| 00:00.00 | kaushal | sure |
| 00:01.06 | p3nguin | f2knight: One underscore causes the variable to be inherited once when a new channel is created by the main channel, where two causes the variable to be inherited indefinitely. |
| 00:01.46 | f2Knight | kaushal, in HTML 101 is a request to switch protocols... so with out more info it would be hard to tell you . Pastebin it with the full execution and make sure you have verbose set high like 10 |
| 00:02.02 | kaushal | ok |
| 00:03.43 | f2Knight | p3nguin, while I am about to try this out, would it work in this resepect... before I call Queue i set some channel variables, caller gets put in queue... when agent picks up (we dial out to the agent from a context, ) I would be able to retrieve the variables I set before calling Queue? If so I just spend 3 days looking for this answer. |
| 00:04.38 | kaushal | f2Knight: http://pastebin.com/RQ7Ck4pR |
| 00:04.49 | p3nguin | f2knight: I would think this will do it. |
| 00:05.39 | f2Knight | kaushal, your problem is not asterisk specfic. it is in regards to your signoma card. |
| 00:05.53 | kaushal | oh i see |
| 00:05.56 | kaushal | good hint |
| 00:06.57 | kaushal | f2Knight: how did you arrived at that conclusion ? |
| 00:06.57 | kaushal | f2Knight: do you have some sort of evidence |
| 00:07.32 | f2Knight | kaushal, I do not use any signoma cards but I know that the t1 interface cards I do use do a span reset every so often. Its a way for the T1-card to make sure what lines are available. Through me for a loop when I first saw it because I thought it was hanging up all the lines but it was not. |
| 00:08.10 | kaushal | f2Knight: I use E1 digital inteface |
| 00:08.17 | f2Knight | kaushal, If your system is working It MIGHT be the same thing that I get on my PRI, and nothing to worry about .. actually if it STOPS then in my case thats an issue. |
| 00:08.54 | f2Knight | kaushal, I think the E1 is the UK version of the US T1. . I would contact signoma directly or look on signoma boards |
| 00:09.18 | kaushal | yes |
| 00:09.44 | kaushal | f2Knight: Thanks for the hint and much appreciated |
| 00:09.52 | f2Knight | kaushal, but the way its spanning on all them I think its the same thing I get on my Digium card, and is part of the PRI driver, and is normal. ... Try finding someone with more info for sure but I think you are safe here. |
| 00:10.10 | kaushal | will contact Sangoma support |
| 00:11.07 | WIMPy | kaushal: Do you have the correct signalling configured? |
| 00:11.24 | WIMPy | 6X is a protocoll error. |
| 00:11.32 | kaushal | WIMPy: you mean in wanpipe1.conf ? |
| 00:12.16 | WIMPy | I'm not famililar with sangoma and what goes where. |
| 00:12.48 | kaushal | WIMPy: not sure i understand your question |
| 00:12.57 | kaushal | Do you have the correct signalling configured? |
| 00:13.00 | *** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net) |
| 00:13.15 | kaushal | are you referring to E1 or T1 digital interface ? |
| 00:13.16 | WIMPy | switchtype |
| 00:14.00 | kaushal | WIMPy: can you please point me to the asterisk wiki about switchtype |
| 00:14.52 | f2Knight | p3nguin, about to give it a try keeping fingers crossed. |
| 00:15.43 | WIMPy | kaushal: You get that information from your telco. |
| 00:15.52 | kaushal | ok |
| 00:16.08 | f2Knight | p3nguin, It looks like DumpChan does not display _ or __ variables at all. |
| 00:16.23 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:17.14 | p3nguin | Well, it should. If the variables are inherited, it will display them. |
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| 00:19.10 | capt_cassimir | Ahoy |
| 00:19.10 | f2Knight | p3nguin, Maybe I am mis understanding ... what I should expect.. I will paste bin |
| 00:19.20 | p3nguin | okay |
| 00:20.47 | p3nguin | My battery is exhausted, so if it takes very long, I'll be gone. |
| 00:25.03 | p3nguin | Here I go. |
| 00:28.47 | f2Knight | p3nguin, http://pastebin.com/Mstw4FzL totally confused. Maybe I am going about the whole thing wrong |
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| 00:38.07 | f2Knight | p3nguin, found that the stripping of _ is common and correct, but that still does not explain why when Queue sends a call to local/1050@agents why it does not display anything in the agents context... ugh. |
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| 00:58.29 | f2Knight | p3nguin, any suggestions? |
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| 01:47.52 | f2Knight | Q: Trying to find a way to account for the actual time that an agent is LIVE in a conversation. Suggestion? I prefer to do this in the dialplan to read variables.. |
| 02:08.14 | f2Knight | Wow with 200 users, I didn't think this would be that hard of an issue. And the net is riddled with either simiular questions relating to local channel and passing variables or just plan no answers... |
| 02:15.11 | *** join/#asterisk jeffspeff (~jeff@c-76-31-128-113.hsd1.tx.comcast.net) |
| 02:17.00 | jeffspeff | i'm having an issue with realtim ldap not refreshing it's cache... i've restarted asterisk, reloaded the core, pruned the peers, etc. I've manually verified the settings, but when i do a "sip show peer <user> load" it brings up old data. particularly the users "*name" |
| 02:17.21 | jeffspeff | any help would be greatly appreciated, i've spent most of my day trying to figure this out... thanks |
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| 02:28.01 | *** join/#asterisk sourcode (~code@ppp-58-11-83-165.revip2.asianet.co.th) |
| 02:29.17 | SunTsu | f2Knight: if you're talking about 200 users here: Remember that it's users from lots of different timezones, and remember that not all are lookng into irc right now. And not everybody is knowledgable, some are only looking for advice themselves. And not everybody knows about agents, so that does not leave many people left from those 200. |
| 02:30.01 | SunTsu | apparently not a single one |
| 02:30.28 | jeffspeff | I wish everybody knew about realtime configs with ldap... or at least one that was looking at a screen. lol |
| 02:30.43 | f2Knight | SunTsu, I understand that, and no doesnt seem like anyone does/ though I got the agents working.. this is more to do with passing of Variables between channels. |
| 02:31.05 | f2Knight | jeffspeff, I have done Asterisk w/ LDAP before.. once. |
| 02:31.28 | SunTsu | f2Knight: I was merely hinting at the fact that 200 is not as much as it seems, considering the circumstances |
| 02:31.30 | f2Knight | jeffspeff, Besides the typical nightmare of LDAP, it creates a LOT of network chatter. |
| 02:32.10 | f2Knight | SunTsu, Oh I know but I have been in the room sense around 1 or 2 my time I think about 6 hours.. just is shocking thats all. |
| 02:32.33 | f2Knight | SunTsu, do you know about Queues and the Local channel driver? |
| 02:32.44 | SunTsu | f2Knight: no, not a single bit |
| 02:33.38 | jeffspeff | f2Knight, any ideas as to my question above? |
| 02:33.42 | f2Knight | SunTsu, well didn't hurt to ask :) |
| 02:35.03 | f2Knight | jeffspeff, have you restarted ldap? |
| 02:35.11 | SunTsu | I'm one of the least knowledgable around here. I know barely enough to keep my asterisk running, with nothing fancy going on |
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| 02:43.45 | jeffspeff | f2Knight, yes, I restarted both servers at the same time, hoping to get rid of any lingering caches... FYI my ldap server is Win$ active directory |
| 02:49.14 | MrTelephone | microsoft exchange is brutal |
| 02:49.17 | f2Knight | jeffspeff, no offense you're not running LDAP, your running AD. While AD is "based" on LDAP, Microsoft has made a lot of changes to it If your running a windows AD and need AD integration. I think 3CX is where you might want to be looking, but then you have some license fees to pay. |
| 02:50.04 | f2Knight | jeffspeff, I can also tell you that it was not a cakewalk with OpenLDAP either. LDAP being a pain in general to configure. The primary question then becomes... what is your reason for LDAP? |
| 02:51.20 | f2Knight | jeffspeff, my original reason was for replication.. but I soon found out that running it that way was not the best performance unless all my servers were in one location. (network traffice it is very "chatty") |
| 02:53.01 | f2Knight | jeffspeff, if it is for Dialplan replication or registering users. I think you are better using the sipfriends table in a mysql server (just run the SQL as a cluster for redundancy), have all your * boxes get their sipinfo from that. Dialplans do not change that often and even Digium now suggests NOT using realtime dialplans. |
| 02:53.36 | f2Knight | jeffspeff, Simply create your dialplan and if you need to sync it with others up have them pull from a primary node every how often you need to pool for it. |
| 02:53.59 | MrTelephone | how long will asterisk keep the users in memory if the sql server goes down? |
| 02:54.01 | jeffspeff | the client is very windows oriented, they currently use microsoft online 365 and the purpose for integrating in AD is for password simplification, and users... we can just add a user to the AD and they're instantly able to make calls, login to computer and all their exchange, messenger, etc is up and going with the same credentials |
| 02:54.39 | MrTelephone | jeffspeff, should you buy office communicator? |
| 02:54.51 | *** join/#asterisk ExpertOrBust (~ExpertOrB@wsip-24-234-159-70.lv.lv.cox.net) |
| 02:55.06 | jeffspeff | communicator is what i meant by messenger... just didn't feel like typing it all... lol |
| 02:58.13 | MrTelephone | I want to setup 2 asterisk machines for load balancing. I have to test if you give an atlas 550 the right hangup cause if it will try a secondary route. |
| 02:58.29 | MrTelephone | anyone here use the atlas 550? |
| 02:58.34 | *** join/#asterisk PrestonL (~Preston@c-76-22-208-42.hsd1.ky.comcast.net) |
| 02:58.47 | PrestonL | I'm trying to link Active Directory to asterisk CAN SOMEONE HELP? |
| 02:59.38 | jeffspeff | PrestonL, not tonight buddy... no jokes, being serious here... the grown ups are talking... |
| 02:59.40 | MrTelephone | Why is everyone wanting to do that all of a sudden |
| 03:00.20 | MrTelephone | jeffspeff. best way to do that is to program a vbscript or somethign to build a sip.conf every 15 minuts or something |
| 03:00.38 | MrTelephone | or better yet. connect to mysql and like f2knight suggested |
| 03:00.58 | MrTelephone | I was going to say 'and like update some tables or something' |
| 03:01.45 | MrTelephone | Not sure how you would get their passwords though.. you would have to store them in reversable format |
| 03:02.02 | MrTelephone | probably not the best security practise out there |
| 03:02.58 | PrestonL | Create a field called "Asterisk Password" in the AD thing? |
| 03:03.27 | Wiretap_Work | wo... jesus |
| 03:03.38 | Wiretap_Work | why so many people tryign to connect craptive directory and asterisk? |
| 03:04.01 | MrTelephone | Get out while you can. I rolled with it now I'm stuck working until midnight 20 times a month figuring things out |
| 03:04.17 | Wiretap_Work | lol |
| 03:04.27 | MrTelephone | I know a good electrical school you can switch careers |
| 03:04.32 | f2Knight | MrTelephone, jeffspeff, a suggestion for the REASON your using AD ... Don't use AD to add users directly. Instead create a simple VB script that does it. you can wrap it up in an HTA webpage if you need pretty, but the point being to capture the account creation first. That way you can add it to AD as normal, and you can use it to create a phone user... how... Most people will never change the phone user passwords so this is good en |
| 03:04.32 | f2Knight | ough.. on that note... |
| 03:04.45 | Wiretap_Work | if you want AD integration, perhaps CUCM or 3CrapXperiences is more suitable for you? |
| 03:05.13 | f2Knight | jeffspeff, why do you have to use the same password as there login? for what reason? |
| 03:05.22 | f2Knight | Wiretap_Work, I suggested that earlier. |
| 03:05.37 | Wiretap_Work | f2Knight: that'd be why I'm resuggestingm it :P |
| 03:05.54 | jeffspeff | it's not my decision... this is what my client wants, and their IT team is all about windows, and there's no convincing them otherwise |
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| 03:06.18 | f2Knight | Wiretap_Work, I think there is $$$ issue with 3CX that makes it prohibitive to most people trying to make a buck. |
| 03:06.42 | f2Knight | jeffspeff, you're not listen to what we have told you. |
| 03:06.57 | jeffspeff | they have an in-house it team that handles the simple things like password creation, user accounts, etc.... |
| 03:07.00 | MrTelephone | If windows is the only option then asterisk is not an option |
| 03:07.03 | Wiretap_Work | jeffspeff: bail then |
| 03:07.07 | Wiretap_Work | yep |
| 03:07.09 | f2Knight | option 1) use 3cx its windows based client pays more money they get gui and intergration done |
| 03:07.18 | Wiretap_Work | if windows is required, asterisk is out |
| 03:07.32 | jeffspeff | I have listened to what you've said, but I'm already balls deep in this. Already extended the AD schema and made many changes along the road to this solution |
| 03:07.37 | MrTelephone | It's like screening resumes.. no highschool no job |
| 03:07.53 | f2Knight | 2) write a vbscript to add users to AD and then add your to your sip.conf. No biggy they two do not need to know each other. |
| 03:08.11 | jeffspeff | ok, i'll give that a shot |
| 03:08.19 | jeffspeff | thanks for all the input |
| 03:08.30 | MrTelephone | That would be your best bet |
| 03:08.42 | MrTelephone | even if there was ldap support it could break in a future release or something |
| 03:08.45 | f2Knight | 3) keep bashing your head and trying to fit the square peg in the round hole.... or make the round hole bigger |
| 03:08.57 | MrTelephone | config file generation scripts would be easy to maintain and rock solid |
| 03:09.04 | f2Knight | jeffspeff, still has not said WHY the passwords have to match |
| 03:09.21 | MrTelephone | Because i setup an email server outside our organization and it was a bitch |
| 03:09.43 | MrTelephone | people can't remember 1 password, 2 causes people to go on stress leave |
| 03:10.03 | MrTelephone | 1 password for email and 1 for logins :( |
| 03:10.04 | jeffspeff | I'm really not sure myself, I think it's more for sip user creation and extensions than for passwords |
| 03:10.15 | jeffspeff | but passwords are part of the problem i'm having right now |
| 03:10.26 | f2Knight | you need to rethink this man |
| 03:10.33 | f2Knight | EVEN IF you had password sync |
| 03:10.39 | jeffspeff | I know |
| 03:10.40 | jeffspeff | lol |
| 03:10.46 | f2Knight | that would mean any time a user changed there password there phones would not work anymore |
| 03:10.54 | f2Knight | you would have to reprovision each phone every time |
| 03:11.10 | jeffspeff | true |
| 03:11.16 | MrTelephone | can you scrap the password part and just go free willy? |
| 03:11.24 | f2Knight | if the IT department has any knowledge they should know how to write a vb script file and if not they should be fired on the spot |
| 03:11.48 | f2Knight | are the phones pre provisioned? |
| 03:11.52 | MrTelephone | change sip port and go passwordless |
| 03:11.57 | f2Knight | will you be running a provisioning service as well? |
| 03:12.16 | f2Knight | MrTelephone, on a private lan.. okay untill you get a half smart user |
| 03:12.21 | MrTelephone | :P |
| 03:12.56 | MrTelephone | I was humouring myself.. |
| 03:13.05 | MrTelephone | I know how he feels though |
| 03:13.30 | MrTelephone | I spent 3 thousand on this new equipment and I'm trying to design a dialplan to use with it and I'm getting nowhere |
| 03:14.03 | f2Knight | MrTelephone, Been there doing that. :) |
| 03:14.45 | f2Knight | jeffspeff, I am typing you a private message with some options .. I think you MIGHT be best served by using a batch logon script if you care to do that then you can add users by AD |
| 03:14.59 | MrTelephone | I have to justify a $6000 dollar asterisk/polycom setup consisting of 12 phones too. I think you can buy a nortel system for $3000 can't you? |
| 03:16.40 | MrTelephone | The sad part is asterisk is 0$. Hopefully they get kickbacks from polycom/atlas and whatever else |
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| 03:17.15 | MrTelephone | It's almost embarrassing to spend thousands on equipment and then come in this channel for free advice. lol |
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| 03:20.43 | f2Knight | MrTelephone, nope no kickbacks. Digium sells T1 Cards and such. |
| 03:20.57 | f2Knight | MrTelephone, they also sell tech support. and a GUI |
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| 03:21.38 | f2Knight | Mr jeff didn't like my suggestions I am afraid |
| 03:21.56 | f2Knight | Sorry when someone is wasting there time they need to know it before they waste more. |
| 03:22.29 | f2Knight | MrTelephone, Nortel is out of business now I think |
| 03:22.51 | f2Knight | MrTelephone, I know they just sold a shit load of pattents. |
| 03:23.35 | f2Knight | MrTelephone, 6Gs for 12 phones... eh okay .. but i don;t like polycoms , besides that whats the problem your having? |
| 03:24.21 | MrTelephone | phones are only 160 a piece |
| 03:24.42 | MrTelephone | poe switch, adtran 924, and a small computer system |
| 03:24.45 | MrTelephone | install fee |
| 03:25.11 | MrTelephone | the conference phone is expensive though |
| 03:25.52 | *** join/#asterisk pinoyskull (vblando@112.205.114.152) |
| 03:25.54 | MrTelephone | well I have 2 asterisk servers and I want the clients to register to whatever endpoint via srv records |
| 03:26.05 | f2Knight | MrTelephone, yea never knew why the conf phones are so damn much |
| 03:26.28 | f2Knight | okay I am following you sofar keep going |
| 03:26.41 | MrTelephone | I created a test iax2 trunk between them. Both boxes have the same sip accounts. So it tries to dial the local box, then the next box, then voicemail |
| 03:27.17 | MrTelephone | i have an atlas 550 which oversubscribes my single t1. Each asterisk box will eventually have a t1 |
| 03:27.51 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 03:28.09 | MrTelephone | It should work in theory but I have some customers using endpoint devices that support t38. Is the iax2 going to make faxing more unreliable then it already is? |
| 03:30.36 | MrTelephone | what if you have a few extensions that dial multiple phones and they are seperated? would you just put a line in like this ? exten => 555555,1,Dial(IAX2/outgoing-iax/601&IAX2/outgoing-iax/602&SIP/601&SIP/602) and see who picks up first? |
| 03:30.51 | tyman | If I wanted a scheduled update of my astdb (for instance), is there a better way than putting an 'asterisk -rx <command string>' in user asterisk's crontab? |
| 03:31.33 | WIMPy | tyman: You could do it via AMI. |
| 03:31.44 | f2Knight | I have not worked with any t.38 devices but from my understanding it runs over a TCP and on its own port which is not SIP or IAX its t38. to another t38 end point. its a tcp datapacket that is reassembled on the other side ... hense the need for t.38 devices on both sides |
| 03:32.40 | tyman | WIMPy: I need to read up on thatâ¦AMI and realtimeâ¦I guess even a simple AGI script would work for this |
| 03:32.55 | MrTelephone | I think i enabled the redundancy option so it sends 2 udp packets for every 1.. I thought that's how it worked |
| 03:32.59 | WIMPy | No. |
| 03:33.09 | f2Knight | MrTelephone, your dial statement would run multiple dial attempts , and yes whoever picks up first gets the call all others are dropped |
| 03:33.19 | WIMPy | AGIs are calld from the dialplan i.e. on an existing call. |
| 03:33.42 | MrTelephone | So for the fax only endpoints I would like to use the Hangup(42) clause or something and hopefully the atlas550 goes to the secondary trunk with the call instead of indicating a busy tone |
| 03:33.54 | tyman | WIMPy: can't you schedule a cron to copy an AGI into the asterisk spool dir? |
| 03:33.55 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 03:34.33 | WIMPy | No, you could create an extension calling an AGI and place a call to that. |
| 03:35.12 | f2Knight | MrTelephone, the better question is how many t.38 fax station are out in the field, and what are they connecting to internally |
| 03:35.29 | MrTelephone | I guess another option would be to loop the call back out the t1 if the user doesn't exist but then your chewing up like 3 timeslots |
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| 03:36.06 | MrTelephone | We run a cable system. the t38 endpoints are arris modems. And I paid for t38 termination (Fax for Asterisk). |
| 03:36.10 | Scorp1us | I am having a heck of a time setting my callerId outboud text. |
| 03:36.21 | f2Knight | MrTelephone, as i recall asterisk is a gateway for t38 just passing it along and does nothing by its self. (though you can use things like soundsp and such ) |
| 03:36.21 | Scorp1us | what do I need to do to set it properly? |
| 03:36.56 | f2Knight | Scorp1us, in sip.conf callerid= "My Test <5551212>" |
| 03:37.06 | Scorp1us | thanks f2Knight |
| 03:37.09 | WIMPy | Scorp1us: Set the callerid of the originating device or use Set(CALLERID(name)=xxx) |
| 03:37.26 | f2Knight | Scorp1us, in dialplan Set(CALLERID(name)="My Test") |
| 03:37.36 | MrTelephone | the new fax for asterisk will terminate the t38. I think it used to only have a pass-through option before |
| 03:37.40 | f2Knight | Scorp1us, in dialplan Set(CALLERID(name)=5551212) |
| 03:37.59 | *** join/#asterisk godmachine-x6 (~godmachin@h18.171.140.67.dynamic.ip.windstream.net) |
| 03:38.00 | f2Knight | MrTelephone, Not sure.. |
| 03:38.07 | Scorp1us | I'm using outbound call files |
| 03:38.13 | f2Knight | MrTelephone, but you still have not said what your dialplan problem is. |
| 03:38.28 | Scorp1us | but CallerID: "Name" <1234> is not working |
| 03:38.32 | f2Knight | Scorp1us, you are welcome , keep in mind your provider might over right it again |
| 03:38.55 | f2Knight | Scorp1us, in dialplan Set(CALLERID(name)=5551212) |
| 03:39.00 | f2Knight | Scorp1us, in sip.conf callerid= "My Test <5551212>" |
| 03:39.07 | f2Knight | Scorp1us, notice where the " are |
| 03:39.12 | Scorp1us | oooh |
| 03:39.16 | MrTelephone | Well I couldn't figure out earlier how to get asterisk to check where the client is actually registered |
| 03:39.18 | Scorp1us | I have to do both? |
| 03:39.31 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 03:39.36 | f2Knight | Scorp1us, depends on how you're setting it |
| 03:39.40 | MrTelephone | If the Dial command fails it goes to the next priority |
| 03:40.22 | MrTelephone | Maybe I should just use heartbeat? |
| 03:40.30 | f2Knight | MrTelephone, you have entered the world of confusion ... here is the EASY way to do it... Asterisk uses sipfriends and sipregs in MySQL |
| 03:40.52 | Scorp1us | ok, will try thanks |
| 03:40.57 | f2Knight | that database is on a 3rd system.. |
| 03:41.10 | MrTelephone | I already do that for voicemail |
| 03:41.17 | f2Knight | that both can access.. now both servers get the sip account info and the reg info from the database. |
| 03:41.28 | MrTelephone | I was afraid mysql going down for an hour.. Will asterisk store the information until it is restarted? |
| 03:41.47 | f2Knight | when _srv records move a call to a different server the DB is accessable from both machines |
| 03:42.22 | f2Knight | if you are worried about mysql going down then you need to look at your equipment |
| 03:42.29 | f2Knight | mysql if very stable more so then asterisk. |
| 03:42.58 | f2Knight | but if your needing failover for mysql, you can set up a mysql cluster. that way if a node fails others pick up the slack. |
| 03:43.15 | f2Knight | You are talking lots of different disaplines to get things running. |
| 03:43.18 | MrTelephone | I can hadnle voicemail going offline for a few minutes but if the realtime goes down then the whole thing stops working |
| 03:43.27 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 03:43.30 | f2Knight | are both servers in same location ? |
| 03:43.33 | MrTelephone | yeah |
| 03:43.39 | f2Knight | at clients location? |
| 03:43.51 | MrTelephone | at our operations center |
| 03:43.55 | f2Knight | are you trying to do failover or load balancing |
| 03:44.02 | MrTelephone | failover |
| 03:44.11 | f2Knight | fail over is easier |
| 03:44.22 | f2Knight | you do understand the differnce right? |
| 03:44.26 | MrTelephone | yes |
| 03:44.37 | f2Knight | then you know you do not use svr records with failover |
| 03:45.01 | MrTelephone | I don't like the idea of heartbeat because of the arp timeouts but maybe that is a better solution |
| 03:45.47 | MrTelephone | Our operation is getting big enough where I might consider a mysql cluster |
| 03:45.59 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:46.17 | f2Knight | failover is simple : build a machine. get it setup .. clone it. Setup timed intervals for DB dumps and for dialplan copying .. *rsync is your friend* set heartbeat up to work on its OWN nic AND a serial port |
| 03:47.00 | f2Knight | I have an fundimential issue with failovers.. that is you run 2 machines all the time and are only ever using one. to me its a double expense and waste of a machine. |
| 03:47.11 | MrTelephone | I agree |
| 03:47.19 | f2Knight | Ohh and your t1.. better get rid of your adtran |
| 03:47.26 | MrTelephone | how come |
| 03:48.32 | f2Knight | your adtran has 1 interface to 1 machine right? |
| 03:48.40 | MrTelephone | it has 5 t1 ports |
| 03:48.44 | MrTelephone | 1 to the telco |
| 03:48.47 | MrTelephone | 1 to a channel bank |
| 03:48.54 | MrTelephone | 2 to asterisk |
| 03:48.59 | MrTelephone | there is supposed to be 2 to the second asterisk box |
| 03:49.12 | MrTelephone | I need to buy another quad t1 card |
| 03:49.31 | MrTelephone | the idea is to not have any switching of cables in the event of a failover |
| 03:49.36 | f2Knight | so long as the adtran can support multiple machines behind it.. I only know of one device off hand that does |
| 03:49.52 | f2Knight | that is redphones |
| 03:49.54 | MrTelephone | This thing is pretty kick ass |
| 03:50.19 | MrTelephone | I spent a few hours onthe phone with adtran guys to figure out what equipment I needed |
| 03:50.19 | f2Knight | http://goo.gl/Y3TXk |
| 03:50.24 | f2Knight | okay this is what I use |
| 03:50.38 | f2Knight | I have 4 asterisk boxes that feed from it |
| 03:51.21 | f2Knight | the point here is that when the boxes go down the t1 device needs to know not to send trafic to it. |
| 03:51.44 | MrTelephone | That looks like a good piece of hardware |
| 03:51.53 | f2Knight | I love it. |
| 03:52.08 | f2Knight | when a node goes down I don't worry |
| 03:52.43 | MrTelephone | Looks rugged |
| 03:52.47 | f2Knight | Anyhow use your HB to monitor on both its own NIC and a serial port... your servers DO have RS232 right |
| 03:53.05 | f2Knight | MrTelephone, it is made of metal not plastic like alot of adtrans |
| 03:53.11 | MrTelephone | Yeah they are pwowerful xeon servers |
| 03:53.38 | MrTelephone | the adtran device routes calls because on did/ani and source port ID's |
| 03:53.45 | MrTelephone | i dont' even need asterisk to run the channel bank |
| 03:54.03 | f2Knight | okay run HA on both interfaces... this way if one gets a failure it looks at the other to make sure.. and you may want to customize a script to assure dial functions I do a random call every 3 min |
| 03:54.55 | MrTelephone | what tests do you do to monitor the serial port? |
| 03:54.58 | f2Knight | So what happens when you get more numbers?? oh yea you have to go program your asterisk servers AND your adtran. thats a good solution. |
| 03:55.20 | MrTelephone | i just route calls based on the source port they come from. wildcard for the phone numbers |
| 03:56.10 | MrTelephone | but I still want all calls going through asterisk for billing so you have to tie up 2 timeslots on the asterisk box if you want to make a call to the channel bank |
| 03:56.19 | f2Knight | I just test to make sure it can send and receive data. kind of to make sure its powered on is all its good for. In case of a busy load and a single failure if the system is on it allows me to check the resource on the nic again. |
| 03:56.20 | MrTelephone | the main reason I got it is so I could start reselling t1's |
| 03:56.54 | MrTelephone | So you have a script connecting to the serial port from the other machine? |
| 03:57.08 | f2Knight | yes |
| 03:57.38 | f2Knight | reselling t1's? no you must get that from a Clec or telco |
| 03:57.50 | MrTelephone | just over subscribing the t1 i have already |
| 03:57.56 | f2Knight | you mean selling DID's? okay |
| 03:58.22 | MrTelephone | there is an office that needs to be hooked up 1km away and wanted to use TDM circuits instead of voip so their fax machines would work |
| 03:58.39 | MrTelephone | yeah, selling dids |
| 03:59.24 | MrTelephone | we just service a small area in the middle of nowhere. not getting rich off it but the more people we can get using what we have makes it cheaper for us |
| 03:59.27 | f2Knight | I still recommend people have a single fax line in house from the POTS. ( I route faxes over it and 911 calls) |
| 04:00.01 | f2Knight | be careful reselling DID's you need to do e911 services. they cost you about 50 bucks a pop to setup per did. |
| 04:00.15 | MrTelephone | On the local network faxing works ok. Faxing works perfect on our channel bank connected to the same t1 card as the telco. |
| 04:00.34 | MrTelephone | don't even get me started on 911 |
| 04:00.37 | f2Knight | you want to be sure that if someone picks up a fax and dials 911 because they fell next to the fax that you don't get sued and lose everything |
| 04:00.55 | MrTelephone | they don't offer 911 here because the homes are not properly addressed |
| 04:01.06 | f2Knight | where are you ? |
| 04:01.11 | MrTelephone | in the bush |
| 04:01.28 | f2Knight | down under? |
| 04:01.36 | MrTelephone | northern ontario |
| 04:01.44 | f2Knight | eh |
| 04:02.10 | f2Knight | that was a joke. |
| 04:02.11 | MrTelephone | there should be database that 911 systems have where you can update address info |
| 04:02.25 | f2Knight | there is not though its a closed racket |
| 04:02.26 | MrTelephone | that kind of shit gets me steaming |
| 04:02.35 | f2Knight | if you provide the numbers you have to provide 911 |
| 04:02.38 | MrTelephone | it's like the ehealth scandel in canada |
| 04:03.10 | MrTelephone | 911 works but we have to give notices saying they have to give them an address |
| 04:03.19 | f2Knight | if you get your DID's from a sip provider they usually can provide you it for about a $1 a month |
| 04:03.23 | MrTelephone | what if they can't talk? what if someone moves the adaptor to another household? |
| 04:03.49 | MrTelephone | use your f'in cellphone if your dying |
| 04:03.51 | MrTelephone | haha |
| 04:04.13 | f2Knight | That is part of the issue. and why voip providers disclose that if you move the device from your register address you have to notify them |
| 04:04.35 | MrTelephone | where is the address informatioN? |
| 04:04.39 | f2Knight | no i don't think its a good solution but its the way it is. |
| 04:05.03 | f2Knight | the addresses to numbers are held by the telco.. mabell |
| 04:05.13 | f2Knight | they do not give access out to it. |
| 04:05.23 | f2Knight | SO you have to PAY to get access to update records |
| 04:05.39 | f2Knight | oh and only SELECT companies can do so ... its a brokerage thing. |
| 04:05.57 | f2Knight | so you pay someone else who pays to have access who updates it for your local area |
| 04:06.07 | f2Knight | because each area has there own database |
| 04:07.23 | MrTelephone | anything more than a dollar per month per did is too much |
| 04:07.54 | MrTelephone | I disconnected a person yesterday for arrears and their boyfriend was abusing her |
| 04:08.03 | MrTelephone | we allow 911 calls but she was too scared to call |
| 04:08.18 | MrTelephone | go figure |
| 04:09.03 | f2Knight | so you turned her phone service off? |
| 04:09.17 | MrTelephone | i put them in a context where it blocks everything except for emergencies |
| 04:09.25 | f2Knight | ohh okay |
| 04:09.51 | MrTelephone | she tried to call the local police shop but it didn't work. I was wondering why she didn't call 911? |
| 04:09.52 | f2Knight | thats actually a good idea... I should add that for disconnected service |
| 04:09.53 | MrTelephone | it works |
| 04:10.04 | MrTelephone | I thought it was another one of those legal requirements |
| 04:10.05 | f2Knight | non-emergency |
| 04:10.14 | f2Knight | umm.. not sure. |
| 04:10.33 | MrTelephone | oh yeah and let them phone our office to ask why their phone doesn't work |
| 04:10.42 | f2Knight | I mean if you don't have service/ you don't have service. Not like the POTS which have hard wired to your house. |
| 04:10.47 | MrTelephone | and if you phone another number my voice comes up saying you need to make a payment |
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| 04:11.02 | MrTelephone | i notice now that Bell canada leaves the lines active when your disconnected |
| 04:11.17 | MrTelephone | when you try and make a call it forwards you to their sales center |
| 04:11.28 | f2Knight | When I turn off a client they get a message that plays to them saying there there account was disconnected and to stay on the line to connect with an operator |
| 04:11.38 | MrTelephone | lol |
| 04:11.43 | MrTelephone | Mine isn't that fancy |
| 04:11.53 | MrTelephone | do you have 24/7 operator? |
| 04:12.13 | f2Knight | no it goes in to a queue |
| 04:12.26 | f2Knight | if its after hours well .. too bad lol |
| 04:12.28 | MrTelephone | due to heavy call volume.... |
| 04:12.38 | MrTelephone | meanwhile its just you and you are sleeping |
| 04:12.40 | MrTelephone | hahah |
| 04:13.25 | f2Knight | no i just tell them its after hours if they need immediate service they dial 900 and it re routes to my cell |
| 04:13.38 | MrTelephone | nice |
| 04:13.42 | f2Knight | in 5 years i have had 1 call |
| 04:13.59 | MrTelephone | if you set things up right the first time your golden |
| 04:14.14 | MrTelephone | I had a bad t1 card that would randomly crash my server.. |
| 04:14.20 | f2Knight | most people take the disconnect message and just pay on the phone with a credit card.. (capture data send it to authorize.net and update my DB) |
| 04:14.22 | MrTelephone | when I first started this business |
| 04:14.37 | f2Knight | MrTelephone, thats why I like external setups :) |
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| 04:15.21 | f2Knight | I am learning how to setup OpenSER now so I can use the other 2 asterisk boxes sitting in stand by all day to handle call load.. not that I need to distrubute but I rather have it used then not. |
| 04:15.25 | MrTelephone | I have to get out of here though. I told this guy I would drop off a laptop for him at midnight and it's 15 after |
| 04:15.38 | MrTelephone | I tried that once but way too difficult |
| 04:15.41 | f2Knight | but right now I am building a SexChat server for a client |
| 04:15.52 | MrTelephone | I didn't have time to work out the logic for voicemail and nat traversal |
| 04:16.04 | f2Knight | night MrTelephone |
| 04:16.11 | *** part/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 04:16.13 | MrTelephone | how many free minutes do you get for that sexchat job? |
| 04:16.36 | MrTelephone | Asterisk for phone switchs and operser for sip registry would be ideal |
| 04:17.13 | MrTelephone | Thanks for the insight on that.. I think failover is the way to go right now. I don't think my server is nearly overloaded |
| 04:17.58 | f2Knight | MrTelephone, 25% of paid calls |
| 04:18.30 | MrTelephone | we'll talk later sometime |
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| 04:18.40 | f2Knight | MrTelephone, @ 1.99 a min and an estimated 1000 min a day between 5 agents to start I am hoping for a large payday |
| 04:18.59 | f2Knight | MrTelephone, but having issues captureing billing info.. arg!!!!!!!!! |
| 04:19.01 | MrTelephone | I still can't beleive people pay for htat |
| 04:19.17 | MrTelephone | lol |
| 04:19.32 | f2Knight | MrTelephone, me either.. but this guys girl friend works for a company and does .40 a min or 24 an hour |
| 04:20.41 | f2Knight | MrTelephone, so yea. I want to get it working if he promotes it even a little, we should have at least 10 callers a day , at 10 min each. thats 200 dollars alone. |
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| 05:01.35 | PrestonL | I need a provider that will bill in 30 second increments at $0.04 cents / min |
| 05:01.37 | PrestonL | ermm |
| 05:01.42 | PrestonL | $0.004 cents / min |
| 05:01.54 | PrestonL | waitt |
| 05:02.03 | PrestonL | $0.0004 dollars / min |
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| 05:02.44 | PrestonL | I run a notification system that does about 200-400 calls / day that are between 30 seconds and 2 minutes a piece. |
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| 05:07.21 | ChannelZ | sharts |
| 05:07.29 | PrestonL | nice |
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| 06:50.43 | Bipul | p3nguin: |
| 06:50.48 | Bipul | p3nguin: ping |
| 06:53.11 | ChannelZ | bling bling |
| 06:54.10 | irroot | morning folks |
| 06:54.22 | kleszcz | morning |
| 07:15.43 | Cobadol | Hello. I got new information about a bug, that I asked here yesterday. |
| 07:18.17 | irroot | cool Cobadol |
| 07:18.17 | Cobadol | I have installed asterisknow. Later change pbx.c and compile asterisk on virtual machine. Then copy "asterisk" to AsteriskNOW real machine and its not work well. That I need to do? This is asterisk output: http://pastebin.com/raw.php?i=fXiTw35U |
| 07:18.59 | Cobadol | irroot: Hello) |
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| 07:21.09 | irroot | what the problem |
| 07:21.37 | ChannelZ | Cobadol: Your paste doesn't really show anything |
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| 07:28.24 | Cobadol | yes. asterisk loads nothing. =( that's bug. some kind of incompatibility, i guess, but have no ideas to resolve it (brainstorming now |
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| 07:46.30 | ChannelZ | Did you build the whole of asterisk the first time yourself? |
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| 08:04.18 | nico | hi |
| 08:04.34 | irroot | lo |
| 08:04.48 | nico | was checking the docs and wanted to be sure about something in dialplan vs ael |
| 08:05.05 | nico | gotoiftime supports adding timezone |
| 08:05.34 | nico | iftime in ael is supposed to but does not validate syntax when TZ is specified |
| 08:05.41 | nico | am I missing something ? |
| 08:05.58 | nico | using https://wiki.asterisk.org/wiki/display/AST/AEL+Conditionals as reference |
| 08:06.15 | nico | exten = 888,n,GotoIfTime(09:00-10:00,mon-fri,*,*,Europe/Paris?888,test2) <= valid |
| 08:06.31 | nico | ifTime(09:00-10:00|mon-fri|*|*|${timezone}) <= invalid |
| 08:07.08 | nico | timezone is set, of course |
| 08:08.33 | kaldemar | "AEL supports if and switch statements, like AEL, but adds ifTime" |
| 08:08.45 | nico | yup |
| 08:08.50 | kaldemar | AEL is like AEL but it is not AEL. |
| 08:09.13 | nico | AEL introduces the ifTime keyword/statement, which works just like the if() statement, but the expression is a time value, exactly like that used by the application GotoIfTime(). See Asterisk cmd GotoIfTime |
| 08:09.28 | kaldemar | thats even better than the * marks on the channel variables page. wiki says that you cannot change caller id in dialplan for example... |
| 08:10.04 | kaldemar | that probably should say "AEL2 supports if and switch statements, like AEL, but adds ifTime". |
| 08:10.26 | nico | using 1.6, I run AEL2 right ? |
| 08:10.35 | Cobadol | ChannelZ: Its installed from image AsteriskNOW on real computer. But I build only "asterisk" itself on virtual machine. Hardware mismatch? hmm... |
| 08:10.35 | Cobadol | Moreover on virtual AsteriskNOW builded "asterisk" module works fine. 90% - hardware mismatch. Ok. Thanks. I will install AsteriskNOW on other real computer and will compile asterisk on it and then inject compiled to target machine. That's that I will try. |
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| 08:13.44 | kaldemar | nico: the wiki example only has four arguments to ifTime. you have the timezone as a fifth. |
| 08:14.19 | nico | kaldemar: yes, but it also says it support the gotoiftime time spec |
| 08:14.23 | kaldemar | nico: pipes instead of commas might also be the problem. |
| 08:14.31 | nico | (tried both) |
| 08:14.47 | nico | so it should accept timezone right ? |
| 08:14.49 | kaldemar | what does dialplan show say about it? |
| 08:15.04 | nico | does not compile, so does not load |
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| 08:16.08 | nico | it creates a gotoiftime when it loads... I guess the iftime defintion is full of crap then |
| 08:16.09 | kaldemar | i'd try it without the timezone. despite the wiki says that it behaves exactly like GotoIfTime. |
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| 08:37.59 | BuenGenio | what does port 2000 do? |
| 08:38.09 | BuenGenio | is there a web interface to manage * ? |
| 08:38.59 | nico | kaldemar: problem is that I need the timezone :) multi offices setup |
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| 08:45.55 | kaldemar | nico: well, i guess you need to use STRFTIME to get the time in the desired zone. |
| 08:46.48 | nico | kaldemar: seems like a nice workaround, thx |
| 08:46.56 | nico | gonna try this |
| 08:49.23 | BuenGenio | any tutorials for setting up a video conference server with * ? |
| 08:50.56 | kaldemar | BuenGenio: asterisk doesn't support video conferencing yet. a third-party app called app_konference does, iirc. |
| 08:51.44 | BuenGenio | is that an addon? |
| 08:52.51 | kaldemar | no. |
| 08:53.56 | kaldemar | not an addon as in "asterisk-addons", but a third-party dialplan application that is not distributed with asterisk. |
| 08:54.43 | BuenGenio | i see |
| 08:54.46 | BuenGenio | dialplan |
| 08:54.48 | BuenGenio | cool |
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| 09:13.54 | *** join/#asterisk b0rf (~b0rf@ip98-164-100-237.no.no.cox.net) |
| 09:14.07 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 09:17.12 | *** join/#asterisk Lawbringer (~Miranda@cpc2-rdng5-0-0-cust632.15-3.cable.virginmedia.com) |
| 09:18.17 | *** join/#asterisk b0rf (~b0rf@ip98-164-100-237.no.no.cox.net) |
| 09:18.55 | Lawbringer | Morning People, Just trying to set up asterisk but got an issue with routing outgoing calls, I've logged in to my provider direct with xlite and can make outgoing calls however when i try via asterisk i get a forbidden message |
| 09:48.38 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 09:48.49 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 09:52.42 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 09:58.52 | dym | Lawbringer: show us config files. |
| 09:58.59 | dym | sip.conf / extensions.conf |
| 09:59.01 | dym | (nopaste) |
| 10:00.29 | Lawbringer | dym, pastebin coming up |
| 10:04.14 | Lawbringer | dym http://paste2.org/p/1532840 |
| 10:06.14 | *** join/#asterisk johnnyasterisk (~johnnyast@62.231.40.243) |
| 10:06.32 | kaldemar | "allow=glaw" ? |
| 10:06.59 | dym | (: |
| 10:07.09 | dym | also, what does "sip show peers" say? |
| 10:07.46 | Lawbringer | it's 1.8 btw |
| 10:07.53 | Lawbringer | so show sip peers right? |
| 10:08.11 | kaldemar | providerPROVIDER in extensions.conf does not match with anything in sip.conf. |
| 10:08.46 | dym | true - names would have to match for calls to route properly |
| 10:09.12 | Lawbringer | dym line 12 |
| 10:09.27 | Lawbringer | i wanted to mask my provider but failed :( |
| 10:09.28 | dym | but extensions.conf |
| 10:09.38 | dym | as kaldemar pointed out |
| 10:10.05 | Lawbringer | hold up which line? |
| 10:10.07 | kaldemar | Lawbringer: change username to defaultuser in sip.conf for the provider. |
| 10:10.28 | dym | Also: why set a context in "general" on sip.conf? |
| 10:10.35 | dym | pretty useless there. |
| 10:10.50 | Lawbringer | dym folling a how too |
| 10:10.55 | Lawbringer | following* |
| 10:11.09 | Lawbringer | just trying to remember what everything does |
| 10:11.16 | dym | yeah, remove it. |
| 10:11.19 | Lawbringer | used asterisk about 6 odd years ago |
| 10:11.24 | kaldemar | change type to peer and secret to remotesecret. you may also need fromuser set. |
| 10:11.50 | dym | in extensions.conf you dial Dial(SIP/${EXTEN}@providername) |
| 10:12.01 | dym | where providername is the name you named it on sip.conf |
| 10:12.16 | dym | in your case provideraql |
| 10:12.49 | Lawbringer | we talking about the reference in 117 linking too line 12? |
| 10:13.06 | dym | no |
| 10:13.07 | dym | 116 |
| 10:13.14 | dym | 117 is oddly formed |
| 10:13.15 | Lawbringer | 116 is commented out |
| 10:13.26 | dym | well, i dont think 117 is right |
| 10:13.38 | dym | you dial SIP/1234@provider |
| 10:13.39 | Lawbringer | 117 is what the howtoo said 116 is what i tried |
| 10:13.46 | dym | use 116 |
| 10:13.49 | dym | and fix it |
| 10:13.51 | dym | :) |
| 10:14.17 | Lawbringer | ok from the top. removed line 2-3? |
| 10:14.30 | dym | 3 |
| 10:14.58 | dym | kaldemar: feel free to way in on anything |
| 10:15.05 | Lawbringer | i haven't played with the codex but are they ok? |
| 10:15.14 | kaldemar | i'd use SIP/devicename/extension in a dial. |
| 10:15.39 | dym | kaldemar: as in SIP/provider/extension? |
| 10:15.46 | *** part/#asterisk johnnyasterisk (~johnnyast@62.231.40.243) |
| 10:15.47 | kaldemar | Lawbringer: you should first disallow all codecs with disallow=all and then allow the codecs you want in the order of preference. |
| 10:15.54 | kaldemar | dym: yes. |
| 10:15.57 | dym | mhhh |
| 10:16.03 | dym | i use it the other way around |
| 10:16.19 | Lawbringer | kaldemar, thats fine but i'm just testing so i'll fiddle with them later when i've got routing right |
| 10:16.20 | kaldemar | so that asterisk won't mistakenly think that is is dialing username@domain. |
| 10:17.49 | dym | Lawbringer: okay, so both Dialout options are valid. kaldemar's version maybe a little more "clean looking", so i'd go for that. |
| 10:18.07 | dym | as in Dial(SIP/providername/${EXTEN}) |
| 10:19.05 | kaldemar | there are fine examples in the current sip.conf.sample. |
| 10:19.35 | dym | just make sure providername matches your type=peer provider section on sip.conf |
| 10:19.44 | kaldemar | http://svn.digium.com/svn/asterisk/tags/1.8.5.0/configs/sip.conf.sample |
| 10:20.37 | Lawbringer | So just "exten => _9XXXX.,Dial(SIP/provideraql/${EXTEN}) should be ok? |
| 10:21.10 | dym | well |
| 10:21.15 | dym | if you want the leading 9 to be dialed |
| 10:21.17 | kaldemar | btw, did you get the forbidden message in your phone or in asterisk? |
| 10:21.18 | dym | which you cut before |
| 10:21.30 | Lawbringer | asterisk |
| 10:21.37 | Lawbringer | phone just says called failed |
| 10:21.42 | dym | ${EXTEN:1} means dont dial the first digit. |
| 10:21.52 | Lawbringer | ahh that could well be why |
| 10:22.10 | dym | so if you dialed 91234 it'd dial 1234 in the sip trunk |
| 10:24.18 | dym | Lawbringer: but you are sure your providers were all connected at that stage? |
| 10:24.22 | dym | check with sip show peers on CLI |
| 10:24.48 | Lawbringer | provider is online phone is offline |
| 10:24.54 | Lawbringer | probably have to wait to resign |
| 10:25.27 | kaldemar | or sip show registry, it will show the state of the outgoing registration. sip show peers doesn't really show any useful information if the peer is static and qualify is not in use. |
| 10:25.49 | Lawbringer | state registered |
| 10:26.13 | dym | well, but in peers you also see the unregistered devices. |
| 10:26.15 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 10:26.17 | dym | So not completely useless |
| 10:26.23 | Lawbringer | still can't dial out but not getting any real errors in asterisk |
| 10:26.32 | dym | Lawbringer: thought your phones were off |
| 10:26.43 | dym | 12:24:48 < Lawbringer> provider is online phone is offline |
| 10:26.53 | Lawbringer | i sign the phone back on |
| 10:26.57 | dym | also: core set debug 9 |
| 10:27.09 | dym | sip set debug on |
| 10:28.54 | Lawbringer | i think it might be a NAT issue |
| 10:29.24 | dym | what makes you say that? |
| 10:29.33 | Lawbringer | dym trying to find the line again |
| 10:29.42 | Lawbringer | it's spewing out quite a bit now |
| 10:29.59 | dym | well, quit the cli and scroll? :D |
| 10:30.02 | dym | or check the logs? : |
| 10:31.25 | Lawbringer | dym problem is i'm not sure what i'm looking for |
| 10:31.51 | dym | Neither do I (: So good luck |
| 10:33.41 | Lawbringer | == Spawn extension (myphones, 9"XXX", 3) exited non-zero on 'SIP/100-00000037' XX my my POTS line i'm testing too |
| 10:37.27 | kaldemar | it's not a NAT issue. |
| 10:38.31 | Lawbringer | ok i had an error in the dial plan i'm back to forbidden |
| 10:39.16 | dym | show us what happens on the dial |
| 10:39.38 | Lawbringer | whats the paste limit in here? |
| 10:39.50 | *** join/#asterisk elfelvin (~elfelvin@87-194-69-88.bethere.co.uk) |
| 10:40.18 | dym | just nopaste it |
| 10:40.20 | dym | lazy git :D |
| 10:40.48 | dym | If you have to ask that question, you most probably will exceed it (: |
| 10:41.14 | Lawbringer | http://paste2.org/p/1532896 |
| 10:41.22 | Lawbringer | well it's only a couple of lines |
| 10:41.31 | Lawbringer | so wondering for future reference |
| 10:41.37 | dym | well |
| 10:41.37 | dym | sip.PROVIDER.com |
| 10:41.47 | Lawbringer | i'm masked that |
| 10:41.49 | dym | does the sip.conf section of your provider say [sip.PROVIDER.com] ? |
| 10:41.57 | Lawbringer | no |
| 10:42.01 | dym | there you go. |
| 10:42.04 | dym | fixit. |
| 10:43.00 | Lawbringer | dym, I thought that was extracted from line 16 or 24 on the previous paste |
| 10:43.14 | Lawbringer | http://paste2.org/p/1532840 |
| 10:43.35 | dym | negative |
| 10:43.43 | dym | afaik |
| 10:43.49 | Lawbringer | so where is it getting that information from |
| 10:43.55 | dym | thats what the contexts are for |
| 10:44.00 | dym | context name |
| 10:44.12 | dym | as in [provideraql] |
| 10:44.26 | dym | extensions.conf Dial(... Line is wrong |
| 10:44.52 | dym | exten => _XXXX.,n,Dial(SIP/provideraql/${EXTEN:1},60) |
| 10:44.53 | Lawbringer | but i have the context there which points to a context in sip.conf which has the URL |
| 10:45.00 | dym | its the context name |
| 10:45.02 | dym | not the url |
| 10:45.03 | dym | that matters |
| 10:45.06 | dym | fix as shown above |
| 10:45.11 | dym | then retry |
| 10:45.12 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 10:45.18 | Lawbringer | thats what it says |
| 10:45.23 | dym | cant be |
| 10:45.27 | Lawbringer | it is |
| 10:45.29 | dym | did you reload? |
| 10:45.32 | dym | core reload? |
| 10:45.33 | Lawbringer | yes several times |
| 10:45.36 | kaldemar | "-- SIP/providerPROVIDER-0000003f" suggests that the used dial command had SIP/providerPROVIDER in it. so in sip.conf there should be [providerPROVIDER]. |
| 10:45.52 | Lawbringer | which it does |
| 10:45.53 | dym | agreed. |
| 10:46.14 | Lawbringer | ok the provider is aql it's not worth masking it :) |
| 10:46.19 | Lawbringer | you knwo ti anyway |
| 10:46.27 | kaldemar | so that's not an issue to fix. now, enable sip debug ans pastebin what you see in CLI. |
| 10:49.06 | Lawbringer | need more history |
| 10:50.02 | kaldemar | use script or tee for example. |
| 10:50.12 | kaldemar | you'll get the output directly to a file. |
| 10:51.03 | kaldemar | asterisk -vvvr | tee asterisk_output.txt |
| 10:51.50 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 10:52.50 | Lawbringer | there is a cunk of ip's i'll need to mask or will it be safe? |
| 10:53.15 | dym | what would we want with your ips? |
| 10:53.36 | Lawbringer | dym i'm not sure you want them but one of the people may be a logger |
| 10:54.37 | dym | then mask them? |
| 10:55.13 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 10:59.31 | *** join/#asterisk m_tadeu (~quassel@89.181.90.166) |
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| 11:12.20 | *** part/#asterisk johnnyasterisk (~johnnyast@62.231.40.243) |
| 11:13.42 | *** join/#asterisk Law (~Miranda@cpc2-rdng5-0-0-cust632.15-3.cable.virginmedia.com) |
| 11:14.29 | Law | sorry dym had a crash |
| 11:19.15 | *** join/#asterisk sourcode (~code@ppp-58-11-83-165.revip2.asianet.co.th) |
| 11:29.18 | Law | http://paste2.org/p/1532959 thats a lot of times it' repeats the number |
| 11:29.55 | Law | couldn't be bothered with IP's after all that dym and kaldemar |
| 11:30.52 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 11:35.05 | kaldemar | Law: what does your peer definition in sip.conf look like now? |
| 11:35.58 | *** join/#asterisk wasanzy_ (~emmanuel@196.201.43.55) |
| 11:36.13 | Law | which bit is the peer definition? |
| 11:36.34 | kaldemar | [providerPROVIDER] |
| 11:37.20 | wasanzy_ | <kaldemar>: how are you? |
| 11:38.01 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:39.10 | Law | kaldemar : http://paste2.org/p/1532981 |
| 11:39.34 | Law | ohh i didn't change it to peer |
| 11:39.38 | Law | it was there right |
| 11:40.25 | Law | not that it makes a differnce |
| 11:40.56 | kaldemar | Law: change it to peer and add fromuser parameter for it. its value is supposed to be the username the provider knows you by. |
| 11:41.53 | kaldemar | and change username parameter to defaultuser. |
| 11:42.38 | Law | nope |
| 11:43.39 | Law | kaldemar : http://paste2.org/p/1532988 now |
| 11:44.35 | kaldemar | not username=defaultuser but defaultuser=yourusername |
| 11:44.41 | Law | ahh |
| 11:45.29 | Law | nope |
| 11:46.04 | Law | defaultuser and fromuser both the same right? |
| 11:49.06 | kaldemar | yes. |
| 11:49.44 | Law | still no dice |
| 11:56.02 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 11:57.45 | *** join/#asterisk johnnyasterisk (~johnnyast@62.231.40.243) |
| 12:03.52 | *** join/#asterisk moos3_ (~rgenthner@cpe-74-75-159-253.maine.res.rr.com) |
| 12:04.14 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 12:05.45 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 12:21.52 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:31.33 | wasanzy_ | hi how can I download the "Asterisk The Definitive Guid 3rd Edition? |
| 12:32.54 | dym | ?book? |
| 12:32.55 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 12:33.05 | dym | grml - forgot the command |
| 12:33.27 | wasanzy_ | ebook |
| 12:34.01 | wasanzy_ | by O'Reilly |
| 12:35.39 | irroot | ~thebook |
| 12:35.39 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 12:35.50 | dym | irroot: cheers :D |
| 12:35.58 | irroot | ~beer dym |
| 12:35.59 | infobot | ACTION pulls out a excellent Guinness for dym |
| 12:36.02 | dym | (: |
| 12:38.42 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 12:38.43 | *** join/#asterisk billmania (~bill@38.98.130.98) |
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| 12:45.58 | *** join/#asterisk mtnbkr (~mtnbkr@75-150-91-17-NewEngland.hfc.comcastbusiness.net) |
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| 12:48.31 | *** join/#asterisk skrusty (~skrusty@83.166.171.74) |
| 12:50.33 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 12:51.22 | asteriskATmarmuD | can someone recommend a small and narrow SIP-softphone for windows 2k? thanks |
| 12:51.23 | skrusty | afternoon |
| 12:52.17 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-hkcxkzoesbsqxdrl) |
| 12:52.22 | Cobadol | asteriskATmarmuD: E-kiga is pretty good and stable with video support |
| 12:53.00 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:53.06 | asteriskATmarmuD | Cobadol: I don't need video, but will take a look. some small and narrow softphone with no extra features at all would be great |
| 12:53.07 | Cobadol | asteriskATmarmuD: quitephone is much ergonomical but in current version unstable |
| 12:53.23 | asteriskATmarmuD | Cobadol: something stable would be great ;) |
| 12:54.58 | Cobadol | asteriskATmarmuD: Take a look on Linphone too |
| 13:03.33 | asteriskATmarmuD | Cobadol: already opened a tab to linphone ;) thanks for your hints! have a nice day! |
| 13:08.05 | *** join/#asterisk Jonalanning (~Jonalanni@108-90-184-88.lightspeed.toldoh.sbcglobal.net) |
| 13:10.16 | WIMPy | Zoiper |
| 13:13.49 | *** join/#asterisk retentiveboy (~pdugas@72.54.144.26) |
| 13:17.07 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 13:18.25 | Jonalanning | Has anyone ever done Alert Info with Polycom Phones? I can't seem to get it to work. |
| 13:19.02 | Cobadol | WIMPy: is there adverts? |
| 13:19.45 | WIMPy | ? |
| 13:20.17 | Jonalanning | adverts?? I am only 5 months into asterisk not sure I know what that is. |
| 13:22.09 | Law | has outgoing call but has lost incoming along the way |
| 13:22.13 | Law | is this progress? |
| 13:22.41 | asteriskATmarmuD | <PROTECTED> |
| 13:22.59 | *** join/#asterisk moy (~moy@209.250.158.226.tor.pathcom.com) |
| 13:27.45 | Cobadol | Jonalanning: advertisment, i mean |
| 13:34.23 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:34.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:41.05 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 13:44.02 | *** join/#asterisk tuxx- (tuxx@pantoff0l.nl) |
| 13:45.02 | Kobaz | what does everyone use these days for high powered long distance wifi through obstructions (trees) |
| 13:46.32 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:47.48 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:48.01 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 13:48.19 | WIMPy | A repeater in the tree, some illegal amplifiers or a higher mast. |
| 13:49.40 | Kobaz | http://store.advancedmart.com/ranoofsipoto.html |
| 13:49.50 | Kobaz | no power between stations |
| 13:50.01 | Kobaz | 1/2 mile between |
| 13:50.18 | leifmadsen | * may cause cancer |
| 13:52.11 | Kobaz | known to the state of florida to kill kittens |
| 13:52.49 | dym | leifmadsen: thatsalie! |
| 13:52.54 | leifmadsen | o.O |
| 13:53.01 | leifmadsen | Kobaz: +1 |
| 13:53.50 | dym | me and * spent several nights together |
| 13:55.30 | Kobaz | leifmadsen: and there's high tension power lines in between the buildings |
| 13:57.13 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 13:57.37 | timeshell | I need some help on occasional "sidetone" issues |
| 13:57.50 | timeshell | Is it possible just turn off sidetown completely in asterisk? |
| 13:58.07 | timeshell | sidetown=sidetone |
| 13:58.22 | leifmadsen | I don't think asterisk provides sidetone.... that's a phoen thing |
| 13:58.23 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 13:58.31 | leifmadsen | I don't even see how asterisk would provide sidetone |
| 13:59.14 | timeshell | it's ip phone side? |
| 13:59.24 | leifmadsen | it's ____ phone side |
| 13:59.36 | timeshell | Not inclined to agree... |
| 13:59.36 | leifmadsen | that's where sidetone is produced |
| 13:59.39 | timeshell | Not on what I've read |
| 13:59.46 | leifmadsen | now the question is, are we talking about the same thing |
| 13:59.49 | timeshell | Maybe I misunderstood |
| 14:00.00 | timeshell | Sidetone: http://forums.asterisk.org/viewtopic.php?f=1&t=18321 |
| 14:00.09 | timeshell | ^^^ I'm having more or less this issue |
| 14:00.26 | _Corey_ | Normally we just call it echo |
| 14:00.33 | leifmadsen | sidetone != echo |
| 14:00.46 | timeshell | It's not on every FXO call though |
| 14:00.56 | _Corey_ | lm: what's the distinction? |
| 14:01.08 | leifmadsen | sidetone is the function of the phone to lightly hear yourself on the phone so that it appears more natural |
| 14:01.13 | timeshell | But on enough calls that's a problem. |
| 14:01.16 | _Corey_ | ah |
| 14:01.32 | leifmadsen | if you talk into a handset on a phone (that is powered) you'll hear yourself slightly |
| 14:01.41 | ectospasm | leifmadsen: did you realize there is no call file info in ~thebook? |
| 14:01.54 | _Corey_ | yeah, i know what you mean... though when you hear it on a delay and it's perceptible it's usually echo, no? |
| 14:01.56 | leifmadsen | who uses callfiles anymore? |
| 14:02.03 | leifmadsen | _Corey_: correct |
| 14:02.07 | _Corey_ | I use callfiles |
| 14:02.10 | ectospasm | leifmadsen: my customer |
| 14:02.27 | *** join/#asterisk luminforce (~luminforc@rrcs-71-42-183-54.sw.biz.rr.com) |
| 14:02.28 | WIMPy | uses them |
| 14:04.14 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:04.53 | timeshell | Any thoughts on my sidetone issue? |
| 14:05.36 | timeshell | There doesn't appear to be any echo to the PSTN side, only on the SIP side of the call. |
| 14:05.39 | timeshell | SIP to SIP are fine. |
| 14:08.15 | Bob_Pierce | I still use call files a fair bit. Is there a newer / more preferred way to perform the same function? |
| 14:09.20 | _Corey_ | timeshell: You should still look at your echo cancellation settings... experiment with the txgain and rxgain and you should be able to eliminate it |
| 14:09.31 | aberrios_ | uses call files too on his IAX2 trunk, although it may be causing some thread locking issue |
| 14:09.32 | timeshell | Already have |
| 14:09.40 | timeshell | Doesn't make any difference at allo |
| 14:10.24 | billmania | When my asterisk 1.8.4.2 dialplan executes Dial("SIP/5103-00000003", "DAHDI/G0/1773XXXXXXX,300,") in new stack and the called party doesn't answer, why do I get "exited non-zero on 'SIP/5103-00000003' in macro" instead of DIALSTATUS and HANGUPCAUSE being set? |
| 14:10.25 | leifmadsen | Bob_Pierce: AMI is the preferred way really, especially if you're doing it all the time |
| 14:10.37 | _Corey_ | is there not a delay associated with the sidetone? I've never had anyone notice it when it's not echo-related |
| 14:11.56 | Bob_Pierce | leifmadsen: so basically have some external process initiate calls through AMI? |
| 14:12.00 | leifmadsen | yes |
| 14:12.06 | leifmadsen | isn't that how you use callfiles anyways? |
| 14:12.23 | leifmadsen | the Shift8 library makes it really easy to interact via PHP |
| 14:13.08 | Bob_Pierce | Basically, except for the callfiles are set for a certain time in the future and then we also make use of the redial attempts being set at 3 hour intervals. |
| 14:13.26 | Bob_Pierce | I'm sure we could replicate some similar logic with an external process |
| 14:15.15 | Bob_Pierce | We're using it for automated reminder calls to our customers of an installation appointment. |
| 14:18.10 | Bob_Pierce | leifmadsen: thanks for mentioning shift8. That looks really good! I had used phpagi before. |
| 14:18.21 | leifmadsen | ya, I just used shift8 recently -- works well |
| 14:21.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:23.56 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 14:30.37 | *** join/#asterisk kfife (~Miranda@kfife.com) |
| 14:31.19 | Linuturk | ok, this is a really weird problem. I was just shown a fax that looks like it didn't spool correctly. some of the text and logo was drawn out down the page, making it look like a barcode. there wasn't any barcode on the original sent to us. What would cause this to happen? |
| 14:31.40 | kfife | Linuturk: Bad paper feeder? |
| 14:32.02 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:32.08 | kfife | Were you receiving from someone else or sending via FFA? |
| 14:32.08 | Linuturk | I can't discount that possibility, but it's a brand new machine |
| 14:32.39 | kfife | Never underestimate how crappy a paper feeder can be. :-) |
| 14:32.42 | irroot | seen it before linuturk i guess its transmission error and the machine tries to rebuild it |
| 14:32.43 | Linuturk | receiving from someone else. the faxes come in over a PRI, hit asterisk, and then are pushed to a sip gateway, where they are passed to the analog machine |
| 14:33.13 | irroot | linuturk using T.38 or G711 |
| 14:33.27 | Linuturk | supposed to be using t.38 |
| 14:33.45 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:33.57 | irroot | ~beer The_Boy_wonder |
| 14:33.57 | infobot | ACTION pours a litre of La Maudite for The_Boy_wonder |
| 14:34.20 | billmania | In asterisk 1.8.4.2 how do I get Dial() to return to the next step in the dialplan when the called party doesn't answer? |
| 14:34.20 | kfife | Linturn: the Sip gateway and Fax are on 'your' managed ethernet? |
| 14:34.48 | Linuturk | yep, the only ethernet link is between the sip gateway and the asterisk server |
| 14:34.56 | jaytee | billmania, use the timeout parameter |
| 14:34.59 | kfife | billmania: ain't that automatimacallly the next thing after your timeout interval |
| 14:36.50 | billmania | jaytee: From the verbose log file I'm using Dial("SIP/5103-00000003", "DAHDI/G0/1773XXXXXXX,300,"). |
| 14:37.15 | billmania | However, what's actually happening after the called party doesn't answer is: |
| 14:37.16 | billmania | exited non-zero on 'SIP/5103-00000003' in macro 'dialout-trunk' |
| 14:37.49 | billmania | (I obfuscated the number before pasting here) |
| 14:37.50 | kfife | billmania: try 30 seconds rather than 300 |
| 14:38.02 | kfife | Just to isolate the trouble |
| 14:38.12 | billmania | kfife: OK. |
| 14:39.11 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 14:40.19 | kfife | Here's a question: Do the following run with the same user-privelege? |
| 14:40.22 | kfife | exten=>test,1,System(/home/ast/myscript.sh) |
| 14:40.22 | kfife | and |
| 14:40.22 | kfife | asterisk*CLI>!/home/ast/myscript.sh |
| 14:41.12 | kfife | The first doesn't want to run, the second works fine |
| 14:45.24 | kaldemar | kfife: yes, but the environment might differ. you might need to defined paths in commands in the script. |
| 14:46.49 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-214-57.w86-204.abo.wanadoo.fr) |
| 14:47.23 | kfife | kaldemar: Thanks. Brilliant |
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| 14:48.07 | merlin8282 | Hi all ! Do you know of an interface that shows the status of many extensions, like BLF does for example on a snom360 ? |
| 14:48.19 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 14:48.52 | merlin8282 | For the moment I'm using ajamdemo.html, but it's not really user friendly (it's not only used by me...) |
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| 14:50.13 | kfife | kaldemar: You were absolutely right. Can I buy you a beer? |
| 14:50.59 | kaldemar | if you insist. :) |
| 14:51.18 | kfife | http://beeradvocate.com/beer/profile/72/763 |
| 14:52.35 | luminforce | we have an issue where a remote peer sends a mid-call invite (session timer) INVITE> <TRYING <OK >ACK, except that the ACK is improperly formatted, so Asterisk sends <OK 3 more times, and then hangs up, but does not send a BYE. shouldn't this BYE be sent anyway, even if there is chance it's not going to be received? |
| 14:52.44 | kfife | Ein schönes Bier |
| 14:54.07 | billmania | kfife: That did the trick. Thanks. Now I have DIALSTATUS and HANGUPCAUSE. |
| 14:54.11 | azv4 | guys, i have a question, we have an old Panasonic KX-TD500 system, and I need to add the capability to transfer calls to our new facility that has a nice shiny new Voip system, can I buy some kind of analog to viop converter and attach it to a few of our extensions, and have the converter unit point to the remote location's voip system, so I can transfer calls? |
| 14:54.59 | kfife | billmania: Great! now you owe me a beer, but I owe kaldemar a beer, so you can just give it to him :-) |
| 14:55.09 | leifmadsen | merlin8282: there is an example on the blog site for the shift8 library |
| 14:55.20 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 14:55.22 | kfife | leifmadsen: Hey there! |
| 14:55.27 | leifmadsen | kfife: omghai! |
| 14:55.27 | billmania | Where do I send it? :-D |
| 14:55.53 | kfife | billmania: my office. Can you have the Warsteiner girls deliver it? |
| 14:56.06 | leifmadsen | heads over to kfife's office |
| 14:56.09 | merlin8282 | leifmadsen: ok, seems good stuff ! thanks |
| 14:56.15 | kfife | leifmadsen: lol |
| 14:56.21 | *** join/#asterisk yahh (~root@122.169.89.18) |
| 14:56.26 | billmania | Of course. Nothing but the best. I'll ask them to also give you "a little something special". |
| 14:56.45 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 14:57.24 | kfife | billmania: :-) |
| 14:57.28 | yahh | i have developed one custom application for asterisk-1.6.2.17.2 |
| 14:57.42 | yahh | it is working ok on centos 5.6 64-bit |
| 14:58.00 | yahh | but it is crashing in centos 5.6 32-bit |
| 14:58.23 | yahh | i tried but couldn't find reason, can someone help me |
| 14:58.39 | Nasga | Hello, i want to manage my queues with "add queue member" to add agents dynamically (and log them on each calls). This look like possible ? (i'm with asterisk 1.2) |
| 15:00.56 | kfife | billmania: They're here. What do I do now?? http://kfife.com/download/pix/KarlAtOffice.jpg |
| 15:01.00 | azv4 | is my question just too far off topic? is there a phone system channel anyone know? |
| 15:01.08 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 15:01.34 | billmania | kfife: Live in the moment, man! |
| 15:01.35 | jaytee | azv4, you might use an FXO gateway with your panasonic extensions |
| 15:02.07 | kfife | billmania: :-) |
| 15:02.10 | azv4 | jaytee, thank you |
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| 15:14.13 | jeffspeff | I'm getting the following error when a sip user registers. ERROR[20348]: res_config_ldap.c:1326 update_ldap: Couldn't modify 'name'='jmarcomb', dn:CN=Jared Marcomb,CN=Users,DC=test,DC=server,DC=local because No such attribute |
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| 15:27.04 | ChannelZ | yahh: did you actually build a new version in 32-bit? |
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| 15:31.22 | yahh | you mean asterisk's new version? No, i have build it with asterisk-1.6.2.17.2. |
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| 15:49.19 | luminforce | we have an issue where a remote peer sends a mid-call invite (session timer) INVITE> <TRYING <OK >ACK, except that the ACK is improperly formatted (and dropped by our OpenSIPS gateway), so Asterisk sends <OK 3 more times, and then hangs up, but does not send a BYE. shouldn't this BYE be sent anyway, even if there is chance it's not going to be received? |
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| 15:51.35 | billmania | I see that Dial() has DIALSTATUS "CANCEL" but my dialplan doesn't get that if I hangup before the call is answered and before the Dial() timeout. Instead, I again get the "exited non-zero on" business. |
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| 16:02.14 | Hive | Hello everyone. Is there anyone with Fax for Asterisk experience around that is willing to chat? |
| 16:02.19 | *** join/#asterisk [psy] (~psy0rz@lounge.datux.nl) |
| 16:02.25 | Hive | Or any experience with * faxing at all haha |
| 16:02.31 | [psy] | is it true that the CDR is still buggy on transfers? |
| 16:02.34 | WIMPy | ~ask |
| 16:02.34 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:02.36 | [psy] | even in 1.8 |
| 16:03.34 | *** join/#asterisk Micc (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 16:03.38 | mnicholson | [psy], yes in some specific cases. Use CEL instead if you expect complex transfer scenarios |
| 16:04.42 | Hive | When does a fax session end and release the session registration from fax for asterisk? |
| 16:04.52 | ssureshot | is there an inexpensive single fxs usb device out there compatible with asterisk? |
| 16:05.32 | *** join/#asterisk Law (~Miranda@cpc2-rdng5-0-0-cust632.15-3.cable.virginmedia.com) |
| 16:06.05 | mnicholson | Hive, when sendfax or receivefax exits or when the channel hangs up |
| 16:06.18 | [psy] | mnicholson when calling to an external number, and then transferring the call to a internal number, the CDR will log it as an internal call. nothing complex about it. tested it with the latest 1.4 and 1.8 versions. |
| 16:06.51 | [psy] | using a freepbx dialplan |
| 16:07.39 | [psy] | i think CDR is completely useless in asterisk |
| 16:08.26 | ssureshot | I always get calls from HR asking if people from specific numbers actually tried to call the calloff line |
| 16:08.33 | ssureshot | thats all I use CDR for |
| 16:08.51 | Hive | mnicholson, I have a php script that I wish to run at the end of every fax. It is in the h extension of my [outboundFax] context. However it appears that my script is occasionally running before the fax thread is complete. Is there a way to say "hey wait until the fax session is completed before doing anything"? |
| 16:09.48 | Hive | yes in my experience CDR table is impossible to understand :-P |
| 16:09.57 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 16:10.54 | mnicholson | [psy], use CEL |
| 16:11.33 | mnicholson | Hive, how are you determining that it is running before the fax thread is complete? |
| 16:13.01 | Hive | Well the script checks the ${FAXOPT(statusstr)} and if it is a success message sends out an email of success. If it is not success it creates a new call file containing the fax file and attempts to send the fax again |
| 16:14.32 | Hive | I only have one fax for asterisk registration and get the error message "Cannot reserve FAX session - session limit exceeded (max: 1)" and "sendfax_exec: Unable to reserve FAX session" |
| 16:14.37 | Hive | So I' |
| 16:14.56 | Hive | I'm assuming that this means it is trying to grab another session for the new fax spawned from the script |
| 16:15.05 | coppice | what would you expect? |
| 16:15.29 | Hive | I would prefer that the previous fax session would end, so that my scrip can then grab up the only session that I have |
| 16:16.02 | Hive | I try not to expect much with faxes over asterisk, because I'm usually wrong :-P |
| 16:16.13 | coppice | well, you could use spandsp and avoid such limits |
| 16:18.51 | [psy] | mnicholson the app only supports CDR csv's , not CEL csv's :( |
| 16:20.19 | *** join/#asterisk Law (~Miranda@cpc2-rdng5-0-0-cust632.15-3.cable.virginmedia.com) |
| 16:24.24 | mnicholson | Hive, spawn the resend script in the background and have it wait a second before adding the call file |
| 16:24.53 | Hive | coppice, thank you for the suggestion. I already have multiple fax for asterisk sessions however, so I am going to try to stick with that. I'm just trying to get the 1 fax lifecycle under control at the moment |
| 16:25.23 | coppice | you actually paid? weird :-) |
| 16:25.50 | Hive | hehe well I didnt pay personally ;-) |
| 16:26.53 | mnicholson | [psy], nothing you can do then. You could perhaps generate your own CSV cdr files from CEL records... |
| 16:30.00 | Hive | mnicholson, if I understand you correctly, I have added in a Wait(40) command before the script is called. |
| 16:30.05 | [psy] | yep i was thinking about that mnicholson .. |
| 16:30.08 | [psy] | thanks anyway |
| 16:31.20 | mnicholson | Hive, im not sure that will work. You can try that, but I was thinking more along the lines of having your script run backgrounded then 'sleep' before sending the retry |
| 16:31.58 | *** join/#asterisk d-_-b- (~d-_-b-@76.75.3.174) |
| 16:32.19 | Hive | Oh interesing. I didn't know about this function :D |
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| 16:45.49 | Hive | Why is it that whenever I want a fax to fail, it work? I suppose this isn't a very good question hehe |
| 16:47.52 | *** join/#asterisk esperanto (~rusty@cpe-65-185-172-226.neo.res.rr.com) |
| 16:48.20 | mnicholson | Hive, fax over sip using gsm, it will fail every time :) |
| 16:50.37 | Hive | mnicholson, it seems you have fixed one of my problems. Thanks :D |
| 16:50.53 | mnicholson | Hive, np |
| 16:53.03 | Hive | Here's the next issue with my system. the script can be called with a System(/usr/bin/php /the/script.php) once. However, it will not execute the script a second time |
| 16:53.23 | mnicholson | what happens the second time? |
| 16:54.58 | Hive | So the process is; attempt to send fax -> fax failure -> run the script (set fax attempts = 2)-> attempt to send fax -> fax failure -> call to System(/usr/bin/php /the/script.php), but a new fax is never sent |
| 16:55.24 | Hive | I checked the .call file that my script creates, and the permissions are fine |
| 16:55.39 | Hive | I just cant figure out why the script will not execute the second time |
| 17:00.21 | *** join/#asterisk irroot (~irroot@197.169.168.170) |
| 17:08.15 | mnicholson | Hive, placing calls via the manager interface is more reliable than the call files. Make sure you are moving the call file into the spool dir and not creating it there or copying it to there. |
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| 17:15.29 | Hive | Uh oh.. I'm not supposed to copy files ther? :x |
| 17:16.07 | dym | Hey all! Any idea why i get this error? Is it an incompatible file? cause it does exist.. http://pastebin.com/t4MgYLcn |
| 17:16.30 | Qwell | dym: Remove the extension. Asterisk will find the best file to use. |
| 17:16.33 | mnicholson | Hive, https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+FIles |
| 17:19.44 | wasanzy_ | Hi all, is it possible to configure Asterisk for an IVR solution? and any doc I can study? thanks |
| 17:19.49 | dym | <PROTECTED> |
| 17:19.51 | Qwell | ~book |
| 17:19.51 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 17:19.55 | Qwell | wasanzy_: Read that. All of it. |
| 17:19.58 | Qwell | Three times. |
| 17:20.16 | wasanzy_ | hmmm, can't I download that book? |
| 17:20.34 | Qwell | I give up. |
| 17:20.52 | wasanzy_ | why? |
| 17:21.10 | Qwell | [...] and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ |
| 17:21.17 | samandiriel | It's a great book, it's what I've been reading. |
| 17:21.18 | Qwell | Did you even bother reading the entire sentence? |
| 17:21.39 | wasanzy_ | I can see buy though |
| 17:23.52 | samandiriel | I've got a question. A common one I'm sure, but nonetheless: I'd like to have a caller record something and then continue processing after the caller has hung up. Is the best way to do this to Dial() a local channel with the F flag, then continue processing below that? |
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| 17:24.09 | samandiriel | wherein I do the recording in the local channel |
| 17:24.56 | wasanzy_ | thank you all, the book is really great |
| 17:25.33 | wasanzy_ | am currently reading "Asterisk The Future Telephony" |
| 17:25.53 | mnicholson | samandiriel, that might work, but you can also continue processing in the 'h' exten |
| 17:26.39 | dym | Can i add a variable to MixMonitor(test.wav) to always generate arbitrary filenames on new records when calling the command? |
| 17:26.49 | dym | is there a variable that spawns a uniqueid or sth? |
| 17:27.10 | samandiriel | thanks mnicholson. which way is more robust? is there any reason to prefer one over the other? |
| 17:27.15 | dym | mhh - why not timestamp :D ill look it up |
| 17:28.20 | dym | urgs - how can i convert ${EPOCH} to realtime? |
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| 17:29.06 | mnicholson | samandiriel, 'h' exten is probably better. It will always get executed upon hangup. Also there are less moving parts when using the 'h' exten |
| 17:30.02 | samandiriel | wotcha. Thanks, mnicholson |
| 17:36.32 | titter | When using leastrecent in a queue with a timeout of 10, after 10 seconds the queue just retries that same member. How would I move on to the next least recent, or is that not possible? |
| 17:39.38 | samandiriel | mnicholson, another question: if the h extension is in a GOSUB'd context, and there is an h exten in the calling context, which gets called? I would think the GOSUB h, as it's lowest in the stack. Might have to experiment on this one, eh? |
| 17:41.11 | titter | I believe the calling context carried the h exten, at least it does in Macro |
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| 17:42.19 | samandiriel | yah, but macro and gosub work differently, I know that |
| 17:43.40 | samandiriel | oh, and I think I found a bug in While(). Either that, or not the behaviour I was expecting. If you call a GOSUB in a While() loop that also has a While() loop, then break out of the outer While(), if your processing returns to that outer While() again it will run the GOSUB'd While() instead. |
| 17:44.20 | samandiriel | I went back and yanked out my While()s and put in gotoif's instead, and got the correct behaviour that way, so I don't think it was my logic |
| 17:46.21 | mnicholson | samandiriel, not sure about the h exten in macro and gosub, best to test it |
| 17:46.44 | mnicholson | samandiriel, as for that while thing, it may be a bug, test with the latest code and if it is still there, report it |
| 17:47.35 | samandiriel | thanks, mnicholson |
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| 17:51.20 | titter | I am having issues with the queue, one all dynamic members take a call once, the queue doesn't start back over from the top. Here is the CLI, http://pastebin.com/WRuwa9vn ... each agent was called once, the last incoming call I just sat in the queue and it never called the first agent again. |
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| 18:03.29 | jeffspeff | I'm running into a problem... ERROR[21991]: res_config_ldap.c:1326 update_ldap: Couldn't modify 'name'='jmarcomb', dn:CN=Jared Marcomb,CN=Users,DC=test,DC=server,DC=local because No such attribute any suggestions on why I'm receiving this? the value is there, otherwise the user wouldn't be able to authenticate. |
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| 18:10.22 | wasanzy_ | in what programming lang, asterisk was written? |
| 18:10.54 | Qwell | COBOL |
| 18:11.58 | singler | isn't asterisk's executables written in asm? :) |
| 18:12.33 | wasanzy_ | and can you write your own module in different lang for it, or write your own program in a different lang for it? |
| 18:13.10 | singler | yes |
| 18:13.20 | wasanzy_ | good |
| 18:14.25 | wasanzy_ | I wish to use C, python or Perl. I hope it support supports them as you mentioned |
| 18:14.38 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 18:15.02 | singler | python and perl are not programming languages |
| 18:15.27 | leifmadsen | wasanzy_: wouldn't looking at the code pretty much tell you what it was written in? |
| 18:15.41 | leifmadsen | or.... even looking at the extension of many of the files.... |
| 18:15.45 | leifmadsen | (that extension being .c) |
| 18:16.18 | wasanzy_ | but they are some times use to write program not only scripts |
| 18:16.41 | wasanzy_ | oh yes, which means C is one of the lang choice |
| 18:17.19 | wasanzy_ | I have no idea how COBOL code looks like that is why |
| 18:17.21 | leifmadsen | if you want to write modules with Asterisk, then you pretty much have to write in C -- it's not a choice |
| 18:17.36 | leifmadsen | wasanzy_: asterisk is writing in COBOL -- Qwell was fscking with you |
| 18:17.40 | leifmadsen | errr |
| 18:17.48 | leifmadsen | s/is writing/is not written/ |
| 18:18.01 | leifmadsen | has to stop doing so many things at once |
| 18:18.13 | wasanzy_ | hmmmm |
| 18:18.17 | wasanzy_ | ok |
| 18:18.19 | leifmadsen | wasanzy_: if you just want to write a script outside of Asterisk and have it triggered (or it trigger things) then you use the AGI or AMI |
| 18:18.26 | leifmadsen | ~agi |
| 18:18.26 | infobot | [~agi] AGI is the Asterisk Gateway Interface. Similar to CGI for web applications, AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), PERL (astperl?), and other languages. See also: https://wiki.asterisk.org/wiki/display/AST/Application_AGI or http://www.voip-info.org/wiki-Asterisk+AGI |
| 18:18.27 | leifmadsen | ~ami |
| 18:18.27 | infobot | well, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 18:19.20 | *** join/#asterisk Maxxed (~Maxxed@216.215.95.118) |
| 18:19.41 | wasanzy_ | good |
| 18:19.47 | wasanzy_ | thank you soo much |
| 18:20.48 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:21.47 | beek | leifmadsen: Make sure that you put in a plug for your book! |
| 18:23.05 | seraphie | wasanzy_: if you are interesed in AGI and are familiar with python, you should check the Asterisk testsuite out from subversion and study it. Many tests uses python's StarPy Fast AGI module. |
| 18:23.14 | jeffspeff | leifmadsen, my issue seems very very similar to a bug report that you worked on ---> http://www.mailfunnel.org/asterisk-bugs@lists.digium.com/2010-10/msg01290.html |
| 18:23.44 | jeffspeff | ERROR[21991]: res_config_ldap.c:1326 update_ldap: Couldn't modify 'name'='jmarcomb', dn:CN=Jared Marcomb,CN=Users,DC=test,DC=server,DC=local because No such attribute |
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| 18:58.20 | f2Knight | Hello fellow Asterisk-ers |
| 18:59.06 | f2Knight | leifmadsen, Good morning, made some progress on my issue from yesterday |
| 19:01.17 | f2Knight | Found an issue that I am confused about.. When using Dial(local/s@somecontext), if the 'somecontext' has an additional Dial statement (such as to establish an outbound call) then when that call terminates it does not "STAY" in the 'somecontext' but rather returns to the calling context. Does this sound normal |
| 19:03.02 | f2Knight | this means doing a Noop(${DIALSTATUS}) right after the Dial statement in 'somecontext' never gets executed but rather returns to the previous calling context and then is handled, from what I see my h statement is executed. |
| 19:06.37 | jeffspeff | can anybody tell me why res_config_ldap.c is trying to modify a users name attribute in update_ldap? |
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| 19:15.11 | ssureshot | WIMPy: I took your advice and hot-rodded my server to add power for the TDM400 since the server didn't have the option for molex power adaptor.. works great!! exciting stuff |
| 19:17.16 | mickecarlsson | jeffspeff: update_ldap - brief Function to update a set of values in ldap static mode |
| 19:17.47 | jeffspeff | static mode? as in trying to write to the ldap database? |
| 19:17.57 | mickecarlsson | Yes -that is what it is for |
| 19:18.05 | mickecarlsson | Look at the source |
| 19:18.15 | jeffspeff | so, why would it be trying to modify a users name? |
| 19:18.20 | mickecarlsson | "update or set a value" |
| 19:18.36 | mickecarlsson | Probably wont, it wants to write it |
| 19:18.41 | mickecarlsson | as in update |
| 19:19.04 | mickecarlsson | it is trying to modify the cn |
| 19:19.07 | mickecarlsson | probably |
| 19:19.17 | jeffspeff | mickecarlsson, i keep getting an error that it can't modify a certain attribute, how do i keep it from doing that? |
| 19:19.51 | mickecarlsson | modify the schema for the ldap source allowing it to be updated |
| 19:20.20 | mickecarlsson | Crank up the debug to see what the attribute is |
| 19:21.04 | jeffspeff | the attribute is name |
| 19:21.16 | jeffspeff | [Jul 21 14:21:08] ERROR[21991]: res_config_ldap.c:1326 update_ldap: Couldn't modify 'name'='jmarcomb', dn:CN=Jared Marcomb,CN=Users,DC=test,DC=server,DC=local because No such attribute |
| 19:21.16 | jeffspeff | sip*CLI> |
| 19:21.39 | mickecarlsson | There is no 'name' in your ldap |
| 19:21.48 | leifmadsen | yes I've had this problem before.... it seems to be a problem with the way the openldap is setup to not allow 'name' to be updated |
| 19:21.49 | mickecarlsson | = No such attribute |
| 19:21.51 | leifmadsen | even if you set it that way |
| 19:21.57 | leifmadsen | or it doesn't have write access to it |
| 19:22.52 | mickecarlsson | DC=local tells me it is an MSAD |
| 19:22.59 | jeffspeff | yep |
| 19:23.28 | mickecarlsson | Hang on, cheking aone thing |
| 19:23.41 | jeffspeff | when i do a "sip show peer jmarcomb load" it shows the name value properly |
| 19:26.57 | mickecarlsson | jeffspeff: check the sample config for res_config_ldap |
| 19:27.17 | mickecarlsson | ; We use the "cn" as the default value for name on the line above |
| 19:27.27 | mickecarlsson | name = cn |
| 19:28.01 | mickecarlsson | Best guess, change name = cn to name = uid |
| 19:28.08 | mickecarlsson | in that config file |
| 19:28.40 | jeffspeff | i have mine set to "name = sAMAccountName |
| 19:29.01 | jeffspeff | why would using UID change it? |
| 19:29.19 | mickecarlsson | Read the file, there is a lengthy explanation in it |
| 19:31.02 | mickecarlsson | sAMAccountName = Pre-Windows 2000 logon name |
| 19:31.49 | jeffspeff | where is this "res_config_ldap" file? |
| 19:32.09 | mickecarlsson | In the source for Asterisk, in configs folder |
| 19:32.30 | mickecarlsson | Or grab it from svnviewer |
| 19:32.56 | mickecarlsson | http://svnview.digium.com/svn/asterisk/branches/1.8/configs/res_ldap.conf.sample?view=log |
| 19:33.32 | mickecarlsson | Hmm, that did not work |
| 19:34.01 | mickecarlsson | Click on View |
| 19:34.39 | mickecarlsson | gtg, might be back later |
| 19:34.58 | jeffspeff | mickecarlsson, thanks for your help... i'll probably still be working on all of this. |
| 19:35.49 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 19:37.03 | *** join/#asterisk Nasga_ (~Nasga@AAmiens-157-1-56-250.w86-196.abo.wanadoo.fr) |
| 19:38.32 | anonymouz666 | leifmadsen: hello. the issue 15609 about the warning "channel.c:952 __ast_queue_frame: Exceptionally long voice queue length queuing to Local" was committed into more than one branch, including the 1.4 - in dec/2009. however, the patch is two lines long and I can't see the code in the latest 1.4.42 |
| 19:38.34 | anonymouz666 | very strange |
| 19:40.27 | azv4 | Does anyone know if a Panasonic KX-TD500 has a management console option where I can access it remotely? |
| 19:40.42 | leifmadsen | o.O |
| 19:40.45 | azv4 | haha |
| 19:41.12 | azv4 | I haven't even looked at the dang thing yet, it is a building away and I am afraid to go look at it! I just started here last week, but I need to enable the switchboard |
| 19:41.22 | Qwell | anonymouz666: You assume the code will stay the same for 2 years. |
| 19:42.48 | anonymouz666 | I am confused |
| 19:44.36 | anonymouz666 | need to see changelog then |
| 19:45.01 | *** part/#asterisk luminforce (~luminforc@rrcs-71-42-183-54.sw.biz.rr.com) |
| 19:47.25 | anonymouz666 | Qwell: batman changed it |
| 19:55.30 | *** join/#asterisk sourcode (~code@ppp-58-11-104-113.revip2.asianet.co.th) |
| 20:02.06 | dym | rHey all. Im using a Sipgate Trunk to dialout and pass the sendernumber with http://pastebin.com/9RkaMegs - for some reason this number is not applied, but the first of the trunknumbers (default number) is transmitted. any idea on that? |
| 20:02.18 | dym | according to the provider they use whatever number you pass to the trunk |
| 20:02.19 | titter | Can someone explain why this queue does not continue on after all members have received 1 call? The next caller just sits in the queue without ringing anyone else. http://pastebin.com/WRuwa9vn |
| 20:05.52 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 20:06.31 | azv4 | can * allow me to manage a Panasonic KT-TD500? |
| 20:08.26 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:09.10 | azv4 | ok no more dumb questions, I'm reading the website! |
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| 20:55.14 | Hive | mnicholson, just wanted to say thanks for your help earlier. My fax setup is working much nicer now :D |
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| 21:02.12 | ChannelZ | titter: what does the config for the queue look like? Maybe the wrapup time is set wonky... |
| 21:03.21 | *** join/#asterisk godmachine-x6 (~godmachin@h18.171.140.67.dynamic.ip.windstream.net) |
| 21:06.22 | titter | ChannelZ: Wrapup is set to 30 |
| 21:16.27 | ChannelZ | hmm |
| 21:22.29 | *** join/#asterisk xpot-mobile (~james@97-117-93-54.slkc.qwest.net) |
| 21:24.46 | titter | Just reloaded it again |
| 21:25.06 | titter | http://pastebin.com/ZAjt5qJM |
| 21:25.11 | titter | thats the queues.conf |
| 21:27.11 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 21:28.10 | titter | Just tested again, waiting 30 seconds and will see |
| 21:29.37 | titter | Works now? |
| 21:29.55 | titter | Oh well. |
| 21:31.21 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 21:46.26 | tiaz | when matching a pattern, is '+' a special character? I notice some other regex-style features ([1-5] being the range, for example) |
| 21:47.48 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 21:50.43 | WIMPy | No, and without the _ everything is literal. |
| 21:52.02 | tiaz | huh. maybe my client isn't sending what I think it is then. thanks |
| 21:53.09 | WIMPy | If it doesn't match, you should see that on the *CLI. |
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| 22:15.58 | *** join/#asterisk cutlass (~cutlass@c-67-165-132-67.hsd1.il.comcast.net) |
| 22:20.58 | cutlass | Regarding SIP registrations when using ARA, does anyone know the relationship between the sip user table's "fullcontact" field, the AstDB's /SIP/registrations/<EXTENSION> key and the data that is show when you perform "sip show peer <EXTENSION>"? |
| 22:22.07 | cutlass | I have seen different things which lead me to believe that none of these values are used in real-time to determine where or if a call is to be delivered to the endpoint |
| 22:22.38 | cutlass | any insight at all would be massively helpful |
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| 22:46.46 | sunfone | Anyone familiar with PRTG network monitoring tool? Researching ways to measure jitter. |
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| 22:59.19 | sunfone | Anyone play with the Polycom VVX 1500 (video phone)? Wondering if I can connect an external camera/mic/projector to it and use a pair to connect two conference rooms. |
| 23:09.58 | cutlass | is this still the recommended way to interact with other Asterisk users? It seems like there is not much activity on here... |
| 23:10.39 | sunfone | Does seem unusually quiet right now. |
| 23:11.09 | sunfone | Normally there are conversations about all kinds of things... the evils of sugar in your diet, local politics, etc. |
| 23:12.14 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
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| 23:36.21 | *** join/#asterisk hardwire (~spencersr@12.17.188.86) |
| 23:36.28 | hardwire | heya hiya hoya! |
| 23:36.32 | hardwire | anybody in hawaii? |
| 23:36.36 | hardwire | the big HI |
| 23:37.55 | hardwire | doesn't look like it :( |
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| 23:52.30 | *** join/#asterisk pdtpatrick (~pdtpdt@mainstwan.farheap.com) |
| 23:52.46 | WiretapWork | hardwire: physical location does not affect our ability to assist you |
| 23:52.48 | WiretapWork | ~ask |
| 23:52.48 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:53.25 | pdtpatrick | Question .. if i wanted to have asterisk use SIP as trunking rather the PRI card.. which right now is set to TRUNK=DAHDI/g1 ... how can i do that? TRUNK=SIP/contextgoes here ? |
| 23:54.55 | pdtpatrick | nvm got it |
| 23:55.35 | ChannelZ | Hawaii isn't very big, just yell out the window.. I'm sure you'll find someone |
