| 00:00.56 | dan__t | Hrm. Can I unset a channel variable? |
| 00:00.57 | Karen_m | do you enable that on the phone or something? |
| 00:01.01 | dan__t | Maybe just set it back to =? |
| 00:01.08 | Karen_m | i have tried to connect via ftp and the port is not responding |
| 00:02.43 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 00:03.37 | nightrid3r | polycom uses TFTP |
| 00:03.57 | nightrid3r | so you need te setup a tftp server to boot from |
| 00:04.11 | Karen_m | how does it discover the tftp server though? |
| 00:04.24 | nightrid3r | dhcp |
| 00:04.58 | nightrid3r | or setup phone manualy |
| 00:05.36 | Karen_m | so everytime the phone reboots, i have to be running a tftp server for it to grab the settings from? |
| 00:05.55 | Karen_m | or can i enable an ftpd on the phone, connect, put the files there and it reads from the phone? |
| 00:07.46 | nightrid3r | tftp is the only thing that will work, not only for rom updates but you can also put your phones config on the tftp server |
| 00:08.29 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 00:09.09 | Karen_m | so that implies i need to run my own dhcp server and not rely on the router my adsl company gave me? |
| 00:09.22 | Karen_m | i'll try and setup a tftp server and see what happens |
| 00:09.40 | ChannelZ | Surely there is a bootstrap URL or something you hit the phone with to direct it to the server of your choice |
| 00:10.05 | ChannelZ | That's how it works on almost all Sipura/Linksys/Cisco |
| 00:11.53 | nightrid3r | yes you can manualy give the phone the address of the tftp server |
| 00:12.02 | ChannelZ | http://www.thinkbright.net/support/video_display.php?id=5 |
| 00:12.10 | nightrid3r | http://help.fonality.com/index.php?title=IP_Phones/Polycom/Polycom_501,_601,_550_%26_650_Factory_Flash |
| 00:13.09 | Karen_m | awesome! thanks i'm watching that video |
| 00:13.17 | Karen_m | thanks nightrider, will check that in a second. |
| 00:24.05 | gogasca | anyone using Tandberg Video phones E20s, C20s, C40s with * ? |
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| 01:34.47 | linuxgecko | Karen_m: whenever i have a box to spare for it, i NEVER rely solely on the dhcp on the ISP modem, or even a router/hub. they goof things too easily, and i get more control iwth a linux nat/dhcp/et al :) |
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| 02:26.49 | Caplain | if i want to plug a pstn phone into my asterisk box would i need an fxo or fxs card? |
| 02:30.46 | Caplain | oh, fxs...they're so expensive |
| 02:32.23 | p3nguin_ | You might just as well get an ATA. |
| 02:32.31 | p3nguin_ | $35 or less, typically |
| 02:35.36 | Karen_m | with my polycom 650, it shows 34 lines in the webpage config, but only 6 buttons for each line. Is there a way to get access to the 7th line and so on somehow on the phone? |
| 02:37.22 | Caplain | p3nguin_, i want to use a fax machine with my mj on gentoo, im about to install vmware :( |
| 02:38.27 | ChannelZ | why do the lines even matter? |
| 02:38.29 | p3nguin_ | I have no idea if fax over voice over internet protocol works in a virtual machine. |
| 02:38.59 | Caplain | well fax uses those squeaky noises...so in theory if the line quality isn't bad then i guess it would work |
| 02:39.05 | Karen_m | ChannelZ, multiple dids and to access the outbound caller id |
| 02:42.18 | ChannelZ | *shrug* it's just an old way of thinking for a SIP phone. In any case, it's probably setup that way to using those expansion console with a zillion buttons on them |
| 02:42.24 | ChannelZ | sidecars or whatever they call them |
| 02:48.02 | *** join/#asterisk banditti (~banditti@unaffiliated/banditti) |
| 02:48.40 | banditti | I have an IP335 and asterisknow. The 335 just is in a boot loop. Any thoughts? |
| 02:49.23 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 02:50.01 | banditti | I am also using the end point manager to gen the files. |
| 02:50.06 | ChannelZ | bogus config... |
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| 02:50.49 | banditti | I did a google search to try and find a how to and couldn't find any. Is the IP335 like another polycom that I can search on? |
| 02:52.52 | Caplain | dsfghjkl;' |
| 02:53.05 | banditti | ? |
| 02:53.15 | Caplain | wrong channel |
| 02:59.37 | Karen_m | wow i got the polycom 650 setup with 6 different lines and it's a beautiful thing!! so amazing :) |
| 02:59.39 | Karen_m | thanks for your help guys |
| 03:12.19 | p3nguin_ | I'm still wondering where is the proper channel for saying that? |
| 03:13.11 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 03:25.26 | a1fa | does anyone know how to make "register" work via outbound proxy? |
| 03:26.09 | WIMPy | Isn't that explained in the samples? |
| 03:26.53 | a1fa | which ones? |
| 03:26.53 | a1fa | no |
| 03:27.09 | WIMPy | No, indeed it doesn't look like it is. |
| 03:27.18 | a1fa | http://www.voip-info.org/wiki/view/Asterisk+SIP+register |
| 03:28.05 | hardwire | doesn't setting a global outbound proxy work for that? |
| 03:28.09 | WIMPy | user@registrar:password:user@proxy/extension |
| 03:28.12 | a1fa | if TK-DFender was here he would know |
| 03:28.12 | hardwire | whic may not be great |
| 03:29.12 | WIMPy | There is no global outboundproxy. |
| 03:29.54 | a1fa | WIMPy: that did not work |
| 03:30.19 | WIMPy | That's the way I did it when I needed to. |
| 03:31.03 | a1fa | register => 4083133377@sip.broadvoice.com:Z!3fGp71:4083133377@192.168.1.1/202 |
| 03:31.47 | p3nguin_ | You really want broadvoice to send calls to 202? |
| 03:32.14 | a1fa | thats not how it works |
| 03:32.28 | a1fa | inbound calls go to a context |
| 03:34.46 | a1fa | WIMPy: this is sending traffic to |
| 03:34.50 | a1fa | the proxy |
| 03:34.55 | a1fa | still no registration |
| 03:35.05 | a1fa | register => 4083133377@sip.broadvoice.com:Z!3fGp71:4083133377@sip.broadvoice.com@192.168.1.1/202 |
| 03:35.11 | a1fa | maybe i need that |
| 03:36.24 | *** join/#asterisk banditti (~banditti@unaffiliated/banditti) |
| 03:37.23 | banditti | I have asterisk and a polycom IP335. When I pick up the phone to dial, it says enter url. Then if I have to dial a second number, it switches to letters and the call won't go through. Thoughts? |
| 03:38.19 | titter | Disable URL dialing |
| 03:38.25 | titter | It's a Polycom option |
| 03:39.02 | banditti | in the cfg file? |
| 03:39.13 | titter | Yes it's in one of the Polycom cfgs |
| 03:39.40 | titter | The option is in the administrators guide for the Polycom |
| 03:45.40 | a1fa | friggin cluster fuck |
| 03:46.07 | WIMPy | An Orgy? |
| 03:46.13 | a1fa | yes |
| 03:46.27 | a1fa | my QUALIFY is unreachable |
| 03:46.32 | a1fa | but i see calls popping up on the log |
| 03:48.17 | a1fa | [Aug 2 03:47:42] NOTICE[9001]: chan_sip.c:21253 handle_request_invite: Sending fake auth rejection for device "CallerID@Broadvoice" |
| 03:48.18 | a1fa | lol |
| 03:49.24 | banditti | When I make the following change from "1" to feature.9.name="url-dialing" feature.9.enabled="0", I can't dial out. |
| 03:51.25 | a1fa | what does this mean? |
| 03:51.26 | a1fa | [Aug 2 03:50:51] NOTICE[9001]: chan_sip.c:21253 handle_request_invite: Sending fake auth rejection for device |
| 03:52.38 | WIMPy | I think, no mathing user found and you have alwaysauthreject (or what it's called) enabled. |
| 03:55.05 | a1fa | yes, but how why? |
| 03:55.14 | a1fa | it works fine if you take the outboundproxy |
| 03:55.40 | WIMPy | Incoming and outgoing are quite different things. |
| 03:55.55 | a1fa | yes |
| 03:55.58 | a1fa | tru |
| 03:56.24 | WIMPy | You probably have to match them on their IP(s). |
| 03:58.34 | a1fa | where? |
| 03:59.29 | a1fa | which setting is that? |
| 03:59.54 | banditti | on my polycom ip335, I get url dialing. When I make the following change from "1" to feature.9.name="url-dialing" feature.9.enabled="0", I can't dial out. Thoughts? |
| 04:00.24 | a1fa | WIMPy |
| 04:00.47 | WIMPy | Create one peer per IP. |
| 04:01.08 | a1fa | i hardcoded sip.broadvoice.com to 1 ip address |
| 04:01.39 | WIMPy | You need to have all IPs they might be sending from. |
| 04:01.53 | a1fa | they are only sending from one IP address |
| 04:02.09 | WIMPy | There's also the fromdomain, but I'm not sure that can be used as the only match. |
| 04:02.26 | WIMPy | It doesn't seem to macht your peer. |
| 04:02.45 | WIMPy | match |
| 04:03.36 | a1fa | i do have fromdomain |
| 04:03.42 | a1fa | set to sip.broadvoice.com |
| 04:04.10 | WIMPy | Maybe that doesn't match? |
| 04:04.28 | WIMPy | If you have it it's an and condition. |
| 04:05.36 | a1fa | ya |
| 04:05.41 | a1fa | it shoud work |
| 04:05.56 | p3nguin_ | I've never had to use fromdomain. |
| 04:06.18 | a1fa | i'll hardset it to an IP address |
| 04:07.52 | WIMPy | I haven't tried it as only condition. Would make things a lot easier if you're less security minded. |
| 04:09.47 | a1fa | odd.. my firewall will enforce the source, i guess |
| 04:10.14 | a1fa | with both things disabled its still rejecting |
| 04:11.36 | a1fa | fromuser, and fromdomain |
| 04:13.17 | p3nguin_ | I don't know if it's really a security-minded approach that's the problem. |
| 04:13.21 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 04:13.37 | a1fa | security throgh obscurity |
| 04:13.47 | Kalidarn | is there a migration guide for 1.4 to 1.8? |
| 04:13.51 | p3nguin_ | If it's obscurity, it's not security at all. |
| 04:13.52 | a1fa | anyway, i'll dick with it tomorrow |
| 04:13.56 | Kalidarn | just wondering if my configs will be broken |
| 04:13.58 | a1fa | p3nguin_: lol |
| 04:15.40 | p3nguin_ | kalidarn: It depends on how complex your system is. If you're using only chan_sip and extensions.conf, the change won't be all that difficult. |
| 04:16.42 | Kalidarn | yeah im using chan sccp b and i just realised they haven't updated for 1.8 yet |
| 04:16.49 | Kalidarn | but are working on it |
| 04:17.04 | p3nguin_ | Only a few apps have changed syntax. A few of the sip.conf options have changed, but backward compatibility should help you. And you'll see warning messages for most things that need changed which you don't change accordingly. |
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| 04:17.24 | Kalidarn | yeah i figured that p3nguin_ |
| 04:17.59 | p3nguin_ | chan_sccp-b is going to be your downfall. They've been working on that for a while and still don't have the release out. I'm patiently waiting for it. |
| 04:18.53 | p3nguin_ | chan_skinny is barely functional in 1.8 branch, so don't even bother trying it. |
| 04:20.33 | Kalidarn | yup |
| 04:20.37 | Kalidarn | there's a proper check though now |
| 04:20.53 | Kalidarn | for if you're running 1.8 it tells you to go back to 1.6 |
| 04:20.59 | Kalidarn | instead of giving a random compile error |
| 04:21.16 | WIMPy | You will get even more with 10. |
| 04:21.17 | p3nguin_ | If you can get the development branch of chan_sccp-b, and you aren't mission critical, that could get you into 1.8. |
| 04:21.33 | p3nguin_ | More errors? |
| 04:21.47 | Kalidarn | its going to be released pretty soon i think though p3nguin_ development is still active |
| 04:21.50 | WIMPy | yes |
| 04:22.06 | Kalidarn | yeah but 10 wont be out for a while yet |
| 04:22.17 | *** join/#asterisk gokulnath (~gokulnath@115.111.177.122) |
| 04:22.24 | Kalidarn | and i only use the LT versions |
| 04:22.29 | p3nguin_ | me too |
| 04:22.40 | WIMPy | The TRUNK version hass essentially been 10 for month, |
| 04:22.48 | Kalidarn | mostly because i can't afford to have it not working :) |
| 04:22.54 | WIMPy | months |
| 04:25.18 | Kalidarn | i guess ill try again when chan-sccp-b 3.1.0 comes out |
| 04:25.21 | Kalidarn | i think its due soon |
| 04:25.39 | Kalidarn | https://sourceforge.net/mailarchive/forum.php?thread_name=917DC428-5799-4F06-B716-AC2D3C6DED7E%40chenetz.net&forum_name=chan-sccp-b-discussion |
| 04:25.45 | p3nguin_ | I don't really NEED anything that 1.8 provides, so I don't care too much about staying in 1.4. |
| 04:26.46 | Kalidarn | yeah me neither |
| 04:27.05 | Kalidarn | its just more that it's the current version and probably gets more attention |
| 04:27.21 | Kalidarn | so ill update when it's ready but until then :P |
| 04:28.12 | Kalidarn | hey here's an interesting question can you up the default volume of say a 7912 phone (a particular one) on restart of asterisk |
| 04:28.47 | Kalidarn | say someone has a hearing condition :P |
| 04:28.50 | p3nguin_ | not to my knowledge |
| 04:29.10 | Kalidarn | ya you have to do it on the handset i think. but that doesn't seem to save upon restarts |
| 04:29.42 | p3nguin_ | If you remember to press the Save key after changing the volume, it should stay across restarts. |
| 04:29.50 | Kalidarn | it doesn't ;) |
| 04:30.10 | p3nguin_ | It's been a little while since I used a 7912 with SCCP, but I thought it did. |
| 04:31.06 | Kalidarn | okay that is odd i restarted it just then and it did |
| 04:31.07 | p3nguin_ | I know it doesn't stay after firmware changes. |
| 04:31.46 | Kalidarn | i dunno if ill buy cisco phones next time round |
| 04:31.50 | p3nguin_ | If I, say, change to SIP and then back to SCCP, it'll be default values on volume (as well as everything else). |
| 04:32.58 | Karen_m | so I've got a polycom 650 setup with my asterisk. How can I get another phone into the mix? Is there a device that will hook into the polycom 650 that will let me add on cordless phones? I'm concerned when a call comes in and I'm not upstairs in the office... is there a way to get the cordless phones using the polycom 650? any ideas? |
| 04:33.08 | p3nguin_ | I personally like the legacy Cisco phones. |
| 04:33.17 | Kalidarn | im sick of their smartnet bullshit |
| 04:33.46 | Kalidarn | but i haven't really looked around at other handsets |
| 04:33.56 | Kalidarn | probably wont update until i need ipv6 |
| 04:34.01 | p3nguin_ | Using sccp-b, I get almost every feature. The SIP images aren't that great, but since they were built for SCCP, that's to be expected. |
| 04:34.11 | WIMPy | Karen_m: You don;t use a phone on a phone. Get a DECT handset with a SIP base station. |
| 04:34.46 | p3nguin_ | Just add another phone to the system. sip.conf |
| 04:34.57 | p3nguin_ | Then create an extension that calls it. extensions.conf |
| 04:35.06 | Karen_m | WIMPy, know of any phones that have dect+sip base station? |
| 04:35.16 | p3nguin_ | You can do parallel dialing or sequential dialing of two phones. |
| 04:35.28 | Karen_m | i'm happy with polycom, i'd like to stay with a good name |
| 04:36.34 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 04:37.32 | WIMPy | Hasn't Kirk become part of Polycom? |
| 04:40.08 | Karen_m | thanks! kirk is |
| 04:40.12 | Karen_m | very nice, thanks wimpy |
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| 06:18.05 | schmidts | good morning |
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| 06:43.56 | ChannelZ | tally ho |
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| 06:57.15 | PhoenixMage | Hi guys, just wondering if chan-sccp-b works with 1.8 yet? |
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| 07:08.18 | ChannelZ | I hear no |
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| 07:24.35 | Kalidarn | PhoenixMage: no |
| 07:24.43 | Kalidarn | it works with a minimum of 1.6 |
| 07:24.56 | *** part/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 07:25.01 | Kalidarn | and 3.0.1 works with 1.4.27 properly now |
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| 07:31.53 | donatas | hey. is it possible to catch and execute some commands on ending call? |
| 07:31.58 | donatas | I mean when call is ended |
| 07:32.18 | WiretapSeven | donatas, use the 't' exten |
| 07:33.04 | donatas | what is means? |
| 07:36.20 | donatas | I need to catch begining of call and execute some scripts. Also I need the same with ending call. |
| 07:36.46 | *** part/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 07:39.10 | donatas | any ideas? |
| 07:39.56 | schmidts | donatas you can use the M option for dial where you can execute a dialplan Macro when the call is answered |
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| 07:40.39 | schmidts | donatas and for the ending you can use the h extension like this exten => h,1,Noop(call is hungup) |
| 07:43.12 | tully` | hey all, i'm trying to record a call on asterisk 1.6. I'm starting the call from a .call file in /var/spool/asterisk/outgoing and then passing it to a macro extension which then dials the 2nd number. Should I use Record() Monitor() MixMonitor() or something else? I can record with automon by pressing *1 but i'd prefer to do it automatically. |
| 07:44.29 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 07:46.23 | tully` | here's my current macro http://pastie.org/2307862 |
| 07:47.22 | donatas | schmidts: maybe it is possible to enable CDR so, that it will add records real-time? |
| 07:47.37 | donatas | because now it puts an entry after hangup |
| 07:48.47 | schmidts | donatas if you mean the normal CDRs they couldnt be changed in the h extension cause they are allready written then |
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| 07:54.06 | donatas | and how about this endbeforehexten |
| 07:54.38 | schmidts | donatas ? dont know what you mean |
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| 07:58.47 | donatas | schmidts: Normally, CDR records are not closed out until after all extensions are finished executing. By enabling this option, the CDR will be ended before executing the h extension so that CDR values such as end and billsec may be retrieved inside of this extension. Defaults to no. |
| 07:58.55 | nunne_ | problem with the dial option M. i'm calling the dial command with M option from within a macro. and in the first macro i set a channel variable CURSTATUS. and then in the macro called by the M-option that CURSTATUS changes. But in reality it doesnt. when i check the value of CURSTATUS after the M-option macro has finnished it has whatever previous value i set to it. Anyone have any idea why? |
| 08:03.08 | *** join/#asterisk frawd (~francois@60.Red-81-38-88.dynamicIP.rima-tde.net) |
| 08:03.24 | jacc0 | #!@%#$@ cisco support |
| 08:03.26 | schmidts | donatas ok i see ;) |
| 08:04.25 | schmidts | nunne_ AFAIK chan vars are only copied to a macro so you wouldnt get a change back to the original var |
| 08:04.34 | schmidts | jacc0 :D |
| 08:05.21 | jacc0 | I need a cisco contract to report a securty flaw in one of there products |
| 08:05.57 | nunne_ | schmidts: i see, gosub is perhaps a better option for me then? |
| 08:06.30 | schmidts | nunne_ IMHO yes |
| 08:06.42 | nunne_ | thanks for the input :) |
| 08:06.50 | schmidts | your welcome |
| 08:17.42 | nunne_ | schmidts: should gosub work from within a macro? because i'm seeing no results :/ i'm calling U(ans) for example and i have an exten => s,n(ans),XXXXX but it doesnt get executed |
| 08:19.48 | schmidts | nunne_ sorry i have never used gosub, dont know |
| 08:19.59 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 08:28.40 | nunne_ | schmidts: after a google-session it seems like they dont really mix well together. looks like i have to migrate to use only GoSub |
| 08:51.32 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 08:57.21 | kaldemar | nunne_: gosub as a dial option takes a context name as an argument and goes to s,1 in it. |
| 08:57.22 | *** join/#asterisk Manu18 (~megairc@lau18-1-82-246-196-145.fbx.proxad.net) |
| 08:57.24 | *** join/#asterisk coppice (~coppice@m121-202-4-103.smartone-vodafone.com) |
| 08:57.41 | Manu18 | bonjour à tous |
| 08:59.06 | Manu18 | Il y a des français svp ? |
| 09:03.58 | *** join/#asterisk Eye_Trust_No1 (~Eye_Trust@84.203.229.194) |
| 09:06.03 | Manu18 | de quoi cela peut venir que mon asterisk ne veut pas tropver l'hote freephonie? |
| 09:10.52 | *** join/#asterisk timahvo1 (~rogue@41.212.123.197) |
| 09:15.29 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
| 09:15.30 | jacc0 | right now it does : <CiscoIPPhoneExecute><ExecuteItem Priority="0" URL="Softkey:Exit"/><ExecuteItem Priority="0" URL="Softkey:Exit"/> |
| 09:15.46 | *** join/#asterisk nickfennell (~nick@cov1.appliansys.com) |
| 09:15.47 | nickfennell | hi all |
| 09:15.56 | nickfennell | Having some audio issues that I'm trying to debug |
| 09:16.09 | nickfennell | Is there a way to make asterisk tell me what the expected inbound audio port will be? |
| 09:18.21 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:20.37 | *** join/#asterisk prash10x (b75260e6@gateway/web/freenode/ip.183.82.96.230) |
| 09:20.55 | Manu18 | please help me |
| 09:21.18 | prash10x | hello pls help |
| 09:21.30 | prash10x | i get echo with speakers |
| 09:21.45 | prash10x | im using yate soft phones |
| 09:27.37 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 09:32.34 | WiretapSeven | how complete is Asterisk's H323 implementation? |
| 09:36.01 | Manu18 | Il y a des français svp ? |
| 09:39.59 | kaldemar | WiretapSeven: it acts as a gateway and may register itself to a gatekeeper with a prefix. |
| 09:40.13 | WiretapSeven | kaldemar, eh? |
| 09:40.24 | WiretapSeven | you can't use H.323 phones with Asterisk? |
| 09:40.31 | kaldemar | Manu18: this is an english speaking channel. |
| 09:40.49 | kaldemar | WiretapSeven: sure you can, in gateway mode. |
| 09:41.15 | WiretapSeven | kaldemar, I basically don't understand H.323 _at_all_ :P just looking at an Avaya phone that doesn't (And will never) support SIP |
| 09:44.28 | Manu18 | kaldemar, i'm french no speak english |
| 09:45.43 | WiretapSeven | Manu18, #asterisk-fr ? |
| 09:46.22 | Manu18 | non |
| 09:55.05 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 09:57.28 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 11:13.28 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 11:13.28 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 11:13.59 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-180.dsl.net.pk) |
| 11:14.55 | mechbangirc | Any one knows a good CentOS/Fedora AMI on Amazon EC2 secifically for asterisk. |
| 11:15.10 | mechbangirc | specifically*? |
| 11:16.41 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 11:25.08 | k3asd` | nickfennell: of nothing :) |
| 11:34.51 | *** join/#asterisk volker- (~volker@84.201.4.133) |
| 11:34.52 | volker- | hi |
| 11:35.41 | volker- | It seems like "include => foo" will everything from foo totally to the end and not at the point of include |
| 11:37.28 | volker- | but I cant find any documentation about it |
| 11:41.46 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.45) |
| 11:57.20 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:00.33 | leifmadsen | volker-: yes, includes are checked AFTER the dialplan in the local context is checked first |
| 12:01.21 | leifmadsen | volker-: includes are checked, in the order they are added, after all dialplan has been checked in the local context. If you want to control the order, then place the local dialplan into another context and include that in the order you wish it to be matched on |
| 12:02.25 | leifmadsen | volker-: http://ofps.oreilly.com/titles/9780596517342/asterisk-DP-Basics.html#asterisk-DP-Basics-SECT-3.8 |
| 12:02.57 | leifmadsen | ~include |
| 12:04.32 | leifmadsen | infobot: include is <reply> The include => directive in the Asterisk dialplan is matched in the order in which they are listed, only checked after the local dialplan in the context is checked first. For more information see http://ofps.oreilly.com/titles/9780596517342/asterisk-DP-Basics.html#asterisk-DP-Basics-SECT-3.8 |
| 12:04.32 | infobot | leifmadsen: okay |
| 12:04.38 | leifmadsen | ~include |
| 12:04.38 | infobot | The include => directive in the Asterisk dialplan is matched in the order in which they are listed, only checked after the local dialplan in the context is checked first. For more information see http://ofps.oreilly.com/titles/9780596517342/asterisk-DP-Basics.html#asterisk-DP-Basics-SECT-3.8 |
| 12:13.37 | Dovid | anyone from Germany here? |
| 12:14.04 | jglauche_ | I am |
| 12:16.14 | tuxx- | hey guys, does asterisk 1.8 have support for digium isdn cards out of the box? or do i need another library for stuff like that (libPRI and dahdi or something?) |
| 12:16.42 | Dovid | jglauche_: Do you have a number that I can make a test call to ? i need to see if I can send blocked calls to german numbers. I need a "real DID" to test to. |
| 12:17.00 | Dovid | tuxx-: Libpri+DAHDI AFAIK |
| 12:17.50 | jglauche_ | what, you dont have a mobile phone ? :) |
| 12:18.14 | Dovid | jglauche_: I am not in Germany. i need to test to a German number |
| 12:18.51 | tuxx- | tnx Dovid |
| 12:19.56 | Dovid | jglauche_: Mind if I make a few test calls to u ? |
| 12:20.01 | jglauche_ | bad atm. |
| 12:20.12 | jglauche_ | is in moving stress. |
| 12:20.23 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:20.48 | Dovid | ok |
| 12:21.06 | Dovid | jglauche_: So take a train up to Amsterdam (only 3 hours) and smoke some.... |
| 12:21.30 | jglauche_ | that stuff makes me actually feel terrible |
| 12:25.01 | volker- | leifmadsen: thanks a lot |
| 12:25.10 | jacc0 | you don't need to go all the way to amsterdam; just across the border there will be a cofee shop |
| 12:25.22 | volker- | but I dislike the behaviour, makes stuff complicated |
| 12:25.33 | jglauche_ | I heard they won't sell to foreigners |
| 12:25.39 | jglauche_ | new laws |
| 12:25.53 | jacc0 | net jet |
| 12:26.27 | jacc0 | they haven't asked me for ID in the cofee shop in the place where I live; Waalwijk |
| 12:26.55 | *** join/#asterisk wonderworld (~ww@port-92-201-144-138.dynamic.qsc.de) |
| 12:26.57 | leifmadsen | volker-: not really... it's quite straight forward |
| 12:27.01 | leifmadsen | and its' worked like that for years |
| 12:27.11 | leifmadsen | it's also the only way to control the order of matches |
| 12:27.32 | volker- | leifmadsen: i include e.g. the echotest and at the end of the extension I reject everything else for security reason |
| 12:27.33 | leifmadsen | (because asterisk "sorts" pattern matches) |
| 12:29.16 | volker- | err, not rejecting, require authentification |
| 12:29.50 | Manu18 | French ? |
| 12:32.13 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:32.58 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 12:39.48 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-yivgbfhpjyqolbnx) |
| 12:40.49 | Manu18 | ya personne pour un sucis avec asterisk ? |
| 12:41.01 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
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| 12:47.51 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 12:47.52 | nunne_ | does anyone know if there is maximum row length in extensions.conf (asterisk1.4) .. because i'm seeing wierd missing )}-errors etc.. and I can see no errors what so ever in the string. |
| 12:48.11 | *** join/#asterisk livemoon (deadsun@115.174.1.214) |
| 12:48.31 | irroot | hi there folks been abducted by the busy |
| 12:58.49 | tzafrir | nunne_, maybe 80? |
| 13:05.07 | Manu18 | : chan_sip.c:12654 sip_reg_timeout: -- Registration for '0952966838@freephonie.net' timed out, trying again (Attempt #350) |
| 13:05.41 | Manu18 | Of or the problem can come? |
| 13:14.42 | *** part/#asterisk prash10x (b75260e6@gateway/web/freenode/ip.183.82.96.230) |
| 13:16.37 | *** join/#asterisk GreyFoxx (~greg@mythtv/developer/GreyFoxx) |
| 13:17.22 | GreyFoxx | I don't suppose anyone here could point me to a link where I could download the linux version of the aastra 'anacrypt' tool for encrypting config files ? |
| 13:20.06 | GreyFoxx | And of course seconds later I find it... nevermind :) |
| 13:20.32 | volker- | is there a way to use TLS for sip communication, where the client needs no certificate? |
| 13:30.16 | *** join/#asterisk timahvo1 (~rogue@41.212.123.197) |
| 13:47.54 | leifmadsen | nunne_: what does the string look like? And yes, there is a length I believe of 253 chars |
| 13:48.13 | volker- | looks like the whole tls support is a little bit buggy |
| 13:50.13 | volker- | if I set "transport=tls,udp" to a friend (which is dynamic) asterisk wont start due of "tcptls.c: Unable to connect SIP socket to IP:5060: Connection refused", where the IP is known from the last connect of it.. |
| 13:51.06 | leifmadsen | nunne_: actually I was thinking of the context name (which is 79 chars, 80 chars - 1 terminating null) |
| 13:51.28 | leifmadsen | nunne_: there is certainly a maximum length, but I forget what it is in 1.4, I think in 1.8 and greater it is something like 4096 or something |
| 13:51.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:53.05 | Dovid | anyone know when 1.10 is gona be released ? like whats the target date? |
| 13:53.27 | leifmadsen | ~asteriskreleases |
| 13:53.31 | leifmadsen | ~asteriskrelease |
| 13:53.33 | leifmadsen | gah |
| 13:53.41 | leifmadsen | it's documented on the wiki |
| 13:53.52 | leifmadsen | ~asteriskversions |
| 13:54.02 | leifmadsen | I never remember the bloody tag |
| 13:54.04 | leifmadsen | Dovid: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:54.44 | Dovid | leifmadesn: Thanks |
| 13:55.41 | Dovid | why is 1.8 going to have a longer life than 1.10? |
| 13:56.07 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 13:56.20 | chazzam | Dovid: there is no 1.10 |
| 13:56.30 | chazzam | and see the second paragraph |
| 13:56.32 | jacc0 | asterisk 10 it iz |
| 13:56.53 | Jcook_5xData | anyone any experience with polycom 650 and remove the echo on the handset |
| 14:00.48 | tzafrir | ~versions |
| 14:00.51 | leifmadsen | Dovid: because 1.8 is a LTS release |
| 14:01.02 | tzafrir | ~asteriskversions |
| 14:01.02 | leifmadsen | ~asteriskversioning |
| 14:01.02 | infobot | well, asteriskversioning is http://www.asterisk.org/asterisk-versions |
| 14:01.07 | leifmadsen | that's it :) |
| 14:01.11 | Dovid | what is LTS? |
| 14:01.17 | leifmadsen | Long Term Support |
| 14:01.18 | tzafrir | Long Term Support |
| 14:01.20 | tzafrir | ~lts |
| 14:01.20 | infobot | methinks lts is Leaning Toothpick Syndrome. |
| 14:01.25 | leifmadsen | the page totally tells you what LTS is |
| 14:01.30 | leifmadsen | which I linked several minutes ago |
| 14:01.55 | leifmadsen | it's described in the 2nd paragraph |
| 14:02.10 | tzafrir | infobot, lts is also Long Term Support: a version that is supported for a longer time period. |
| 14:02.10 | infobot | tzafrir: okay |
| 14:02.22 | leifmadsen | and the graphic on the page even shows the life cycle of the versions |
| 14:02.34 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:03.20 | jacc0 | lol @infobot |
| 14:04.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:04.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:06.35 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 14:13.19 | *** part/#asterisk clintc (~clintc@n128-227-109-39.xlate.ufl.edu) |
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| 14:19.36 | cusco | ChannelZ: I didn't |
| 14:19.37 | cusco | hi |
| 14:19.46 | cusco | came back now to research about it |
| 14:20.10 | cusco | so I still have 1 cable connected to the second PRI card |
| 14:20.26 | cusco | dahdi_tool says that all 4 spans on that card are 'internally clocked' |
| 14:20.30 | cusco | now I want to change that |
| 14:20.31 | cusco | argh! |
| 14:20.53 | cusco | how can I change that? |
| 14:21.14 | *** join/#asterisk sourcode (~code@ppp-58-8-211-220.revip2.asianet.co.th) |
| 14:25.13 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
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| 14:26.52 | Kobaz | anyone have an example of having different rings on a polycom depending on certain conditions... ie: external call vs internal call |
| 14:28.59 | raden | Katty, :D :D :D :D |
| 14:29.10 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:35.52 | *** join/#asterisk saisoma (~saisoma@client105.jdcc.edu) |
| 14:38.15 | Katty | ohai |
| 14:38.17 | Katty | hugs raden |
| 14:38.33 | saisoma | hey guys, ver 1.8.5.0, i want to automate updating a voicemail message on a remote system. there is, of course, a login sequence. I'm using call files to generate the call. how can i send the digits to the remote end? I've tried senddtmf in the context and the D option on dial, had it dial my phone for testing, but all i get is silence |
| 14:38.52 | raden | how is Katty ? long time noo typee.... |
| 14:40.09 | Katty | bout the same as always |
| 14:40.11 | Katty | how're you raden |
| 14:40.48 | raden | Pretty good working on buying another house or duplex for an investment |
| 14:40.54 | raden | lil girl just turned 4 |
| 14:41.59 | raden | still single lol ... that will probably never change |
| 14:42.48 | jaytee | Kobaz, I'm doing that with Polycoms. |
| 14:43.54 | raden | Katty, catch yea later dear I have a meeting in 15 min I forgot about :( take it easy :D :D :D |
| 14:49.32 | jaytee | Kobaz, here's how I'm doing it: http://pastebin.com/GXGtuEE8 |
| 14:49.57 | *** join/#asterisk theron (~theron@ip244.scolloc.lh.net) |
| 14:57.32 | *** join/#asterisk TeknoJuce (~TeknoJuce@xbmc/staff/TeknoJuce) |
| 14:58.41 | Kobaz | ah very cool |
| 15:00.48 | Kobaz | http://runforyourlives.com/# |
| 15:00.54 | saisoma | hey guys, ver 1.8.5.0, i want to automate updating a voicemail message on a remote system. there is, of course, a login sequence. I'm using call files to generate the call. how can i send the digits to the remote end? I've tried senddtmf in the context and the D option on dial, had it dial my phone for testing, but all i get is silence |
| 15:01.21 | leifmadsen | saisoma: your question is still on my screen |
| 15:01.33 | leifmadsen | D() option should work fine, I've used it many times |
| 15:04.10 | volker- | come on, the official asterisk-1.8 package for debian is not compiled with srtp support!?! |
| 15:04.37 | saisoma | leifmadsen: thanks. rgr. i'll see if I can figure out what's wrong. If not, I'll do a pastebin of my test. Thanks!! |
| 15:07.58 | pabelanger | volker-: which OS |
| 15:08.16 | pabelanger | We don |
| 15:08.20 | pabelanger | heh |
| 15:08.29 | volker- | pabelanger: debian |
| 15:08.35 | pabelanger | version? |
| 15:08.53 | volker- | stable - 6.0.2 |
| 15:09.45 | volker- | Asterisk 1.8.3.3-1digium1~squeeze |
| 15:11.37 | pabelanger | volker-: upgrade to 1.8.4+ |
| 15:12.19 | pabelanger | it was add in 1:1.8.4-1digium1~squeeze |
| 15:12.31 | volker- | ok, thanks |
| 15:15.09 | saisoma | OK, still stumped:(. pastebin on trying to send digit tones through automated call file here: http://pastebin.com/N1bHbJzZ |
| 15:17.29 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 15:22.08 | cusco | hi |
| 15:22.18 | cusco | so I'm still stumped |
| 15:22.31 | cusco | two PRI Digium cards, each 4 span |
| 15:22.42 | cusco | the first is fine, the second shows that timing source: internal clock |
| 15:23.01 | cusco | I'm not able to set the timing source on spans 5 to 8 |
| 15:24.37 | cusco | thing is, as soon as I restart dahdi, spans from 1 to 4, are yellow and 5 to 8 OK. |
| 15:24.40 | cusco | then dmesg shows: |
| 15:24.52 | cusco | http://paste.debian.net/124879/ |
| 15:25.25 | *** join/#asterisk nfi|ermes (~sdff@host25-9-static.224-95-b.business.telecomitalia.it) |
| 15:25.35 | nfi|ermes | hi all |
| 15:25.47 | nfi|ermes | can anyone help me to debug a module problem ? |
| 15:27.54 | *** join/#asterisk anskywalker (~anskywalk@200.69.247.189) |
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| 15:39.24 | anskywalker | hello... i got an asterisk with 3 sip trunks from the same external server. i made a peer with the ip of that external server as hostname. so the calls for the 3 diferent trunks go to the context of that peer. how can i make a peer that matches the calls that comes for only one of the trunks? (the calls for the 3 trunks will come ever from the same server with the same ip addres) |
| 15:40.46 | *** join/#asterisk kdekorte (~kdekorte@63-163-137-113.iprev.kci.net) |
| 15:40.59 | dym | anskywalker: make 3 peer accounts in sip.conf |
| 15:41.05 | dym | and set context=XXX for each |
| 15:42.06 | anskywalker | ok... i had that, but the calls goed to the default context |
| 15:42.39 | anskywalker | because the registration ip address of the external sip server |
| 15:42.53 | chazzam | match a DID that each trunk is dialing? |
| 15:43.05 | anskywalker | its different than the ip address that make the calls |
| 15:46.34 | anskywalker | how can i see the recived calls's did in asterisk console? |
| 15:46.58 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 15:47.08 | anskywalker | because i made some tries and it never matched |
| 15:47.39 | *** join/#asterisk irroot (~irroot@41.55.91.229) |
| 15:47.44 | neurosys | Can anyone recommend a [right] click to dial app that runs with IE/windoze and *?? |
| 15:47.47 | anskywalker | im not writing peer settings properly |
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| 16:00.05 | volker- | strange. my client is not connected to the asterisk pbx due of lack of sip-tls support. but its able to call out |
| 16:00.56 | volker- | so unencrypted authentication does work, even if transport= is set to tls |
| 16:03.27 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-243-205.w81-51.abo.wanadoo.fr) |
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| 16:06.28 | dym | anskywalker: dont pm |
| 16:06.41 | dym | people on channel can help you. |
| 16:07.39 | anskywalker | ok sorry |
| 16:09.05 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 16:15.41 | volker- | I don't get it. transport= in sip.conf should say which SIP transport is allowed. If it is not allowed by it, there shouldn't be any password authentication |
| 16:17.10 | *** join/#asterisk WiretapWork (~wiretap@unaffiliated/wiretap) |
| 16:17.27 | *** join/#asterisk cyborg-one (1000@188-115-162-27.broadband.tenet.odessa.ua) |
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| 16:30.05 | p3nguin_ | anskywalker: What does "sip trunk" mean to you? |
| 16:31.13 | irroot | p3nguin_ elephants sip water with trunks :P |
| 16:31.21 | irroot | eve folks 10% here |
| 16:31.55 | Kobaz | to most people, it means a sip link supporting multiple channels |
| 16:32.33 | p3nguin_ | I'm not asking about most people; I'm asking what it means to anskywalker so I can determine what he really needs to do. |
| 16:32.56 | p3nguin_ | He said he has three sip trunks from the same provider. |
| 16:33.08 | p3nguin_ | So I need anskywalker to tell me what he thinks that means. |
| 16:33.31 | Kobaz | sip accounts? |
| 16:33.39 | Kobaz | true, there's some ambiguity |
| 16:33.41 | p3nguin_ | I'm asking anskywalker. |
| 16:33.44 | Kobaz | i know |
| 16:33.47 | Kobaz | i'm filling in for him |
| 16:37.53 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 16:40.01 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 16:49.04 | cusco | I just found a mispelling in a README :p |
| 16:49.44 | p3nguin_ | I've found several errors in them, but no one else cares. |
| 16:50.26 | p3nguin_ | I also found a misspelling in your complaint about misspelling. |
| 16:51.36 | cusco | crawls into a whole |
| 16:53.32 | cusco | hole |
| 16:57.20 | chuckf | a whole hole? |
| 16:57.55 | tzafrir | cusco, what mispellings? what version? |
| 16:58.27 | p3nguin_ | misspelling |
| 16:58.38 | p3nguin_ | What irony. |
| 16:59.03 | leifmadsen | p3nguin_: which bug reports? |
| 16:59.15 | p3nguin_ | mispelling... when you are pelling something incorrectly |
| 16:59.23 | p3nguin_ | leifmadsen: What bug? |
| 16:59.45 | leifmadsen | p3nguin_: the ones you reported for the misspellings you're saying no one cares about |
| 17:00.00 | tzafrir | leifmadsen, there's there's a reference to doc/tex/asterisk/ right at the start of the README |
| 17:00.14 | leifmadsen | tzafrir: ok, feel free to assign the issue to me |
| 17:01.09 | cusco | tzafrir: in latest dahdi, but its really nbot important |
| 17:01.33 | cusco | im just wondering, while reading the readme, /proc/dahdi/1 should have text in it, right? |
| 17:01.39 | cusco | /proc/dahdi/1: empty |
| 17:01.43 | tzafrir | cusco, latest: 2.4.1.2 or svn trunk? |
| 17:01.57 | cusco | ../dahdi-linux-complete-2.4.1.2+2.4.1# |
| 17:02.11 | cusco | its really not important |
| 17:02.18 | tzafrir | That's not really the latest, but the README didn't really change since in 2.4 |
| 17:02.23 | cusco | ok |
| 17:02.37 | cusco | im having real bad issues with dahdi atm |
| 17:02.44 | tzafrir | dahdi-linux, right? There's also a README in dahdi-tools |
| 17:02.51 | p3nguin_ | leifmadsen: I didn't file any bug reports for text documents. When I see one, I mention it here and the most common response has something to do with how it isn't important or it's not a priority. If you want bug reports for misspellings and grammatical errors in the docs, I'll probably file in the future. |
| 17:02.56 | cusco | it was the one in dahdi tools tzafrir |
| 17:02.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:03.09 | cusco | and its a ./configure mispelled as ./ocnfigure |
| 17:03.14 | cusco | nvm |
| 17:03.20 | tzafrir | cusco, yes, it should have at least a "title" line |
| 17:03.28 | p3nguin_ | I find typing errors in sample files and docs from time to time. |
| 17:03.35 | cusco | tzafrir: all 8 are empty |
| 17:03.46 | cusco | :( |
| 17:03.48 | cusco | why? |
| 17:04.49 | tzafrir | ispell wants to replace DAHDI with MAHDI |
| 17:05.22 | cusco | hehe whatever mahdi means .. |
| 17:05.43 | cusco | my problem: all 4 spans on second card show as 'internally clocked' |
| 17:06.20 | cusco | but all 4 spans from 1st card are fine |
| 17:07.32 | cusco | dahdi_scan|grep syncsrc - 1st four set to 1, and other 4 set to 0 |
| 17:08.03 | cusco | I can only mess with it later at 3 am or so.. its a production server |
| 17:08.07 | leifmadsen | p3nguin_: ok, because if it isn't reported it's something that someone isn't likely to drop what their doing to fix. When they say, "not important" I imagine it's in reference to, "It's not important to me this very second" which is the only time something is going to get fixed if it is mentioned here |
| 17:08.57 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 17:08.59 | leifmadsen | if you guys find issues with text documents, please just file an issue and assign it to me as I'd be happy to fix them, but it's unlikely I'm going to drop what I'm doing to fix a misspelling unless I'm already working on the document -- I'd rather batch a bunch of fixes in one commit. |
| 17:09.42 | p3nguin_ | I also didn't want to clog up the bug tracker with teaching people how to write English more betterer. |
| 17:09.56 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 17:10.40 | ruben23 | hi guys i have asterisk server and used voip to call but the problme is on the client end which i call there volume is very low, what could be an issue with it..? |
| 17:10.43 | ruben23 | any idea..? |
| 17:11.26 | tzafrir | cusco, http://en.wikipedia.org/wiki/Mahdi |
| 17:12.05 | paulc | I'll take DAHDI over Mahdi any day of the week :-) |
| 17:12.08 | leifmadsen | p3nguin_: I like grammar nazi's, so feel free to create issue s:) |
| 17:12.15 | chuckf | p3nguin_: but that's what the bug tracker is for. worst case is Leif looks at it after you submit it and says 'I'm not gonna deal with this'. |
| 17:12.22 | p3nguin_ | leifmadsen: nazis |
| 17:12.27 | leifmadsen | suck it ;) |
| 17:12.30 | p3nguin_ | :D |
| 17:15.21 | paulc | leifmadsen: Caller A connected to agent B via Queue. If Caller A gets redirected somewhere (currently via console, ultimately via AMI, functionality=same or different?), you wouldn't expect the h extension to fire in the redirected-to context, for caller A, right? |
| 17:16.12 | tzafrir | cusco, I can't find "ocnfigure" in dahdi-tools |
| 17:16.14 | leifmadsen | ummm.... hmmmm I'm not sure -- I think I would expect 'h' to fire at the end of the call in the context that the channel is executing |
| 17:16.32 | leifmadsen | paulc: I would expect the functionality to be the same, yes |
| 17:17.16 | paulc | leifmadsen: I've only tested with console "channel redirect", need to test with actual AMI to confirm.. but I can reproduce the issue with h firing at the wrong time (and not at the end of the call) |
| 17:17.45 | paulc | it's work-around-able by redirecting to a context that just Goto's another, and the h in THAT context fires right at the end of the call.. but it seems "wrong", doing what it's doing |
| 17:18.06 | Qwell | tzafrir: ./ocnfig.status |
| 17:18.25 | paulc | I'll test with AMI then file a bug if it still seems wrong.. |
| 17:20.33 | tzafrir | OK |
| 17:21.10 | Qwell | tzafrir: While you're there, there's also a 'th' |
| 17:21.29 | Qwell | "onto a subtree rather than on th real system." |
| 17:22.27 | tzafrir | Why didn't ispell find that? |
| 17:22.41 | Qwell | tzafrir: Thorium |
| 17:23.07 | p3nguin_ | an element? |
| 17:23.34 | tzafrir | th |
| 17:23.39 | tzafrir | Th, actually |
| 17:23.41 | p3nguin_ | Then it should be Th rather than th, and ispell still should have complained. |
| 17:23.43 | p3nguin_ | yeah |
| 17:24.07 | *** join/#asterisk gravin (~gravin@61.76.50.60.brf01-home.tm.net.my) |
| 17:24.18 | Qwell | shrugs |
| 17:24.35 | Qwell | Maybe for things like "4th" |
| 17:24.50 | Qwell | Probably an ispell quirk |
| 17:25.02 | p3nguin_ | It really takes several humans who knows how to spell things to catch all the errors. |
| 17:25.16 | Qwell | "rd" is valid, "nd" is not. hrm, maybe not then |
| 17:25.39 | *** join/#asterisk Karen_m (~karen@d50-99-60-236.abhsia.telus.net) |
| 17:25.51 | p3nguin_ | I once saw "1th" written on a web page. |
| 17:26.21 | Qwell | I would be more impressed if it was spelled out as "oneth" |
| 17:26.29 | p3nguin_ | firth? |
| 17:26.37 | Karen_m | hello everyone, is there a way to use non-sip-based dect phones, plugging them into the polycom somehow and have it extend to the cordless? Last night a guy suggested the kirk polycom's, which is what I'm looking at but i want to make sure there isn't a module or addon that we can get to do this with the polycom 650 |
| 17:26.45 | p3nguin_ | People puzzle me. |
| 17:26.57 | Qwell | Karen_m: no.. |
| 17:27.03 | p3nguin_ | karen_m: You don't connect phones on other phones. |
| 17:27.23 | p3nguin_ | (2334.46) <p3nguin_> Just add another phone to the system. sip.conf |
| 17:27.24 | p3nguin_ | (2334.56) <p3nguin_> Then create an extension that calls it. extensions.conf |
| 17:27.26 | p3nguin_ | (2335.15) <p3nguin_> You can do parallel dialing or sequential dialing of two phones. |
| 17:27.28 | Karen_m | well with the USB jack in the back of the phone, I thought maybe they had a usb->old phone module or something lol |
| 17:27.37 | Qwell | no |
| 17:27.57 | Karen_m | ok thakn you guys, i just wanted to make sure before money was spent :) |
| 17:28.14 | p3nguin_ | Either way, you'll be buying the phone. |
| 17:28.29 | p3nguin_ | Just use it properly instead of trying to piggy back it onto another phone. |
| 17:28.55 | Karen_m | which line would you recommend from polycom, the kirk or spectralink? |
| 17:29.10 | Karen_m | reason i ask, someone mentioned the polycom soundpoint 650 and it is amazing :) |
| 17:29.15 | Karen_m | so, i'm asking the pro's :) |
| 17:29.18 | p3nguin_ | What happened to SoundPoint? |
| 17:29.32 | cusco | tzafrir: sorry: it reads: ./ocnfig.status --recheck |
| 17:29.46 | *** join/#asterisk jkroon (~jkroon@dsl-241-238-75.telkomadsl.co.za) |
| 17:29.51 | p3nguin_ | You don't want another SP? |
| 17:29.59 | tzafrir | cusco, yeah. Fixed in trunk |
| 17:30.07 | cusco | ok |
| 17:30.16 | cusco | Im still wondering why my /proc/dahdi/1 is empty |
| 17:30.18 | Karen_m | Soundpoint has a dect phone? |
| 17:30.26 | cusco | I guess that is part of my current problem |
| 17:30.31 | Karen_m | i need cordless :) |
| 17:30.46 | tzafrir | cusco, what device do you have? |
| 17:30.51 | p3nguin_ | I don't know what they offer because I don't use Polycoms. |
| 17:30.59 | tzafrir | Any interesting messages in the kernel logs? |
| 17:31.08 | tuxxie | how can i check to see if a number starts with 71 and is only 4 digits long from with in a micro. ie. macro(test,callerid,71xx,....) |
| 17:32.35 | tuxxie | or mocro(test,$(callerid),71xx) |
| 17:33.58 | Karen_m | so if i have a dect phone, and the polycom 650 setup to use the same DID... if I initiate a call with the cordless, with the polycom 650 detect the call and let me lift the receiver and hear the conversation without having to transfer to another extension? |
| 17:36.18 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 17:37.45 | *** join/#asterisk Bipul (~bipul@unaffiliated/bipul/x-4918593) |
| 17:42.56 | cusco | tzafrir: pci:0000:02:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen) |
| 17:42.57 | cusco | pci:0000:04:01.0 wct4xxp+ d161:0410 Wildcard TE410P (4th Gen) |
| 17:43.18 | cusco | tzafrir: dmesg shows plenty |
| 17:44.03 | cusco | http://paste.debian.net/124886/ |
| 17:44.26 | cusco | right now I have both telcos on 1st PRI card |
| 17:44.38 | cusco | and 1 cable on 2nd PRI card |
| 17:44.51 | cusco | I will uninstall dahdi and re-install latter tonight |
| 17:44.54 | cusco | and see what happens |
| 17:45.12 | cusco | I suspect that make uninstall was not performed before installing latest release |
| 17:45.50 | _Corey_ | Karen_m: When you lift the receiver on your Polycom in your scenario you will hear dialtone... |
| 17:46.23 | _Corey_ | one has nothing to do with the other |
| 17:46.50 | Karen_m | how do you get them 'joined', always requiring a transfer? |
| 17:47.31 | _Corey_ | Transfer doesn't join calls... you should probably research your phone's conference button |
| 17:49.44 | paulc | Karen_m: "Joining" phones like that is an analog thing.. with SIP you'd have to transfer and/or conference the calls.. (I'm trying to think if you could do some cleverness with shared line appearances, but my brain's failing me this morning) |
| 17:49.55 | p3nguin_ | karen_m: No, IP phones do not work this way. When you do a parallel dial, both phones will ring, but the first one that you pick up wins the call. If you then want the other phone to have the call, you can either transfer the call or park the call and pick it up on the other phone. |
| 17:51.03 | Karen_m | oh i see, thanks paul! |
| 17:51.10 | *** join/#asterisk MrNemus (~MrNemus@firewall.drgutah.com) |
| 17:56.35 | paulc | Karen_m: I have to run to a meeting, but back in 30 or less.. if you want to talk more about your setup/requirements, we can chat |
| 18:02.06 | Karen_m | ok thank you! |
| 18:08.24 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 18:08.57 | *** join/#asterisk ickmund_ (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 18:19.03 | tuxxie | I am using the following "exten => s,2,GotoIf($["${CALLERID(num):0:3}" = "310"]?local,7139,1)" to redirect calls starting with 310 to a local extension. Is there a way to say if callerid(num) is 310 and not larger than 4 digits? |
| 18:20.29 | anskywalker | p3nguin_: sorry, i was out of the office... i have 3 sip accounts in an X external servers and i configured it as peers to make calls trough it and to recive calls |
| 18:21.48 | p3nguin_ | anskywalker: If you use type=peer in each sip account, it will match by IP/port. If you use type=user or type=friend, it will match the incoming calls by username or defaultuser (depending on asterisk version). |
| 18:22.40 | p3nguin_ | anskywalker: Why are you using three SIP accounts from the same host? |
| 18:23.16 | anskywalker | here we have a lot of sip accounts from a local server |
| 18:24.20 | anskywalker | we have to link some of that sip accounts to our asterisk server |
| 18:26.11 | anskywalker | i configured the server with asteriskgui |
| 18:26.20 | anskywalker | this is one of the sip trunk entries |
| 18:26.21 | anskywalker | <PROTECTED> |
| 18:26.22 | anskywalker | <PROTECTED> |
| 18:26.22 | anskywalker | <PROTECTED> |
| 18:26.22 | anskywalker | <PROTECTED> |
| 18:26.22 | anskywalker | <PROTECTED> |
| 18:26.22 | anskywalker | <PROTECTED> |
| 18:26.26 | anskywalker | <PROTECTED> |
| 18:26.28 | anskywalker | <PROTECTED> |
| 18:26.30 | anskywalker | <PROTECTED> |
| 18:26.32 | anskywalker | <PROTECTED> |
| 18:26.34 | anskywalker | <PROTECTED> |
| 18:26.36 | anskywalker | <PROTECTED> |
| 18:26.38 | anskywalker | <PROTECTED> |
| 18:26.40 | anskywalker | <PROTECTED> |
| 18:26.42 | anskywalker | <PROTECTED> |
| 18:26.44 | anskywalker | <PROTECTED> |
| 18:26.46 | anskywalker | <PROTECTED> |
| 18:26.48 | anskywalker | <PROTECTED> |
| 18:26.49 | serafie | anskywalker: ~pastebin |
| 18:26.56 | p3nguin_ | anskywalker: Next time use a pastebin instead of flooding the channel with crap that most of us here don't want to see. |
| 18:26.58 | serafie | grr. pastebin.com |
| 18:26.59 | KavanS | christ |
| 18:27.01 | KavanS | pastebin * 8 |
| 18:27.10 | KavanS | I literally just woke up from a coma from all that txt |
| 18:27.11 | KavanS | thank you! |
| 18:27.21 | KavanS | I'm cured! :) |
| 18:27.27 | anskywalker | okey sorry, im new wit this |
| 18:28.09 | anskywalker | http://pastebin.com/PV9naZG4 |
| 18:28.14 | anskywalker | there you have |
| 18:30.33 | anskywalker | all the calls come from "190.xxx" address |
| 18:30.50 | serafie | anskywalker: is that from users.conf? There is no "type" line at all. |
| 18:31.48 | anskywalker | yes, it's in users.conf |
| 18:31.55 | p3nguin_ | ~users.conf |
| 18:31.55 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 18:32.11 | _Corey_ | haha |
| 18:32.12 | _Corey_ | nice |
| 18:32.47 | serafie | You will have to paste more than that. |
| 18:34.16 | anskywalker | with the standard trunk configuration, all the calls go to the default context |
| 18:35.02 | anskywalker | i created another trunk with "190.xxx" ip address as hostname (the ip from the calls come) |
| 18:35.22 | anskywalker | and now the calls recived go to that trunk context |
| 18:36.13 | anskywalker | now i want to recive the calls from one trunk in an particular context |
| 18:36.25 | anskywalker | and i don't know how |
| 18:36.29 | paulc | karen_m: And I'm back :-) |
| 18:38.05 | p3nguin_ | anskywalker: I already told you how. |
| 18:38.11 | p3nguin_ | (1321.48) <p3nguin_> anskywalker: If you use type=peer in each sip account, it will match by IP/port. If you use type=user or type=friend, it will match the incoming calls by username or defaultuser (depending on asterisk version). |
| 18:38.51 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 18:38.57 | Karen_m | paulc, i'm using voip.ms to connect both subaccounts to each line and created a ring-group, which works on the incoming. The outgoing, i don't know how to join them yet :(. I haven't ordered the phone cordless phone yet, but i guess I just hit join and it lets me say line2 or something |
| 18:39.14 | p3nguin_ | So use type=user or type=friend for each account on that same host. You'll be able to have multiple accounts from that same IP address and match by the user name. |
| 18:39.45 | p3nguin_ | karen_m: Are you using Asterisk? |
| 18:39.50 | anskywalker | ok, will try |
| 18:40.00 | Karen_m | they are, yes |
| 18:40.05 | p3nguin_ | karen_m: Why do you use subaccounts in voip.ms? |
| 18:40.07 | anskywalker | p3nguin_: lots of thanks! (= |
| 18:40.50 | paulc | karen_m: So what do you want to do on outbound? Place a call out, but have multiple local phones in on the conversation? |
| 18:41.36 | Karen_m | they have a restriction on how many lines per account |
| 18:42.10 | p3nguin_ | Huh? Who has this restriction? |
| 18:42.36 | p3nguin_ | If you use Asterisk, you only need one account on VoIP.ms to send all calls to your Asterisk system. |
| 18:42.41 | Karen_m | on outbound, i want to make a call on either phone and be able to lift the other receiver to switch back and forth |
| 18:42.45 | jeffspeff | i'm having trouble receiving a fax... It appears to me that the problem is where it does an Auto fallthrough after answering the call... Any help would be greatly appreciated... http://pastebin.com/W6snPgdC <=== the sip debug info http://pastebin.com/t18N9U1i <=== the part of extensions.conf for inbound and faxing. thanks |
| 18:43.46 | paulc | karen_m: You're not going to be able to do that without either transferring, or parking and retrieving. |
| 18:44.04 | p3nguin_ | karen_m: You have already been given the answer about that. IP phones DO NOT work that way. You can either transfer the call to the other phone, or you can park the call and pick it up from the other phone. It is not possible to pick up a random second handset and join a call in progress like an analog phone does. |
| 18:44.10 | Karen_m | parking and retrieving means, put on hold and then open the other line? |
| 18:44.19 | p3nguin_ | Parking means PARK the call. |
| 18:44.26 | Karen_m | ok |
| 18:44.35 | p3nguin_ | Park it, then pick it up from another handset. |
| 18:44.56 | paulc | karen_m: Yes. Transfer = Hit transfer on phone A, dial phone B's extension number, hangup or announce it then hangup. Parking = hit transfer, dial 700, listen to the digits read back (7.. 0.. 1..) then go to phone B and dial 701 to pick up the call (typically) |
| 18:45.06 | p3nguin_ | phone A parks the call. It goes into a parking space, and asterisk tells you where it is. |
| 18:45.17 | Karen_m | paulc, oh i see! |
| 18:45.22 | p3nguin_ | phone B then calls the parking space number to pick up the call. |
| 18:45.43 | Karen_m | neat, very neat |
| 18:45.46 | Karen_m | thank you gentlemen |
| 18:46.29 | leifmadsen | Karen_m: http://ofps.oreilly.com/titles/9780596517342/asterisk-SysAdmin.html |
| 18:52.18 | *** join/#asterisk vfabi (~fabi@host-static-109-185-198-45.moldtelecom.md) |
| 18:55.27 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:00.57 | *** join/#asterisk hexanol (~etienne@modemcable094.94-70-69.static.videotron.ca) |
| 19:03.20 | *** part/#asterisk MrNemus (~MrNemus@firewall.drgutah.com) |
| 19:07.43 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-susvdwbvowmeovfp) |
| 19:09.48 | Qwell | gogasca: spy! :p |
| 19:09.53 | Qwell | (I kid) |
| 19:10.17 | gogasca | qwell: why ? |
| 19:10.24 | Qwell | nothing |
| 19:10.31 | gogasca | ok :) |
| 19:18.08 | *** join/#asterisk epart (~Administr@121.97.63.210) |
| 19:18.13 | epart | hi |
| 19:18.19 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:18.47 | epart | any body knows about astGUIclient |
| 19:20.10 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:21.53 | tzafrir | epart, IIRC it is an unmaintained software |
| 19:25.03 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 19:25.50 | telnettech | Question.....what are some of the web based GUI for montioring ACD queues on Asterisk that people are using???? |
| 19:26.52 | epart | wat do u mean?? |
| 19:28.36 | telnettech | I have an ACD setup using asterisk. I am looking for something that is web based so that managers can log in and montor productivity of their departments |
| 19:29.15 | telnettech | i dont want to get some software that requires me to rebuild the ACD |
| 19:29.59 | telnettech | i just want it to take the Asterisk queues and the MySQL database info to show the stats for the queues to management |
| 19:30.51 | leifmadsen | telnettech: I've had pretty good luck with QueueMetrics |
| 19:30.51 | Nugget | telnet is eeeeeeevil! |
| 19:32.35 | telnettech | thank you leifmadsen |
| 19:32.43 | telnettech | any others out there????? |
| 19:32.55 | leifmadsen | none that I have used and liked |
| 19:33.19 | leifmadsen | also the people at Loway have been fairly impressive in terms of support |
| 19:33.58 | leifmadsen | I also built and tested against a 1.6.2 based system early on, adding it after building the system (because the client didn't request any interface prior) and it worked out quite well |
| 19:36.08 | *** join/#asterisk joeflyde (~Gregory@63.239.202.68.cfl.res.rr.com) |
| 19:37.10 | telnettech | leifmadsen: Does it require to be built on the same asterisk server or is it installed on another server that points back to the asterisk server??? |
| 19:37.27 | leifmadsen | telnettech: why so many question marks? |
| 19:37.42 | telnettech | just curious |
| 19:37.56 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:38.00 | leifmadsen | I understand, but why are you finishing so sentences like you're being chased by a coyote? |
| 19:38.07 | leifmadsen | s/so/your |
| 19:38.29 | leifmadsen | telnettech: http://queuemetrics.com/faq.jsp |
| 19:43.06 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:43.52 | telnettech | thank leif....i am looking at them now |
| 19:44.10 | Karen_m | which sntp server should i use? |
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| 19:51.04 | Karen_m | time-a.nist.gov :) |
| 19:53.13 | chuckf | I like to use servers from the pool.ntp.org project |
| 19:53.23 | jaytee | same here |
| 19:53.46 | Karen_m | why? |
| 19:54.00 | chuckf | no worries about one server going down. |
| 19:54.13 | beardy | Because it consists of a lot of members, and it the Right answer. |
| 19:54.17 | chuckf | if it can't reach the first, it tries a second, third, etc |
| 19:54.20 | beardy | is* |
| 19:54.23 | _Corey_ | .gov is pretty broke these days... who knows what will happen :) |
| 19:54.48 | chuckf | Karen_m: check out www.pool.ntp.org |
| 19:55.20 | Karen_m | so you just set the sntp server to: pool.ntp.org and it handles if a server is broke, tries second and so on? |
| 19:55.30 | estranger | hey, how do I show whos in what conference bridges from the cli ? |
| 19:56.03 | chuckf | Karen_m: yes |
| 19:56.11 | paulc | estranger: meetme list <confid> I think |
| 19:56.30 | estranger | meetme list .. just found it too |
| 19:56.37 | estranger | was trying sip show / dahdi show ect |
| 19:56.42 | chuckf | it is (basicly?) a round robin dns setup for ntp servers |
| 19:57.49 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 19:58.04 | nny | is sip show channelstats considered accurate now? |
| 19:58.19 | chuckf | you always point to pool.ntp.org and they will point you to a recently tested ntp server on their list. So out of 5 hits to the site, you will likely get results back from 5 different ntp servers |
| 19:59.08 | chuckf | and if you use your webbrowser and don't put www before pool.ntp.org you will get a random web site each time. |
| 20:00.24 | chuckf | (if a web server is running on the ntp server you are redirected to that is) |
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| 20:04.50 | *** part/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 20:04.59 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 20:05.56 | tuxxie | What is wrong with this statment "ExecIf("${LEN(${NUMBER})})" = "4"]?Macro(callmenu,${NUMBER},external,${NUMBER})" |
| 20:06.48 | WIMPy | There's no $[ |
| 20:07.04 | tuxxie | :) |
| 20:07.07 | tuxxie | oops |
| 20:09.50 | *** part/#asterisk epart (~Administr@121.97.63.210) |
| 20:10.52 | tuxxie | WIMPy: it should use macro Callmenu if ${NUMBER} is 4 digits. correct? |
| 20:11.48 | WIMPy | I think so. |
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| 20:29.19 | tuxxie | I am still having issue with the following" ,ExecIf("$[{LEN(${NUMBER})})" = 4]?Macro(callmenu,${NUMBER},external,${NUMBER}) ". I would like to sent calls that $NUMBER is for digits to Callmenu. but its locking up my dial plan for any call even those above 4 digits. |
| 20:30.25 | tuxxie | is my syntax proper? |
| 20:38.48 | pdtpatrick | Question .. anyone know of an alternative to openvxi ? maybe Festival ? |
| 20:41.20 | paulc | tuxxie: Is there a $ missing between $[ and {LEN? Should be: $["${LEN(${NUMBER})}" = "4"]?... |
| 20:41.39 | paulc | and possible mis-placed quotes |
| 20:42.06 | billmania | pdtpatrick: Are you looking for only a speech synthesizer or something which provides all three of recognition, synthesis and telephony? |
| 20:42.10 | tuxxie | paulc: are the quotes needed |
| 20:42.28 | jaytee | anyone here using sip session timers? |
| 20:42.31 | WIMPy | yes and no, but I'm not sure the saces around th e= might do some harm. |
| 20:42.50 | pdtpatrick | billmania, i would like when users call in, they could get check on orders so this would translate the information into speech and give it back to customer |
| 20:42.51 | WIMPy | tuxxie: no |
| 20:42.58 | WIMPy | jaytee: yes |
| 20:43.06 | paulc | tuxxie: I don't think the quotes are necessarily needed, but they're useful for when one side of the evaluation could equate to empty |
| 20:43.11 | tuxxie | thanks |
| 20:43.33 | billmania | I'm sort of familiar with festival and know that it's probably the best known open source speech synthesizer. |
| 20:43.38 | jaytee | WIMPy, do you use it in your general section or on a peer by peer basis? |
| 20:43.53 | WIMPy | Indeed, but LEN will always produce a result. |
| 20:44.01 | billmania | I also have read about speech synthesis with asterisk and understand that it has support for festival. |
| 20:44.02 | pdtpatrick | billmania.. mind guiding me? And do you think what i am trying to do could be accomplished? |
| 20:44.03 | WIMPy | jaytee: general |
| 20:44.20 | billmania | So, I believe the answer to your question is "Yes, festival is appropriate". |
| 20:44.39 | billmania | pdtpatrick: I have no actual experience using festival with asterisk but know that it's possible. |
| 20:44.53 | jaytee | WIMPy, if I might ask, what is your session-expiry= value? |
| 20:45.43 | billmania | pdtpatrick: If you only want to read back an account number or an order quantity, festival would be overkill. |
| 20:45.56 | pdtpatrick | billmania, what would you suggest? |
| 20:46.07 | WIMPy | jaytee: 600 (the minimum allowed) |
| 20:46.21 | pdtpatrick | billmania, basically I would like them to call in and then get a status on their order |
| 20:46.29 | billmania | You can create canned speech templates as boilerplate and then let asterisk fill in the important bits (account numbers and quantities) by using the existing asterisk applications to speak the numbers. |
| 20:47.16 | billmania | pdtpatrick: Start with this page: http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
| 20:47.56 | billmania | pdtpatrick: Then look at the applications: Festival(), Playback(), SayDigits() and SayNumber(). |
| 20:48.25 | pdtpatrick | okay cool |
| 20:48.42 | billmania | Do you plan to use AGI() to retrieve the information about the orders? |
| 20:49.51 | pdtpatrick | probably |
| 20:50.09 | pdtpatrick | i think im going to have to write a bash or python script to grab data possibly from a database |
| 20:50.25 | billmania | OK. You can actually cause those applications to be called from within your AGI script. |
| 20:50.34 | pdtpatrick | nice |
| 20:50.53 | pdtpatrick | have you any examples? if not I can piece it together |
| 20:51.49 | billmania | I don't have any, but there are plenty available, starting at http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 20:52.13 | pdtpatrick | Thanks a bunch |
| 20:52.14 | pdtpatrick | :) |
| 20:53.40 | billmania | pdtpatrick: Sure. Good luck. |
| 20:53.53 | jaytee | WIMPy, thanks |
| 20:56.50 | billmania | pdtpatrick: And BTW, I vote for Python. :) |
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| 20:57.37 | jaytee | I've been checking CDR records and I've got two calls inbound over the last month and a half that lasted over 6 hours and both calls are from a client who is also running an Asterisk we installed. I'm implementing session timers now in hopes it will kill the stuck calls well before Flowroute's 6 hour limit. |
| 20:59.32 | WIMPy | Do you have rtptimeout set? That should trigger a lot fester. |
| 21:00.33 | jaytee | no, i didn't set rtptimeout |
| 21:07.28 | pdtpatrick | billmania, i might actually end up doing it in python haha. Should be simple enough. |
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| 21:24.29 | jeffspeff | I'm having an issue with SIP fax detection... in my dialplan when it gets to exten=fax,1,Goto(fax.......) I just get an auto fallthrough on the channel. voice calls work fine, i have set faxdetect=yes in sip.conf... any suggestions? |
| 21:28.39 | dym | jeffspeff: Faxing via SIP Trunks. Recieving or Sending is still not flawless. |
| 21:29.27 | *** join/#asterisk wasanzy (~emmanuel@196.201.43.55) |
| 21:29.30 | jeffspeff | dym, true, but 1 out of 30 test faxes should work shouldn't they? |
| 21:30.28 | dym | i wouldnt necesarily count on it |
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| 21:34.16 | cerienjean | hi all - I have an issue with the asterisk gui |
| 21:34.28 | cerienjean | I have installed it, configured, make checkconfig is ok |
| 21:34.49 | cerienjean | but when I go to the page, I just get a digium logo and no login/password prompt |
| 21:34.55 | cerienjean | any ideas why ? |
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| 21:44.06 | billmania | cerienjean: Asterisk GUI? Do you mean the FreePBX GUI or do you mean "make menuselect"? |
| 21:44.20 | billmania | (in the asterisk source collection) |
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| 21:52.09 | sunfone | Anyone have anything good/bad to say about Aastra 6731i? Especially wrt presence? |