| 00:01.37 | sebastianpersic | Are they all the same depending to which format I choose |
| 00:02.08 | sebastianpersic | wav, ulaw,sln etc. |
| 00:02.23 | WIMPy | Off course |
| 00:02.42 | sebastianpersic | what's the point of having different formats in the system |
| 00:02.54 | sebastianpersic | for kb/s rate? |
| 00:03.11 | WIMPy | To avoid transcoding if possible. |
| 00:03.44 | sebastianpersic | transcoding for different sip phones? |
| 00:03.59 | WIMPy | For different formats. |
| 00:04.28 | WIMPy | There are many CODECs to choose from. |
| 00:05.28 | sebastianpersic | Yes, but if I ask it this way. When must the system choose either of it.. In which situations |
| 00:06.44 | WIMPy | If the format that's used on a call exists as a file, that's the first choice. |
| 00:08.06 | sebastianpersic | Ok, so this can depend of what kind of call that is made for example. |
| 00:09.40 | WIMPy | yes |
| 00:15.42 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 00:22.32 | jeffspeff | having an issue with realtime sip in mysql database. for some reason, it's not recognizing that i have any codecs specified in 'allow' |
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| 00:48.26 | *** join/#asterisk k1ng__ (~k1ng@unaffiliated/k1ng) |
| 00:48.28 | k1ng__ | hello |
| 00:48.43 | k1ng__ | is there anyway to use alpanumeric? |
| 00:49.09 | k1ng__ | is there anyway to use alpanumeric on extension? |
| 00:49.25 | WIMPy | yes |
| 00:49.27 | jpsharp | If your endpoints can dial using alphanumeric, yes. |
| 00:50.41 | k1ng__ | i tried to create an extension with alpanumeric but i didnt allow me |
| 00:51.08 | WIMPy | Asterisk doesn't care. |
| 00:51.20 | k1ng__ | freebpx |
| 00:51.32 | WIMPy | ~freepbx |
| 00:51.32 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
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| 01:54.25 | dampjam | Hey - when I hook a grand stream 286 SIP device up to asterisk, and have an old-style phone attached/dial….. is it the grand stream device that buffers the dtmf when I'm dialing numbers, or does asterisk get each number I dial as I dial them? |
| 01:59.59 | WIMPy | The ATA. |
| 02:00.30 | WIMPy | I don't know if there are any where you can disable that. |
| 02:10.35 | [TK]D-Fender | it isn't something to "disable". You can have is dial immediately upon going offhook and do the rest in-call with * but that'll wreck your CDR, etc |
| 02:12.33 | dampjam | So I researched more - turns out I'm looking for something called "early-dial" and want asterisk to send the 484 back after each digit (incomplete)….. |
| 02:12.41 | dampjam | But I'm not sure that's a mode asterisk supports ---- |
| 02:13.16 | WIMPy | It would be something to disable. |
| 02:13.42 | WIMPy | Yes, Asterisk supports that. |
| 02:14.01 | WIMPy | But I don;t know if any ATA exists that can do it. |
| 02:14.23 | WIMPy | And it's usually called overlap dialling. |
| 02:19.33 | dampjam | My grand stream 286 claims to support it --- |
| 02:20.23 | jpsharp | It does, but I never got it to work right. |
| 02:20.27 | jpsharp | But that was a few years ago. |
| 02:21.32 | camelCase | allowoverlap=yes|no |
| 02:21.49 | camelCase | is what you are looking for |
| 02:24.39 | dampjam | Nice -- that worked perfectly -- for the record allowoverlap=dtmf did not work with the grand stream, had to be yes. |
| 02:26.00 | dampjam | I have one of those really old rotary phones, hooked up to a 4-prong -> rj11 adapter, hooked up to a dp -> dtmf adapter, hooked up to a grand stream 286, hooked up to asterisk, joined to google voice. Now I can make VOIP calls with 1950s technology. |
| 02:28.40 | camelCase | did you get the adapter from a cat in austraillia? |
| 02:29.43 | camelCase | i bought one built in a biscuit jack housing based on some form of PIC |
| 02:30.11 | dampjam | yeah - same guy |
| 02:30.14 | camelCase | cool |
| 02:30.20 | camelCase | it works well eh? |
| 02:30.46 | camelCase | when i have time I am getting a tattoo of an old bakelite rotary with crossbones |
| 02:30.47 | dampjam | well - dialing is too slow, so the grand stream 286 tries to dial when you're 1-2 digits into the number if you're dialing anything larger than a 1.... |
| 02:30.56 | camelCase | ouch |
| 02:31.10 | camelCase | i have a tdm card |
| 02:31.12 | dampjam | even setting my internal # to 111 didn't let me dial fast enough. |
| 02:31.14 | dampjam | So turning on allow overlap is necessary |
| 02:31.25 | camelCase | and a cisco asa |
| 02:31.28 | camelCase | *not that |
| 02:31.30 | dampjam | ah, nice. |
| 02:31.31 | camelCase | ata |
| 02:31.36 | camelCase | but i do have an asa |
| 02:31.53 | camelCase | it's a tdm400p |
| 02:32.46 | jpsharp | Does the system have enough juice to actually ring the phone? |
| 02:33.00 | dampjam | yup |
| 02:33.06 | camelCase | yeah |
| 02:33.08 | dampjam | very loudly |
| 02:33.20 | camelCase | magicjack dont |
| 02:33.22 | WIMPy | You may run in to strange effects if you dial out to a SIP ITSP that way, BTW. |
| 02:33.23 | camelCase | :-/ |
| 02:33.57 | camelCase | WIMPy: why? |
| 02:34.02 | WIMPy | That is from SIP to SIP. All other combinations seem to be ok. |
| 02:34.27 | camelCase | you may have problems if you use teliax |
| 02:34.30 | camelCase | :P |
| 02:34.58 | WIMPy | Because Asterisk can't do outgoing overlap on SIP and unlike other combinations of channels it doesn't collect digits by itself when going from SIP to SIP. |
| 02:36.30 | camelCase | but, you dial the number and it is passed from your * box to the ITSP |
| 02:36.42 | camelCase | or were you saying going directly from the ata to the itsp? |
| 02:37.04 | WIMPy | No, that's exactely what does not work. |
| 02:37.09 | WIMPy | Via Asterisk. |
| 02:37.38 | WIMPy | From the ATA to an ITSP is a question of what that ITSP supports. |
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| 02:59.51 | lkthomas | hey guys |
| 03:00.31 | lkthomas | does asterisk have any plugin to monitor the T1 situation ? |
| 03:02.06 | volga629 | Hello Everyone, I am looking solution for nat and sip, I have vpn server which open source is openswan and site to site branch tunnel terminates on this box. User connecting to asterisk through this tunnel. I put qualify 3000 to delay checks, is user need set to nat=yes, what else can done to improve little bit connection because latency not great at all |
| 03:04.44 | volga629 | DNAT on this box add more to final latency, because of processing |
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| 03:08.23 | Narkov | I'm not that familiar with the revision tracker. Can anyone tell me what (if any) Asterisk release revision 190350 is in? |
| 03:09.11 | Narkov | Revision 190350 is the HTTP Digest auth for the AMI. I can't find it in Debian Asterisk 1.6.2.9 |
| 03:09.15 | [TK]D-Fender | Narkov: http://lists.digium.com/pipermail/asterisk-commits/2009-April/033039.html |
| 03:10.12 | Narkov | Thanks [TK]D-Fender. So that means it is in 1.6.2 ? Would you know why I can't see the feature in the Debian release of 1.6.2.9 ? |
| 03:14.54 | [TK]D-Fender | Narkov: http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ |
| 03:15.11 | [TK]D-Fender | Date: Thu Apr 23 15:37:16 2009 |
| 03:15.44 | [TK]D-Fender | That should mean it was in there |
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| 03:17.11 | Narkov | [TK]D-Fender: That's the confusing part. I can't see any reference to "r190349" or "r190350" in the change logs |
| 03:17.19 | *** part/#asterisk dampjam (~walter@68.33.9.147) |
| 03:17.51 | Narkov | ...and my limited understanding suggests it may not have been "released" yet. |
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| 03:24.31 | [TK]D-Fender | Narkov: from the changelog in 1.6.2.10 I scanned back for that range and it should be in there I would think... |
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| 03:26.59 | Narkov | [TK]D-Fender: That is the thing, in the 1.6.2.10 changelog there is no reference to 190350 or anything to do with the patch. Could it not be "released" yet? |
| 03:27.13 | Narkov | Perhaps it may have been skipped in the change log? |
| 03:27.49 | [TK]D-Fender | Didn't mention it explicitly. Not sure if it got "bundled". I never really looked at revision details quite so much before.... |
| 03:30.04 | Narkov | How would you confirm it is actually in the release (or indeed any release)? |
| 03:58.45 | leifmadsen | Narkov: svn log on the tag probably |
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| 04:54.35 | Marquel | morning. is it possible to use the state of a single dahdi channel in a hint? or are there just spans of channels supported? |
| 04:55.51 | [TK]D-Fender | yes |
| 04:59.50 | Marquel | [TK]D-Fender: honestly, you can confuse a boolean logic brain. |
| 05:00.21 | [TK]D-Fender | Former |
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| 05:01.08 | Marquel | [TK]D-Fender: would you please be so kind as to hint me to a howto? hint,DAHDI/<channel> did not work. |
| 05:01.36 | [TK]D-Fender | Marquel: It does. |
| 05:02.06 | Marquel | interesting... :/ |
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| 05:26.31 | DND | hi guys]\ |
| 05:28.05 | DND | i have asterisknow with freepbx. i check on cdr and i have this incoming at around 1am: http://pastebin.com/gByPDj52 there's another incoming which the source is a valid number but the destination is "s" so what does "s" and "i" means? |
| 05:49.16 | beebeeep | DND: 'i' means |
| 05:49.29 | beebeeep | ... 'invalid' extension |
| 05:50.18 | beebeeep | 's' means start or unspeciefied xtension, i.e. for incoming INVITE without DID number |
| 05:51.43 | DND | hmm |
| 05:53.55 | beebeeep | if your inbound calls are placed into 's' extension, mb you just forgot to specify callback extension in trunk registration string |
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| 07:13.30 | bulkorok | hi |
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| 08:43.30 | k610 | can i delete-disable extensions.ael if I only use extensions.conf |
| 08:46.18 | wdoekes | yes |
| 08:46.39 | wdoekes | I'd even recommend cleaning out all of /etc/asterisk and leaving only those files that you've found out you need |
| 08:47.01 | wdoekes | (asterisk.conf, modules.conf, sip.conf, extensions.conf, ...) |
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| 08:57.36 | pbxMan | hello |
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| 09:17.40 | kresp0 | hi |
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| 09:31.45 | dym | Not sure if my question was answered yesterday, but how do i react to this: "attempt_thread: Call failed to go through, reason (3) Remote end Ringing" - i'd like to execute a system command in case this happens |
| 09:48.04 | wdoekes | dym: dial a local channel where you do the dial and handle timeouts there |
| 09:49.34 | dym | wdoekes: i am. but how can i react to a timeout? I know how to set a dial timeout, but then it doesnt use the following priority |
| 09:50.21 | wdoekes | it should go to the next prio after a "failed" dial.. and timeout counts as failure |
| 09:51.24 | dym | I have a system call which is not initiated right in the next priority |
| 09:53.16 | *** join/#asterisk mattp (~mattp@fuchs.unfinished.org.uk) |
| 09:53.31 | wdoekes | is the spool timeout shorter than the dial timeout? |
| 09:54.54 | dym | whats the default dial timeout? |
| 09:54.57 | dym | if none is set |
| 09:55.22 | wdoekes | I think we're talking about waittime, which is 45.. |
| 09:56.08 | wdoekes | but if you're testing this, you will probably have lowered your dial timeout to 5 seconds |
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| 10:01.06 | dym | wdoekes: i've set it to 5, the call terminates, but omits the next priority |
| 10:04.40 | wdoekes | you should probably up the verbosity and debug and pastebin that |
| 10:27.02 | tompaw | Guys, can you recommend any G729 compatible softphone for OSX? |
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| 10:30.27 | thejpster | Hello. I'm running Asterisk 10.8.0 on a Beagle Board with some success. I can connect my N900 over SIP and place calls, yada yada. I'm having no luck with the Console audio though - I just hear loud clicks and nothing else, whether dialling out to the N900 or dialling the demo extension. Has anyone seen this before? |
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| 10:38.36 | sgimeno | hi! my asterisk is ocassionally responding to some registers with 482 Loop Detected, what could be causing this? |
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| 11:07.43 | as001 | Hello, I have strange problem with my Asterisk PBX. When call comes from PSTN (dahdi/2) i do Dial(Dahdi/1,40,hH) in dialplan (ring my ordinary phone) but if person who called me hangup in those 40 secs before I answer the call (example: he hangs up after 20 sec) my Dahdi/1(ordinary phone) rings to 40 secs like it did not receive hangup. When I answer the call and hang up after everything is fine. Can you help me with this issue ? |
| 11:10.01 | as001 | I have busydetect=yes and busycount=4 and it is strange that everything works fine if I answer the call no metter who hang up, but when person who call hang up before I answer my phone continue to ring until timeout from Dial command. |
| 11:11.02 | as001 | Shouldn't it stop ringing when calling party hang up no metter how did i set timeout in Dial command ? |
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| 11:13.51 | tompaw | Is there a definitive list of countries, where G729 patents are valid (e.g. hosting.lv codecs are illegal)? |
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| 11:24.57 | WIMPy | tompaw: That might even depend on which version. |
| 11:25.31 | WIMPy | But I'm pretty sure you are supposed to believe it's everywhere, so noone will tell you. |
| 11:25.36 | tompaw | :o We've got infrastructure in us, panama and eu and I was hoping to find a simple list or even a map on Sipro Lab website. |
| 11:26.18 | tompaw | WIMPy: yeah but we both know it doesn't work that way. I think patent holders should be obliged by the law to make that information clear and public. |
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| 11:26.23 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 11:28.10 | WIMPy | I think they will fight VERY hard to not do so. |
| 11:28.34 | WIMPy | sgimeno: A DNS failure |
| 11:30.35 | sgimeno | WIMPy: can u elaborate more please? |
| 11:30.56 | *** join/#asterisk tvoip (b4f2a1f5@gateway/web/freenode/ip.180.242.161.245) |
| 11:31.03 | tvoip | hello experts.... |
| 11:31.28 | tvoip | anyone knows about device or addon apps for tapping |
| 11:31.36 | WIMPy | sgimeno: You're talking to yourself because you can't find out who to talk to. |
| 11:31.38 | tvoip | or asterisk can do that ? |
| 11:32.10 | WIMPy | tvoip: yes |
| 11:32.28 | sgimeno | WIMPy: uhm, but it's a register with an ip as domain, not a hostname |
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| 11:32.44 | tvoip | give me reference please... |
| 11:33.24 | WIMPy | sgimeno: Are you sure it's related to that? |
| 11:33.57 | WIMPy | tvoip: ChanSpy, ExtenSpy |
| 11:34.24 | tompaw | WIMPy: it is required by every other aspect of law, isn't it? To be expressed clearly and accessible for everyone. |
| 11:34.55 | sgimeno | WIMPy: see http://pastebin.com/fyMBuWhx |
| 11:34.57 | tvoip | WIMPy: thank you so much Boss.... |
| 11:35.51 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:35.53 | WIMPy | sgimeno: Unlikely I can see that. |
| 11:36.06 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 11:36.24 | sgimeno | yeah, I had never seen a 482 to a REGISTER |
| 11:37.51 | WIMPy | I have. |
| 11:38.07 | WIMPy | Every time I've been offline. |
| 11:39.29 | WIMPy | Wow. I got that pb. |
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| 11:40.31 | WIMPy | What is trying to register from 172.17.4.88? |
| 11:41.20 | sgimeno | its a softphone |
| 11:41.59 | sgimeno | 172.17.4.80 |
| 11:42.26 | WIMPy | I don;t know when a 482 is triggered. Maybe it recycles some ID that's supposed to be unique. |
| 11:42.46 | sgimeno | it could be |
| 11:42.55 | sgimeno | it rarely happens |
| 11:43.02 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 11:43.18 | sgimeno | i'll check chan_sip |
| 11:43.35 | sgimeno | thank for your time WIMPy |
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| 12:35.16 | Lantizia | Hey, say I've got two DID's pointing to the same conference room... one DID is french, the other DID is english... if I have both sound packs installed - can I make the person calling the french number hear the french versions of the sound files? |
| 12:35.54 | [TK]D-Fender | Lantizia, its your dialplan, you can do whatever yuo want with them |
| 12:36.06 | [TK]D-Fender | Lantizia, including changing the CHANNEL's language |
| 12:36.18 | Lantizia | so there's something I can put in the dialplan to change sound pack? |
| 12:36.34 | Lantizia | that's what I though - know what the variable is called? |
| 12:36.39 | Lantizia | *thought |
| 12:36.44 | [TK]D-Fender | "core show function CHANNEL" |
| 12:36.51 | Lantizia | ok |
| 12:38.40 | Lantizia | i see it thanks |
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| 12:49.02 | jeffspeff | having an issue with realtime sip in mysql database. for some reason, it's not recognizing that i have any codecs specified in 'allow' |
| 12:49.48 | [TK]D-Fender | jeffspeff, the field order matters. You are required for the allow field to be AFTER the deny field |
| 12:50.18 | jeffspeff | [TK]D-Fender, oh wow, that would explain it. |
| 12:50.24 | thejpster | I'm getting clicking instead of sound with chan_alsa audio on a beagleboard. Currently compiling chan_alsa.so with extra bits of debug in. What magic do I have to do for ast_debug(...) calls to print out? |
| 12:50.38 | jeffspeff | [TK]D-Fender, thanks. i'll change that and test |
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| 12:54.52 | dym | When executing a call via Dial, is it possible to get the used Channelid from that call without waiting for it to jump to another priority, or the call beeing bridged somewhere? |
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| 12:59.10 | WIMPy | The predial handlers might give you access to them. |
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| 13:01.31 | [TK]D-Fender | dym, When? do what with it? Where? |
| 13:02.17 | thejpster | I've added code to dump frames supplied to alsa_write() to a file. If I play that file with aplay, it sounds OK. That rules out codec issues and now it looks like an ALSA setup issue. |
| 13:02.32 | jeffspeff | dym, you might look at using a macro in the dial command... Dial(SIP/1234,,(m)foo) |
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| 13:37.44 | Sidrov | Hello all |
| 13:37.57 | Sidrov | is there any default limit in asterisk for concurent calls ? |
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| 13:40.39 | jeffspeff | Sidrov, no |
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| 13:41.38 | jeffspeff | concurrent voice paths is dependant on your max lines or whatever you use, or if you're using SIP, then you're only restricted by bandwidth and local system resources (RAM, processor, etc.) |
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| 13:51.03 | Sidrov | jeffspeff, thank you |
| 13:52.02 | Sidrov | is there any way I can use two providers for outbound, and in case one line isn't acceptinc calls to route through the other one ? |
| 13:52.28 | leifmadsen | Sidrov: yes, just set it up in your dialplan to fail over to the secondary provider |
| 13:53.52 | Sidrov | leifmadsen, how, if i place call to SIP/provider1 then i should retry to SIP/provider2 |
| 13:54.10 | Sidrov | i was wondering if i can make it in sip.conf and dial SIP/mygateway |
| 13:54.16 | leifmadsen | look at the ${DIALSTATUS} variable and do a GotoIf() or ExecIf() |
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| 13:54.22 | leifmadsen | Sidrov: no, it doesn't work that work |
| 13:54.26 | leifmadsen | s/that work/that way/ |
| 13:54.54 | Sidrov | i know |
| 13:56.00 | leifmadsen | if you knew then I'm not sure why you asked the question |
| 13:56.49 | Sidrov | i know u have typed with / / |
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| 13:56.56 | Sidrov | but, your info is helpful |
| 13:56.57 | Sidrov | thanks |
| 13:57.29 | leifmadsen | np |
| 13:57.33 | leifmadsen | Basically: |
| 13:57.35 | leifmadsen | Dial(...) |
| 13:57.48 | leifmadsen | ExecIf($[${DIALSTATUS} = CONGESTION]?Dial(....)) |
| 13:57.55 | leifmadsen | or GotoIf(), or whatever other logic you want |
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| 13:58.47 | Sidrov | better ExecIf($[${DIALSTATUS} != ANSWER]?Dial(....)) |
| 13:58.54 | Sidrov | to avoid other reasons |
| 13:59.11 | leifmadsen | well, != ANSWER & != BUSY |
| 13:59.23 | leifmadsen | busy is a valid reason to not redial too |
| 13:59.35 | Sidrov | yep, u re right |
| 13:59.46 | WIMPy | ITheer are other occasions wehere a retry doesn't make sense. |
| 14:00.04 | leifmadsen | right, but that is left as an exercise to the implementor :) |
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| 14:00.17 | Sidrov | :) |
| 14:00.32 | Sidrov | we'll catch exceptions on the way |
| 14:01.29 | WIMPy | You can't get it completey right with Asterisk anyway. |
| 14:02.20 | Sidrov | then ? WIMPy have any suggestion ? |
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| 14:02.58 | WIMPy | For what? |
| 14:06.36 | Tom_123 | when you place a call, and you hear ringing in your own headset, what is that ringing called |
| 14:06.38 | Tom_123 | ring-back? |
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| 14:29.37 | thejpster | OK, fixed it. The default start_threshold was set to period_size, which is PERIOD_FRAMES*4 = 320 frames. I wound this up to 1280 frames and snd_pcm_state stops saying the card is in XRUN all the time and the audio comes out clear. |
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| 16:42.21 | zenmaster | Hi, how can I check the status of my carrier in asterisk? |
| 16:42.59 | Qwell | make a call? |
| 16:43.08 | zenmaster | Just see if it is connected or not. |
| 16:43.16 | zenmaster | I can't seem to remember. |
| 16:43.19 | Qwell | You can set qualify=yes in your peer, but that won't tell you a whole lot. |
| 16:43.27 | Qwell | Just whether they're responding to certain type of traffic. |
| 16:43.54 | zenmaster | Yeah, I don't think that is what I'm looking for. |
| 16:44.01 | zenmaster | I am refering to the proper command in the cli. |
| 16:44.04 | zenmaster | :) |
| 16:44.12 | leifmadsen | define "proper" |
| 16:44.27 | zenmaster | Found what I was looking for. Thanks any ways. :) |
| 16:44.52 | zenmaster | "sip show peers" |
| 16:45.02 | leifmadsen | that doesn't really tell you anything other than you configured a sip peer |
| 16:45.05 | Qwell | That doesn't really show a.. |
| 16:45.06 | Qwell | that |
| 16:45.15 | zenmaster | leifmadsen: That is all i was looking for. |
| 16:45.26 | leifmadsen | that's not what you asked though |
| 16:45.31 | Qwell | not even close, heh |
| 16:46.40 | cmendes0101 | lol |
| 16:47.06 | zenmaster | Sure it was to me. |
| 16:47.24 | zenmaster | :D |
| 16:49.28 | zenmaster | leifmadsen: Nice to meet the man behind the book. How is life? |
| 16:49.55 | navaismo | all is related with semantic |
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| 16:51.46 | chris_n | any opinions on line cards from here: http://www.nicherons.com/ |
| 16:52.03 | leifmadsen | zenmaster: not bad, just working away on chef stuff for work |
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| 16:53.48 | zenmaster | So I had a question for you. If you go to setup a call center. Do you build something completely from scratch, or do you use something like Vici,Aastracr,GoAutodial,Elastix,Newfies etc...? |
| 16:54.00 | Qwell | zenmaster: That depends. Do you actually mean a call center? |
| 16:54.27 | Qwell | Or perhaps you mean a steamship? |
| 16:54.29 | zenmaster | A facility that has numerous agents, who are recieving inbound say for technical support. |
| 16:54.56 | zenmaster | Or what about for telemarketing, say a "Press 1". |
| 16:55.04 | zenmaster | What do the asterisk gods do. |
| 16:55.20 | Qwell | We write our own configs. |
| 16:55.29 | zenmaster | Ok, so you do it from scratch. |
| 16:55.49 | Qwell | That isn't for everyone though, certainly. |
| 16:56.20 | zenmaster | Right. |
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| 16:57.25 | leifmadsen | ya I've never used a package system (like freepbx, vicidial etc) to build anything |
| 16:57.44 | leifmadsen | if you don't know how to build it from scratch, my opinion is you shouldn't be building call centres |
| 16:58.11 | zenmaster | Right. |
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| 17:32.14 | navaismo | see this video http://vimeo.com/47682405 |
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| 21:05.34 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.8.0 (2012/09/13), 1.8.16.0 (2012/09/13), 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 |
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| 22:40.46 | nickgaw | Hi, anyone here know of a debian apt repository that contains freepbx so I can install it on my debian system? |
| 22:46.11 | navaismo | Hi nickgaw here is a link with some instructions in taht link the author say: "Using Debian packages to install Asterisk and FreePBX would not be the most optimised way but it does the job when you want a quick installation. FreePBX is not yet in the Debian repository but it is available through xorcom.com." |
| 22:46.22 | navaismo | the link is http://www.thewireframecommunity.com/node/20 |
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