't need more than core set verbose 3 and core set debug 1. I turned on debug mode and tried an agent login and logoff. However, when I looked into debug and messages, there are lots of chan_sip.c and a few cdr_addon_mysql.c but no occurrence at all of res_config_mysql.c What is happening? Do I have to explicitly load it? *CLI module show like res_config_mysql Module Description Use Count res_config_mysql.so MySQL RealTime Configuration Driver 0 1 modules loaded This should also be fine. You could also try catching me on irc, just look for atis_work or atis_home in #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 voicemail
On Tue, Oct 14, 2008 at 1:00 PM, Chris Rowson [EMAIL PROTECTED] wrote: Hi folks, I'm working on a solution using the Asterisk voicemail component and wondered if anyone knew the answer to this question please? I understand that Asterisk saves voicemail to /var/spool/asterisk/voicemail/context/mailbox/INBOX/ but I wondered if * creates the file in memory (or tmp/or wherever) and then loads the completed file into that directory, or if it writes the file to the directory directly, appending it till the recording is finished? Sorry to reply to my own post! I notice a tmp directory at /var/spool/asterisk/voicemail/context/mailbox/tmp/ I'm wondering if this is where the file is created, and then moved to the INBOX folder perhaps? I'm not sure about voicemails, perhaps they have temporary storage in /tmp/, however there's more general option for asterisk. See man asterisk, there's command -t which could be passed at asterisk startup, then asterisk will write all files in /var/spool/asterisk/tmp (allocating empty filename before), and after recording finishes it will move them to correct location. 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_log to mySQL backport to 1.4
Hi John,
On Tue, Oct 14, 2008 at 3:36 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
if you have applied everything correctly - queue_log file shoudln't
have any more lines (except init when restarting asterisk).
Thanks Atis.
I see what you are saying. In the patch for logger.c,
The code to write to mysql is there except that we need to perform
ast_check_realtime(queue_log).
I guess ast_check_realtime() is looking into extconfig.conf and
searching for
queue_log = mysql,db1
Ok, after another check this seems wrong. For me it's:
queue_log = mysql,asteriskcdrdb,queue_log
that is engine,database,table
If it's wrong, you should see some warnings when asterisk is starting up.
Another idea that came into my mind is, that (if this config doesn't
still work) you might have to do make dist-clean within
asterisk-addons after reinstalling asterisk, and then configure, make,
make install. It's because addons do use headers from installed
version of asterisk, and they might not have correct declarations.
Also, you mentioned that you checked /var/log/asterisk/messages,
however i think debug is written into file called debug. Anyway you
can enable full in logger.conf and get everything there. To debug
this you shouldn't need more than core set verbose 3 and core set
debug 1.
Regards,
Atis
which is there in my extconfig.conf already.
Can any Asterisk developers enlighten me on this?
void ast_queue_log(const char ...)
{
+ char qlog_msg[8192];
+ char time_str[16];
+
+ if (ast_check_realtime(queue_log)) {
va_start(ap, fmt);
+ vsnprintf(qlog_msg, sizeof(qlog_msg), fmt, ap);
va_end(ap);
+
+ snprintf(time_str, sizeof(time_str), %ld,
(long)time(NULL));
+ ast_store_realtime(queue_log, time, time_str,
+ callid, callid,
+ queuename, queuename,
+ agent, agent,
+ event, event,
+ data, qlog_msg,
+ NULL);
+ } else {
+ if (qlog) {
+ AST_LIST_LOCK(logchannels);
+ va_start(ap, fmt);
+ fprintf(qlog, %ld|%s|%s|%s|%s|,
(long)time(NULL), callid, queuename, agent, event);
[...]
+ }
}
-Original Message-
From: Atis Lezdins [mailto:[EMAIL PROTECTED]
Sent: Monday, 13 October 2008 8:02 PM
To: Lee, John (Sydney)
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] realtime queue_log to mySQL backport to
1.4
Hi John,
On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
http://ftp.iq-labs.net/queue_log-
1.4/asterisk_queue_log_realtime_1.4.19.patch
Haven't you forgotten this one? ;)
if you have applied everything correctly - queue_log file shoudln't
have any more lines (except init when restarting asterisk).
Regards,
Atis
This uses standardized realtime/mysql library from asterisk addons.
For it to support SQL inserts in 1.4, you would also need to apply
both patches from (1 for asterisk, another for asterisk-addons)
http://ftp.iq-labs.net/realtime_store_destroy-1.4/
This will later allow you to upgrade to 1.6 and having everything
working without patching.
I have patched in asterisk 1.4
. main/logger.c
. include/asterisk/config.h
. main/config.c
I have patched in asterisk-addons 1.4
. res/res_config_mysql.c
I have re-installed asterisk and asterisk-addons.
I created a database called db1 and in there created a table called
queue_log as per instruction
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
I changed /etc/asterisk/extconfig.conf to add the following line:
[settings]
queue_log = mysql,db1
I changed /etc/asterisk/res_mysql.conf to add the following:
[general]
dbhost = localhost
dbname = db1
dbuser = user
dbpass = password
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
1) However, whenever I perform an agent login, no row is written to
table queue_log. I checked /var/log/asterisk/queue_log and a new
entry
is written there.
2) I set debug to 10 on the console in asterisk and re-did the test
but
there were no error messages in /var/log/asterisk/messages.
3) I set debug on in mysqld and there are no information for
inserting
into table queue_log, except the cdr logging as below.
Tcp port: 0 Unix socket: (null)
Time Id CommandArgument
081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1
2 Connect [EMAIL PROTECTED] on db1
081013 16:00:32 1 Query INSERT INTO cdr_log ...
081013 16:01:42 1 Query INSERT INTO cdr_log ...
Is there anyone who
Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4
Hi John, On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: http://ftp.iq-labs.net/queue_log- 1.4/asterisk_queue_log_realtime_1.4.19.patch Haven't you forgotten this one? ;) if you have applied everything correctly - queue_log file shoudln't have any more lines (except init when restarting asterisk). Regards, Atis This uses standardized realtime/mysql library from asterisk addons. For it to support SQL inserts in 1.4, you would also need to apply both patches from (1 for asterisk, another for asterisk-addons) http://ftp.iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. I have patched in asterisk 1.4 . main/logger.c . include/asterisk/config.h . main/config.c I have patched in asterisk-addons 1.4 . res/res_config_mysql.c I have re-installed asterisk and asterisk-addons. I created a database called db1 and in there created a table called queue_log as per instruction http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL I changed /etc/asterisk/extconfig.conf to add the following line: [settings] queue_log = mysql,db1 I changed /etc/asterisk/res_mysql.conf to add the following: [general] dbhost = localhost dbname = db1 dbuser = user dbpass = password dbport = 3306 dbsock = /var/lib/mysql/mysql.sock 1) However, whenever I perform an agent login, no row is written to table queue_log. I checked /var/log/asterisk/queue_log and a new entry is written there. 2) I set debug to 10 on the console in asterisk and re-did the test but there were no error messages in /var/log/asterisk/messages. 3) I set debug on in mysqld and there are no information for inserting into table queue_log, except the cdr logging as below. Tcp port: 0 Unix socket: (null) Time Id CommandArgument 081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1 2 Connect [EMAIL PROTECTED] on db1 081013 16:00:32 1 Query INSERT INTO cdr_log ... 081013 16:01:42 1 Query INSERT INTO cdr_log ... Is there anyone who can help me? -- 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] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'
On Fri, Oct 10, 2008 at 10:50 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: Sorry to post a C compile error on this mailing list but this is Asterisk related. Basically, I was following http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queu e_logging to patch logger.c and Makefile in Asterisk 1.4.* in order to write queue_log to mySQL database. When I ran make, it complained: In function `write_mysql_logger': [...] /usr/src/asterisk-1.4.21.2/main/logger-mysql.c:98: undefined reference to `mysql_error' [...] collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 In my modified Makefile, I already had the line: ASTCFLAGS+=-I/usr/include/mysql and I found that mysql.h is already in /usr/include/mysql. I also already had mysql-client installed. In logger-mysql.c, there is already a line at the front of the program: #include mysql.h Any thoughts? This looks really old and weird. I could suggest using realtime queue_log backport from 1.6 which i'm currently using. http://ftp.iq-labs.net/queue_log-1.4/asterisk_queue_log_realtime_1.4.19.patch This uses standardized realtime/mysql library from asterisk addons. For it to support SQL inserts in 1.4, you would also need to apply both patches from (1 for asterisk, another for asterisk-addons) http://ftp.iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. 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 on using DMZ
On Thu, Oct 9, 2008 at 6:38 AM, C. Savinovich [EMAIL PROTECTED] wrote: I am tinkering with a new router, a Linksys wrt610n dual-band, etc. But the when I connect it, the softphones(x-lite) on the computers don't even register. After a couple of hours of hassle, I found out that if I dmz the router to the computer I am using, the softphone starts to work. Problem is, there are about 6 computers in this office, all using x-lite. Can anybody suggest what to do here to so that I can enable all 6 computers connected to this router? You should forward different ports for each softphone, and change ports in each of them. As i remember, x-lite uses 5060 and 8000-8005, so forward those to first computer, then change settings for x-lite on second computer (5061 and 8006-8010) and forward them to second ip, etc. DMZ is just alias for forward all ports to one ip, so not much use. As alternative you can set up VPN on router and asterisk box, so asterisk will treat all internal addresses as local. Regards, Atis Thanks CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Sent: Wednesday, October 08, 2008 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] retransmitting NAT Hi, What does retransmitting NAT means? I have a client that uses SPA 942, and his phone sometimes cannot be called. i did a sip sebug and i keep on seeing retransmitting NAT. on the realtime it shows that it is registered, so when i try to call it , asterisk thinks it is still online so it tries to reach it instead of saying it's unavailable, [Oct 9 11:10:33] -- Called 103100 it stops there until it reached the timeout i set then it will say unavailable. is there a way that realtime will know that the phone is not registered anymore? or what could be causing the retransmitting of NAT? has anyone encountered the same prob? thank you regards, nhadie ___ -- 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
