| 00:07.46 | *** join/#asterisk Janos (~Janos@186.176.87.144) |
| 00:09.38 | Janos | hey there, question, is there any way to get the initial context where the call was place in the dialplan ? |
| 00:10.38 | p3nguin | core set verbose 3 |
| 00:10.44 | p3nguin | Make the call. Watch the console. |
| 00:10.53 | WIMPy | At that time in ${CONTEXT} |
| 00:14.51 | Janos | WIMPy, what if the call has gone through more than one context ? will ${CONTEXT} == last_context or ${CONTEXT} == initial_context ? |
| 00:15.29 | WIMPy | The current context |
| 00:15.43 | WIMPy | If you want the first one later, you have to save it. |
| 00:16.28 | Janos | WIMPy, ok that works, on last question, inside a macro, what will i get with ${CONTEXT} ? |
| 00:17.17 | *** part/#asterisk JacobIsre (~JacobIsre@99-190-176-190.lightspeed.hstntx.sbcglobal.net) |
| 00:17.31 | WIMPy | Not sure on that one. |
| 00:17.31 | Janos | will it be ${CONTEXT} == macro-mycontext ? or will it be the context before getting in the macro ? |
| 00:17.57 | Janos | WIMPy, kk, thanks a lot, i´ll just save it, that will do the trick |
| 00:18.00 | WIMPy | Isn't there a MACRO_CONTEXT or something? |
| 00:18.28 | Janos | mm no idea :P |
| 00:20.44 | Janos | i was think about migrating everything to lua since it´s a bit more familiar for me, but i wonder about issues that may arise, anyone has any experience with it ? |
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| 01:02.41 | jstapleton | Upgraded to 10.4.0, Google Talk inbound is giving an error on 1/3 accounts: chan_gtalk.c:2082 gtalk_parser: Remote peer reported an error, trying to establish the call anyway. Anyone else seen this? I would think that the upgrade broke things but considering 2/3 talk accounts work, maybe Google broke things?!? |
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| 02:14.46 | din3sh | looking for a decent billing solution for asterisk 1.8, any suggestions? |
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| 02:35.30 | sawgood | A few days ago, I was watching call completion from the CLI (Asterisk 1.8.9.x). When the calls completed, I would see the voicemail to email process flow by on the screen (I could see the SMTP transaction messages (200,250, etc), but now I do not see anything on the CLI (even tough the voicemail to email process is still working) |
| 02:35.42 | sawgood | any tips on how I was able to see the SMTP stuff the other day, but not now? |
| 02:38.32 | ChannelZ | hmm I've never seen it |
| 02:38.48 | sawgood | cool .. I'm not loosing my mind |
| 02:39.10 | sawgood | But, inside of the CLI ... I saw SMTP traffic flow ... (different from what is in /etc/mail/maillog) |
| 02:39.38 | WIMPy | I don;t see how that could be possible, |
| 02:40.08 | sawgood | I did not have tail running on a file ... I was inside of the CLI the whole time |
| 02:40.11 | ChannelZ | Sounds like something went haywire and wrote to a file descriptor it shouldn't have or I have no idea. |
| 02:40.12 | WIMPy | Unless you send the mail via an AGI that implements SMTP an provides the debug output. |
| 02:40.59 | sawgood | It was so cool because as the voicemail to email process was happeing, I was seeing the handshaking between the Asterisk and sendmail process ... to the SMART HOST |
| 02:42.02 | WIMPy | Asterisk doesn't do SMTP. |
| 02:42.18 | sawgood | ok .. I'm sorry for the flood |
| 02:43.09 | ChannelZ | I don't use sendmail, does it get chatty on stderr or something, and somehow that was getting injected into his shell? |
| 02:44.15 | WIMPy | I haven't seen anything like that, but it may be possible to make it do so. |
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| 02:45.22 | WIMPy | I'm not sur if the submission process will do the SMTP part if you have SuperSafe disabled. |
| 02:45.24 | sawgood | The reason I know this was in the CLI ... is because my mailcmd= statement in /etc/asterisk/voicemail.conf (had some incorrect switches add to it) ... and I would see the CLI tell me it was not able to process the mailcmd because of a bad operator statement |
| 02:45.29 | WIMPy | +e |
| 02:47.02 | sawgood | ah ... inside the mailcmd= ... I have the -v option listed |
| 02:47.31 | sawgood | I think I was see the -v process of sendmail scrolling on the screen (maybe I was not at the CLI) |
| 02:47.58 | sawgood | I was at the CLI (Because I could see the rest of the Asterisk syntax fly by on the screen after the mailcmd statement was done |
| 02:48.05 | WIMPy | That makes a lot more sense. |
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| 02:51.19 | sawgood | I am just so happy vm2email is working ... |
| 02:56.05 | sawgood | Do the new Digium phones (the high end model) ... really now support SLA (line sharing ... like/similar to a key system)? |
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| 03:32.42 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 03:32.58 | linocisco | hi all |
| 03:33.13 | linocisco | can (http://www.ipcomsupply.com/product/product_detail.php?productid=product-0704261957480113 ) give one FXO and 4 FXS ? |
| 03:33.56 | linocisco | I have one analog FXO line and want to share among 2 or 4 FXS analog extension. I want to know which cheapest device i should use |
| 03:44.42 | Janos | hey there, seeing this error in my console, everything is working great but still would like to know what is it, http://pastebin.com/KnTY5ydj |
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| 03:51.23 | sawgood | linocisco: The Grandstream HT503 has one FXO and one FXS for under 50 bucks |
| 03:51.34 | sawgood | The HT502 has two FXS ports |
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| 03:58.35 | linocisco | sawgood, thanks. let me check it out |
| 03:59.26 | linocisco | sawgood, I want analog one FXO and 2 FXS analog please. not one to one. it doesnt mean any difference |
| 04:04.16 | linocisco | hi all, apart from my previous question, I have another question, if i use http://www.voiplink.com/Grandstream_GXW4104_IP_Analog_Gateway_p/grandstream-gxw4104.htm, do I need asterisk sytem also? |
| 04:07.41 | linocisco | softswitch vs IP PBX? |
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| 06:19.12 | din3sh | morning peeps |
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| 06:21.27 | schmidts | good morning |
| 06:21.37 | din3sh | mrning schmidts |
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| 08:21.02 | v0lZy | hellow |
| 08:28.53 | ChannelZ | OH HAI! |
| 08:33.59 | v0lZy | quick question... how to get avialable modules |
| 08:34.02 | v0lZy | core show... somethign? |
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| 08:35.12 | v0lZy | ah |
| 08:35.13 | v0lZy | module show |
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| 08:36.43 | kaldemar | v0lZy: it does not show modules that are not loaded though. |
| 08:37.04 | v0lZy | how do i see those? |
| 08:37.08 | v0lZy | those i can load |
| 08:37.38 | kaldemar | ls -la /usr/lib/asterisk/modules |
| 08:40.41 | v0lZy | isnt there an asterisk command to also do it? |
| 08:41.19 | v0lZy | i don thave it that way. |
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| 08:42.20 | v0lZy | fount them |
| 08:43.17 | v0lZy | im trying to do this |
| 08:43.19 | v0lZy | http://www.fosslc.org/drupal/node/643 |
| 08:43.25 | v0lZy | but i dont know wha tmodulesi need for this |
| 08:43.55 | v0lZy | i see app_mixmonitor.so |
| 08:43.58 | v0lZy | i guess i need to enable that one |
| 08:44.38 | v0lZy | not loaded in my modules (doesnt show up in rasterisk 'module show' |
| 08:46.32 | kaldemar | v0lZy: modules.conf for asterisk defines what is loaded on start. if you have autoload=yes, a noload might prevent the module from loading. |
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| 08:49.50 | v0lZy | im using askoziapbx |
| 08:49.56 | v0lZy | and by default, it doesnt include that module |
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| 08:50.05 | v0lZy | i mean, its in the directory |
| 08:50.07 | v0lZy | just not loaded |
| 08:52.07 | kaldemar | you probably have "noload => app_mixmonitor.so" in modules.conf or some file included from it. |
| 08:52.17 | v0lZy | let me see |
| 08:52.49 | v0lZy | autoload = no |
| 08:52.56 | v0lZy | each is loaded individually |
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| 09:12.05 | *** join/#asterisk WebSprocket (~WebSprock@93-97-23-210.zone5.bethere.co.uk) |
| 09:12.25 | WebSprocket | Hello, We have a PBX In a Flash server here, running purple, nothing fancy. |
| 09:12.45 | WebSprocket | We have a remote office that connect to the server's external IP |
| 09:13.09 | *** join/#asterisk nickfennell (~nick@unaffiliated/nickfennell) |
| 09:13.11 | WebSprocket | they cannot ring between eachother in that office, nor can they recieve inbound calls. Outbound calls are crystal clear. |
| 09:13.12 | nickfennell | hi all |
| 09:13.20 | WebSprocket | Any thoughts on this would be appreciated |
| 09:13.21 | nickfennell | Anyone know why Cisco phones would not always clear a sip channel in use |
| 09:13.41 | WebSprocket | nickfennell our Linksys SPA921s are bad for that |
| 09:13.53 | v0lZy | kaldemar: how does call recording work in asterisk |
| 09:14.03 | v0lZy | i mean... say that i configure this mixmonitor |
| 09:14.13 | v0lZy | according to this. http://www.fosslc.org/drupal/node/643 |
| 09:14.43 | v0lZy | my user could press |
| 09:14.56 | v0lZy | *3 while talking with someone |
| 09:15.01 | v0lZy | and asterisk would start to record? |
| 09:15.46 | kaldemar | did you actually read that page? |
| 09:16.50 | kaldemar | ~sipnat |
| 09:16.50 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 09:16.53 | kaldemar | WebSprocket: ^ |
| 09:18.51 | WebSprocket | ty :) |
| 09:29.44 | v0lZy | kaldemar: yes i did but i dont understand some things.. or more like i have questions regarding |
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| 09:42.14 | kaldemar | v0lZy: enabling a feature like automixmon makes asterisk detect the DTMF in the audio stream and start the mixmon app with an audio hook and record the call. |
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| 09:54.20 | v0lZy | is this DTMF heard on the other side as well? probably, right? |
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| 10:07.32 | din3sh | only the one activating the recording i think hears it |
| 10:09.33 | v0lZy | and how do i playback those things' |
| 10:09.43 | v0lZy | it puts them in my voicemail or something? |
| 10:10.25 | kaldemar | v0lZy: "When the recording ends, under /var/spool/asterisk/monitor, you should see a file..." |
| 10:10.43 | v0lZy | yes |
| 10:10.54 | v0lZy | i can script that to get sent via email or something |
| 10:10.58 | kaldemar | what you do with them after that is your business. |
| 10:11.06 | v0lZy | ah, ok, so it only deposits them |
| 10:12.30 | din3sh | kaldemar: how can i stop cisco phones from reinviting in asterisk 1.8.x |
| 10:12.32 | din3sh | ? |
| 10:12.39 | kaldemar | din3sh: with a hammer |
| 10:12.43 | din3sh | directmedia=no doesnt do it |
| 10:12.44 | din3sh | loll |
| 10:13.09 | din3sh | any specific kindof hammer? |
| 10:13.11 | din3sh | :D |
| 10:13.22 | kaldemar | sledge, preferrably. |
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| 10:14.34 | din3sh | :o |
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| 10:15.09 | din3sh | i'll try that on couple of cisco's which have blank lcds |
| 10:15.10 | din3sh | :p |
| 10:15.45 | din3sh | seriously though directmedia=no doesnt do it |
| 10:15.46 | din3sh | :S |
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| 10:17.06 | kaldemar | din3sh: is it a phone sending re-invites? |
| 10:26.41 | din3sh | yes |
| 10:27.24 | kaldemar | din3sh: directmedia affects how asterisk behaves, not how a phone does. |
| 10:28.03 | din3sh | Retransmitting #10 (no NAT) to 192.168.6.65:5060: SIP/2.0 487 Request Terminated |
| 10:28.48 | din3sh | after this i get the evil critical sip retransmit error |
| 10:30.55 | din3sh | for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 10:31.09 | din3sh | am unsure if this also causes call drops |
| 10:32.37 | kaldemar | is the phone at that address? |
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| 10:37.58 | din3sh | yes |
| 10:38.41 | Faustov | I got 2 servers, each with an IVR, where menu on server1 leads to a menu on server2, then there is an option to return to the ivr on server1, which works, but there is no sound for about 10 seconds - I'm thinking this is some anti-looping mechanism, any idea how to solve it? |
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| 10:42.49 | din3sh | Kaldemar: i get the error only when the phone tries to call another phone but cancels the call |
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| 10:43.44 | din3sh | dont know if my wording is correct, but the phone doesnt seem to match the 487 cancel request |
| 10:45.52 | din3sh | dont even know if its an re-invite issue |
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| 12:02.44 | SuperNull | hey guys, is there a way to add src ip of call to mysql_cdr ? |
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| 12:32.02 | [TK]D-Fender | SuperNull, UserData <- |
| 12:37.12 | SuperNull | TK you are always here :) |
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| 13:58.59 | eduzimrs | hi, im trying to place a call with SRTP media using polycom IP550, and asterisk is saying "chan_sip.c: Multiple audio streams are not supported" ideas? |
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| 14:50.02 | Katty | helllloooooooooooo nurse. |
| 14:51.12 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:51.12 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:01.20 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:09.04 | zamba | i have a diva 2.01 isdn card.. can this work as a BRI card? |
| 15:10.01 | [TK]D-Fender | zamba, https://www.google.ca/#hl=en&output=search&sclient=psy-ab&q=diva+2.01+isdn+asterisk&oq=diva+2.01+isdn+asterisk&aq=f&aqi=q-w1&aql=&gs_l=hp.3..33i21.411.3287.0.3416.11.9.0.0.0.1.770.3840.3-2j0j3j2.7.0...0.0.j4gJD-TsChM&pbx=1&bav=on.2,or.r_gc.r_pw.r_cp.r_qf.,cf.osb&fp=4f0b47acf7722ea9&biw=1600&bih=883 |
| 15:10.12 | Qwell | is it a BRI card? |
| 15:11.15 | zamba | [TK]D-Fender: i interpret that as a yes? :) |
| 15:11.32 | [TK]D-Fender | You mean the gigantic page of results talking all about it? |
| 15:11.38 | [TK]D-Fender | zamba, Well... if you insist... |
| 15:12.40 | zamba | yay, cool |
| 15:13.16 | zamba | you don't agree? |
| 15:13.54 | [TK]D-Fender | Don't know... haven't touched it. Let me find a thermometer |
| 15:19.37 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:19.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.4.0 (2012/05/02), 1.8.12.0 (2012/05/02), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:19.46 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 15:19.57 | Qwell | You can? Well good. |
| 15:20.14 | ghostmediapro | i can't sorry typo |
| 15:20.26 | ZenMaster | Morning guys. |
| 15:20.33 | ZenMaster | Last day of the week woohoo! |
| 15:20.39 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 15:20.57 | [TK]D-Fender | ghostmediapro, *'s H323 channel driver's and configs are all documented right there |
| 15:20.58 | *** mode/#asterisk [+b *!*chatzilla@*unaffiliated/t-dot-zilla/x-2830497] by Qwell |
| 15:20.58 | *** kick/#asterisk [vipkilla!~north@pdpc/sponsor/digium/Qwell] by Qwell (bye) |
| 15:21.14 | [TK]D-Fender | ghostmediapro, Show us what you've actually tried and maybe we'll have something to contribute |
| 15:22.17 | ghostmediapro | well before we go back in time, i've i few things found online, but get stock at entering the setting in the phone, especially the gateway or gatekeep |
| 15:22.57 | ghostmediapro | [TK]D-Fender: please provide a link and all reread, that information, maybe something new will stand out |
| 15:23.37 | [TK]D-Fender | ghostmediapro, it's all in the tarball |
| 15:25.09 | [TK]D-Fender | ghostmediapro, And if you're stuck entering something into your phone.. then that is between you & your phone, not * |
| 15:25.10 | ghostmediapro | i'm not using the direct asterisk distro |
| 15:25.38 | [TK]D-Fender | ghostmediapro, Doesn't matter what you isntalled as long as you're configuring it yourself with a normal * release |
| 15:25.42 | ghostmediapro | thanks, i'll just keep googleing |
| 15:27.33 | *** part/#asterisk ghostmediapro (~Trupsalms@184.78.109.95) |
| 15:27.50 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 15:31.48 | *** join/#asterisk carrar (tim@osburn.com) |
| 15:35.16 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v017-224.mobile.uci.edu) |
| 15:39.46 | leifmadsen | Qwell: +1 |
| 15:40.29 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:41.31 | Qwell | O.o |
| 15:53.33 | SuperNull | anyone willing to attempt a sip udp connect to see if my firewall is working nicely ? ;) |
| 15:54.26 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 15:55.40 | saliak | I'm interesting in setting up asterisk to receive faxes. I found these instructions (http://www.voip-info.org/wiki/view/Asterisk+Fax+to+email). it looks like it requires spandsp. how do i check if i have that installed? |
| 15:57.14 | Qwell | saliak: If you didn't install it, you don't have it. |
| 15:58.12 | coppice | it might me there from an RPM/DEB dependency |
| 16:02.24 | *** join/#asterisk hc (hc@pdpc/supporter/active/hc-e) |
| 16:02.26 | hc | hi alll |
| 16:02.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:02.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:02.54 | hc | is it normal that an asterisk console does not react after a "channel originate" until the call is picked up or times out? |
| 16:03.25 | hc | when i enter 'core show channels' immediately after a 'channel originate', i won't get any immediate output |
| 16:03.42 | hc | (tested on 1.8 as well as 1.10.4) |
| 16:06.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:07.28 | *** join/#asterisk engrxyz (~tyrtyrty@cpc3-basl7-0-0-cust252.basl.cable.virginmedia.com) |
| 16:07.36 | saliak | I could have sworn that i've seen it somewhere in asterisk |
| 16:07.49 | saliak | is there a "show …" command to verify if it's installed? |
| 16:11.15 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 16:22.25 | zamba | what are the steps for setting up a BRI connection? |
| 16:23.42 | zamba | there's no capi.conf? |
| 16:25.25 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:25.47 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
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| 16:42.59 | TechSmurf | Is there a way to install *NOW via usb flash? |
| 16:46.19 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 16:46.52 | *** join/#asterisk vastina (~vastina@70.99.103.122) |
| 16:48.31 | pabelanger | TechSmurf: with some work you could |
| 16:49.43 | TechSmurf | It looks like the only hitch is the installer being so insistent on finding a cdrom |
| 16:50.43 | TechSmurf | but the installer isn't offering a terminal to mount up the flash drive either |
| 16:53.32 | zamba | ok, giving this up |
| 16:53.51 | zamba | can someone please provide me with some help setting up my ISDN BRI connection? |
| 16:54.35 | TechSmurf | Qwell: fix it. |
| 16:54.41 | zamba | i have an eicon diva 2.01 S/T card |
| 16:54.58 | zamba | i'm running on ubuntu and the hisax module is automatically probed |
| 16:55.18 | Qwell | TechSmurf: install CentOS 5, install the Digium Asterisk repos, yum install asterisk |
| 16:55.36 | Qwell | kickstart makes USB stupid |
| 16:55.43 | zamba | i have probed divacapi myself and then i got the /dev/ttyIX devices |
| 16:55.46 | TechSmurf | agrees |
| 16:55.50 | zamba | but i don't know how to proceed |
| 16:56.28 | TechSmurf | copies the .iso to the usb key so the centos installer can mount it loopback |
| 16:56.30 | zamba | anyone care to shed some light on this? |
| 17:01.26 | Kobaz | lights a candle |
| 17:02.15 | n3hxs | hurrys to turn off the gas. |
| 17:02.16 | Kobaz | zamba: there's several bits of docs on the voip wiki if i remember |
| 17:04.29 | zamba | Kobaz: yeah, but there's lots of different ways of doing it |
| 17:04.40 | Kobaz | which ever way works would be the best one |
| 17:04.42 | zamba | Kobaz: and for someone who hasn't done this before, it's very confusing |
| 17:05.04 | zamba | and much of the documentation is dated 2005 :p |
| 17:05.13 | Kobaz | i've never set up bri, but i would imagine its very similar |
| 17:05.18 | Kobaz | to pri |
| 17:05.35 | zamba | do my steps so far seem logical? |
| 17:05.42 | Kobaz | insert card, compile dahdi drivers, make install, and then edit dahdi.conf |
| 17:05.54 | zamba | dahdi? |
| 17:05.56 | Kobaz | and then dahdi_cfg -vv, and then edit chan_dahdi.conf to match |
| 17:06.06 | zamba | where did that come from? |
| 17:06.35 | Kobaz | dahdi = digium asterisk hardware device interface |
| 17:06.36 | zamba | so maybe dahdi is what i need? |
| 17:06.44 | Kobaz | it's the interface between cards and asterisk |
| 17:06.51 | zamba | oh.. ok.. so that's related to the digium cards? |
| 17:06.58 | zamba | oh.. ok.. i SHOULD use dahdi? |
| 17:07.00 | Kobaz | and compatible cards |
| 17:07.15 | Kobaz | like rhino, sangoma (which has additional drivers) |
| 17:07.35 | Kobaz | what card do you have again |
| 17:07.45 | zamba | eicon diva 2.01 S/T card |
| 17:08.04 | zamba | it's *OLD* |
| 17:08.27 | zamba | but it uses the hisax kernel module and it was detected there, so maybe.. ? |
| 17:08.40 | Kobaz | oh |
| 17:08.43 | Kobaz | eh |
| 17:09.03 | Kobaz | dahdi doesn't know anything about cards other than ones that implement the dahdi/zaptel api |
| 17:09.22 | Kobaz | you'll probably need to not load the kernel module for anything to work |
| 17:09.39 | zamba | ok, so dahdi isn't the way i should try to set up this, then |
| 17:10.00 | Kobaz | well asterisk doesn't talk to kernel modules other than dahdi (unless you compile in additional drivers) |
| 17:10.51 | zamba | i'm reading write-ups of people who has gotten my card working |
| 17:10.56 | zamba | so it -should- be doable |
| 17:10.58 | Kobaz | you may need like misdn or something |
| 17:11.09 | Kobaz | again i can't help a whole lot since i've just used very modern pri cards |
| 17:12.07 | *** join/#asterisk skirmisha (~vk@46.47.82.189) |
| 17:12.12 | skirmisha | hi guys |
| 17:12.24 | skirmisha | does asterisk support ECM |
| 17:12.35 | skirmisha | for T38 |
| 17:13.49 | zamba | http://www.trend-watcher.org/archives/asterisk-setup/ |
| 17:13.55 | zamba | and that's 7 years ago |
| 17:14.38 | zamba | and that uses chan_modem.. ? |
| 17:15.05 | zamba | but there's no /etc/asterisk/modem.conf here |
| 17:15.37 | Kobaz | you might need to use asterisk 1.2 or something as a gateway |
| 17:15.56 | zamba | *shrug* |
| 17:17.30 | Qwell | or get something that isn't 10 years old |
| 17:18.13 | Kobaz | yeah |
| 17:18.25 | Kobaz | you'll spend like $1000 in time, to save buying a $200 card |
| 17:20.27 | TechSmurf | hmm |
| 17:20.43 | TechSmurf | Qwell: I think I just figured out how to make kickstart play nice |
| 17:20.53 | Qwell | oh? |
| 17:21.13 | Qwell | I've spent about 0 time on it. I've had people ask before though. |
| 17:21.28 | TechSmurf | I'll detail it if it works |
| 17:21.40 | Qwell | k, feel free to msg |
| 17:27.09 | TechSmurf | newp, newp. it's chock full of fail |
| 17:31.00 | *** join/#asterisk n0tk (~n0tk@gateway/tor-sasl/n0tk) |
| 17:31.28 | Katty | hello my asterisk does not work at all how to fix plz???? |
| 17:31.57 | talntid | Katty, i will fix it. i will send you asterisk-fix.exe. please run it on your windows desktop. |
| 17:34.31 | *** part/#asterisk hc (hc@pdpc/supporter/active/hc-e) |
| 17:35.34 | TechSmurf | mkay, this is getting silly. *digs out a usb dvd drive* |
| 17:36.08 | *** join/#asterisk michael-i (~anonymous@204.11.231.178.static.etheric.net) |
| 17:36.24 | Qwell | TechSmurf: note that if you use a USB DVD drive, it'll tell you that it couldn't find the drive after boot. Just hit okay and it'll work. CentOS is dumb sometimes. |
| 17:36.46 | Qwell | it never bothers to init them the first time around... |
| 17:37.13 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:38.47 | michael-i | Hi all. I'm setting up Asterisk to interface with an AudioCodes Mediant 1000. I have incoming and outgoing calls over SIP working but incoming always lands in my "default/strangers" context. How can I associate the incoming calls from that IP to a specific context? http://pastebin.com/AUCCcHCJ |
| 17:40.22 | ChannelZ | You need to configure a sip peer such that it matches where your incoming calls come from |
| 17:40.28 | michael-i | (they should land in cloud-incoming but the traffic isn't being associated with my peer definition) |
| 17:40.52 | michael-i | ChannelZ: In that paste, I thought I had. Something's missing/wrong though |
| 17:41.08 | ChannelZ | Dunno, we didn't see a call. |
| 17:41.14 | ChannelZ | As a peer it should be matching by IP |
| 17:42.04 | ChannelZ | But someone else will step in and help I'm sure, I'm just passing by at the moment sorry |
| 17:45.33 | TechSmurf | Qwell: Actually, I tried that, and it still went wonky |
| 17:45.44 | TechSmurf | after partitioning it decided it couldn't remount the iso |
| 17:46.29 | TechSmurf | I'm burning the dvd now |
| 17:46.48 | TechSmurf | I feel so retro |
| 17:47.25 | [TK]D-Fender | michael-i, ditch the domain and show us the new config along with a filed call with SIP DEBUG enabled |
| 17:50.26 | *** join/#asterisk sickgrinder (~sickgrind@office2.vicidial.com) |
| 17:53.00 | *** join/#asterisk navaismo (~navaismo@201.123.34.101) |
| 17:56.55 | ZenMaster | [TK]D-Fender: I have a quick questiontino bro-suf. |
| 17:57.08 | [TK]D-Fender | just ask in public |
| 17:57.13 | ZenMaster | So my outbound is working fine, those tests are complete. My SIP provider provided me a number..... |
| 17:57.19 | [TK]D-Fender | Anyone with an answer might chime in |
| 17:57.24 | ZenMaster | Inbound is saying that it is not setup. Is that on me or is that on my SIP provider? |
| 17:57.56 | ZenMaster | Well when I call the number that is. |
| 17:58.07 | [TK]D-Fender | "Inbound is saying that it is not setup" ? |
| 17:58.17 | [TK]D-Fender | What? How? |
| 17:58.20 | ZenMaster | Yeah pre-recorded, |
| 17:58.35 | [TK]D-Fender | Do you see the call come in at all with SIP DEBUG enabled? |
| 17:58.47 | ZenMaster | I have no tried that. |
| 18:00.16 | zamba | [TK]D-Fender: do you have any idea how i can set up my eicon diva ISDN card? |
| 18:00.25 | zamba | [TK]D-Fender: should i be using mISDN? |
| 18:01.51 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:02.08 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-pmohgpjmethlcgcf) |
| 18:02.40 | [TK]D-Fender | zamba, never touched BRI before |
| 18:03.45 | zamba | [TK]D-Fender: oh, ok |
| 18:05.27 | Katty | resists Your Mom Joke. |
| 18:05.39 | Katty | twitches. |
| 18:14.30 | *** join/#asterisk ghostmediapro (~Trupsalms@184.78.109.95) |
| 18:15.24 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 18:15.27 | cusco | hi |
| 18:15.44 | cusco | Taking a call from PRI, I use Progress(); then Queue() |
| 18:15.52 | cusco | and if teh call is not answered within 30 seconds |
| 18:15.57 | cusco | it is hugup |
| 18:16.00 | cusco | hungup |
| 18:16.10 | cusco | is this some asterisk setting, or telco? |
| 18:16.22 | ghostmediapro | [TK]D-Fender, i've done some reading and i see that h323 support is there, does asterisk have the capibilty to act as a gatekeeper? and how |
| 18:16.34 | [TK]D-Fender | cusco, you should be showing us the call so we can tell you instead of guessing. |
| 18:16.37 | _Corey_ | cusco: Answer() the channel |
| 18:17.08 | cusco | _Corey_: I would rather only answer when a queue member does answer, avoiding costs to the caller |
| 18:17.24 | [TK]D-Fender | ghostmediapro, * supports H.323 phones and there are already sample configs for this. Come back with an actual attempt to implement it and a failure to debug |
| 18:17.31 | _Corey_ | cusco: You have to accept that you're subject to the calling party's dial timeout then |
| 18:17.41 | *** join/#asterisk vastina (~vastina@70.99.103.122) |
| 18:18.03 | _Corey_ | cusco: They're not going to let the call stay ringing/in progress forever |
| 18:18.08 | ghostmediapro | can you please provide a starting direction |
| 18:18.41 | [TK]D-Fender | ghostmediapro, the samples are alerady in the tarball. |
| 18:18.55 | [TK]D-Fender | use them and modify for your tests |
| 18:21.38 | ghostmediapro | which location, damn you uses to tell me googles your friend, now it been quite sometime since i've been here on irc, and i still can't get a str8 answer |
| 18:22.25 | [TK]D-Fender | in the main asterisk tarball or "addons" if you're in an older version using] |
| 18:22.27 | [TK]D-Fender | that |
| 18:22.43 | [TK]D-Fender | There is a config filder that is blatantly obvious if you actually look for even a second |
| 18:22.46 | ghostmediapro | i've been googling h323 and asterisk for the past 7 hours, and nothing stands out on connect ip h323 protocol base ip phone |
| 18:22.55 | [TK]D-Fender | Did I say GOOGLE? |
| 18:22.56 | [TK]D-Fender | NO. |
| 18:23.00 | [TK]D-Fender | TARBALL <------ |
| 18:23.28 | [TK]D-Fender | Look at the sample configs they already gave you for those channel drivers |
| 18:23.42 | cusco | [TK]D-Fender: cause 102 - the call: http://paste.debian.net/169464/ |
| 18:24.08 | cusco | _Corey_: the calling party may have a timeout of 30 seconds if not answered? |
| 18:24.22 | cusco | I don't think thats the case..., tho its possible |
| 18:24.47 | cusco | 102 - define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE |
| 18:26.12 | _Corey_ | cusco: You can't really control their behavior past a certain point... |
| 18:27.21 | ghostmediapro | asterisk-10.4.0 , addons dir, ooh323c, no config or simple files |
| 18:31.23 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 18:33.59 | [TK]D-Fender | I'm staring at an h323 config file in the tarball right now |
| 18:35.20 | cusco | _Corey_: so if that is true; I can try on the same number take out the PROGRESS() and dial(something-that-does-not-answer)... and it also should hangup at 30 seconds |
| 18:35.55 | cusco | or really use: Wait(35) |
| 18:38.15 | ghostmediapro | \asterisk-10.4.0\asterisk-10.4.0\addons |
| 18:38.38 | ghostmediapro | \ooh323c |
| 18:38.44 | ghostmediapro | no config |
| 18:39.51 | [TK]D-Fender | <[TK]D-Fender> in the main asterisk tarball or "addons" if you're in an older version using that <---- |
| 18:40.14 | [TK]D-Fender | <[TK]D-Fender> There is a config filder that is blatantly obvious if you actually look for even a second <----- |
| 18:40.18 | [TK]D-Fender | You aren't even looking |
| 18:40.43 | ghostmediapro | yes |
| 18:40.51 | [TK]D-Fender | You are NOT on an older version and are not paying attention or loking 1 inch further than you seem to feel you have to get away with |
| 18:40.53 | ghostmediapro | i'm looking in the tar |
| 18:41.13 | ghostmediapro | i just downloaded the tar and extracted its contents to browse |
| 18:41.35 | ghostmediapro | i wanna figure this out |
| 18:41.37 | [TK]D-Fender | And you are restricting your search to 1 stupid sub-folder. |
| 18:41.40 | [TK]D-Fender | ^^^^^6 |
| 18:41.55 | ghostmediapro | so i don't mind looking, but its not where you say it is |
| 18:41.57 | [TK]D-Fender | CONFIGS <------- |
| 18:42.16 | ghostmediapro | i'll give you the whole dir |
| 18:42.35 | ghostmediapro | \asterisk-10.4.0\asterisk-10.4.0\addons\ooh323c\src\h323 |
| 18:42.47 | [TK]D-Fender | I did not ay to look only in there |
| 18:42.52 | ghostmediapro | no conf file |
| 18:43.57 | ghostmediapro | chan_ooh323.conf.sample |
| 18:44.03 | ghostmediapro | is that the file |
| 18:44.43 | ghostmediapro | h323.conf.sample |
| 18:45.25 | [TK]D-Fender | there are MULTIPLE channel drivers for H.323 |
| 18:47.00 | TechSmurf | Qwell: whatcha know, it installed from dvd without a hitch. *stabby* |
| 18:47.49 | *** join/#asterisk c4t3l (~c4t3l@c-76-30-80-232.hsd1.tx.comcast.net) |
| 18:49.49 | cusco | _Corey_: I just used Dial(SIP/bla); and seen the phone ringing and did not answered it |
| 18:49.55 | cusco | I did not use progress() tho |
| 18:50.00 | cusco | but the call lasted far longer |
| 18:50.21 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 18:51.18 | cusco | wel, using progress() it also lasted more than 30 secs |
| 18:51.34 | cusco | so if I use progress but do not choose to send audio, it will last? |
| 18:51.50 | cusco | dies sip dial generate inband ringing or something? |
| 18:52.16 | cusco | I'm trying to understand why the channel can be kept alive if the call is not picked up, but not if I'm sending early audio ? |
| 18:53.05 | *** join/#asterisk censor (~you@p57BD74D6.dip.t-dialin.net) |
| 18:53.07 | censor | hi all |
| 18:54.03 | censor | i finally got a mISDN-compatible ISDN card, after i was eventually not able to compile kernel modules for my old Fritz!Box USB |
| 18:54.34 | censor | now i'm wondering how to get it to work with Asterisk?! Running Debian squeeze/wheezy with Asterisk 1.8 |
| 18:55.00 | censor | but there doesn't seem to be a chan_misdn, although i read everywhere that's it's now included by default? |
| 18:55.20 | WIMPy | censor: http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 18:55.58 | *** join/#asterisk _pll (~anonymous@200.110.25.33) |
| 18:56.09 | censor | WIMPy ah, that clears it up, thanks! |
| 18:56.23 | _pll | Hello, I have a weird issue with MixMonitor. |
| 18:56.45 | censor | quick dumb question: is it possible to have mISDNv2 emulate CAPI, so i can stick to my old setup? |
| 18:57.01 | _pll | I am getting tons of "audiohook.c:311 audiohook_read_frame_both: Failed to get 160 samples from write factory 0xb721291c" when I enable cor set debug 1 |
| 18:57.09 | _pll | read and write |
| 18:57.20 | _pll | I have been googling for the root cause without avail. |
| 18:57.29 | WIMPy | censor: The git version can do CAPI, but I wouldn;t recommend that. |
| 18:58.30 | *** part/#asterisk ghostmediapro (~Trupsalms@184.78.109.95) |
| 18:58.31 | censor | WIMPy thanks, i'll give it a shot, i can always go back to LCR then ;) |
| 18:58.58 | WIMPy | Either LCR or dahdi. Depending on what you expect. |
| 19:00.23 | censor | i've only worked with CAPI so far - it's just my server here at home with ISDN, nothing bog |
| 19:00.25 | censor | *big |
| 19:00.39 | censor | what would be the diff between LCR and dadhi? |
| 19:01.04 | WIMPy | There's a feature list a few pages in. |
| 19:01.20 | WIMPy | dahdi has more features, but also more surprises. |
| 19:01.22 | censor | right, sorry =) |
| 19:02.18 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 19:09.36 | saliak | I'm trying to install "asterisk-app-fax" under ubuntu with apt-get (to install fax support), but I can't seem to find it (I have universe added). anyone had luck installing it? |
| 19:09.38 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 19:10.51 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 19:13.04 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 19:30.29 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 19:30.29 | *** mode/#asterisk [+o angler] by ChanServ |
| 19:37.54 | *** join/#asterisk sewellj (~chatzilla@64.93.111.89) |
| 19:44.45 | sewellj | Can anyone offer some help fixing a zaptel problem in AsteriskNOW? |
| 19:45.30 | [TK]D-Fender | Show us the problem and maybe... |
| 19:45.31 | [TK]D-Fender | ~pb |
| 19:45.31 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:45.32 | [TK]D-Fender | ^^ |
| 19:46.28 | sewellj | http://pastebin.com/8PqMRiFu |
| 19:47.19 | [TK]D-Fender | sewellj, "ztcfg -vvvv" |
| 19:47.27 | sewellj | stby |
| 19:48.17 | sewellj | http://pastebin.com/dkG5ktp8 |
| 19:48.39 | [TK]D-Fender | looks fine, now asterisk -gvvvvvvvvvc |
| 19:52.55 | sewellj | it's still chugging away |
| 19:53.07 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:53.07 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:01.30 | sewellj | had to do it again, exceeded my putty scrollback |
| 20:02.53 | WIMPy | Just do a 'module unload chan_zap' and take the output of 'module load chan_zap'. |
| 20:03.03 | WIMPy | Or what it was called way back then. |
| 20:03.45 | sewellj | http://pastebin.com/QLg6V6a5 |
| 20:04.09 | sewellj | WIMPy: I will try |
| 20:05.48 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 20:06.40 | sewellj | http://pastebin.com/WDLGh7Hm says No such file or directory on the .so, is the file simply missing? |
| 20:07.31 | WIMPy | obviousely it's not that old that you're using zaptel. it's chan_dahdi. |
| 20:09.54 | sewellj | http://pastebin.com/1hiLqreB |
| 20:12.43 | WIMPy | wonders why it mentions zapata.conf. |
| 20:13.04 | sewellj | I don't have a zapata.conf in my /etc/asterisk |
| 20:13.55 | sewellj | locate just finds the example |
| 20:14.32 | WIMPy | Maybe this is a half zap half dahdi version? |
| 20:15.21 | gtTuna | I have a CDR question -- In my auto attendent menu, for some phone tree options, I may end up calling multiple extensions. Using cdr-custom, and custom fields, how can I set a userfield to the extension of the phone that ended up picking up the call |
| 20:15.56 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 20:16.44 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:16.46 | sewellj | WIMPy: I'm considering ditching this old version (1.4!!!) it was started by another guy in our lab but he's away and I don't know why he didn't try something newer. It could be that we only have an old hand me down computer to run on. |
| 20:17.48 | WIMPy | I certainly wouldn't want to use anything less than 1.8. |
| 20:17.54 | gtTuna | im running 1.6.x btw |
| 20:18.32 | gtTuna | how big of a pain is it/will it be to upgrade to 1.8 ?, because I think theres a solution to my issue w/ MASTER_CHANNEL in 1.8.x |
| 20:19.01 | WIMPy | Should be straight forward. |
| 20:21.32 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 20:26.01 | sewellj | WIMPy and Fender: thanks for your time. I'm going to try to get a newer computer and install AsteriskNOW 2.0. It would be nice to have the GUI. |
| 20:27.08 | p3nguin | That's what YOU think. |
| 20:28.39 | *** join/#asterisk d_preston215 (~dpreston@50-73-214-237-philadelpia.hfc.comcastbusiness.net) |
| 20:29.22 | d_preston215 | How do I go about monitoring two extensions at once? |
| 20:29.37 | d_preston215 | Primarily with chanspy. |
| 20:29.50 | p3nguin | ChanSpy monitors channels, not extensions. |
| 20:29.58 | p3nguin | For extensions, you use ExtenSpy. |
| 20:30.10 | d_preston215 | Ok. |
| 20:30.24 | p3nguin | And you won't be able to do two extensions at one time from the same channel using either app. |
| 20:31.03 | p3nguin | ExtenSpy can only accept one extension as app data. |
| 20:31.53 | d_preston215 | What about two channels extensions in a conference? |
| 20:32.10 | d_preston215 | channels/extensions in a chanspy/extenspy conference? |
| 20:33.11 | p3nguin | Channels don't have extensions. |
| 20:33.35 | p3nguin | If you want to spy on a channel, use chanspy. |
| 20:33.44 | p3nguin | If you want to spy via extension, use extenspy. |
| 20:34.04 | p3nguin | To spy on a channel involved in a conference, you can use either app. |
| 20:34.38 | p3nguin | You can ExtenSpy() the extension used to get into that conference, or you can ChanSpy() the channel in the conference. |
| 20:34.43 | d_preston215 | I understand that. So you are saying that there is no reasonable way for one extension to spy on two extensions at the same time? |
| 20:34.53 | p3nguin | That doesn't even make sense. |
| 20:35.14 | p3nguin | (1531.02) <p3nguin> ExtenSpy can only accept one extension as app data. |
| 20:35.29 | d_preston215 | Ok. |
| 20:35.56 | p3nguin | However, both ExtenSpy and ChanSpy can listen to two CHANNELS which are bridged together. Would that help? |
| 20:38.04 | d_preston215 | Possibly. What about spygroups? Would that be an option as well? |
| 20:39.32 | *** join/#asterisk mbrit (~mbrit@186.120.97.195) |
| 20:39.48 | p3nguin | It does the exact same thing, except groups channels together and limits the spying to those channels only. |
| 20:41.11 | d_preston215 | Ok. |
| 20:41.21 | p3nguin | You can listen to multiple channels separately. Did I misunderstand that you wanted them together? |
| 20:42.05 | p3nguin | When you spy on multiple channels separately, you can cycle through the channels of a specific prefix or group. |
| 20:43.30 | d_preston215 | I wanted a person to be able to listen to 2 channels together. |
| 20:43.46 | p3nguin | You'll only be able to do that if those two channels are currently bridged together. |
| 20:44.11 | p3nguin | If the two channels in question are on a call together, you can certainly spy them as one. |
| 20:45.27 | p3nguin | If you have two separate channels as parts of two separate calls, the only thing I can think of is to use two instances of ChanSpy and dump them both into a meetme where you are waiting and listening. |
| 20:45.57 | d_preston215 | Ok. |
| 20:47.00 | p3nguin | You can originate the spies with a Local channel and Application ChanSpy YourChannelHere |
| 20:47.33 | p3nguin | channel originate Local/spy@meetme Application ChanSpy SIP/whatever |
| 20:47.38 | p3nguin | something like that. |
| 20:47.46 | p3nguin | and then |
| 20:47.54 | p3nguin | channel originate Local/spy@meetme Application ChanSpy SIP/something-else |
| 20:48.04 | *** join/#asterisk mynoon (~my@46.20.109.147) |
| 20:48.23 | p3nguin | You would have already been in the meetme waiting. |
| 20:50.39 | *** join/#asterisk mynoon (~my@46.20.109.147) |
| 20:53.44 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:54.00 | talntid | p3nguin, i know what the issue is on my original PBX I was testing on |
| 20:54.07 | talntid | not sure how to fix it 100% yet, but I know what it is |
| 20:54.20 | p3nguin | Yes? |
| 20:54.20 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:55.07 | talntid | there are 2 pbx's on the network... 192.168.1.2 and 192.168.1.3.. |
| 20:55.16 | talntid | .2 is the main pbx, and .3 is specifically for the auto call stuff.. |
| 20:55.50 | talntid | when .3 sends call out, it's not getting the connection back to .3 soon enough... know what I mean? |
| 20:56.14 | talntid | its going to .2, cause that's what sip traffic is set to go back into... it eventually gets there.. just not fast enough.. |
| 20:56.58 | p3nguin | I see. SIP traffic has to take the long path. |
| 20:57.49 | talntid | yeah |
| 20:58.44 | fubada | talntid: you need to do iproute2 policy routes |
| 20:58.50 | fubada | do you want me to show you how? |
| 20:58.57 | talntid | yes |
| 20:59.00 | fubada | ok |
| 20:59.08 | fubada | redhat based distro? |
| 20:59.26 | talntid | the main pbx, is redhat based, yes, (.2) |
| 20:59.33 | talntid | the one I need traffic to get to, is ubuntu |
| 21:00.05 | fubada | wait im sorry, i thought you mean you have a single box with two interfaces and its choosing the wrong default ateway |
| 21:00.12 | talntid | nope |
| 21:00.14 | fubada | i read your problem incorrectly |
| 21:00.16 | fubada | sorry man |
| 21:00.19 | talntid | no worries |
| 21:02.11 | p3nguin | I'm interested to know why traffic can't go directly to/from the system where the work is being done. |
| 21:02.31 | talntid | we've got 1 static ip addrerss |
| 21:03.41 | p3nguin | yeah... |
| 21:03.41 | talntid | so, traffic goes out from 1 pbx, it comes back in, and appears to first be hunting to the first pbx first |
| 21:04.04 | p3nguin | That sounds like a faulty nat. |
| 21:04.10 | talntid | i agree |
| 21:04.24 | p3nguin | Imagine instead of two PBXs, you had two phones. |
| 21:04.25 | talntid | currently combing through the firewall |
| 21:04.26 | fubada | ive ditched the gui, and now im setting up * from scratch, the rpm for asterisk 1.8 installs a lot of noise into /et/asterisk, is it wise to rm -rf and start with a blank dir? |
| 21:04.32 | p3nguin | You call out from phone1... |
| 21:04.33 | talntid | exactly my argument to the IT guy |
| 21:04.45 | fubada | p3nguin/talntid |
| 21:04.46 | p3nguin | And return traffic would go to phone2 instead? |
| 21:05.18 | talntid | seems so, but that phone2 then says "not mine" and puts it back in the pool.. and phone1 picks it up |
| 21:05.29 | talntid | fast enough so ahuman won't notice, but a computer can |
| 21:05.35 | p3nguin | I doubt it would work like that. |
| 21:06.01 | talntid | yeah, but that's what it feels like |
| 21:14.35 | *** part/#asterisk newtonr (~newtonr@nat/digium/x-pmohgpjmethlcgcf) |
| 21:17.32 | *** join/#asterisk vastina (~vastina@70.99.103.122) |
| 21:22.34 | fubada | is it recommended to start with an empty /etc/asterisk directory |
| 21:22.40 | fubada | and create each file by hand |
| 21:23.07 | fubada | its overwhelming to look at the config that gets installed by the asterisk epel pkg |
| 21:23.26 | talntid | i edit it, using the examples i need... |
| 21:23.30 | talntid | then remove everything else |
| 21:23.53 | fubada | im like doing rm -rf first, then writing each config file |
| 21:23.59 | fubada | but theres no doumentation for that route |
| 21:25.00 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:28.58 | p3nguin | Refer to the sample files. |
| 21:29.10 | fubada | theres just an insane amoutn of noise |
| 21:30.38 | p3nguin | For the most part, the sample files are a good place to start, with exception of the extensions.conf. |
| 21:31.02 | fubada | i started with users.conf, added two extensions to it |
| 21:31.07 | p3nguin | ~users.conf |
| 21:31.07 | 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. |
| 21:31.11 | p3nguin | Wrong file! |
| 21:31.14 | fubada | dude |
| 21:31.19 | fubada | this project has THE WORST |
| 21:31.21 | fubada | documentation |
| 21:31.23 | fubada | and structure |
| 21:31.26 | p3nguin | ~book |
| 21:31.26 | 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://www.asteriskdocs.org/ or see ~buybook |
| 21:31.28 | fubada | god fuckign aweful |
| 21:31.31 | p3nguin | Read the book yet? |
| 21:31.35 | Qwell | or the wiki? |
| 21:31.37 | Qwell | ~asterisk wiki |
| 21:31.37 | infobot | i heard asterisk wiki is http://www.voip-info.org/wiki-Asterisk |
| 21:31.40 | fubada | looking at it now |
| 21:32.23 | p3nguin | Asterisk I can work with. Getting this damn system to be able to boot up... not so much. |
| 21:33.30 | p3nguin | I'm fighting with an issue where the scsi devices are not loaded until AFTER the root device is tried to be used. |
| 21:33.45 | p3nguin | After it falls back to initram prompt, then the device loads. :/ |
| 21:34.38 | Qwell | p3nguin: sounds like the CentOS installer |
| 21:34.48 | p3nguin | It's a bother. |
| 21:34.51 | Qwell | That's exactly what happens with USB CD drives. |
| 21:35.07 | p3nguin | I can't figure out how to add wait time so the bus can load first. |
| 21:45.52 | *** join/#asterisk dorphalsig (be9399cc@gateway/web/freenode/ip.190.147.153.204) |
| 21:45.55 | dorphalsig | Hello |
| 21:46.26 | *** part/#asterisk drhonk (~drhonk@static.57.112.4.46.clients.your-server.de) |
| 21:47.53 | dorphalsig | I have a SPA3k acting as a bridge between pstn and my * box. When a caller hangs up before the ivr is finished, my spa keeps the line open and asterisk routes the call as if somebody was still on the other side |
| 21:48.21 | [TK]D-Fender | dorphalsig: Then you SPA isn't detecting the line hangup. |
| 21:48.29 | [TK]D-Fender | ~cds |
| 21:48.29 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 21:48.31 | [TK]D-Fender | ^^^ |
| 21:48.38 | [TK]D-Fender | Ask your telco to enable this |
| 21:48.45 | [TK]D-Fender | And check the options on your SPA |
| 21:48.52 | [TK]D-Fender | This isn't an * side issue... |
| 21:48.53 | dorphalsig | I dont think they even know what that is :( |
| 21:49.02 | [TK]D-Fender | Think less, call more |
| 21:49.11 | dorphalsig | ok |
| 21:50.13 | dorphalsig | So while I get an answer from my telco is there anyway to tweak the spa? |
| 21:50.31 | [TK]D-Fender | Go google up the terms I've given you |
| 21:55.03 | dorphalsig | UTFGs |
| 21:59.32 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xfbtdzgpjwuznvzz) |
| 22:04.00 | zamba | what has modem.conf been replaced with? |
| 22:04.33 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:07.30 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:09.25 | WIMPy | zamba: Nothing. |
| 22:10.38 | zamba | WIMPy: oh.. i haven't got it here |
| 22:10.51 | zamba | WIMPy: i actually found an ISDN card i'm currently testing - Eicon Diva 2.0 |
| 22:11.17 | zamba | WIMPy: according to some google searches it's supposed to be working.. do you have any suggestions to how i should go about getting this working? |
| 22:11.46 | zamba | the linux kernel automatically uses the hisax module |
| 22:11.56 | WIMPy | What drivers are available for that thing? I guess it's CAPI only? |
| 22:12.13 | WIMPy | Asterisk cannot make use of I4L. |
| 22:12.37 | WIMPy | But if it can be used with misdn, you're in business. |
| 22:12.45 | zamba | how can i check these things? |
| 22:12.55 | zamba | any tools i can utilize? |
| 22:13.06 | zamba | i believe it can be used with misdn, yeah |
| 22:13.15 | WIMPy | If you don;t have any information about the card, try to load drivers. |
| 22:13.47 | zamba | the modules are loaded |
| 22:14.08 | zamba | the hisax module was loaded automatically |
| 22:14.27 | WIMPy | You need to ensure they are not loaded. |
| 22:15.02 | zamba | oh, ok |
| 22:16.12 | zamba | removing the modules now |
| 22:16.58 | zamba | btw was mISDNcore, mISDNipac and mISDNinfineon all loaded automatically |
| 22:17.02 | zamba | so maybe that's a good sign? |
| 22:17.57 | WIMPy | Maybe, buy you must make sure not to load multiple drivers. |
| 22:18.08 | zamba | they're all removed now |
| 22:19.47 | zamba | what then? |
| 22:20.55 | WIMPy | Check if the misdn modules support that card. |
| 22:21.19 | WIMPy | You can check with misdn_info after loading the drivers. |
| 22:21.33 | zamba | i have no misdn_info |
| 22:21.45 | WIMPy | Install mISDNuser |
| 22:22.24 | MrTelephone | if you tcpdump a udptl stream would you only see traffic in one direction? |
| 22:22.35 | zamba | WIMPy: is this the isdnutils package in ubuntu? |
| 22:22.46 | WIMPy | probably |
| 22:23.05 | WIMPy | Or it could be the isdn4k utils. |
| 22:23.53 | zamba | no such package in ubuntu, i'm afraid |
| 22:24.43 | WIMPy | I meant to say that "isdnutils" could be isdn4k-utils or mISDNuser. I have no idea. |
| 22:25.20 | zamba | http://www.misdn.org/index.php/Main_Page |
| 22:25.24 | zamba | it's this, yeah? |
| 22:25.43 | WIMPy | yes |
| 22:25.57 | zamba | "From Linux Kernel 2.6.27 on, mISDN v2 will be already included by kernel.org" |
| 22:26.06 | zamba | running 3.2.0 here |
| 22:26.28 | WIMPy | That obviousely doesn't include utilities. |
| 22:26.35 | zamba | true, true |
| 22:26.38 | zamba | downloading now |
| 22:28.20 | zamba | and compiling |
| 22:29.14 | zamba | # misdn_info |
| 22:29.17 | zamba | Cannot open mISDN due to Address family not supported by protocol. (Does your Kernel support socket based mISDN?) |
| 22:29.51 | WIMPy | Is the module loaded? |
| 22:29.59 | zamba | which module? hisax? it's loaded, yeah |
| 22:30.05 | zamba | but you told me earlier to remove all modules? :) |
| 22:30.19 | WIMPy | No. isdn4linux / hisax NUST NOT be loaded. |
| 22:30.19 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 22:30.32 | WIMPy | You need mISDN and the hardware modules to it. |
| 22:31.09 | zamba | mISDN_core? |
| 22:31.40 | zamba | yup.. loaded that module and got a message that i need to load card drivers.. |
| 22:31.48 | WIMPy | yes |
| 22:32.00 | zamba | is this something that's included in the kernel? |
| 22:32.07 | zamba | or do i need to get these drivers from somewhere else? |
| 22:32.35 | WIMPy | They are in the kernel. |
| 22:33.14 | WIMPy | But I don;t know which one might fit your card. |
| 22:33.37 | zamba | got a few under drivers/isdn/hardware/eicon/ |
| 22:33.53 | zamba | divacapi, divadidd, diva_idi, diva_mnt and divas |
| 22:35.42 | zamba | tried all the different and then issuing misdn_info after loading each.. but still the same error |
| 22:36.23 | WIMPy | That's not looking good then. |
| 22:36.41 | zamba | but it *should* work.. |
| 22:36.55 | WIMPy | Who says so? |
| 22:36.56 | zamba | http://www.trend-watcher.org/archives/asterisk-setup/ |
| 22:37.26 | zamba | i have exactly the same card |
| 22:37.55 | zamba | <PROTECTED> |
| 22:37.57 | WIMPy | That uses ISDN4Linux and that was only supported in very early versions of Asterisk. |
| 22:37.58 | zamba | urr |
| 22:38.04 | WIMPy | 1.0 or maybe 1.2. |
| 22:38.05 | zamba | http://pastie.org/3932943 |
| 22:38.12 | zamba | that's the output of lspci -v |
| 22:38.26 | zamba | so nothing has superseded isdn4linux for asterisk? |
| 22:38.38 | WIMPy | "mISDNinfineon"? |
| 22:38.59 | WIMPy | No, chan_moden was sinly discontinued. |
| 22:39.19 | zamba | hm |
| 22:39.21 | WIMPy | And misdn is the replacement for misdn which was the replacement for I4L. |
| 22:39.49 | zamba | what are your thoughts about mISDNinfineon? |
| 22:40.11 | WIMPy | That's what you pasted from lspci. |
| 22:40.16 | WIMPy | So that should be the driver you need. |
| 22:40.27 | zamba | oh! |
| 22:40.43 | MrTelephone | do you know the default rtp timeout off the top of your mind? |
| 22:40.48 | zamba | http://pastie.org/3932956 |
| 22:40.53 | zamba | is this success? |
| 22:40.54 | *** join/#asterisk Silvestre_id (~Silvestre@186.204.140.234) |
| 22:41.00 | WIMPy | So it will probably work, I guess. |
| 22:41.10 | zamba | WIMPy: what do i do from here? |
| 22:41.16 | WIMPy | yes |
| 22:41.28 | WIMPy | You need LCR. |
| 22:41.38 | WIMPy | That was linked from the same page. |
| 22:43.00 | WIMPy | MrTelephone: Quite high. 30s or maybe even more. |
| 22:43.19 | zamba | don't like the lack of binary packages, but i'll live :) |
| 22:43.28 | WIMPy | Plus an extra second for an off-by-one-error. |
| 22:43.43 | MrTelephone | i'm trying to figure out a t38 negotiation with arris modems. asterisk says TRYING.. and i see packets going towards the endpoint. exactly 5 seconds later asterisk replies with 488 not acceptable here. |
| 22:44.58 | MrTelephone | endpoint gives the t38 information, asterisk responds with sip TRYING. asterisk sends packets to the proper port. after the fallback to voice/rtp there is communication both ways.. |
| 22:46.20 | MrTelephone | I'm wondering if asterisk is waiting for packets or something from the modem telephone adapter. It shouldn't be waiting for a SIP message because it said it was Trying. Funny it just doesn't send the packets and send a 200 OK or ACK |
| 22:48.57 | Silvestre_id | Hi. I'm using Asterisk 1.8.9.2. I made a call and I waited the called party begins to ring. I dropped the call while it rings and my channel doesn't go to h extension. Someone known how i can go to h extension? |
| 22:49.57 | zamba | WIMPy: lcr installed |
| 22:50.39 | WIMPy | Ok, so it's about interface.conf then. |
| 22:51.02 | WIMPy | Should be straight forward to pick the right exaple. |
| 22:51.38 | zamba | hm.. but maybe we should rewind a bit here.. what is the difference between NT and NE mode? |
| 22:51.54 | zamba | i do lcr query and i see that the card supports TE-mode |
| 22:52.08 | WIMPy | Network Termination or Terminal Equipment |
| 22:52.11 | zamba | that's what i want? |
| 22:52.29 | WIMPy | If you want to connect it to a line, yes. |
| 22:52.35 | zamba | we get our telephony through two NT boxes from our provider (telenor) |
| 22:52.47 | zamba | so we have four lines into our offices |
| 22:52.57 | zamba | so this is all good? |
| 22:53.09 | WIMPy | two lines, four channels :-) |
| 22:53.17 | zamba | yup |
| 22:53.19 | WIMPy | yes |
| 22:53.23 | zamba | i meant four channels :) |
| 22:53.53 | zamba | but, and this was something i meant to ask my provider earlier, how do the different NT boxes differ on the phone numbers? |
| 22:53.59 | zamba | because we have 20 phone numbers available to us |
| 22:54.16 | zamba | and four channels |
| 22:54.38 | zamba | it makes sense that the provider has terminated 10 on each NT box, or? |
| 22:55.08 | WIMPy | Not neccessarily. |
| 22:55.18 | zamba | oh? how can it be handled instead? |
| 22:55.19 | WIMPy | Buth lines might share the same block. |
| 22:55.35 | zamba | ah, ok |
| 22:55.45 | zamba | because i want to use one of the NT boxes for testing asterisk |
| 22:55.50 | WIMPy | Do you have DDI? Do you know if the lines ate ptp or ptmp? |
| 22:55.54 | zamba | have no idea |
| 22:56.01 | zamba | never heard about DDI before |
| 22:56.19 | WIMPy | Is it a block of 20 consecutive numbers? |
| 22:56.43 | zamba | yup |
| 22:57.11 | WIMPy | Then it's most likely one trunk over two ptp lines. |
| 22:57.18 | zamba | ok |
| 22:57.23 | WIMPy | ptp means that you can only connect one device. |
| 22:57.25 | zamba | how should i configure my end then? |
| 22:57.40 | WIMPy | As ptp |
| 22:58.07 | WIMPy | The issue is that you won't know on which line a call will arrive. |
| 22:58.08 | zamba | what's the difference between Ext and Int here? |
| 22:58.18 | zamba | ah, i see |
| 22:58.34 | zamba | that's why we have this dinosaur of an telephony server in the offices already |
| 22:58.44 | zamba | because it has both NT boxes terminated |
| 22:58.52 | zamba | and i guess that handles exactly this |
| 22:58.55 | WIMPy | int is for interfaces with telephones connected, managed by LCR. |
| 22:59.05 | zamba | well.. i will have no int then |
| 22:59.09 | zamba | right? |
| 22:59.16 | WIMPy | right. |
| 22:59.31 | WIMPy | You only have one interface. |
| 22:59.35 | WIMPy | Or is it a multi port card? |
| 22:59.47 | zamba | only one interface |
| 23:00.14 | WIMPy | That's only one external. |
| 23:00.57 | zamba | which options should i set for the ext stanza then? |
| 23:00.59 | WIMPy | But I have internal ports configured as ext anyway, because I don't use LCRs extension management, but do it all via Asterisk. |
| 23:01.04 | WIMPy | But that's nothing you have to care about. |
| 23:01.12 | zamba | i can't do all via asterisk as well? |
| 23:01.14 | WIMPy | Most probably ptp. |
| 23:01.26 | WIMPy | Sure |
| 23:01.46 | zamba | unless lcr is the better method....? |
| 23:02.07 | WIMPy | But LCR has some handling for phone extensions. But even if you connect phones to it, you don't have to use it. |
| 23:02.26 | zamba | i want something that's easy and simple to set up |
| 23:02.38 | WIMPy | To do what? |
| 23:02.41 | zamba | and that works :) |
| 23:02.47 | zamba | to dial, basically |
| 23:02.53 | zamba | and receive calls |
| 23:03.02 | WIMPy | To/from where? |
| 23:03.08 | zamba | and then these calls will be sent over SIP internally in the offices |
| 23:03.21 | zamba | so this asterisk box will be the gateway between SIP and ISDN, or whatever you call it |
| 23:03.35 | zamba | if that makes sense? |
| 23:03.38 | WIMPy | I'm not sure it's in any release, yet, but the current LCR also supports SIP. |
| 23:03.48 | WIMPy | But I haven't tried that. |
| 23:03.54 | zamba | but i still want to run asterisk |
| 23:04.06 | zamba | you mean LCR -instead- of asterisk now, right? |
| 23:04.21 | WIMPy | Yes. That should be possible. |
| 23:04.27 | zamba | yeah, but i still want to run asterisk |
| 23:04.37 | zamba | and just use the isdn as a channel |
| 23:04.40 | WIMPy | But I'm (alsmost) only using it as channel for Asterisk. |
| 23:04.44 | zamba | yeah |
| 23:04.56 | zamba | so we can skip the lcr step then? |
| 23:05.07 | WIMPy | nope |
| 23:05.24 | WIMPy | Asterisk can't use mISDN2. |
| 23:05.33 | WIMPy | You need to do it via chan_lcr. |
| 23:05.35 | zamba | ok, so we need LCR as a gateway? |
| 23:05.37 | zamba | ok |
| 23:06.09 | WIMPy | Actually a chan_misdn2 exists, but is not available and won't work with current Asterisk versions anyway. |
| 23:06.14 | WIMPy | exactely. |
| 23:08.43 | zamba | hm.. there's no chan_lcr available here either |
| 23:09.00 | zamba | i need to compile that as well? |
| 23:09.10 | WIMPy | That's made by lcr if it finds an asterisk installation. |
| 23:09.17 | zamba | ah, ok |
| 23:09.20 | zamba | nice |
| 23:09.36 | zamba | but what do i need to configure in interface.conf? |
| 23:09.45 | WIMPy | Wehn you install lcr, you should also get chan_lcr. |
| 23:09.54 | zamba | just [Ext] and then just ptp? |
| 23:10.16 | *** join/#asterisk joobie (~joobz@unaffiliated/moo0o0ooo00o0o0o) |
| 23:10.24 | WIMPy | [some name] portnum 0 extern ptp |
| 23:10.37 | zamba | yup, that's what i've got |
| 23:10.49 | WIMPy | The name is what you use to dial from Asterisk. |
| 23:11.17 | zamba | Dial(<name>/$EXT), yeah? |
| 23:11.19 | WIMPy | Dial(lcr/some name/number/options) |
| 23:11.25 | zamba | yeah |
| 23:11.35 | zamba | do i need to edit routing.conf? |
| 23:11.46 | WIMPy | For the other way, you need the routing.conf. |
| 23:11.47 | zamba | i don't want to use lcr for this, right? |
| 23:11.53 | zamba | ah, ok |
| 23:12.24 | WIMPy | Yes, just a : remote application=asterisk context=dialplan context |
| 23:12.37 | WIMPy | under [main] that is. |
| 23:12.43 | WIMPy | Only those two lines. |
| 23:13.08 | WIMPy | Yes, just a ": remote application=asterisk context=dialplan context" |
| 23:13.15 | WIMPy | (the : is part of that line) |
| 23:13.42 | zamba | one line is ': remote application=asterisk' and the other is 'context=dialplan context'? |
| 23:14.00 | zamba | got it |
| 23:14.03 | WIMPy | Nope that's the 2nd line. The first is [main] |
| 23:14.13 | zamba | ah :) |
| 23:15.01 | WIMPy | I really need to write the howto section. |
| 23:15.08 | zamba | yeah, you do ;) |
| 23:15.54 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 23:16.02 | WIMPy | And hope the whole thing will be supported in the future. |
| 23:16.53 | zamba | hm.. error here |
| 23:17.00 | zamba | did i interpret your line too literally? |
| 23:17.02 | WIMPy | Where? |
| 23:17.17 | zamba | http://pastie.org/3933084 |
| 23:17.20 | zamba | that's when doing lcr fork |
| 23:17.39 | WIMPy | I guess it doesn't like the space. |
| 23:17.58 | WIMPy | Yu should put the Asterisk dialplan context there where you want the calls to go. |
| 23:18.21 | *** join/#asterisk drhonk|afk (~drhonk@static.57.112.4.46.clients.your-server.de) |
| 23:18.26 | *** part/#asterisk drhonk|afk (~drhonk@static.57.112.4.46.clients.your-server.de) |
| 23:20.42 | zamba | yeah, changed it to from-isdn |
| 23:21.03 | zamba | oh |
| 23:21.04 | zamba | it's running |
| 23:21.23 | zamba | but no chan_lcr |
| 23:21.30 | WIMPy | Congatulations. You can now forget about LCR and use Asterisk. |
| 23:21.42 | WIMPy | Oh |
| 23:22.08 | WIMPy | Then the configure must have missed your Asterisk installation. |
| 23:22.14 | zamba | but maybe i cheated, since i pulled down a deb of lcr..? |
| 23:22.39 | zamba | instead of compiling it myself |
| 23:22.46 | WIMPy | The Asterisk module needs to match your Asterisk version. |
| 23:23.27 | WIMPy | So that may be hard to make sure with packages. |
| 23:23.38 | zamba | yeah |
| 23:23.42 | zamba | i'll try building it myself |
| 23:23.50 | zamba | but it's not bootstrapped.. or whatever it's called |
| 23:24.06 | WIMPy | What do you mean? |
| 23:24.18 | zamba | no configure script |
| 23:24.34 | WIMPy | There is a configure script, sure. |
| 23:24.58 | zamba | git clone git://git.misdn.eu/lcr.git/ |
| 23:25.02 | zamba | no configure script there |
| 23:25.10 | WIMPy | That will look for optional support, like for Asterisk. |
| 23:25.37 | WIMPy | Ah, git version. |
| 23:25.48 | zamba | guess i need aclocal and all that stuff for that |
| 23:25.54 | WIMPy | No you need to run tehe autoconf script first. |
| 23:26.16 | zamba | yeah, but i need aclocal and stuff for that |
| 23:26.18 | WIMPy | autogen.sh |
| 23:26.25 | WIMPy | yes |
| 23:26.31 | zamba | so i'm pulling down that now |
| 23:26.39 | WIMPy | If you take the tarball, you only have to ./configure. |
| 23:26.50 | zamba | yeah |
| 23:27.02 | zamba | need asterisk-dev as well, i guess+ |
| 23:27.15 | WIMPy | Probably |
| 23:28.05 | zamba | ah, damn |
| 23:28.06 | WIMPy | Well, pretty certain. |
| 23:28.12 | zamba | syntax error |
| 23:28.16 | zamba | let's go for the tarball |
| 23:28.23 | zamba | where's that? |
| 23:28.41 | zamba | ah, got it |
| 23:28.42 | WIMPy | http://linux-call-router.de/ |
| 23:29.13 | zamba | this needs to be supported directly in asterisk ;) |
| 23:29.24 | zamba | i agree with you :p |
| 23:29.45 | WIMPy | Would be nice, yes, but I guess noone will do so. |
| 23:30.25 | zamba | *sigh* |
| 23:30.34 | zamba | doesn't look like it wants to compile |
| 23:30.57 | zamba | http://pastie.org/3933128 |
| 23:30.59 | WIMPy | Andreas is thinking about dropping support for chan_lcr for the same reason chan_capi isn't available for Asterisk 10 and chan_misdn2 isn't available at all. |
| 23:30.59 | zamba | and so on |
| 23:31.24 | WIMPy | No curses(-dev) installed? |
| 23:31.34 | zamba | yup, just installed it |
| 23:32.33 | WIMPy | Constantly patching for the ever changing Asterisk APIs gives ppl quite some work and some frustration. |
| 23:34.52 | zamba | argh.. what now? |
| 23:35.06 | zamba | i HATE compiling :p |
| 23:35.10 | WIMPy | Where are you? |
| 23:35.26 | zamba | http://pastie.org/3933137 |
| 23:36.15 | WIMPy | That's exactely one of there API changes. |
| 23:36.25 | *** join/#asterisk DarthExpeditor (~IceChat9@c-24-11-175-127.hsd1.mi.comcast.net) |
| 23:36.27 | WIMPy | Wat versions of Asterisk and LCR do you have? |
| 23:36.43 | zamba | 1.8.10 |
| 23:36.53 | zamba | 1.7 of lcr |
| 23:37.08 | *** join/#asterisk wtf911 (~wtf911@cpe-67-251-103-137.stny.res.rr.com) |
| 23:37.16 | WIMPy | Use 1.10 of lcr |
| 23:37.22 | zamba | i'm using the asterisk version that comes with ubuntu 12.04 |
| 23:37.30 | zamba | oh! |
| 23:37.32 | zamba | haha |
| 23:37.57 | zamba | that was my bad |
| 23:38.25 | WIMPy | That's what happens when numbers are sorted as strings. |
| 23:38.53 | zamba | yeah :) |
| 23:41.49 | zamba | *sigh* |
| 23:42.23 | zamba | http://pastie.org/3933164 |
| 23:42.32 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:42.57 | zamba | i HATE compiling |
| 23:43.15 | WIMPy | Must be te distro :-) |
| 23:43.27 | WIMPy | You installed mISDNuser from source? |
| 23:43.30 | zamba | yes |
| 23:43.33 | zamba | the git source |
| 23:43.42 | zamba | .. maybe use the tarball here as well :) |
| 23:43.50 | WIMPy | No |
| 23:43.59 | zamba | what then? |
| 23:44.10 | WIMPy | The git version is good. |
| 23:44.17 | WIMPy | Not sure what it's missing there. |
| 23:44.41 | WIMPy | diggs around... |
| 23:47.16 | WIMPy | Interesting. I find mISDN_debug_init in mISDNuser, but not in lcr. |
| 23:48.34 | WIMPy | Hmm, yes, it has been removed from lcr. |
| 23:48.43 | WIMPy | Wait. |
| 23:50.05 | WIMPy | That's a little strange. That function appears in mISDNuser for the old misdn1. |
| 23:50.31 | WIMPy | Do you perhaps have a olde version of mISDNuser installed as well? |
| 23:51.53 | WIMPy | Hmm. Not sure how old that version actually is. |
| 23:52.11 | zamba | from source? |
| 23:52.29 | zamba | the /usr/local/include/mISDN/ dir is dated from today |
| 23:52.36 | zamba | from when i compiled and installed it |
| 23:52.39 | zamba | from the git repo |
| 23:52.47 | WIMPy | Hmm. No the mISDNuser version where I find that function seems to work with misdn2. |
| 23:53.20 | WIMPy | But it doesn;t exist in the current verison. |
| 23:53.51 | WIMPy | What git repo did you use? |
| 23:55.47 | zamba | git://git.misdn.eu/mISDNuser.git/ |
| 23:56.05 | WIMPy | Sounds good. |
| 23:58.03 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 23:59.31 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |