| 00:00.31 | weinerk | that is kinda what I concocted - instead of dialing an external number I dial another extention / agi - that orginates a separate call and only if succesfull - all the legs are connected. |
| 00:00.48 | weinerk | but it seems a bit ugly and not sure what else could be wrong with that. |
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| 00:30.04 | Freeaqingme | weinerk, once you have figured it out, I'd be interested in your solution ;) |
| 00:30.19 | weinerk | nny: thanks. i am having a hard time to find samples with bridge to make it work. |
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| 00:32.04 | Freeaqingme | weinerk, the bridging is the easy part, you'll have to set up the other channel first |
| 00:32.19 | weinerk | exactly |
| 00:33.09 | Freeaqingme | and as said, I think you should use ami originate for that |
| 00:33.21 | Freeaqingme | (I have an ami script for you if interested) |
| 00:33.39 | weinerk | Freeaqingme: this is the thing - I kinda got it working - as discribed above - but I suspect this is suboptimal... It dials each destination for x seconds until someone answers and accepts call and then connects... But ... |
| 00:33.52 | Freeaqingme | but? ;) |
| 00:34.10 | weinerk | Freeaqingme: thanks - I would love to look at the script |
| 00:34.41 | WiretapWork_ | weinerk, the way commercial click-to-dial stuff sems to work |
| 00:34.44 | WiretapWork_ | seems to work* |
| 00:34.46 | weinerk | ... but - it is nasty to have to dial another extention and to pass all context and I am not sure what else can break... |
| 00:35.08 | WiretapWork_ | is that it originates to the hardphone first, then once that is answered originates to the destination and bridges the channels immediately |
| 00:36.03 | Freeaqingme | weinerk, https://github.com/Freeaqingme/zf2/blob/asterisk/library/Zend/Telephony/Asterisk/Ami/Adapter/Tcp.php |
| 00:36.19 | Freeaqingme | not all commands have been implemented (this should be fairly easy), but I think the socket handling etc is done |
| 00:36.26 | weinerk | thanks, will check it now |
| 00:36.59 | Freeaqingme | and apparently my tab settings suck, sorry for that |
| 00:39.02 | weinerk | for perl there are actually a couple modules already |
| 00:39.03 | weinerk | http://cpan.uwinnipeg.ca/htdocs/Asterisk-AMI/Asterisk/AMI.html |
| 00:39.31 | Freeaqingme | there is. Thought you were using php |
| 00:39.36 | Freeaqingme | (May have made that up myself) |
| 00:40.31 | weinerk | I am :-) |
| 00:40.57 | Freeaqingme | then the perl modules arent really useful for you, are they? :P |
| 00:41.00 | *** join/#asterisk phoenixsampras (~xman@static-190-181-38-119.acelerate.net) |
| 00:41.05 | phoenixsampras | dahdi show status |
| 00:41.05 | phoenixsampras | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 00:41.05 | phoenixsampras | Wildcard TDM410P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 00:41.05 | weinerk | but my first love :-) |
| 00:41.15 | phoenixsampras | i cant receive calls, why? |
| 00:41.31 | phoenixsampras | when i look for channels, there is noone |
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| 00:45.50 | weinerk | also I found this: http://www.freepbx.org/trac/browser/freepbx/branches/2.9/amp_conf/htdocs/admin/libraries/php-asmanager.php |
| 00:47.04 | WiretapWork_ | phoenixsampras, your coding is unknown... and I highly doubt you're seeing 0dB |
| 00:47.23 | WiretapWork_ | sounds like either a config error at your end, or nobody provisioned your line |
| 00:48.35 | Freeaqingme | weinerk, yes, but the code is <php5 and truly ugly |
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| 00:51.41 | WIMPy | phoenixsampras: What do you mean by there is no channel? |
| 00:58.45 | phoenixsampras | WIMPy: when i do , dahdi show channels, nothign is shown |
| 00:59.17 | WIMPy | Then you haven't configured any. |
| 01:01.32 | phoenixsampras | WIMPy: how to do that please, its s simple TDM410P card? |
| 01:02.31 | WIMPy | dahdi/system.conf and asterisk/chan_dahdi.conf |
| 01:03.00 | phoenixsampras | ohh |
| 01:03.37 | phoenixsampras | where are those files located please |
| 01:03.55 | WiretapWork_ | in /etc/ |
| 01:04.02 | WIMPy | Probably /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf |
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| 01:06.39 | phoenixsampras | WIMPy: ok now should i restart asterisk? /etc/init.d/asterisk restart |
| 01:06.40 | phoenixsampras | ? |
| 01:08.20 | phoenixsampras | WIMPy: restart. no channels yet :( |
| 01:11.14 | phoenixsampras | oh nice, but now i just have 2 channels insted of 4 |
| 01:14.29 | phoenixsampras | still the inbound calls does in |
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| 01:50.49 | dr00d | hi all - just wondering if anyone can tell me how to run asterisk dial plan code from a shell script ? |
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| 02:39.07 | Ghidorah_ | This is probably a commonly asked question but does anyone know if it is possible to have redundant asterisk servers with analog phones? |
| 02:41.04 | Freeaqingme | Ghidorah_, you can cluster asterisk servers, you can connect analog phones to asterisk (through an ATA). I'm not sure that answers your question though |
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| 02:42.17 | Ghidorah_ | Freeaqingme: Thanks for your response. From my understanding I can have a FXO/FXS card in the asterisk server. But if they're clustered how would i connect both to the same analog block? |
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| 02:42.45 | Ghidorah_ | Assuming there are two servers in the cluster. |
| 02:43.05 | Wiretap_Work | Ghidorah_, use an ATA instead? |
| 02:44.20 | Ghidorah_ | Wiretap_Work and Freeaqingme thank yoU! |
| 02:44.22 | Ghidorah_ | you* |
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| 02:52.08 | Wiretap_Work | WIMPy, my research shows me that directed pickup actually _does_ need some changes made in asterisk |
| 02:56.22 | dr00d | im guessing that no answer to my question means its not possible ... |
| 02:57.13 | Wiretap_Work | missed your question dr00d |
| 02:58.02 | dr00d | ok - all i want to do is call an extension, and hang up - but from a shell script as i need to do this from vmnotify in voicemail.conf |
| 02:59.58 | Wiretap_Work | dr00d, use AMI and a php CLI script |
| 03:03.17 | dr00d | ook thanks - are you able to elaborate on this at all ? i can google ami tho |
| 03:03.41 | Wiretap_Work | AMI = asterisk manager interface |
| 03:05.24 | dr00d | ok ill take a look thanks |
| 03:07.09 | Wiretap_Work | dr00d, you want to originate |
| 03:07.47 | dr00d | ok - it looks too hard :( to use ami - what i want to do is very simple there must be an easier way |
| 03:09.05 | dr00d | ive found an ami originate example but im not sure how to implement this tho |
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| 03:10.06 | russellb | dr00d: you can drop a file into /var/spool/asterisk/outgoing |
| 03:10.14 | dr00d | oh |
| 03:10.24 | russellb | it's like AMI originate, but you just create a file and move it into that directory |
| 03:10.36 | russellb | there is a recipe that talks abotu it in the asterisk cookbook on http://ofps.oreilly.com |
| 03:11.05 | russellb | you can also do ... # asterisk -rx "channel originate Local/123@foo application NoOp" |
| 03:11.13 | russellb | there are a few different originate recipes, check them out |
| 03:11.21 | russellb | goes to bed |
| 03:11.28 | dr00d | ok thanks ill go take a look |
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| 03:57.13 | dr00d | is anyone able to help me get this going ? -> http://mikecathey.com/code/vmnotify/vmnotify.txt |
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| 05:19.39 | g00gle | hello room |
| 05:19.45 | g00gle | need help on jabber |
| 05:22.26 | dr00d | hi g00gle - i cant help but hi anyway :) |
| 05:22.46 | g00gle | hi good to get a response.. i m new to IRC |
| 05:23.04 | dr00d | im sure there will be someone in here that can help ya |
| 05:25.03 | g00gle | waiting for someone to help me |
| 05:25.23 | g00gle | i want to know how to receive text messages from buddy to my context |
| 05:26.06 | g00gle | is it true that only two of us talking in this room out of 212 people? |
| 05:26.35 | ChannelZ | Nope. At least 2 of us aren't actually people. |
| 05:27.12 | g00gle | hi ChannelZ |
| 05:27.39 | ChannelZ | ahoy |
| 05:28.02 | g00gle | would you be able to answer my question? |
| 05:29.14 | ChannelZ | not directly. But where do you intend for these texts to actually go? |
| 05:30.11 | g00gle | i have to gtalk.conf and jabber.conf accounts go to google-in context |
| 05:30.32 | g00gle | there I have s,1,Answer() and few NoOp statements |
| 05:30.44 | g00gle | i do have JABBER_RECEIVE statement . |
| 05:31.03 | l1nuxman | my voicemails voicemail length is wrong... |
| 05:31.28 | g00gle | but if i IM the asterisk buddy from my personal IM ... in debug mode i can see the xml in CLI |
| 05:32.55 | g00gle | but i don't see the dialplan in context is executed |
| 05:33.10 | ChannelZ | it probably only works during a call |
| 05:34.03 | g00gle | but whats the purpose of JABBER_RECEIVE? is it not intended to exchange text messages to client |
| 05:34.10 | g00gle | JabberSend as well |
| 05:36.26 | ChannelZ | Aren't those dialplan apps? |
| 05:37.05 | g00gle | yes i am using them in dialplan |
| 05:37.10 | g00gle | inside google-in context |
| 05:37.58 | ChannelZ | But it sounds like you're just sending a random IM.. you're not actually IN the dialplan yet. |
| 05:39.13 | g00gle | yes i am sending random IM .. say HELO |
| 05:39.17 | ChannelZ | I'm not sure to be honest but I doubt it's setup that receiving a text triggers an extension in the dialplan |
| 05:39.38 | g00gle | but from the dialplan i can send message to my buddy |
| 05:39.44 | g00gle | using JabberSend |
| 05:39.44 | ChannelZ | If however you have an extension that you dial, in that exten you could use JabberSend and Receive to do things |
| 05:39.58 | g00gle | i see |
| 05:40.27 | g00gle | but if I send IM it shows as xml which means its received by asterisk |
| 05:40.56 | ChannelZ | Sure. Asterisk can't prevent a remote client from sending it things. |
| 05:42.32 | g00gle | hmm so send IM will not trigger any dialplan execution? |
| 05:43.04 | ChannelZ | not from what I can tell just browsing the docs. I haven't played with the gtalk stuff in some time |
| 05:44.13 | g00gle | i see |
| 05:44.42 | ChannelZ | Just throwing a dart here, but maybe it generates an AMI event that you could trap and then have it enter the dialplan.. to what end I dunno but that's your problem :) |
| 05:47.51 | g00gle | ok i will dig into it |
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| 06:38.53 | schmidts | good morning |
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| 06:42.46 | jacc0 | hi all |
| 06:43.21 | jacc0 | I was doing an interoperability with a siemens HG1500 |
| 06:43.44 | jacc0 | I think it is due to asterisk not being rfc complient |
| 06:44.00 | jacc0 | will file a bug report today |
| 06:48.13 | jacc0 | it has to do with an invite with session desription that supports both SRTP(SAVP) and RTP(AVP); when asterisk is not configured for SRTP it will not start an RTP session. |
| 06:48.54 | schmidts | jacc0 can you tell me the issue number? |
| 06:49.06 | jacc0 | I will tell you later on |
| 06:49.11 | jacc0 | <PROTECTED> |
| 06:49.19 | jacc0 | I still need to file the bug |
| 06:49.24 | schmidts | ok ;) |
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| 07:05.31 | Wiretap | hey guys, how would I use a literal 'x' in my dialplan (I'm trying to implement x-cisco-serviceuri-blfpickup-) |
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| 07:09.02 | ChannelZ | don't use a pattern |
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| 07:17.27 | Wiretap | ChannelZ, so this would work instead? "x-cisco-serviceuri-blfpickup-_." |
| 07:18.01 | ChannelZ | No |
| 07:18.29 | Wiretap | I need to match the extension after that string, that's my dilemma |
| 07:18.48 | Wiretap | its "x-cisco-serviceuri-blfpickup-<EXTEN>" |
| 07:19.16 | Wiretap | (and no, you can't change the string in the phone) |
| 07:19.57 | ChannelZ | use a . instead of the x |
| 07:20.37 | Wiretap | I was under the impression you couldn't match past a '.' |
| 07:21.07 | ChannelZ | actually I wonder if you can put characters in a [] class |
| 07:21.47 | Wiretap | not and have it work :P |
| 07:22.18 | jacc0 | just use s,1,gotoIF($["${EXTEN:29}"="x-cisco-serviceuri-blfpickup-"]?ciscoserviceuri) |
| 07:22.36 | ChannelZ | it worked here |
| 07:22.57 | ChannelZ | _[x]barf,1,NoOp(YAY) |
| 07:23.09 | ChannelZ | matches xbarf but not barf or tbarf or anything else |
| 07:23.37 | jacc0 | https://issues.asterisk.org/view.php?id=19281 |
| 07:24.24 | ChannelZ | how is that related..? |
| 07:25.34 | Wiretap | its my fail lol XD |
| 07:27.11 | ChannelZ | eh? |
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| 07:27.25 | Wiretap | got it |
| 07:27.26 | *** join/#asterisk Maxus2 (~Maxus2@59.191.225.49) |
| 07:27.36 | jacc0 | it is not related |
| 07:28.08 | jacc0 | it's an answer to the question of schmitdts: jacc0 can you tell me the issue number? |
| 07:28.37 | Maxus2 | Hi People, how do i create an outbound sip registration with realtime? |
| 07:28.44 | ChannelZ | non-sequitor |
| 07:28.56 | jacc0 | good question Maxus2 |
| 07:29.28 | ChannelZ | doesn't it still pay attention to some stuff in sip.conf? |
| 07:29.33 | jacc0 | I've been doing that yesterday and added a line in sip.conf |
| 07:30.02 | Wiretap | exten => _[x]-cisco-serviceuri-blfpickup-.,1,Pickup(${EXTEN:29}&${EXTEN:29}@ext-local&${EXTEN:29}@from-internal&${EXTEN:29}@from-internal-xfer&${EXTEN:29}@from-did-direct&LC-${EXTEN:29}@from-internal&LC-${EXTEN:29}@from-internal-xfer&FMPR-${EXTEN:29}@from-internal&FMPR-${EXTEN:29}@from-internal-xfer&FMPR-${EXTEN:29}@from-did-direct) |
| 07:30.03 | Wiretap | exten => _[x]-cisco-serviceuri-blfpickup-.,n,Hangup |
| 07:30.10 | jacc0 | ~pb |
| 07:30.10 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 07:30.11 | Wiretap | yes, its freepbx, but it works now :P |
| 07:30.13 | jacc0 | :P |
| 07:30.19 | Wiretap | jacc0, two lines isn't flooding |
| 07:31.27 | jacc0 | okay, so making one line line that wraps to miltipe lines is not flooding, I'll keep that in mind |
| 07:31.30 | jacc0 | :p |
| 07:31.40 | Wiretap | jacc0, sorry, does it? big screen |
| 07:32.05 | jacc0 | hehehe, np |
| 07:32.28 | schmidts | jacc0 sorry i was away could you send me the issue number again |
| 07:32.38 | schmidts | sorry found it ;) |
| 07:32.48 | jacc0 | <PROTECTED> |
| 07:32.49 | ChannelZ | Use The Scrollbar, Luke |
| 07:32.57 | jacc0 | hehehe |
| 07:33.33 | schmidts | ChannelZ the scrollbar i used |
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| 07:34.35 | schmidts | jacc0 could you append also a debug output, cause only out of the sip messages its hard to say why this doesnt work ;) |
| 07:35.15 | jacc0 | I'm work ing on that; will update the issue later this day |
| 07:35.22 | schmidts | ok ;) |
| 07:35.37 | jacc0 | chan_sip.c: Can't provide secure audio requested in SDP offer |
| 07:35.43 | jacc0 | that was what is in cli |
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| 07:37.09 | jacc0 | just edit chan_sip.c and search for " Can't provide secure audio requested in SDP offer" and it's very clear why it fails(imho) |
| 07:38.27 | jacc0 | it just stops when it finds SAVP (even if there is also a AVP) |
| 07:39.30 | jacc0 | I'm going to make a quickfix for this specific sutuation: replacing SAVP with AVP in the invit string |
| 07:39.39 | jacc0 | *invite |
| 07:40.01 | jacc0 | quick and very dirty |
| 07:40.04 | jacc0 | :p |
| 07:40.09 | jacc0 | just like me |
| 07:40.57 | schmidts | jacc0 i am not sure about this cause in your invite there is only an AVP and this should match if (!strcmp(protocol, "SAVP")) |
| 07:41.16 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 07:41.40 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59.dsl.dyn.forthnet.gr) |
| 07:41.49 | schmidts | s/should/shouldnt/ |
| 07:42.27 | jacc0 | sorry, maybe I included the wrong pcap |
| 07:42.58 | schmidts | ah ok ;) |
| 07:45.00 | jacc0 | no, it's the right one |
| 07:45.26 | schmidts | hmmm strange |
| 07:45.32 | jacc0 | look at the message body of 23 4.670000 |
| 07:45.59 | jacc0 | 0000 52 54 50 2f 41 56 50 RTP/AVP |
| 07:46.02 | schmidts | ok i got it its down below |
| 07:46.04 | Maxus2 | <jacc0> sorry missed what you said irc crashed, have you go any info on the registing sip devices in realtime? |
| 07:46.13 | schmidts | you have first avp and then SAVP |
| 07:46.27 | jacc0 | yes |
| 07:46.44 | jacc0 | and when astrisk finds the SAVP it just stops |
| 07:46.58 | jacc0 | but there is als a AVP |
| 07:47.16 | jacc0 | and asterisk should be able to do that |
| 07:47.21 | Maxus2 | sorry, registering outbound |
| 07:47.38 | jacc0 | @Maxus2: just add a line to sip.conf |
| 07:48.05 | jacc0 | @Maxus2: I don't know how to do it in the database and doubt it is even posible |
| 07:48.15 | Maxus2 | cant be done in the DB? |
| 07:48.45 | Maxus2 | oh okay |
| 07:49.58 | schmidts | jacc0 the question is if asterisk should do a fallback to nonsecure AVP if SAVP fails but i think this would be ok, cause the client offers both |
| 07:50.23 | jacc0 | true |
| 07:50.56 | jacc0 | I don't know if there is a RFC about this |
| 07:51.28 | jacc0 | can anyone point me to it? I'll will read it and add the relevant part to the bugreport |
| 07:52.49 | jacc0 | if the RFC tells not to fall back I'll report it to Siemens |
| 07:53.40 | jacc0 | as more and more PBXes support SRTP this will get a bigger issue |
| 07:53.54 | schmidts | i will take a look at the rfc later, do you know which are the right params for diff to create a patch? i only use svn diff normaly :D |
| 07:54.42 | sezuan | good topic. Can anyone recommend a srtp capable client for linux? |
| 07:58.04 | schmidts | jacc0 i have uploaded a patch to your issue, please try it |
| 07:58.35 | schmidts | moment this is wrong |
| 07:59.23 | schmidts | now it should do whats expacted ;) |
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| 08:00.36 | jacc0 | :) |
| 08:00.44 | jacc0 | did I say I love you already? |
| 08:00.49 | jacc0 | I love you |
| 08:00.54 | schmidts | not today but thanks ;) |
| 08:01.00 | schmidts | first try it and then say this |
| 08:01.06 | jacc0 | will do |
| 08:01.13 | jacc0 | I'll get to it right away |
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| 08:10.45 | morbidwar | hello, i have a question. I have a analog gsm gateway (channel bank) that send signals in dtmf. on an analog phone shows the CID but the asterisk did not recive it. could somebody help me configure the asterisk in order to see the CID? |
| 08:11.05 | schmidts | jacc0 btw http://www.ietf.org/rfc/rfc4566.txt part 5.14: "A session description may contain a number of media descriptions" so its ok to use both in one invite but it didnt find if its ok to fallback if the client offers this |
| 08:13.47 | jacc0 | I guess it hsould match supported media descriptors |
| 08:13.53 | jacc0 | *should |
| 08:15.00 | schmidts | i like this * (asterisk) should :D |
| 08:15.19 | jacc0 | :) |
| 08:17.21 | jacc0 | okay, the siemens guy just opend up some ports so I can register to his PBX again |
| 08:18.03 | jacc0 | I was @ location yesterday but it's a 2,5 hour drive |
| 08:19.22 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 08:19.24 | kaushal | Hi |
| 08:19.38 | kaushal | is there a step by step guide to install asterisk SCF ? |
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| 08:22.34 | zkn | Hi, can anyone remind me if and if yes then in what chapter it was covered in the Definitive Guide how to pepare Linux filesystem properly for Asterisk? I remember reading about it but I cant seem to find it in The Book anymore.. |
| 08:23.01 | zkn | maybe I read this somewhere else... |
| 08:25.27 | zkn | ok, found it now :) |
| 08:25.37 | Wiretap | kaushal, did jacc0 deliver your message? |
| 08:25.48 | zkn | it is in Distribution Installation, obviously.. |
| 08:26.58 | jacc0 | @wiretap: what message? |
| 08:27.08 | Wiretap | sangoma |
| 08:27.59 | jacc0 | there is an update available |
| 08:28.11 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 08:28.41 | jacc0 | @kaushal: I've send you a PM |
| 08:29.37 | jacc0 | so chan_woomera is no longer supported; you should install SMGv3.2 and register it to asterisk as a normal sip client |
| 08:30.03 | jacc0 | didn't test it jet |
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| 08:38.57 | schmidts | jacc0 i asked olle about the rtp description cause i didnt know this very well and he said this invite is not an alternative its the offer for two simultaneous streams |
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| 08:44.40 | schmidts | jacc0 this patch could work for this case but we found another problem. the second media description with SAVP didnt offer a port: m=audio 0 RTP/SAVP 8 0 18 4 98 99 |
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| 08:48.05 | schmidts | jacc0 i think there is a deeper problem with this and it depends on the hipath, cause they want two simultan streams of the same type audio but asterisk can only handle multiple streams if they are different like audio + video and not audio+audio |
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| 08:54.20 | jacc0 | Okay, thnx , will report this to siemens |
| 08:56.05 | schmidts | jacc0 we have searched in the rfcs and this type of offer is valid but the missing port could not work. |
| 08:56.33 | schmidts | but there is a depper problem with asterisks media architecture not honoring multiple streams of the same type |
| 09:02.34 | jacc0 | okay, but why should it? |
| 09:02.55 | jacc0 | it's just a normal audio call |
| 09:03.05 | jacc0 | from an analog phone |
| 09:03.15 | jacc0 | no special features |
| 09:03.42 | schmidts | jacc0 what do you mean? |
| 09:03.51 | jacc0 | why would a call fro |
| 09:03.54 | jacc0 | sorry |
| 09:04.10 | jacc0 | why would the siemens want 2 audio streams to a singe analog phone? |
| 09:04.29 | jacc0 | I see no reason for that |
| 09:04.51 | schmidts | jacc0 i dont know ;) maybe thats their kind of fallback way if a client doesnt support SRTP |
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| 09:14.24 | *** join/#asterisk DoberMann (~james@2a01:e35:8b44:84c0::2) |
| 09:14.34 | DoberMann | hi |
| 09:14.56 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 09:15.40 | DoberMann | i'm stuck configuring cisco 7937g phone using chan_skinny : i can see devices, call from/to them, but i have no sound despite the fact that codec seems to be negotiated properly |
| 09:16.12 | DoberMann | core show shannels sho me " Blocking in: ast_waitfor_nandfds |
| 09:16.20 | DoberMann | if this is an arror |
| 09:16.23 | DoberMann | *errot |
| 09:16.26 | DoberMann | *error |
| 09:17.59 | DoberMann | asterisk 1.6.2.5-0ubuntu1.3 / ubuntu 10.04.2 if that matters |
| 09:19.02 | Wiretap | DoberMann, what you want is a log with debug on |
| 09:19.57 | DoberMann | i have some, but i can't find what's wrong |
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| 09:21.01 | Wiretap | fraid I don't know how SCCP works very well |
| 09:21.03 | DoberMann | do you know if 7937g devices are working with chan_skinny or do you recommand switching to chan_sccp ? |
| 09:27.16 | Wiretap | DoberMann, chan_sccp-b is much more advanced, but precludes you using ast-1.8 |
| 09:27.28 | Wiretap | which you will want if you want to use any of the features of the SIP models |
| 09:30.51 | DoberMann | yes, but upgrading asterisk is not an option for now, and my needs are basic (only call from/to, no need to transfert , ...) and theorically filled by chan_skinny |
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| 11:14.49 | *** join/#asterisk MikeH (~mikeh@109.224.134.207) |
| 11:15.46 | MikeH | Hi, I'm having troubles with callerid. I want asterisk to pass callerid from an analogue interface (digium card) to th eventual sip destination (via IVR) |
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| 11:35.01 | kaldemar | MikeH: what kind of analog interface? FXO or FXS? |
| 11:35.46 | MikeH | Fox |
| 11:35.50 | MikeH | Fxo |
| 11:36.16 | kaldemar | do you have usecallerid=yes and callerid=asreceived configured for the channel? |
| 11:37.12 | MikeH | The first yes, second no |
| 11:40.10 | MikeH | Just added the second under both chan-dahdi and dahdi-channels for the appropriate channel, with no luck |
| 11:41.00 | MikeH | kaldemar: Is there something to do with the IVR that could be causing it? The callerid on the interface is working ad I can see the number in the console |
| 11:41.09 | MikeH | As |
| 11:41.39 | kaldemar | so you get the caller id in your asterisk from the analog line? |
| 11:42.23 | MikeH | Yes |
| 11:42.33 | kaldemar | then something in your dialplan may be stripping it. what exactly does your IVR do? |
| 11:42.43 | MikeH | Just not on the destination sip phone. I simply get 'unknown' |
| 11:44.45 | kaldemar | if you enable sip debug for the peer, do you see the caller id? |
| 11:47.21 | MikeH | I'm not sure how to do that. |
| 11:47.46 | MikeH | I just changed inbound route to and extension instead of IVR, and the issue remains |
| 11:47.54 | BarthezZ | Hmm, I'm having a little challenge... For handling an incoming call from Dahdi it sometimes needs to be ringing locally connected phones (SIP/16X) but also some mobile phones (DAHDI/g1/_X.)... Now the dahdi channel needs to have a non-default CID, but the internal phones need to get the external CID from the callee, so Dial(SIP/161&DAHDI/g1/06X) doesn't work |
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| 11:48.19 | kaldemar | MikeH: are you using freepbx? |
| 11:48.25 | MikeH | Yes |
| 11:49.24 | kaldemar | MikeH: you better ask in #freepbx then, because it configures asterisk. |
| 11:50.01 | MikeH | I have done that also, however I cannot see anything within freepbx to configure this, so assumed it was an asterisk issue. |
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| 11:56.02 | BarthezZ | hmm, can i do a simultanious Dial both with different variables? |
| 11:56.20 | MikeH | Oh, that's weird. |
| 11:56.52 | MikeH | kaldemar: Callerid works if I call in to the gsm interface that does not have an inbound route configured |
| 11:57.07 | kaldemar | MikeH: freepbx makes asterisk configurations, and uses an AGI script to dial. the people at freepbx know its internals better. |
| 11:58.00 | kaldemar | MikeH: i can promise you that asterisk will successfully handle caller id in a case like yours if configured to do so. |
| 11:58.41 | kaldemar | MikeH: other than that, all i can say is that something in the dialplan prevents the caller id from being shown, at that is a freepbx issue. |
| 11:59.07 | MikeH | Ok thanks |
| 12:00.02 | morbidwar | Hi all, i have a problem with CID, if i connect an analog phone to the analog gsm gw it's showing cid but the asterisk didn't recive, i modified the chan-dahdi.conf with usecallerid=yes and callerid=asreceived, however i understant that the gsm gateway is sending DTMF signals, so do i need to modify the dahdi_channels.conf from signalling=fxk_ks to signalling=dtmf ? |
| 12:08.29 | kaldemar | morbidwar: no, leave the signalling option as is. there are separate parameters for cidsignalling and cidstart. |
| 12:09.53 | morbidwar | kaldemar: could you give me a hint? |
| 12:11.06 | leifmadsen | I think he just did |
| 12:11.33 | kaldemar | morbidwar: i already gave you the actual parameters. |
| 12:11.41 | morbidwar | :) |
| 12:11.44 | morbidwar | thanks |
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| 12:29.43 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 12:29.48 | Dovid | anyone from Italy here ? |
| 12:30.27 | stope | if I call SLATRUNK(line1) and its already in use, then ${SLATRUNK_STATUS} returns 'failed' but if I have SLATRUNK(line2) right after and its not in use, when complete it also returns 'failed' |
| 12:30.46 | stope | should the 2nd SLATRUNK return 'success'? |
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| 12:31.32 | *** join/#asterisk smooth_penguin (~smooth_p@59.162.86.164) |
| 12:31.38 | smooth_penguin | hey, need some help :\ |
| 12:31.48 | smooth_penguin | "PRI Error on span 0: We think we're the CPE, but they think they're the CPE too" |
| 12:31.54 | smooth_penguin | any idea whats happening here |
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| 12:34.24 | fish-bulb | smooth_penguin: do you have a loopback in or connected to another device that is set for CPE signalling? |
| 12:35.17 | fish-bulb | that just means that your side is set to pri_cpe, and so is the other side. One should be acting as the network. If it is Asterisk that should be, then it needs to be set to pri_net |
| 12:39.48 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:40.15 | smooth_penguin | fish-bulb: this was working for a long time |
| 12:40.21 | smooth_penguin | its suddenly acting like this |
| 12:40.35 | fish-bulb | smooth_penguin: what is the line connected to? |
| 12:40.46 | smooth_penguin | its an E1 pri line |
| 12:41.01 | smooth_penguin | with 64 channels |
| 12:41.13 | fish-bulb | the telco may have put the line into loopback mode for some reason |
| 12:41.17 | fish-bulb | have you spoken with them yet? |
| 12:42.00 | smooth_penguin | fish-bulb: getting in touch |
| 12:42.14 | smooth_penguin | thanks for helping :) |
| 12:42.43 | fish-bulb | if it's been working and now is doing that then something must have changed, and if it wasn't the configuration or the physical connection on your side then it is more than likely some goofiness on the telco's part |
| 12:43.02 | fish-bulb | smooth_penguin: sure thing, let me know what they say |
| 12:45.04 | smooth_penguin | yep, well the pri cards show red |
| 12:45.11 | smooth_penguin | but asterisk show them green |
| 12:45.15 | smooth_penguin | or OK |
| 12:45.47 | smooth_penguin | http://pastebin.com/tQ19qZvU |
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| 13:06.48 | smooth_penguin | whats odd is incomming calls via the pri are working |
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| 13:14.20 | WIMPy | smooth_penguin: You got a loopback on your line? |
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| 13:20.45 | smooth_penguin | WIMPy: no, but the provider probably did - it seems to be fixed now, waiting for a reply from them |
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| 13:22.36 | Jasnejac | Hi all. I have a problem with callerid that has me stumped at the moment. I havbe two asterisk boxes SIP trunked together and also IAX trunked. If I dial from one to the other using IAX the callerid is maintained. If I do using the SIP trunk callerid is lost. Any clues please? |
| 13:23.00 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 13:25.58 | WIMPy | You configured a callerid in the sip peer? |
| 13:27.09 | Jasnejac | no |
| 13:27.57 | Jasnejac | I think I'm probably missing something obvious |
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| 13:38.01 | smooth_penguin | fish-bulb: you around? |
| 13:38.07 | fish-bulb | yep |
| 13:39.53 | *** join/#asterisk jkroon (~jkroon@dsl-241-248-179.telkomadsl.co.za) |
| 13:40.19 | jkroon | hi guys, weird question, is there an easy way to store voicemail in a relational database instead of files/imap? |
| 13:41.18 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 13:41.23 | fish-bulb | jkroon: that is pretty common, check out https://wiki.asterisk.org/wiki/display/AST/Storing+Voicemail+in+PostgreSQL+via+ODBC |
| 13:41.48 | fish-bulb | don't know if I would exactly call it easy, but it's not that tough |
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| 13:42.36 | fish-bulb | you can use ODBC with a lot more databases than just PostgreSQL of course |
| 13:43.03 | jkroon | fish-bulb, i was about to say that it's quite easy to get the config from a DB, but storing the actual voicemail recordings in the DB itself might be a different matter. |
| 13:44.54 | jkroon | after mysql support |
| 13:46.02 | jkroon | looks very promising thanks. |
| 13:46.17 | fish-bulb | np |
| 13:50.40 | jkroon | argh, requires * 1.8 ... ok well, that's it. motivation to upgrade. Just not right now. |
| 13:51.47 | *** join/#asterisk zorp75ck (~zorp75ck@146.186.115.70) |
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| 13:52.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:52.22 | leifmadsen | asterisk 1.8 is the new black |
| 13:52.38 | jkroon | hehe, leifmadsen jip. except for the SIP over TCP issues :p |
| 13:52.38 | fish-bulb | jkroon: the same idea should work for previous versions, as long as it includes res_odbc. The syntax may just be a little different |
| 13:52.42 | leifmadsen | jkroon: that works in 1.6.2 as well |
| 13:53.03 | leifmadsen | jkroon: if you don't have access to ODBC voicemail, you don't have SIP over TCP either |
| 13:53.14 | jkroon | hmm, there goes the upgrade motivation, but it sounds like there has been some Local/ updates as well. |
| 13:53.29 | leifmadsen | shrugs |
| 13:53.34 | jkroon | leifmadsen, no, but i've been fearful of a 1.8 * upgrade because of that and other issues. |
| 13:53.44 | leifmadsen | been working great for me in several deployments |
| 13:53.52 | leifmadsen | guess it depends what you use |
| 13:55.12 | jkroon | jip, i'd like to find some time to test and i've finally managed to get a test environment going for myself, which will help a LOT. |
| 14:00.06 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
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| 14:00.36 | eXcAliBuR | does anyone know of a decent web app for automated calling? |
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| 14:15.08 | jkroon | eXcAliBuR, vicidial is the only one i know of. |
| 14:15.17 | jkroon | i'm not sure i like it though. |
| 14:15.47 | schmidts | war dialing is allways you shouldnt like ;) |
| 14:16.16 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:16.45 | *** join/#asterisk Polysics (~polysics@host153-70-dynamic.41-79-r.retail.telecomitalia.it) |
| 14:16.57 | Polysics | hello |
| 14:17.15 | jkroon | ok, let me rephrase, i hate auto-dialers with a passion, it's good business though and i don't blame voip providers for going after call centers - i do hate those that make it their business to operate call centers. should you not have moral concerns with providing these guys with good solutions then vicidial is probably the best floss solution available to you. |
| 14:17.17 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:17.42 | jkroon | *outbound sales driven call centers. |
| 14:17.57 | jkroon | just realized inbound support call centers are not that bad :p |
| 14:18.09 | Polysics | what is the proper way to bring a third channel in a call, please? |
| 14:18.18 | Polysics | two people are speaking and athird joins |
| 14:18.28 | _Corey_ | people need to know that there ARE auto warranty extensions options dammit |
| 14:18.54 | *** part/#asterisk morbidwar (~ovidiu@81.196.150.82) |
| 14:18.55 | jacc0 | @PLYSICS: MEETME |
| 14:18.59 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 14:19.01 | jacc0 | sorry for the caps |
| 14:19.25 | Aut0ExeC | perhaps i'm a weebit offtopic here but does anyoen know of a public ssh server that I can tunnel into for like 10 mins? |
| 14:21.49 | leifmadsen | jkroon: heh ya I make a business on inbound call centres, and refuse business to those looking to build outbound call centres |
| 14:22.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:23.23 | leifmadsen | Polysics: there is a recipes in the cookbook that lets you use DTMF to play back a prompt -- it could probably be configured to do what you want. However that is more complicated than necessary. Most SIP phones have a conferencing button so that would work fine. The other alternative is to have the two calls on separate lines, and transfer everyone to a conference room. |
| 14:24.11 | Polysics | leifmadsen: how would the transfer happen? problem here is not giving the command, as i use adhearsion to control that |
| 14:24.18 | Polysics | butthe transfer itself |
| 14:24.31 | Polysics | i am on 1.8 |
| 14:25.03 | Polysics | and using a web-based interface |
| 14:25.04 | leifmadsen | just use the AMI Transfer command |
| 14:25.31 | Polysics | will go read up |
| 14:27.14 | *** join/#asterisk magronez (~magrao@unaffiliated/magrao/x-2903) |
| 14:28.05 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:28.05 | *** part/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 14:30.26 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 14:30.27 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:31.36 | *** join/#asterisk rjizzle (~rj@nat/digium/x-lqdfxktjrswximpm) |
| 14:35.39 | chazzam | its an rjizzle |
| 14:35.52 | rjizzle | it is |
| 14:36.25 | Polysics | where can i find docs on that Transfer command, please? |
| 14:36.48 | leifmadsen | manager show commands |
| 14:36.52 | leifmadsen | from the asterisk CLI |
| 14:38.40 | Polysics | wiki is missing quite a few of those :-( |
| 14:39.07 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 14:39.38 | aberrios | anyone had trouble with Asterisk 1.8 skipping over Queue Agents that are on softphones but are free to receive a call? |
| 14:40.41 | *** join/#asterisk Khratos (~Khratos@66.128.60.148) |
| 14:40.52 | Khratos | Good morning |
| 14:41.34 | leifmadsen | Polysics: which wiki? |
| 14:41.44 | Polysics | voip-info |
| 14:43.35 | leifmadsen | Polysics: ya that site is more often than note out of date |
| 14:43.36 | leifmadsen | try this: https://wiki.asterisk.org/wiki/display/AST/AMI+Actions |
| 14:43.52 | leifmadsen | more generally: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Command+Reference |
| 14:44.46 | Khratos | leifmadsen: can anyone post to the official wiki? |
| 14:45.01 | leifmadsen | Khratos: you can at the very least comment |
| 14:45.14 | leifmadsen | I think it is more restricted who can edit the wiki in order to keep a high quality level |
| 14:45.32 | leifmadsen | there are work flows though, so you make changes in your personal work space, then have those changes "merged" into the wiki I believe |
| 14:45.40 | leifmadsen | some things in the wiki are also dynamically generated from the code |
| 14:45.48 | leifmadsen | so that would be where you'd make the changes (like application reference) |
| 14:46.41 | Khratos | I see. And, can contributions be sent to an email? |
| 14:48.32 | leifmadsen | I don't think so |
| 14:48.38 | leifmadsen | you'd be best to do them in your personal work space |
| 14:48.56 | leifmadsen | the wiki is kind of "licensed" like code contributions |
| 14:49.06 | leifmadsen | which I think is the reason things can't just be accepted via email |
| 14:49.25 | leifmadsen | once you had a personal work space setup with changes, you could always email the asterisk-doc mailing list for review |
| 14:49.33 | leifmadsen | malcolmd: am I on the right path here? |
| 14:49.49 | leifmadsen | I think the one thing is we haven't totally established a work flow yet for contributions |
| 14:49.50 | leifmadsen | :) |
| 14:50.58 | Khratos | Understood. I'll work there, then. Things on voip-info are starting to look prehistoric |
| 14:51.05 | leifmadsen | agreed |
| 14:51.23 | leifmadsen | the official wiki is designed to stay more up to date with resources from digium |
| 14:51.43 | leifmadsen | with workflows we can make sure content is relevant and of a high quality |
| 14:51.52 | leifmadsen | I'd be happy to review anything you might wish to submit |
| 14:52.06 | leifmadsen | in some cases simple changes can be acknowledged through the commenting system on each page |
| 14:52.54 | *** join/#asterisk ketema (~ketema@2001:470:5:138:219:e3ff:fe09:5b25) |
| 14:54.08 | *** part/#asterisk ketema (~ketema@2001:470:5:138:219:e3ff:fe09:5b25) |
| 14:54.15 | *** join/#asterisk cusco (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 14:54.18 | cusco | hi folks... |
| 14:54.30 | cusco | We have a "A580 IP |
| 14:54.32 | cusco | oops |
| 14:54.40 | cusco | "Gigaset A580" |
| 14:55.18 | cusco | some siemens wireless phones, each 'central' supports up to 6 registered peers, and 3 simultaneous calls at the same time |
| 14:55.21 | cusco | (so they say) |
| 14:55.21 | jacc0 | bye all!! |
| 14:55.39 | cusco | I have only two phones (260 and 270) |
| 14:55.44 | cusco | they both recieve calls |
| 14:56.06 | cusco | but when dialing ... I'm getting: [May 12 11:12:13] WARNING[9907] chan_sip.c: username mismatch, have <270>, digest has <260> |
| 14:56.20 | cusco | not always.. sometimes it works, but that is happening a lot latelly |
| 14:56.53 | *** join/#asterisk zkn (~zkn@82.131.33.142.cable.starman.ee) |
| 14:59.26 | cusco | http://paste.debian.net/116745/ |
| 14:59.36 | cusco | both peers have the same from address: |
| 15:00.16 | fish-bulb | cusco: do you have "insecure" set to anything in sip.conf for that device? |
| 15:00.41 | cusco | hold... |
| 15:02.06 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:02.39 | cusco | fish-bulb: no |
| 15:02.42 | cusco | its not set |
| 15:02.47 | cusco | should it? |
| 15:03.02 | fish-bulb | cusco: try setting insecure=invite, and then reload sip |
| 15:03.07 | cusco | ok |
| 15:06.01 | fish-bulb | that comes up whenever an invite contains a different username than the authentication digest. It is rejected because it is a security issue in most cases, but since this is a local trusted device then it should be ok. You will run into it pretty much anytime you have multiple devices with different usernames that that are reaching Asterisk through one user |
| 15:08.49 | fish-bulb | if that makes sense... |
| 15:09.52 | *** join/#asterisk byronc (~byron@byron.theclarkfamily.name) |
| 15:14.17 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-hchbhryuyvwslrfj) |
| 15:15.38 | *** join/#asterisk sereal (~jjrh@2001:4830:16ca:1:21a:6bff:fe6a:3cd2) |
| 15:20.32 | eXcAliBuR | I'm trying to get AMI over HTTP working following along in the book, but the webserver isn't responding to my request... did i miss something? |
| 15:21.23 | eXcAliBuR | i enabled it in manager.conf and http.conf |
| 15:21.57 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:22.36 | eXcAliBuR | i restarted my asterisk... should i restart something else? |
| 15:23.54 | serafie | eXcAliBuR: run http show status on the CLI |
| 15:24.13 | *** join/#asterisk Polysics (~Luca@host153-70-dynamic.41-79-r.retail.telecomitalia.it) |
| 15:24.16 | Polysics | hello again |
| 15:24.27 | Polysics | apparently AMI does not have a Transfer command :-) |
| 15:24.31 | Polysics | any pointers? |
| 15:24.53 | sereal | to include a something into a conf file does it need to be done in the [general] section, or can one do it anywhere? |
| 15:27.10 | sereal | no doesn't look like it matters |
| 15:27.13 | WIMPy | Polysics: "redirect" |
| 15:27.22 | eXcAliBuR | serafie: says it's enabled and bound |
| 15:27.31 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 15:28.05 | serafie | bound to what IP and port? |
| 15:28.24 | Polysics | so, to allow a third party to join a conversation, i put him in a MeetMe room, then Redirect both to the same extension? |
| 15:28.29 | eXcAliBuR | 8088 localhost |
| 15:28.40 | Polysics | or is the idea something else? |
| 15:28.44 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 15:28.51 | serafie | well, that doesn't matter as much as that you make sure the bindaddr is the machine you're accessing it *from* and not necessarily not the IP of the machine asterisk is running on. |
| 15:29.01 | WIMPy | Polysics: yes |
| 15:29.10 | eXcAliBuR | hmmm |
| 15:30.19 | serafie | (unless they are one and the same, obviously) |
| 15:31.09 | cusco | fish-bulb: thanks mate |
| 15:33.18 | fish-bulb | cusco: no problemo |
| 15:50.36 | Polysics | WIMPy, that way, though, i need to keep a lot of spare extensions around and keep track of which are in use currently, correct? |
| 15:50.46 | Polysics | not that big of a deal with realtime, i think |
| 15:51.32 | WIMPy | You don't need any extensions, just some meetme rooms. |
| 15:52.50 | *** join/#asterisk sphenxes (~sphenxes@85-127-213-25.dynamic.xdsl-line.inode.at) |
| 15:53.13 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-076.mobile.uci.edu) |
| 15:54.35 | jkroon | on a pri link - does the two systems negotiate whether to use ulaw/alaw or does it need to be configured on both sides? |
| 15:54.56 | WIMPy | Nedds to be configured |
| 15:55.31 | jkroon | what would be the audible effect should they mismatch? |
| 15:56.05 | WIMPy | never tried that |
| 15:56.10 | jkroon | let's say my side is configured as alaw and the remote end is ulaw, how would my side sound different from what it should, and how would the remote end sound different? |
| 15:56.27 | WIMPy | Not sure how much distortion that gies. |
| 15:56.48 | Polysics | there is meetme too |
| 15:56.58 | Polysics | i meant, realtime meetme |
| 15:57.02 | jkroon | has a client that's complaining about voice quality using a Digium B410P card -> g729 (SIP) on the remote end of the peer. |
| 15:57.04 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 15:57.29 | coppice | jkroon: it sounds pretty awful if you decode alaw as ulaw or vice versa |
| 15:58.06 | Polysics | i can just read the meetme table from adhearsion and see empty rooms |
| 15:58.08 | jkroon | coppice, thanks. client just confirmed that the "remote end" is also in alaw, however, i'm still suspect ... |
| 15:58.13 | Polysics | and can also easily add more if needed |
| 15:59.09 | Polysics | WIMPy, doesn't Redirect want an extension though? |
| 16:00.46 | *** join/#asterisk sip123 (~sip123@196.29.186.102) |
| 16:01.02 | fish-bulb | jkroon: there was an old bug that was similar to that. Not sure at all when it was resolved, but it was at some point. you know what version of DAHDI they are using? |
| 16:01.16 | fish-bulb | or I guess they could be using mISDN |
| 16:01.51 | WIMPy | Polysics: Oh, right, yes. But you can use only one extension and a variable, if you want. |
| 16:02.21 | sip123 | test |
| 16:02.44 | Polysics | one extension with a variable saying which conf to move to, i suppose - sounds easy enough |
| 16:03.41 | WIMPy | Polysics: That was what I was thinking about. Bout you can also just put the conf # in to a wildcard extension. Many ways to do it. |
| 16:05.16 | jkroon | fish-bulb, dahdi 2.4.1 |
| 16:05.23 | jkroon | asterisk 1.6.2.17.2 |
| 16:06.05 | jkroon | the only additional info i've got is that apparently the recordings are crystal clear. |
| 16:06.28 | jkroon | libpri 1.4.11.5 |
| 16:06.51 | jkroon | remote end of the link is a quintim gateway |
| 16:07.11 | jkroon | the other PRI link on the quintim gateway goes to a siemens PABX. |
| 16:07.20 | WIMPy | Do you have distortions in both directions? |
| 16:07.50 | jkroon | not sure about the ast side ... it goes out to a SIP provider ... only purpose of * is to record the calls. |
| 16:08.07 | jkroon | if it's a ulaw/alaw mismatch I would expect issues in both directions though. |
| 16:08.20 | WIMPy | indeed |
| 16:08.36 | WIMPy | But they would be different. |
| 16:09.48 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:10.08 | jkroon | perhaps because the client doesn't know it's distorted on the other side too ? |
| 16:12.27 | *** join/#asterisk sip123 (~sip123@196.29.186.102) |
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| 16:16.58 | sip123 | #centos |
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| 16:34.16 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:38.20 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 16:39.29 | nicola_pav | hello. i have asterisk as a guest machine on proxmox |
| 16:39.49 | nicola_pav | while i have iptables and fail2ban on the hosting proxmox machine |
| 16:40.05 | nicola_pav | i still see hits on the VM with wrong password flooding all the full file |
| 16:40.16 | nicola_pav | how can i enable iptables or fail2ban on the vm? |
| 16:40.32 | nicola_pav | any hints please? |
| 16:45.24 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
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| 16:52.12 | draeath | "reload" doesn't seem to be reloading the AMI configuration. Is this normal? |
| 16:55.47 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:00.03 | paulc | how about "manager reload" ? |
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| 17:08.52 | *** join/#asterisk kaushal (~kaushal@115.246.242.75) |
| 17:08.54 | kaushal | Hi |
| 17:09.13 | kaushal | Any step by step guide to install Asterisk SCF ? |
| 17:11.05 | kaushal | How do i check version of asterisk running on the box ? |
| 17:11.21 | Deeewayne | kaushal, there is a #asterisk-scf channel which might be more helpful |
| 17:11.57 | kaushal | Deeewayne: ok |
| 17:12.09 | kaushal | How do i check version of asterisk running on the box ? |
| 17:12.35 | _Corey_ | asterisk -V |
| 17:13.06 | kaushal | _Corey_: Thanks |
| 17:13.21 | kaushal | is Asterisk 1.8.3.3 the latest stable release ? |
| 17:14.17 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-076.mobile.uci.edu) |
| 17:16.04 | kaushal | _Corey_: i see it as 1.8.4 |
| 17:16.22 | Khratos | where does 'busy', 'unavailable', etc. greetings get saved when using ODBC? |
| 17:16.42 | nicola_pav | under fail2ban in jail.conf |
| 17:16.45 | _Corey_ | kaushal:http://www.asterisk.org/downloads |
| 17:16.51 | nicola_pav | can i specify more than one entry in the logpath? |
| 17:17.01 | nicola_pav | i can use * |
| 17:17.17 | kaushal | I tried installing it on CentOS following the document https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29 |
| 17:17.22 | nicola_pav | but what about monitoring two different files? |
| 17:17.32 | nicola_pav | can i use more than one path? what is the syntax please? |
| 17:17.34 | nicola_pav | thanks |
| 17:18.09 | kaushal | but i see it as 1.8.3.3 and not 1.8.4 |
| 17:18.16 | kaushal | Any Clue ? |
| 17:19.41 | Deeewayne | svn list http://svn.asterisk.org/svn/asterisk/tags ... 1.8.4 is the latest tag of the 1.8 branch |
| 17:21.03 | *** join/#asterisk zkn (~zkn@82.131.33.142.cable.starman.ee) |
| 17:21.46 | kaushal | ok |
| 17:22.16 | eXcAliBuR | i'm spinning in a circle... i'ma put my objective and let your lovely ppl steer me... |
| 17:22.51 | eXcAliBuR | Objective is to call a list of phone numbers and alert them of a cancellation. |
| 17:25.26 | eXcAliBuR | can it loop through a list of numbers in a file calling them? |
| 17:26.57 | _Corey_ | excalibur: You probably want to look into generating call spool files with an external script |
| 17:26.59 | kaushal | Deeewayne: so is there a way to get 1.8.4 tag using YUM package management in CentOS ? |
| 17:27.10 | *** join/#asterisk josta (~josta@pdpc/supporter/active/josta) |
| 17:27.15 | _Corey_ | excalibur: there is an example call spool file in the asterisk sources |
| 17:29.54 | kaushal | or do i need to use the source package to get 1.8.4 ? |
| 17:29.56 | Deeewayne | kaushal, yes if there was a 1.8.4 rpm in a repository somewhere. Maybe digium has it the repo(s) used by asterisknow. I dont know. |
| 17:30.38 | kaushal | Deeewayne: ok |
| 17:31.02 | kaushal | so 1.8.3.3 is also good to go ? |
| 17:31.04 | eXcAliBuR | I know that i can make a .call file and asterisk will execute it |
| 17:31.29 | eXcAliBuR | my head is starting to hurt again |
| 17:31.30 | eXcAliBuR | :{ |
| 17:32.07 | Deeewayne | kaushal, you don't _need_ to use a package. yum install subversion ; svn checkout http://svn.asterisk.org/svn/asterisk/tags/1.8.4 asterisk ; cd asterisk ; ./configure ; make ; make install |
| 17:32.54 | kaushal | Deeewayne: yes that helps me |
| 17:33.00 | kaushal | Deeewayne: Thanks again |
| 17:33.07 | kaushal | Much appreciated |
| 17:33.18 | *** join/#asterisk MiteshShah (~chatzilla@115.119.205.34) |
| 17:33.18 | Deeewayne | np |
| 17:37.19 | *** join/#asterisk pdtpatrick_ (~pdtpatric@mainstwan.farheap.com) |
| 17:37.56 | MiteshShah | how to install vicidial n asterisk in ubuntu |
| 17:38.19 | Qwell | ~google vicidial asterisk ubuntu |
| 17:38.41 | Deeewayne | eXcAliBuR, when your head hurts enough to pay a consultant you can ask in #asterisk-consultants |
| 17:39.10 | eXcAliBuR | :} |
| 17:40.44 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 17:46.12 | malcolmd | leifmadsen: yes, your'e on the right track (from hours ago) :D |
| 17:46.22 | malcolmd | er.. you're |
| 17:47.24 | russellb | malcolmd: how dare you typo an IRC message. |
| 17:48.16 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:49.53 | ferdna | my calls wont go into s.... i have to specify my whole number as an extension for my calls to come in... |
| 17:50.00 | eXcAliBuR | why do i over complicate things? |
| 17:50.20 | eXcAliBuR | 2 hrs left of work yay and i have gotten nothing done |
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| 18:31.46 | eXcAliBuR | how many calls at the same time would be too many for asterisk ? |
| 18:31.59 | eXcAliBuR | i have a xeon process 3.0GHZ 1GB memory |
| 18:33.08 | *** join/#asterisk dprophit (dprophit@229.201.205.68.cfl.res.rr.com) |
| 18:34.25 | dprophit | I just tried to download the 64bit ISO for asteriskNOW and the rpm kmod-dahdl-linux-fwload-vpmadt* rpm was corrupt if anyone from Digium is in here. |
| 18:35.49 | fish-bulb | Qwell: have you heard of that? |
| 18:35.57 | JonathanRose | eXcAliBuR: That really depends on a lot of factors. |
| 18:36.45 | dprophit | fish-bulb I'm going to modify the kickstart so I can do a media scan before install and i'm re-downloading the 64bit just in case but.. yah corrupt rpm in the ISO |
| 18:36.53 | JonathanRose | What kinds of channels, encoding, applications running on those calls, monitoring, there's just lots of stuff that can bulk up a call. |
| 18:37.52 | fish-bulb | dprophit: I'll check it out. Did you run a checksum on the iso prior to burning it? I'm looking for a md5 to compare it to |
| 18:37.57 | eXcAliBuR | just a sip call out to play a file and then hang up |
| 18:38.17 | Qwell | fish-bulb: Yes. packages.asterisk.org is being rebuilt ATM. |
| 18:38.24 | Qwell | oh wait, on the ISO? |
| 18:38.40 | dprophit | yah qwell. I just tried to burn the 64bit iso |
| 18:38.41 | Qwell | Yeah, checksum it |
| 18:43.11 | dprophit | I don't even see it at http://packages.asterisk.org/centos/5/current/x86_64/RPMS/ |
| 18:44.09 | dprophit | kmod-dahdl-linux-fwload-vpmadt032-2.3.0.1-1_centos5.2.6.18_194.11.1.el5.x86_64.rpm |
| 18:47.50 | dprophit | Can I just remove all the kmod-dahdl-linux-fwload* rpms from my iso so that the installer can complete the centos install? |
| 18:47.53 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 18:48.56 | dprophit | I'll replace them from the current rpm packages |
| 18:52.29 | sereal | is the syntax for using 'same' like this |
| 18:52.29 | sereal | exten = 1234,1,Function(), exten = same,n,Function() |
| 18:52.50 | sereal | Or can someone point me to some documentation |
| 18:52.59 | sereal | google doesn't seem to be telling me anything |
| 18:53.18 | Qwell | dprophit: No. |
| 18:54.00 | serafie | sereal: same = n,App() |
| 18:54.06 | leifmadsen | same => n,App() |
| 18:54.19 | leifmadsen | sereal: you can learn all about this at ofps.orelly.com |
| 18:54.22 | leifmadsen | ofps.oreilly.com |
| 18:54.51 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-076.mobile.uci.edu) |
| 18:55.03 | dprophit | reattempting the ISO burn again |
| 18:55.03 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 18:55.33 | sereal | thanks |
| 18:55.42 | fish-bulb | dprophit: did you run the checksum and compare it to the md5 or sha1 on the site? |
| 18:56.38 | fish-bulb | dprophit: I just downloaded and opened the iso and I see the package on the disc |
| 18:56.51 | dprophit | no I ignored that. I don't even have an MD5 checker on this windows machine. I'll install one |
| 18:58.04 | fish-bulb | I got 01331d4e8a9a11253dcf65a31b91fe8d, which matches |
| 18:58.34 | *** part/#asterisk JonathanRose (~jonathan@nat/digium/x-pxauucrbvuppeoex) |
| 18:58.38 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-pxauucrbvuppeoex) |
| 19:00.27 | dprophit | 01331d4e8a9a11253dcf65a31b91fe8d as well |
| 19:00.33 | dprophit | Now I'm going to burn and try to install again |
| 19:00.45 | dprophit | I chose option 3 to build asterisk + gui |
| 19:00.47 | fish-bulb | k, it could be an issue with the burner or media then |
| 19:01.16 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust646.15-3.cable.virginmedia.com) |
| 19:01.25 | dprophit | chose to verify media after burning this time |
| 19:01.26 | fish-bulb | once you have it burnt, do you see the file there under /AsteriskNOW/ ? |
| 19:02.09 | dprophit | Before burning, yes. I use PowerISO nice util for creating and modifying ISO's |
| 19:03.19 | dprophit | Is the skype add-on from digium pretty easy to install after purchase? |
| 19:05.05 | fish-bulb | dprophit: yup. super easy even |
| 19:05.16 | dprophit | lol |
| 19:05.44 | fish-bulb | there are packages built for AsteriskNOW as well |
| 19:05.48 | dprophit | I spent all day yesterday trying to get the gui going. I did not love me |
| 19:05.58 | dprophit | it did not* |
| 19:06.42 | fish-bulb | lol, GUIs can be temperamental |
| 19:07.34 | fish-bulb | if you have any problems with installing Skype for Asterisk then give Digium a call, it comes with free install support |
| 19:07.54 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 19:07.58 | leifmadsen | they're the only ones who can really debug the problems |
| 19:08.12 | dprophit | One thing I found confusing on their site is how many licenses I need say if I had 3 incomming and two out going channels |
| 19:08.29 | leifmadsen | I think that requires 5 licenses |
| 19:08.37 | Qwell | 3 plus 2 is indeed 5 |
| 19:08.47 | leifmadsen | wins at addition! |
| 19:09.07 | fish-bulb | from my experience though, between the readme, manual and FAQ, every possibility is covered in the docs |
| 19:09.37 | _Corey_ | yeah, it's easy |
| 19:10.06 | fish-bulb | dprophit: it works a little differently than that. It counts based on concurrent calls, so all of your licenses go in a pool |
| 19:10.15 | _Corey_ | some of the options for the skype users aren't documented but they're pretty obvious |
| 19:11.06 | fish-bulb | so, it's best to think of it as what is the highest number of concurrent calls you expect, rather than incoming or outgoing if they aren't going to be used concurrently |
| 19:11.22 | dprophit | Our phone traffic is spurradic. I'll gamble on 3 for the moment |
| 19:12.01 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:4d0e:6913:1385:4ec1) |
| 19:13.14 | fish-bulb | k. It is trivial to add another license later. It requires no configuration changes, just registering the new license |
| 19:13.58 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:14.28 | bmoraca_work | so, as I understand it, allworx phones are simply rebranded Aastra phones. do they use the same config scripts as aastra phones? |
| 19:17.20 | dprophit | Qwell if this iso gets the GUI working finally, I'm gonna kiss the first fat lady I see |
| 19:18.28 | fish-bulb | dprophit: were you install FreePBX/Asterisk GUI over top of an existing installation before? |
| 19:18.36 | dprophit | thanks the guys from digium for helping yesterday in the #asterisk-gui channel |
| 19:18.53 | leifmadsen | w00t |
| 19:19.23 | dprophit | no I went the rpm install route + gcc build for gui the first time, then RPM installs for everything... still didn't work. So now trying the ISO version |
| 19:22.46 | kfife | leifmadsen: Any idea how much work woudl it be to patch automon to optionally include an audible confirmation? I can kludge up a workaround, but it's such a good idea, that I'd gladly spring for a bounty. |
| 19:25.12 | dprophit | wow the GUI finally =) Guess Asterisk 1.8 and GUI have issues |
| 19:25.32 | dprophit | I'm using 1.6 fromt he ISO |
| 19:25.54 | sereal | is it #include "filename" or just #include filename ? |
| 19:26.20 | paulc | sereal: I think either works - it's referenced in the same config files |
| 19:27.51 | sereal | sorry what do you mean referenced in the same config files? |
| 19:28.06 | serafie | dprophit: \o/ success! |
| 19:28.21 | sereal | you mean if it's in /etc/asterisk/ I don't need the quotes? |
| 19:28.27 | dprophit | seeks out the first fat lady to kiss =) |
| 19:28.58 | dprophit | Do you have a bookmark for the docs to GUI, serafie? |
| 19:29.10 | serafie | Heh, docs? |
| 19:29.20 | dprophit | with pretty pictures =) |
| 19:29.32 | serafie | that's a good idea... someone should do that. |
| 19:29.32 | dprophit | https://wiki.asterisk.org/wiki/display/AST/Asterisk+GUI found it |
| 19:29.45 | dprophit | oh how barron this is |
| 19:29.47 | dprophit | lol |
| 19:30.38 | dprophit | So I guess I don't feel so bad asking setup questions now since I don't have docs to follow |
| 19:30.52 | dprophit | VoIP phone Digit Map = hrmmmm |
| 19:31.08 | serafie | nah, don't. Just feel bad about threatening to ditch Asterisk based on the quality of it's GUI. :) |
| 19:31.14 | dprophit | I can prolly skip this since my code is 1+ |
| 19:31.19 | serafie | *its |
| 19:31.39 | kfife | serafie: nice save |
| 19:31.43 | dprophit | hah |
| 19:32.02 | dprophit | Make it known that GUI is not ready for 1.8 and all should be well in camelot |
| 19:32.20 | serafie | GUI is fine with 1.8. GUI is not ready to be installed outside AsteriskNOW. :P |
| 19:32.44 | serafie | Definitely needs some improvement in the error-reporting and troubleshooting departments. |
| 19:32.45 | kfife | serafie: Your getting you're grammer to good two often and its refreshing. |
| 19:32.52 | serafie | but 1.8 compatibility is fine *now* |
| 19:32.53 | dprophit | k maybe that should go towards the requirements statement prior to install on the wiki |
| 19:33.05 | dprophit | haha kfife |
| 19:33.11 | serafie | *brainsplode* |
| 19:33.53 | sereal | Everyone told me that asterisk gui's and webinterfaces are more complicated than just doing the configs by hand |
| 19:34.32 | leifmadsen | the configs that are generated are likely more complicated |
| 19:35.37 | sereal | I just heard that simply using the gui/webinterface was more complicated |
| 19:35.55 | sereal | well complicated to the point that it would be just as easy to learn how to do it by hand. |
| 19:36.08 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 19:36.46 | JonathanRose | I imagine it's probably a matter of perspective. |
| 19:36.54 | *** join/#asterisk iscario (~div@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 19:36.58 | leifmadsen | depends what you're trying to do I suppose ;) |
| 19:37.05 | JonathanRose | For some people, wizards are the best things ever. |
| 19:37.17 | JonathanRose | For others, they are banal. |
| 19:38.31 | JonathanRose | Among Linux communities, people are just kind of used to looking at big text files full of options. |
| 19:38.50 | JonathanRose | And scripting languages. |
| 19:38.53 | JonathanRose | And all that stuff. |
| 19:39.10 | dprophit | Global OutBound CID syntax of xxx-xxx-xxxx ok? |
| 19:39.37 | *** join/#asterisk iscario (~div@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 19:40.51 | dprophit | the on hovers should include a syntax example. Let me get to the tracker and make a feature request. |
| 19:41.28 | WIMPy | Or a "what if?" button as in SMIT. |
| 19:44.55 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053225.ics.uci.edu) |
| 19:44.58 | dprophit | serafie, are there issues in chrome with the form submit? |
| 19:45.20 | sereal | is asterisk gui like a webinterface? or a actual gui? |
| 19:45.31 | dprophit | web forms |
| 19:45.37 | sereal | ah okay. |
| 19:45.50 | kfife | AMI socket? |
| 19:46.00 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:46.13 | kfife | IOW, it can live on a different host? |
| 19:46.22 | kfife | like a VM? |
| 19:46.41 | dprophit | I can't get the general preferences to save |
| 19:48.26 | seraphie | dprophit: there used to be issues in Chrome, but they were fixed. |
| 19:49.25 | dprophit | General Preferences => Save button at the bottom of the page with red checkmark will not make a form submit action |
| 19:49.52 | dprophit | cancel button works |
| 19:50.47 | dprophit | same issue in firefox 4.01 |
| 19:51.51 | dprophit | I can't even create a dial plan. The div layer pops up. Try to click save.. no love |
| 19:53.26 | seraphie | OH, this is AsteriskNOw |
| 19:53.31 | seraphie | it doesn't have that fix yet. |
| 19:53.51 | dprophit | Kinda hard to setup using GUI when the submit button doesn't work =O |
| 19:53.59 | seraphie | rpms should be coming out *very* soon. |
| 19:54.08 | seraphie | Qwell: do you know when? ^ |
| 19:54.26 | Qwell | seraphie: Soon(TM) |
| 19:54.46 | Qwell | seraphie: After I get these VMs back up. Asterisk 1.8.4 is first, then that. |
| 19:55.00 | seraphie | dprophit: I can tell you what to change if you want. |
| 19:55.15 | seraphie | Or you can use Firefox. :) |
| 19:55.59 | dprophit | ok It said I needed a calling rule before I could create a dialplan so that's where I'm at. Deciding how to allow US + Canada to travel through skypeout |
| 19:57.29 | dprophit | _Nxxxxxxxxxx would that be alright? |
| 19:58.10 | fish-bulb | looks like one too many x's |
| 19:58.23 | dprophit | counts to ten heh |
| 19:59.07 | Qwell | Only if you want people to call the 000 prefix. |
| 19:59.16 | dprophit | heh |
| 20:00.04 | dprophit | I want xxx-xxx-xxxx which I hope just covers us and canada to travel through skype out. I want to maybe use a pin code to allow someone to dial international skype out rates |
| 20:04.05 | dprophit | Tried to setup trunk. I guess the save button doesn't work anywhere, seraphie =( |
| 20:04.38 | dprophit | is that sip.conf |
| 20:04.42 | seraphie | Firefox. or I can tell you what files to edit... |
| 20:04.54 | seraphie | the GUI puts trunks in users.conf |
| 20:04.56 | dprophit | I'm using firefox 4.01 |
| 20:04.59 | dprophit | ok |
| 20:05.08 | seraphie | OK, Firefox 3, then. :-\ |
| 20:05.50 | dprophit | so in brackets is the provider name, right? |
| 20:06.32 | QuantumSchema | Hey all! |
| 20:06.52 | QuantumSchema | Does anyone here happen to know about using mpg123 for streaming MOH? |
| 20:06.55 | seraphie | dprophit: honestly, you need to wait until the rpms come out or use another browser. |
| 20:06.57 | QuantumSchema | I'm puzzled at the moment. |
| 20:07.38 | diatonic | dprophit, serafie Firefox 4.01 works fine for me on FreePBX 2.8.1.4 |
| 20:07.48 | QuantumSchema | the MoH config is correct (i think), the mpg123 process is starting and is listed under ps. I still hear no audio on the client. |
| 20:07.49 | QuantumSchema | :-( |
| 20:08.17 | dprophit | I chose option 3. Asterisk 1.6 + GUI |
| 20:08.43 | *** join/#asterisk michael-i (~michael@204.11.230.58.static.etheric.net) |
| 20:08.49 | dprophit | I'll just use my time to read |
| 20:09.38 | diatonic | dprophit, serafie Ugh, thought I was reading in #freepbx |
| 20:09.47 | QuantumSchema | This is the musiconhold.conf for it...... http://pastebin.com/60nRVayw |
| 20:09.59 | seraphie | dprophit: another option would be renaming your config directory and checking out asterisk-gui from svn and replacing it with the updated version. |
| 20:10.21 | seraphie | Or just replace the html and js files |
| 20:10.28 | dprophit | remember how evil that was to me yesterday? =O |
| 20:10.35 | seraphie | which wouldn't even require you to change the directory structure |
| 20:10.46 | seraphie | if you're only replacing the static content, nothing would change. |
| 20:11.12 | dprophit | ok I'll do a local working copy onto my pc using tortoisesvn |
| 20:11.45 | seraphie | cp tempdir/config/*.html /var/lib/asterisk/static-http/config/ && cp tempdir/config/js/*.js /var/lib/asterisk/static-http/config/js |
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| 20:45.40 | ferdna | my calls are not being sent to 's'... |
| 20:45.48 | ferdna | i need to specify full number |
| 20:45.54 | ferdna | is there a work around for this? |
| 20:48.34 | *** join/#asterisk sonic (~paul@2001:470:1f09:11d6::2600) |
| 20:50.58 | ChannelZ | SIP? |
| 20:51.15 | *** join/#asterisk Freeaqingme_ (~dolf@dsl-083-247-011-232.solcon.nl) |
| 20:51.19 | ChannelZ | Most ITSPs send to an extension matching your DID so you can figure out where the call came from. |
| 20:51.40 | Freeaqingme_ | Does anybody have info that would reflect the future contents of his page? ("Generating billing information from CEL"): https://wiki.asterisk.org/wiki/display/AST/Generating+Billing+Information+from+CEL |
| 20:51.44 | ChannelZ | 's' is only for calls that come in with no extension |
| 20:54.54 | paulc | ferdna: are you registering with them using a register => line? |
| 20:55.11 | paulc | ferdna: and do you have multiple DIDs with them (on a single account)? |
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| 21:14.08 | dprophit | seraphie are you still @ work? |
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| 21:57.55 | *** part/#asterisk sn9 (BO37@gimpelevich.san-francisco.ca.us) |
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| 21:59.24 | ferdna | paulc, yes i am using "register =>" line and no i just have one account |
| 21:59.26 | sn9 | is there a general-purpose voip channel, not necessarily related to asterisk? |
| 22:00.05 | paulc | ferdna: Do you have an extension number on the end of the line? register => username:password@provider/target-extn-here |
| 22:00.06 | WIMPy | #doomed? |
| 22:00.14 | paulc | May 21? |
| 22:00.28 | paulc | sn9: Try #vuc |
| 22:00.36 | ferdna | paulc, no i don't |
| 22:00.46 | sn9 | thanx |
| 22:00.47 | *** part/#asterisk sn9 (BO37@gimpelevich.san-francisco.ca.us) |
| 22:01.02 | paulc | ferdna: Add one and it'll land on that extension in the context you defined in [general] |
| 22:02.03 | paulc | ferdna: It's different with multiple DIDs on the same account - in that case you want your provider to send you the right thing... but for single account/DID operation, it should work fien |
| 22:02.08 | paulc | fine* |
| 22:04.01 | ferdna | paulc, so there is no way to just use "s"? |
| 22:04.53 | ChannelZ | he just told you how |
| 22:05.52 | paulc | ferdna: If you don't end your register with /extn-number-here, you should land in s. If you specify /some-extn-number, you'll land there. |
| 22:06.07 | ferdna | ChannelZ, i will have to define an ext for incoming calls... i dont want that.. |
| 22:06.37 | ferdna | paulc, yeah i understand that... however i havent specified and register provider/EXT |
| 22:06.47 | ferdna | and thus all calls should be picked up by 's' |
| 22:06.47 | ChannelZ | So you don't want to tell the calls where to go, but you want them to go to a specific place... |
| 22:07.03 | ferdna | ChannelZ, i just want them to go to 's' |
| 22:07.06 | ChannelZ | No, they will go wherever your provider sends them to |
| 22:07.19 | ChannelZ | which is usually your DID. Who cares anyway? |
| 22:07.53 | ChannelZ | make exten => yourdid,1,Goto(s,1) |
| 22:08.38 | ferdna | ChannelZ, awesome =) |
| 22:08.49 | ferdna | paulc, ChannelZ , thanks =) |
| 22:08.54 | ChannelZ | But as he said, you can also ask if your provider to send them to 's' in your register line |
| 22:09.17 | ChannelZ | whether or not they do is another story but.. |
| 22:10.05 | ChannelZ | meant 'ask of', not 'ask if'. Oops |
| 22:10.53 | ferdna | :) |
| 22:10.55 | ferdna | thanks |
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| 22:33.56 | Ineluctable | How do I install chan_gtalk in asterisknow? I installed iksemel, but am confused. Do I still do the ./configure , menuselect thing? |
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| 22:43.15 | pdtpatrick_ | is there a way to reset a user's voicemail message to the default message in asterisk or do i have to call the mailbox, delete the message and then use default? |
| 22:43.38 | WIMPy | Delete the files? |
| 22:44.16 | pdtpatrick_ | that would be located where please? |
| 22:44.47 | WIMPy | In the users mailbox dir. |
| 22:45.18 | WIMPy | Don't remember what the files are called, but it's obvious if you take a look. |
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| 22:45.42 | *** mode/#asterisk [+o leifmadsen] by verne.freenode.net |
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| 22:53.09 | pdtpatrick_ | @WIMPy are you talking about voicemail.conf ? |
| 22:53.25 | pdtpatrick_ | or there's another location where the inbox files are kept |
| 22:53.38 | WIMPy | No, the directory where the voicemail is stored. |
| 22:54.16 | pdtpatrick_ | let me check |
| 22:54.32 | pdtpatrick_ | i think it might be /var/spool/asterisk/voicemail ?? |
| 22:56.39 | WIMPy | And thn the context and mailbox your interested in. |
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| 23:08.30 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 23:18.05 | pdtpatrick_ | Thanks WIMPy |
| 23:19.11 | pdtpatrick_ | Question: Where are greetings stored? For instance when a user calls a number and it says "Thank you for calling so and so corporation, if you know the party's extension please dial .... " .. where's that usually kept if anyone can help please |
| 23:21.15 | el3slave | i think ./var/lib/asterisk |
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| 23:37.29 | paulc | pdtpatrick_: extension 1234 in the voicemail context default will be in /var/spool/asterisk/voicemail/default/1234 - the files you want are "busy" or "unavail" |
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| 23:45.18 | pdtpatrick_ | el3slave: would that be under sounds in /var/lib/asterisk/sound ?? |
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| 23:51.12 | pdtpatrick_ | where in asterisk do u set properties for how a call is managed when you are dialed? |
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| 23:55.24 | paulc | pdtpatrick_: in /etc/asterisk/extensions.conf |
| 23:55.50 | paulc | and yes, your auto attendant and other IVR related sounds come from /var/lib/asterisk/sounds |