Re: [asterisk-users] How often to restart Asterisk...
I've actually had an AGI script that Asterisk never closed the fork for. It was testing a particular feature so it was pretty badly written. Ended up consuming a lot of resources. No idea why Asterisk hated that script, though. Failed to kill it every time. But would continue on the dial plan after I sent back the data it needed and supposedly ended the program. Never had it happen since On Jan 12, 2013 12:56 PM, Steve Edwards asterisk@sedwards.com wrote: On Sat, 12 Jan 2013, Stelios Koroneos wrote: The biggest issue i have faced in term of stability is badly written AGI scripts that tend to hog resources and bring systems down in the end. Veering off topic, but still curious :) Since an AGI only exists for part of the life of a single call, how does it accumulate enough resources to be a problem? -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon SIP trunking Field Trial
Does anyone have a good contact for their sales? I've attempted calling their Enterprise sales a few times and was just spun around in circles. Having a sales rep I can just call would be awesome. - Logan On Fri, Jan 4, 2013 at 1:36 PM, Michael L. Young myo...@acsacc.com wrote: - Original Message - From: Matthew J. Roth mr...@imminc.com At least Verizon maintains a consistent customer experience. ; ) Overall, we've found the service to be reliable and stable, but when there are problems or changes needed you're dealing with Verizon and the w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y. Haha... that is funny... it is sooo true. Well, you are right. Once it is working, it is usually pretty stable. Just a pain in the butt when things are not working. Hopefully we can get through the Field Trial and that is all I have to worry about for a while. Thanks Matthew for all the encouragement as I go down this temporary (I hope) unpleasant path. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Logan Logan Bibby, CEO Ke*o*bi Communications Tuscaloosa, Alabama -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timeout(absolute) not working on transfer
Geoff,
I believe its actually TIMEOUT(absolute)=value. The function name is case
sensitive.
- Logan
On Dec 30, 2012 9:53 AM, Geoff Lane ge...@gjctech.co.uk wrote:
Hi All,
Asterisk 1.4.22.1 on CentOS 5
I've configured my dialplan to limit the maximum call length on
outgoing calls. I've done this as I get the first hour of each call
free with my bundle but I pay through the nose if the call goes over
an hour.
Family members who live overseas sometimes ask me to transfer them to
UK landline numbers, which is fine by me as it doesn't cost me
provided they don't exceed the hour limit. However, I noticed a few
days ago that a call from my son (who lives in Australia) that I
transferred didn't time out.
Relevant snippets of extensions.conf follow.
The incoming (via SIP) call fetches up at the following:
exten = [munged],1,Goto(main,1)
exten = main,1,Log(NOTICE, Prefilter: call from ${CALLERID(num)})
exten = main,n,PrivacyManager(2,10)
exten = main,n,GotoIf($[${PRIVACYMGRSTATUS} = FAILED]?withheld,1)
exten = main,n,Log(NOTICE, Incoming call from ${CALLERID(num)})
exten = main,n,GotoIf($[${BLACKLIST()}]?banned,1)
exten = main,n,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten = main,n,Dial(${rgMain},${RINGTIME},t)
exten = main,n,Log(NOTICE, Call from ${CALLERID(num)} sent to voicemail)
exten = main,n,VoiceMail(main@default)
To transfer the call, I press # then dial the number, which is in the
form of 01nnn nn, and so should fetch up at the following:
exten = _01.,1,SET(Timeout(absolute)=3540)
exten = _01.,n,Dial(${UKGeographical}/${EXTEN},,g); send anything
preceded with 01 to UKGeographical
Am I missing something (e.g. Timeout(absolute) doesn't apply to
transferred calls) or can anyone spot something else that's allowing
the call to continue past the 59 minute set limit?
TIA,
--
Geoff
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Re: [asterisk-users] Timeout(absolute) not working on transfer
No problem! Doubt check through a test extension. I don't want to be entirely wrong. ;) - Logan On Dec 30, 2012 12:12 PM, Geoff Lane ge...@gjctech.co.uk wrote: On Sunday, December 30, 2012, Logan Bibby wrote: I believe its actually TIMEOUT(absolute)=value. The function name is case sensitive. Many thanks. I've changed my dialplan accordingly. -- Geoff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
I'm a +1 for the change, should it come to a vote. I realize the benefits of bottom-posting, especially when posting inline. But top-posting keeps things in reverse chronological order so any reader could catch up quickly on any missed messages in the chain. A new reader scrolls to the bottom and reads up. - Logan On Dec 29, 2012 7:22 PM, Pete Mundy p...@fiberphone.co.nz wrote: On 30/12/2012, Steve Edwards wrote: On Sat, 29 Dec 2012, Don Kelly wrote: 2. How do we change rule #5? -1. + -1 from me too! Ie I dislike top-posting on mailing lists and if a democratic approach was taken to rule changes (I have no idea is this is the case?) then I would vote against the change. Just my 2c since we're discussing it. Pete -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
I suppose I'm one of the few people that remember the content of threads by subject and easily catch up... I'm also on my phone 99% of the time time and the way Gmail lays out emails makes top-posting beneficial to me. On Dec 29, 2012 8:57 PM, Richard Kenner ken...@gnat.com wrote: I realize the benefits of bottom-posting, especially when posting inline. But top-posting keeps things in reverse chronological order so any reader could catch up quickly on any missed messages in the chain. A new reader scrolls to the bottom and reads up. What's there to catch up with if you don't first read what the person is replying to? Do you think that everybody remembers every thread. Of what value is it to see something like No, that didn't work. *before* a description of what it was that didn't work. When people reply to an email, it's their responsibility, whether they top-post or bottom-post to remove unnecessary old message and keep just what's necessary to understand the email. One of the problems with top-posting is that it makes it easier to forget to do this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?
I'm a fan of your method. I haven't had good luck with GotoIfTime in the past. A lot of my dialplan is actually handled by an AGI script. I've always found that to be the easiest. On Thu, Dec 27, 2012 at 2:00 PM, Doug Lytle supp...@drdos.info wrote: Ernie Dunbar wrote: This appears to boil down to always remember to test it at the time that it becomes relevant. But if I was the kind of person who always remembered to do things at the right time, then there would never be a need for computers to do jobs like this in the first place. I no longer use GotoIfTime for these events. I do database lookups based on date. At the beginning of each year, our HR department releases the holiday schedule and I enter them into the database. All inbound calls query the database to see if there is a match and jump to the appropriate sub-routine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Logan Logan Bibby, CEO Ke*o*bi Communications Tuscaloosa, Alabama -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
I have a huge logrotate config file and I use Webmin to manage it all. Actually, Webmin is a good all-around system management tool, in my opinion. On Dec 4, 2012 9:12 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On 12-12-04 10:02 AM, Danny Nicholas wrote: IIRC log rotate only rolls the files in /var/log/asterisk, not /var/log/asterisk/cdr-csv You need to configure logroate with the path and filename. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
It is facing the outside world, but I just use SSH's port forwarding. :) On Dec 4, 2012 10:43 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Tuesday 04 December 2012, Logan Bibby wrote: I have a huge logrotate config file and I use Webmin to manage it all. Actually, Webmin is a good all-around system management tool, in my opinion. Just not on a box with an outside-world IP address, though . -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue_log into MySQL - best practices
Have you considered using something like Splunk to aggregate your log files and store a copy for later analysis? Even if you want it to be available to someone, say a remote customer, via a web panel, I believe you could even have Splunk put it into another database or make a view in Splunk's database. I believe that might work. On Nov 22, 2012 2:01 AM, Dmitry mbike200...@yahoo.com wrote: Hi, I use asterisk 1.8. Currently I use a perl daemon to parse queue_log into MySQL. It works reliably. But I know that there is a method ( http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL and http://work.mikeboylan.com/asterisk-queuelog-to-mysql) to write to MySQL directly with app_mysql which has a DEPRECATED status. My question is: What is the best/preffered approach to put queue_log into MySQL in asterisk 1.8 and up? 1) To use external daemons to parse /var/log/queue_log? 2) To use the deprecated app_mysql? the status does not guarantee that this application will be in the future 3) To use odbc to access mysql? but I could not find a procedure for it. And I doubt it is possible. BR, Dmitry Pavlenko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Managing complex setups with Asterisk
What about just setting up a database which stores your data however you want then generate static files from that data or creating views for realtime (where appropriate)? That's how I do it with my company's system. To keep things not so complicated, I have AGI scripts. Keeps things clean and is a little more flexible and powerful. - Logan On Nov 8, 2012 12:41 AM, martin f krafft madd...@madduck.net wrote: also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.07.2340 +0100]: What is your point of pain? Right now we do most of the configuration, provisioning, and system management outside of asterisk. My systems are already managed automatically, thankfully no longer with Puppet. ;) I am only talking about configuration of Asterisk, whether in /etc/asterisk or some sensible external data source. My point of pain is the complexity due to a couple of special cases, e.g. - Roaming users, i.e. no 1:n relation between sites and users; - Multiple devices per user (some want them all to ring, some want individual extensions but shared voicemail, …) - Keeping track of the mappings between incoming calls (from SIP providers) and extensions to ring (using incoming contexts and extension groups for that) - Keeping track of which extension uses which outgoing trunk - … With a logical naming scheme, a policy and include files, this is all working. But it's very error-prone and there is a bit of redundancy in the information, so I was wondering if there wasn't a better way. Either way, don't manually build your 6th machine. Start from fresh using some sort of automated tool (chef / puppet). This will help you get on the right path. The new machine for the 6th site is up and running (provisioning (not image-based) took less than half an hour). What now? ;) -- martin | http://madduck.net/ | http://two.sentenc.es/ science without religion is lame, religion without science is blind. -- albert einstein spamtraps: madduck.bo...@madduck.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bypass queue wrapup time
I don't think you can. But you could set it to a lower value like 3 seconds and give your operators a feature key to pause themselves in the queue if they need extra work time. - Logs On Oct 29, 2012 12:15 PM, Mitch Claborn mitch...@claborn.net wrote: In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? -- Mitch -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
I had the same problem for a while. I found replacing fax machines with a scanner and either an email-to-fax program or just web-based faxing had better results. I don't want to tell you the gateway I used because they turned out pretty badly in the end. But there is hope! - Logan On Oct 4, 2012 8:29 AM, Brett Lehrer brett.leh...@solarismed.com wrote: I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking service over a DSL line solely dedicated to VoIP usage. For both incoming and outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful of reasons. Is it natural to have this many problems on a completely digital configuration? I'm trying to cut our analog phone line (because it's so expensive), but some fax machines just don't seem to ever accept a fax. Many of the failures are on the same numbers, forcing me to fall back to an old analog fax machine just to make sure it actually gets through. Has anyone else had any similar experiences, or is this indicative of a failure in the setup on my end (or even the trunking service)? Brett Lehrer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reuse h extension?
I have a status context with a hangup extension. All my h calls go there. - Logan On Sep 29, 2012 4:32 AM, Stefan at WPF stefan.at@googlemail.com wrote: I have 2 contexts, however both have the same h extension. Currently I am doing copypaste for the h extension - is there a better way? Can I somehow reference a h extension, so I have to create/modify it only once? Thanks for any hint! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reuse h extension?
I do. I call the Hangup application in priority 1 so I can send calls there without needing to call it. Then the h extension goes to status,hangup,2. - Logan On Sep 29, 2012 4:36 AM, Stefan at WPF stefan.at@googlemail.com wrote: How do you redirect all h calls to your status context? Thanks :-) 2012/9/29 Logan Bibby lo...@keobi.com I have a status context with a hangup extension. All my h calls go there. - Logan On Sep 29, 2012 4:32 AM, Stefan at WPF stefan.at@googlemail.com wrote: I have 2 contexts, however both have the same h extension. Currently I am doing copypaste for the h extension - is there a better way? Can I somehow reference a h extension, so I have to create/modify it only once? Thanks for any hint! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reuse h extension?
I don't have it readily available, but it would be something like this [status] exten = hangup,1,Hangup same = 2,NoOp(Hangup) ; do further processing here [default] exten = 1234,1,Answer ; other priorities same = Goto(status,hangup,1) ; - actually hangs up the channel exten = h,1,Goto(status,hangup,2) ; - processes a channel not hung up by the dialplan On Sep 29, 2012 6:08 AM, Stefan at WPF stefan.at@googlemail.com wrote: Thanks Logan. Can you send an extract of your extensions.conf, how you do that? 2012/9/29 Logan Bibby lo...@keobi.com I do. I call the Hangup application in priority 1 so I can send calls there without needing to call it. Then the h extension goes to status,hangup,2. - Logan On Sep 29, 2012 4:36 AM, Stefan at WPF stefan.at@googlemail.com wrote: How do you redirect all h calls to your status context? Thanks :-) 2012/9/29 Logan Bibby lo...@keobi.com I have a status context with a hangup extension. All my h calls go there. - Logan On Sep 29, 2012 4:32 AM, Stefan at WPF stefan.at@googlemail.com wrote: I have 2 contexts, however both have the same h extension. Currently I am doing copypaste for the h extension - is there a better way? Can I somehow reference a h extension, so I have to create/modify it only once? Thanks for any hint! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
I agree. A script that read the spool directory, sent enough files to equal 10, wait a few seconds, check again and move more would do the trick. - Logan On Sep 27, 2012 11:27 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09/28/2012 03:01 AM, Patrick Archibald wrote: Hi, Is there a way to move 100 .call files in to /var/spool/asterisk/outgoing/ at once and have Asterisk call at maximum 10 at a time? Afaik that is not possible. Wouldn't it make more sense to move call files in batches of 10 to outgoing/? Regards, Patrick -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
MyISAM would be best, in my opinion. The features that cause the little bit of performance overhead in InnoDB wouldn't be necessary for CDR storage. - Logan On Sep 25, 2012 4:15 PM, Matt Hamilton mistral9...@hotmail.com wrote: Which one (InnoDB or MyISAM) is preferred for CDR as far as write performance is concerned? Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
Very good point. For revenue critical data like CDRs, being ACID compliant is important. MyISAM is compliant. And like InnoDB, can have the features making it compliant turned off. On Sep 25, 2012 6:12 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09/25/2012 11:18 PM, Logan Bibby wrote: MyISAM would be best, in my opinion. The features that cause the little bit of performance overhead in InnoDB wouldn't be necessary for CDR storage. Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not. More information here: http://en.wikipedia.org/wiki/**ACID http://en.wikipedia.org/wiki/ACID https://blogs.oracle.com/**MySQL/entry/comparing_innodb_** to_myisam_performancehttps://blogs.oracle.com/MySQL/entry/comparing_innodb_to_myisam_performance Regards, Patrick -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peculiar problem with failover provision.
Why not use the DIALSTATUS channel variable to determine if a fail over is necessary? - Logan On Sep 24, 2012 6:00 AM, Thomas Kenyon dig...@sanguinarius.co.uk wrote: I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/**extension) n,Dial(SIP/provider-2/**extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peculiar problem with failover provision.
I think a lot of people leave it out in examples for simplicity's sake. It doesn't instil proper practices in folks' heads. - Logan On Sep 24, 2012 12:06 PM, Eric Wieling ewiel...@nyigc.com wrote: You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the internet do it the same way. It is still wrong. When you do a Dial on the dialplan you need check the value of DIALSTATUS or HANGUPCAUSE before dialing again. Both variables will give you some indication of why the first call ended. Then your dialplan logic can decide how to proceed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: Monday, September 24, 2012 7:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peculiar problem with failover provision. I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/extension) n,Dial(SIP/provider-2/extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get SIP Response Code and use it to change destination.
If you're using below 1.8, there isn't a way. The DIALSTATUS channel
variable can give you a little, but not with those response codes.
However, if you're using 1.8, there's some hope: you can use
${HASH(SIP_CAUSE,channel)} (where channel is the destination channel,
not source) to read the SIP response code.
For my setup, I have an OpenSIPS sever that handles the lower level logic
such as failure routes. I find it a lot amiable to deal with than Asterisk
for that sort of thing.
- Logan
On Sep 23, 2012 5:17 PM, Jarek Jarzebowski jarek.jarzebow...@gmail.com
wrote:
Hello,
I need to do such a simple thing:
1. Dial SIP/123
2. If I get for example 503 - jump to Dial SIP/789
3. If I get for example 403 - jump to Playback(...)
The real question is:
how can I get SIP Responses and use it in dialplan?
Regards,
Jarek
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Re: [asterisk-users] accept email and make phone call?
If you're using sendmail and receive e-mail directly to your box, you could create a user and add a .forward file that pipes the e-mail to a script which access the Asterisk Manager interface or something of the like. There's lots of tutorials on both. Good luck! On Thu, Sep 20, 2012 at 12:31 PM, Joseph Acquisto j...@j4computers.comwrote: Any ideas on how asterisk could accept an email (such as an email to SMS or num...@mybox.org sort of thing) and make a phone call to a specific number and make an announcement? I imagine the first part is the big question. joe a. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Logan Logan Bibby Ke*o*bi Communications Mobile: (205) 394-0424 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
