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| 00:15.11 | killfill | hm.. whats rxwink?.. |
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| 03:06.21 | killfill | hm.. |
| 03:07.07 | killfill | how do i see in what version did ringtimeout did apear the first time in zapata.conf? (or chan_dahdi.conf) |
| 03:07.29 | killfill | its the option i need, and in production, i got an old version. need to know exactly what version is it. |
| 03:08.36 | killfill | svn says " 6470 markster ;ringtimeout=8000" |
| 03:08.45 | killfill | but thats from 2005.. :S |
| 03:08.55 | killfill | in asterisk 1.4.4 i see its ignoring that option... |
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| 03:49.56 | Yurik | Anybody experienced heavy jitter on 2-way calls (originated by a combination of chan_local of extension number), but only for Playbacks â voice works absolutely fine |
| 03:55.02 | Yurik | asterisk 1.6.1, dedicated server (not a xen or vmware instance) |
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| 04:19.58 | kyoshi | i'm using asterisk 1.4.1 in real time and i want to use Set(CDR(MyColumn)=1) however, it writes the CDR data to the Master.csv text file but not to the CDR on the MySQL DB. How can I ensure that it is written to the MYSQL CDR table? |
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| 04:26.02 | kyoshi | bad connection :( |
| 04:26.03 | kyoshi | i'm using asterisk 1.4.1 in real time and i want to use Set(CDR(MyColumn)=1) however, it writes the CDR data to the Master.csv text file but not to the CDR on the MySQL DB. How can I ensure that it is written to the MYSQL CDR table? |
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| 04:41.09 | bbryant | kyoshi, 1.4.1 is an old version |
| 04:41.20 | kyoshi | yea i know |
| 04:41.41 | kyoshi | but right now i have an entire platform build around it so im now yet ready to step it up |
| 04:44.46 | [TK]D-Fender | kyoshi: And everything that is broken back then... is still broken... and there are a lot of CDR fixes for cutom filed that has happened in later versions |
| 04:46.54 | Kevin` | can anyone recommend some voip providers? |
| 04:47.03 | Qwell | ~itsplist-us |
| 04:47.04 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 04:48.34 | Qwell | hmm |
| 04:49.14 | Qwell | infobot: no, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 04:49.14 | infobot | okay, Qwell |
| 04:54.29 | kyoshi | [TK]D-Fender: the cdr's write perfectly to the mysql db, its just the custom setting in the extensions (also realtime from the db), i'd like to put in an extra column in the cdr table for specific things when calls come in. it writes fine to the Master.csv but not to the cdr table in the db. |
| 04:55.29 | Yurik | hmm... even chan_local's /j flag does not help.. still jittering awfully on Playback :S |
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| 05:00.03 | drmessano | Hmmm |
| 05:00.23 | kyoshi | i was just hoping that there was a function to write to the db cdr as well as the text file cdr |
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| 05:03.34 | [TK]D-Fender | kyoshi: what part of "some of that shit was BROKEN back then" have I not made clear? |
| 05:05.28 | kyoshi | so are you saying its fixed in the newer version or not fixed in the newer version? |
| 05:06.55 | *** join/#asterisk Ryan09 (n=Ryan09@mail.intersoft.net.au) |
| 05:07.51 | Ryan09 | Hi, I'm having a lot of trouble getting dahdi to detect hangups reliably. |
| 05:08.47 | Ryan09 | I have loadzone and default zone set in system.conf, I have busydetect=yes and busycount=4 in chan_dahdi.conf |
| 05:09.11 | Ryan09 | I have the correct country code set in indications.conf |
| 05:09.14 | drmessano | Anyone know a good tutorial on SED? |
| 05:09.36 | [TK]D-Fender | kyoshi: Just curious, but are you actually of Japanese heritage by any chance? |
| 05:09.38 | Ryan09 | I'm using an openvox a400p |
| 05:10.28 | [TK]D-Fender | Ryan09: Good luck getting Telstra to enable CDS on your line... |
| 05:10.30 | [TK]D-Fender | ~cds |
| 05:10.30 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 05:10.40 | Ryan09 | The carrier doesn't support polarity switching. Does anyone have any ideas on how I can improve hangup detection? |
| 05:10.50 | [TK]D-Fender | Ryan09: this is not indications.conf |
| 05:11.35 | [TK]D-Fender | Ryan09: you can try "callprogress=yes", which BTW is nearly synonymous with "hangupmycallswhenleastconvenient=yes" |
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| 05:12.20 | Ryan09 | lol, ok |
| 05:12.31 | Ryan09 | What is not indications.conf? |
| 05:13.31 | kyoshi | tkd: nopes |
| 05:13.35 | kyoshi | its a title |
| 05:16.33 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 05:18.51 | [TK]D-Fender | kyoshi: Oh well... that might have explained your complete lack of a properly formed future tense :p |
| 05:19.28 | [TK]D-Fender | kyoshi: So let me re-iterate : CDR shit was BROKE in early 1.4 versions and it took them a long time to FIX them |
| 05:19.55 | [TK]D-Fender | kyoshi: So stay there if you want, but that plus all the vulnerabilities, etc make your systems targets |
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| 05:25.05 | kyoshi | hehehe |
| 05:25.31 | kyoshi | how about past formed as well? the japanese language sucks with time frame references |
| 05:25.38 | kyoshi | fortunately english is my primary language |
| 05:25.47 | kyoshi | ok so 1.4 is shot |
| 05:25.50 | kyoshi | that sucks for me |
| 05:26.47 | kyoshi | gonna be hard tho cause we spend months rewriting the code and stripping it down |
| 05:27.12 | kyoshi | pretty much no vunerabilities and very lean |
| 05:27.19 | kyoshi | also has very few features and functions |
| 05:27.58 | kyoshi | bah i should have them recode 1.6 anhows |
| 05:28.00 | kyoshi | anyhows |
| 05:30.10 | [TK]D-Fender | KyNot 1.4... 1.4.1 <- |
| 05:30.37 | kyoshi | ? |
| 05:31.05 | [TK]D-Fender | [00:26]<kyoshi>i'm using asterisk 1.4.1 in real time and i want to use Set(CDR(MyColumn)=1) however, it writes the CDR data to the Master.csv text file but not to the CDR on the MySQL DB. How can I ensure that it is written to the MYSQL CDR table? |
| 05:31.20 | [TK]D-Fender | KyYou just said the entire 1.4 BRANCH is shot. no, only the EARLY releases |
| 05:31.25 | kyoshi | ok well i mis typed then |
| 05:31.38 | kyoshi | it writes fine to the mysql |
| 05:31.49 | kyoshi | but |
| 05:33.43 | kyoshi | try this |
| 05:33.44 | kyoshi | i'm using asterisk 1.4.1 in real time and it writes the CDR data to the Master.csv text file and to the CDR on the MySQL DB. |
| 05:33.44 | kyoshi | i want to use Set(CDR(MyColumn)=1) however, it does not write the cusomtization to MyColumn in the table. How can I ensure that it is written to the MYSQL CDR table/column as i would like? |
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| 05:34.03 | kyoshi | i think thats a bit more to the point of what i need |
| 05:34.41 | kyoshi | sorry its 1.4.21 |
| 05:34.44 | kyoshi | just checked |
| 05:35.03 | [TK]D-Fender | kyoshi: Holy shit, NOTHING you say comes out right |
| 05:35.13 | kyoshi | tell me about it. its a bad night |
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| 05:35.26 | [TK]D-Fender | on that note... bed time... I'm off |
| 05:35.27 | kyoshi | keep in mind, im relaying info from someone else |
| 05:35.29 | [TK]D-Fender | BBL |
| 05:35.38 | kyoshi | from another language too |
| 05:35.43 | drmessano | Damnit |
| 05:35.55 | [TK]D-Fender | 15th-hand miscommunicated information... LOVELY |
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| 05:36.30 | kyoshi | its not that far off, just 2 hands over |
| 05:36.46 | kyoshi | from database admin (cantonese) to me to here |
| 05:37.09 | kyoshi | and my cantonese aint great either |
| 05:38.50 | Ryan09 | ...ok, so assuming telstra wont be at all helpful, and apparently don't provide any form of CDS, should I be using fxo_ls instead of fxo_ks? Does it make any difference? |
| 05:44.10 | *** join/#asterisk gladier (n=bburgess@pedobear.molested.me) |
| 05:45.52 | gladier | hey guys, just looking for some help with my tftp setup - i have a heap of snom300 phones that pump out this strange error on my tftp server, but i dont really have the issue with cisco phones - the error i get is Serving SIPDefault.cnf to 20.0.0.26:2052 | recvmsg: Connection refused | tftpd_file.c: 926: recvfrom: Connection refused |
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| 06:53.45 | ^shark_ | if i bought an analog card that means i will have to connect to my analog phones & also the telco should provide me an analog connection, right? |
| 06:54.24 | carrar | telco will provide you with whatever you order |
| 06:54.51 | carrar | What is "analog card" |
| 06:55.12 | tzafrir_laptop | ~fxsfxo |
| 06:55.13 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 06:55.20 | ^shark_ | i meant to say an analog interface card |
| 06:55.35 | carrar | What card |
| 06:55.36 | tzafrir_laptop | right, but there are two types of them. See above |
| 06:56.08 | carrar | analog is vauge |
| 06:56.11 | tzafrir_laptop | Or rather: two types of ports |
| 06:56.20 | carrar | Please be specific |
| 06:56.24 | carrar | model number |
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| 06:58.46 | ^shark_ | ok the reason i am asking this is, we have analog phones & would like to have a call center. I either have to do away with the panasonic kx-ts500mxw phones or purchase voip digital phones but i am not sure which kind of interface card to purchase and type of connection i will have to get from the telco |
| 06:59.12 | ^shark_ | the call center will be under asterisk |
| 07:00.03 | carrar | What kind of connection do you want from the telco? |
| 07:00.29 | carrar | How many concurrent calls are you doing with this call center |
| 07:00.36 | ^shark_ | i have no idea what kind of connections i can get from the telco |
| 07:01.13 | ^shark_ | the call center is not yet up but would prefer to do as much as 100+ calls |
| 07:02.15 | carrar | So you know if you go with PRI's thats 23 concurrent calls |
| 07:02.23 | carrar | so 100/23 |
| 07:02.29 | carrar | number of pri's you will need |
| 07:02.59 | carrar | So you need at least 5 PRI's of voice |
| 07:03.15 | carrar | to do 100 concurrent calls |
| 07:03.27 | carrar | or get a SIP Provider |
| 07:04.26 | drmessano | carrar: Know a good place I can find a list of characters I need to escape with sed? |
| 07:04.55 | drmessano | Some are obvious.. some not so much |
| 07:04.57 | carrar | man sed? |
| 07:05.06 | drmessano | heh |
| 07:05.10 | ^shark_ | carrar: a sip provider would be a company connecting my phones to the voip network, right? |
| 07:05.16 | carrar | http://www.grymoire.com/Unix/Sed.html |
| 07:05.23 | drmessano | Im so sick of reading 20 page docs on SED |
| 07:05.27 | carrar | drm, what are you trying to do? |
| 07:05.28 | drmessano | Read that too |
| 07:05.35 | ^shark_ | carrar: i am in Uganda east africa and i do not think we have pri if i may be correct |
| 07:06.09 | carrar | ^shark_, sip provider would connect to your Asterisk box, I think you have some work to do before turning up your Asterbox |
| 07:06.40 | drmessano | Im redoing a lot of my master PBX setup guide and I am on a kick to sed a lot of the configs vs nano + edit for something simple |
| 07:06.50 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:07.04 | drmessano | I've learned a lot in a few hours, but its just DRAWN out finding simple shit |
| 07:07.06 | yang | hello mort_gib |
| 07:07.14 | drmessano | or, I guess.. what I think is simple |
| 07:07.31 | ^shark_ | carrar: ok i would like to get my own connection from the telco |
| 07:07.32 | mort_gib | Morning yang |
| 07:08.47 | ^shark_ | thinks he needs to read more about PSTN lines |
| 07:10.36 | carrar | drm, just use vi |
| 07:11.00 | carrar | unless you know what you are replacing |
| 07:11.38 | carrar | sed & awk can do ANYTHING however |
| 07:12.44 | drmessano | I know what I am replacing |
| 07:12.46 | drmessano | Thats the point |
| 07:13.49 | KyleK | ^shark_: whats the telephone company over there? |
| 07:15.04 | ^shark_ | KyleK: Uganda telecom, & MTN are the only ones with fixed lines |
| 07:15.25 | ^shark_ | PSTN is different from VoIP right? |
| 07:15.53 | KyleK | yea |
| 07:16.17 | KyleK | Public Switched Telephone Network |
| 07:17.17 | ^shark_ | KyleK: PSTN is for analog only right? |
| 07:17.58 | KyleK | well in north america we can get PRI's which is part of the telephone network but its still digital |
| 07:20.04 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 07:20.09 | ^shark_ | KyleK: if i got a PSTN lineis there anything else i can use apart from PRI |
| 07:21.24 | KyleK | well what the telco gives you determines what you use, if they give you 100 analog lines, you need FXO channel banks that add up to 100 channels |
| 07:22.07 | kaldemar | ^shark_: PSTN is a network with many techniques, ask your telco what they offer. |
| 07:23.53 | *** join/#asterisk SebastianS (n=schu@dsl-static-111.212-5-200.telecom.sk) |
| 07:24.49 | ^shark_ | kaldemar: ok |
| 07:28.51 | drmessano | kaldemar: Sounds like a chick I used to "date" |
| 07:29.54 | ^shark_ | ok over here i am told we use E1 connections |
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| 07:38.21 | kaldemar | drmessano: sounds horrifying. |
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| 07:46.31 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 07:47.33 | ck_28 | Morning all |
| 07:47.54 | ck_28 | i have a problem using an automated call |
| 07:48.04 | ck_28 | iam using a call file to send a fax |
| 07:48.22 | ck_28 | but the call file send the call twice |
| 07:48.29 | ck_28 | any one can assist me |
| 07:48.55 | ck_28 | my configuration and debug located at http://pastebin.com/d44755e8a |
| 07:49.19 | ck_28 | which leads to busy |
| 07:51.05 | kaldemar | you're calling the same extension twice. call files first call the given channel, then connect it to the given extension upon answer. your channel and extension point to the same extension in your dialplan. |
| 07:51.09 | *** join/#asterisk rhassing_work (n=rob_work@ti152.telin.nl) |
| 07:52.07 | ck_28 | kaldemar i recognized that but i did find a solution to start upon answer |
| 07:52.09 | kaldemar | ck_28: change context in the call file to fax-tx and extension to s and try again. |
| 07:52.22 | ck_28 | kaldemar i will try |
| 07:52.27 | kaldemar | now you're trying to connect the receiving end to itself. |
| 07:52.59 | *** join/#asterisk jsolis (n=Jimmy@190.43.79.89) |
| 07:53.17 | jsolis | hey guys |
| 07:53.56 | jsolis | i try to use sip.conf with realtime but asterisk dont show me the peers when i use sip show peers |
| 07:54.48 | kapr | hey guys i have two sip accounts on my main server i want that any calls comming from server-A to my main server should use first SIP account and calls comming from server-b should use 2nd SIP account only Main server can make outgoing SIP calls |
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| 08:08.10 | ck_28 | kaldemar thanks bro it worked |
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| 09:22.38 | *** part/#asterisk jsolis (n=Jimmy@190.43.79.89) |
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| 09:59.43 | ck_28 | how to convert gif or jpg file to tif format with res 204x196 in order to be sent using asterisk fax,i know how to convert from pdf to tif |
| 09:59.47 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 09:59.52 | j_kroon | would somebody please take a look at https://issues.asterisk.org/view.php?id=15181 ? |
| 10:00.15 | j_kroon | i'm pretty sure it's a bug in asterisk, but russel is making me wonder. |
| 10:00.24 | j_kroon | his argument makes sense but it doesn't seem to work in practise. |
| 10:01.11 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 10:04.06 | ck_28 | i used gs -q -sDEVICE=tiffg3 -sPAPERSIZE=letter -r204x196 -dNOPAUSE -sOutputFile=call.tif c.jpg but failed |
| 10:06.05 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
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| 10:14.54 | ^shark_ | i have a panasonic kx-ts500mxw analog phone is there a way i cna use it to make voip calls if i had an E1 connection from the telco? |
| 10:20.59 | j_kroon | hi, it seems the handling of frames changed in main/file.c and res/res_convert.c was never updated to comply. who would be able to assist me in cooking up a patch? |
| 10:21.43 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
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| 10:22.58 | *** join/#asterisk bionoid (i=terje@mesyah.org) |
| 10:23.58 | kaldemar | ^shark_: if you plug it into an ATA or an analog interface in your asterisk box, you can make VoIP calls with it. |
| 10:24.21 | bionoid | Hello Gurus. I have a TDM400 that -- users report -- does not always free the Zap channel when remote user hangs up the conversation. The thing is, it works every time I test it, - could this somehow be caused by old/outdated SIP client software? |
| 10:31.17 | kaldemar | if remote means someone in the zap end, then no. |
| 10:41.22 | tzafrir_laptop | bionoid, FXO? |
| 10:42.18 | tzafrir_laptop | If so: which PSTN provider are you connected to? Which country? |
| 10:46.26 | ^shark_ | kaldemar: if i got any one of these ATAs and plugged into the asterisk box, then i would need a switch to transfer the traffic to and from the analog phone |
| 10:46.42 | ^shark_ | <PROTECTED> |
| 10:47.11 | ^shark_ | <PROTECTED> |
| 10:50.44 | j_kroon | tzafrir_laptop, how's your knownledge on the usage of stuff in main/file.c ? |
| 10:51.07 | tzafrir_laptop | I know a bit of it |
| 10:51.35 | bionoid | tzafrir_laptop: Yes, FXO -- Telenor, Norway |
| 10:52.38 | tzafrir_laptop | for starters, try setting hanguponpolarityswitch=yes and answeronpolarityswitch=yes in chan_dahdi.conf (or zapata.conf) |
| 10:54.17 | bionoid | tzafrir_laptop: Ah! Thank you. |
| 10:54.29 | kaldemar | ^shark_: yes, you need a network for the ATAs. any ordinary ethernet switch will do. |
| 10:56.30 | ^shark_ | kaldemar: the digital phones will also be able to call analog phones and viceversa? |
| 10:56.58 | *** join/#asterisk chendy (n=chatzill@58.251.103.75) |
| 10:57.17 | *** join/#asterisk ixremedy (n=ixremedy@client130-7.emplot.net.ua) |
| 10:57.28 | ixremedy | hello everyone |
| 10:58.16 | kaldemar | ^shark_: ATA is an adapter that makes an analog phone a VoIP phone. everything that you connect to asterisk, will be able to dial call each other. |
| 10:58.25 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:58.45 | ^shark_ | kaldemar: ok thanks mate |
| 10:58.53 | ixremedy | i've got a question whether or not asterisk supports the SIP MESSAGE mechanism and it is possible to use it as a messaging server where everyone uses REGISTER, NOTIFY and stuff |
| 10:59.25 | ixremedy | i couldn't find any concise answer to that anywhere |
| 11:00.34 | atis_work | ixremedy: what do you want to do with REGISTER and NOTIFY? |
| 11:01.02 | atis_work | it supports SIP registrations, and you can subscribe to certain events and receive NOTIFY |
| 11:02.15 | ixremedy | with REGISTER i want to get guys registered to the queues of interest, than when someone sends out a message, he posts it to asterisk and the asterisk generates a NOTIFY to everyone subscribed to the queue so that everyone can fetch the message |
| 11:02.44 | ixremedy | or the MESSAGE itself can contain the message body |
| 11:02.51 | ixremedy | in the payload section |
| 11:03.18 | atis_work | ixremedy: well, users would have to be allowed to register in sip.conf/users.conf |
| 11:03.25 | atis_work | and there is kind of custom device states |
| 11:03.42 | atis_work | however i'm not sure that you can send arbitary data in sip body |
| 11:04.00 | ixremedy | there is a special container called MESSAGE for that |
| 11:04.23 | atis_work | you're probably better with that on some specific queue server like jabber or activemq |
| 11:04.30 | ixremedy | can those fields from sip.con/users.conf be kept in MYSQL so that i dinamically change the users? |
| 11:04.54 | atis_work | yes |
| 11:05.00 | atis_work | read on realtime engine |
| 11:05.20 | atis_work | but try the custom sip body and device states first.. |
| 11:05.23 | ixremedy | you see, i don't see any particular reason why install anything like jabber if there is a specific container in SIP just for that purpose |
| 11:05.59 | atis_work | ixremedy: probably it's way better configurable for that purpose than asterisk |
| 11:06.08 | atis_work | but of course you can try :) |
| 11:06.22 | ixremedy | RFC 3428 |
| 11:06.22 | atis_work | asterisk is meant to provide some state information, voicemail count, etc.. |
| 11:06.57 | ixremedy | certain queues will be video and voice, jabber's not gonna fit |
| 11:07.07 | atis_work | ooh |
| 11:07.21 | atis_work | try to bug oej about that, but i'm not sure he likes word video :SD |
| 11:07.22 | atis_work | :D |
| 11:07.44 | atis_work | ah, he's not in #asterisk.. |
| 11:07.52 | ixremedy | the prime point is RFC 3428 MESSAGE now |
| 11:08.07 | ixremedy | that askterisk can do video and voice i am aware |
| 11:08.15 | ixremedy | but i am not sure about MESSAGE |
| 11:09.08 | ixremedy | and another things is, ok lets assume it does MESSAGE, that would it be possible to NOTIFY everytone SUBSCRIBED of the reception of a new message? |
| 11:11.27 | atis_work | ixremedy: well, there's something in chan_sip.c regarding RFC3428 and MESSAGE.. so probably something should work :) |
| 11:20.26 | ixremedy | bad is that there is no document that says whats in there |
| 11:20.59 | *** join/#asterisk wonderw (n=ww@dslb-092-072-171-185.pools.arcor-ip.net) |
| 11:22.04 | wonderw | hi, stupid question, but i wasn't able to find an answer: how would i immediatly dial a different extension, if the one i am dialing is busy? i.e. not using Wait to do it? thanks |
| 11:22.39 | oej | Asterisk doesn't really support MESSAGE |
| 11:23.07 | oej | We do support sending MESSAGE by using the sendtext() dialplan application |
| 11:25.29 | kaldemar | wonderw: check the contents of DIALSTATUS variable in the next priority. if it is BUSY, make the call. |
| 11:26.17 | ixremedy | thats bad |
| 11:26.45 | ixremedy | how hard do you think it is to realise that in the code of asterisk if i harness our programmers to do it? |
| 11:29.07 | ixremedy | or what the price would be to prioritise the development of that feature in digium? |
| 11:30.37 | wonderw | kaldemar: wouldn't asterisk stop going on in the dialplan after not beiing able to reach the first number? |
| 11:34.16 | kaldemar | wonderw: no, it will move to the next priority if it exists and priority jumping is not in use. |
| 11:41.24 | ixremedy | does the said mean that Asterisk wouldn't understand an incoming MESSAGE? |
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| 12:05.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:28.21 | gregd | hi guys, I was just wondering, is it possible (extensions.conf) to create a dialplan in which there is a welcome message being played and at the same time (in the background) Dial command is executed. If successful, after the welcome message the call should be transfered? |
| 12:30.01 | gregd | where should I look for it, which command to investigate? |
| 12:32.35 | *** join/#asterisk Aiatek (n=Asterisk@200.88.112.75) |
| 12:33.00 | kaldemar | i'd try that first with a call file and app Bridge. |
| 12:34.12 | kaldemar | but a single queue would probably do that for you, with the welcome message as the queue's moh. |
| 12:35.24 | gregd | hmm |
| 12:36.47 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:36.51 | gregd | thanks |
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| 12:51.58 | rhassing_work | I'm having some troubles with dnsmgr. When I disable it in dnsmgr.conf, I get the following message in my console: "doing dnsmgr_lookup for 'sip.voipbuster.com'" When I enable it, I get problems with the DNSSRV records at voipbuster and get the following error messages: http://pastebin.com/d6a3e7100 |
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| 13:06.17 | *** join/#asterisk jeeger (n=user@192.68.211.161) |
| 13:07.08 | jeeger | Greetings! I'm trying to create a script that calls several people in a row and asks them for a password. I want to quit calling after the first person enters the correct password. I am somewhat mystified as how to do that, though. Any hints? |
| 13:07.10 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:08.29 | leifmadsen | jeeger: sounds like a While() loop and Originate() would do that |
| 13:08.33 | leifmadsen | (on 1.6.x) |
| 13:08.57 | jeeger | Um, originate? |
| 13:09.02 | leifmadsen | alternatively, it would just be a script that would execute calls via AMI |
| 13:09.07 | leifmadsen | (Asterisk Manager Interface) |
| 13:09.15 | leifmadsen | jeeger: core show application Originate |
| 13:09.44 | jeeger | come again? I can't find Originate() in the docs |
| 13:10.07 | leifmadsen | jeeger: I don't even know what docs you're reading |
| 13:10.24 | jeeger | The asterisk book |
| 13:10.35 | leifmadsen | jeeger: I mentioned 1.6.x. That book is based on 1.4.x. |
| 13:10.42 | jeeger | Ah, okay. |
| 13:11.30 | [TK]D-Fender | AMI originate or CLI via System(), AGI(), etc |
| 13:11.46 | leifmadsen | System() yuck |
| 13:11.56 | [TK]D-Fender | jeeger: AND I already told you how to do his alst week. |
| 13:12.48 | [TK]D-Fender | jeeger: 1 call file or originate witha local channel which dials your fixed call-out list consecutively using M() dial option to validate the answer. |
| 13:13.06 | [TK]D-Fender | jeeger: And just repeat in that local channel exten till you get a hit. |
| 13:13.35 | jeeger | [TK]D-Fender: The problem is how to get the return value from the "query" dialplan (Authenticate etc.) to the "calling" dialplan (the one running Dial()) |
| 13:14.41 | [TK]D-Fender | jeeger: M() <----------- |
| 13:14.46 | jeeger | Ah, yes just saw it. |
| 13:14.49 | [TK]D-Fender | jeeger: "core show application dial" |
| 13:14.56 | [TK]D-Fender | jeeger: Now please go read the directions |
| 13:24.36 | *** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 13:25.17 | ^shark_ | ok got a quote from the telco and they are charging a standard fee for all the 30 channels, does asterisk use all the 30 channels? if not how many channels does asterisk use? |
| 13:25.50 | ^shark_ | .... quote for a E1 PRI |
| 13:26.45 | [TK]D-Fender | ^shark_: it uses as many of the 30 possible as you TELL it to |
| 13:27.35 | ^shark_ | [TK]D-Fender: but there is no problem if it uses less of the 30 channels? |
| 13:29.02 | mmlj4 | ^shark_: not unless the channel police start asking questions |
| 13:29.15 | jaytee | sounds like buying a 12 cylinder Jaguar and only running it on 8 cylinders |
| 13:29.24 | ^shark_ | Ah! okay |
| 13:29.28 | [TK]D-Fender | ^shark_: No, but you should math what your telco is providing |
| 13:29.29 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 13:29.31 | [TK]D-Fender | match* |
| 13:29.49 | ^shark_ | and what is IAVI |
| 13:30.03 | ^shark_ | <PROTECTED> |
| 13:30.13 | [TK]D-Fender | ^shark_: Full PRI, which is normal |
| 13:30.22 | ^shark_ | <PROTECTED> |
| 13:32.37 | ^shark_ | the telco has allocated E1 pilot number for 3 months to enable smooth transition. I do not know what they mean by this, was hoping someone might help |
| 13:33.08 | [TK]D-Fender | ^shark_: Single DID. Not much to say for this. they are porting your old # to it? |
| 13:35.38 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 13:35.46 | ^shark_ | [TK]D-Fender: okay since i have not paid for this connection the DID has not been provided, I am wondering if there is anything i need to do on Asterisk to enable my users use the DID with their extensions |
| 13:36.36 | *** join/#asterisk grEvenX (n=even@apb91b.ip.ssc.net) |
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| 13:36.43 | [TK]D-Fender | ^shark_: this is all just dialplan. You need to arrange what you're buying from your telco |
| 13:38.22 | ^shark_ | <PROTECTED> |
| 13:38.29 | kaldemar | nasty res_timing_pthread screwed up my iax audio. |
| 13:39.40 | [TK]D-Fender | ^shark_: shathey provide DID's you decide what to DO when the call arrives against them |
| 13:40.13 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:40.48 | ^shark_ | <PROTECTED> |
| 13:41.37 | [TK]D-Fender | ^shark_: never heard of |
| 13:42.21 | ^shark_ | <PROTECTED> |
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| 14:01.33 | *** join/#asterisk thomasrr (n=scroogey@195-240-213-212.ip.telfort.nl) |
| 14:01.35 | thomasrr | hello |
| 14:01.54 | thomasrr | i am having issues with receiving calls that when the receiver hangs up the phone |
| 14:02.11 | thomasrr | the caller doesn't get disconnected |
| 14:04.49 | thomasrr | do i need to add a Hangup(); after the dial() line? |
| 14:05.15 | thomasrr | what i am doing is dialing all my hardphones |
| 14:05.35 | ^shark_ | !DID |
| 14:05.49 | ^shark_ | my apologies ^^ |
| 14:06.11 | thomasrr | like Dial(SIP/office1&SIP/office2&SIP/gigaset); |
| 14:06.17 | thomasrr | only when i pick up |
| 14:06.20 | thomasrr | and then hung up |
| 14:06.30 | thomasrr | the caller (like my cell phone) still thinks there is a connection |
| 14:06.32 | thomasrr | so doesnt hangup |
| 14:06.35 | thomasrr | how can i fix that? |
| 14:07.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:07.09 | tzafrir_laptop | ~DID NOT |
| 14:07.09 | infobot | from memory, did not is a telephony provider you do not want to have to deal with |
| 14:10.46 | *** part/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 14:10.56 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 14:11.13 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:11.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:11.19 | MaliutaLap | bed for me ... I shouldn't click while this tired |
| 14:12.23 | thomasrr | ugh |
| 14:12.30 | thomasrr | stuff works partly here now :( |
| 14:12.38 | thomasrr | i can get called but now calling outside keeps failing |
| 14:13.00 | *** join/#asterisk gregd (n=gregd@98.142.208.61) |
| 14:14.13 | gregd | hi, could someone explain me what does this error mean: http://pastebin.com/m2ee253f9 |
| 14:20.56 | *** join/#asterisk cesar_CR (n=cesar@201.195.239.11) |
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| 14:27.59 | [TK]D-Fender | gregd: Means "I should be looking at SIP debug" |
| 14:32.29 | eppigy | ZING |
| 14:33.05 | [TK]D-Fender | TRABAJO |
| 14:34.16 | eppigy | 8[] |
| 14:35.12 | jaytee | hehe |
| 14:42.54 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
| 14:43.17 | paulius | If I pay someone, could they try to get rid of my echo issues on the SPA-3102? It's driving people in my household crazy. |
| 14:45.36 | coppice | paulius: replace it with something that works |
| 14:45.53 | paulius | I've had lots of people praise that thing and say that I was the clueless one. |
| 14:46.29 | coppice | try googling for echo and spa3102 |
| 14:46.54 | paulius | done |
| 14:48.19 | [TK]D-Fender | paulius: try following the guides no voxilla like i've mentioned |
| 14:49.19 | ixremedy | anyone knows about RFC 3428 MESSAGE? |
| 14:49.20 | *** join/#asterisk gregd (n=gregd@80-41-215-234.dynamic.dsl.as9105.com) |
| 14:51.09 | [TK]D-Fender | ixremedy: * does not support SIP IM |
| 14:52.15 | paulius | I've been trying them out for a long time. |
| 14:52.28 | paulius | I don't think any of the pages mentionned changing the impedence in the regional tab |
| 14:52.32 | paulius | Doing this helped me a lot. |
| 14:52.45 | paulius | I realized that the non-regional tab impedence is the impedence to the local phone |
| 14:52.50 | paulius | and my echo is when going to telcos |
| 14:52.58 | paulius | if I call from a SIP to the phone it's perfect |
| 14:53.37 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 14:54.33 | timeshell_atwork | Happy Monday! |
| 14:58.21 | *** part/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 14:58.26 | jeeger | [TK]D-Fender: So, I finally settled on the AMI interface for my problem. Many thanks for all your pointers in the past. |
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| 15:12.37 | *** join/#asterisk omer (n=_omer@119.152.102.236) |
| 15:12.41 | omer | hello |
| 15:12.46 | *** join/#asterisk outtolunc (n=me@166.129.225.73) |
| 15:13.48 | omer | I was using Asterisk 1.2.14 ... I upgraded it to 1.4.24 and Now I am getting [May 25 10:33:05] WARNING[8996]: rtp.c:889 ast_rtcp_read: RTCP Read too short |
| 15:15.51 | ixremedy | how costly is it to add support for RFC3428 MESSAGE? |
| 15:16.18 | [TK]D-Fender | ixremedy: costly? LOL |
| 15:16.28 | ixremedy | means we are ready to invest |
| 15:17.17 | [TK]D-Fender | ixremedy: this will require plenty of work and it will get outdated fast as * evolves. this is inviting a dea-end solution. You may want to reconsider *'s role in your solution |
| 15:17.55 | omer | any help on [May 25 10:33:05] WARNING[8996]: rtp.c:889 ast_rtcp_read: RTCP Read too short ? |
| 15:18.12 | ixremedy | plenty of work - 10 000 man/hours? |
| 15:18.47 | *** part/#asterisk paulius (n=paulius@unaffiliated/paulius) |
| 15:20.28 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.1) |
| 15:20.28 | ixremedy | i think what it will really require is a code change to Asterisk-IM |
| 15:20.38 | ixremedy | which is a module |
| 15:24.44 | *** join/#asterisk atis_lap (n=atis_lap@193.238.212.171) |
| 15:26.10 | [TK]D-Fender | ixremedy: O RLY? where is this module? |
| 15:26.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:27.37 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
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| 15:28.46 | *** part/#asterisk psantos (n=psantos@213.63.47.2) |
| 15:28.49 | ixremedy | http://www.ohloh.net/p/AsteriskIM |
| 15:29.33 | ixremedy | and moreover why is it that much work to do in Asterisk to introduce SIP MESSAGE? |
| 15:32.50 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 15:33.11 | ixremedy | Does anyone want to earn a little bit on that?:) |
| 15:35.23 | ixremedy | there is a definite need to have asterisk being able to work with full profile SIP RFC 3428 messages over TCP |
| 15:42.53 | [TK]D-Fender | ixremedy: So far nothing tells me that has anything to do with * support of SIP IM |
| 15:43.05 | [TK]D-Fender | ixremedy: that talks about something else doin all the work |
| 15:43.50 | [TK]D-Fender | ixremedy: As for someone qualified to do this, try contacting OEJ ; Olle E. Johannson |
| 15:44.05 | [TK]D-Fender | ixremedy: and "earn a little bit", will be a lot actually. |
| 15:45.25 | *** join/#asterisk s14ck (n=s14ck@ccscliente154.ifxnetworks.net.ve) |
| 15:51.13 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:51.35 | ixremedy | what is the price of that "a lot"?:) |
| 15:51.58 | ixremedy | the interesting paremeters are time and money :) |
| 15:53.15 | ixremedy | what is his email address? i'll write him an email |
| 15:57.08 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 16:06.44 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 16:15.38 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-22-rbry-bb-1.dynamic.isadsl.co.za) |
| 16:17.07 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:21.08 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-198-22-rbry-bb-1.dynamic.isadsl.co.za) |
| 16:24.35 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 16:25.28 | ixremedy | sent a completed form on the digium web site regarding the voiced issue and got no reply back yet |
| 16:25.46 | ixremedy | interesting if there is an email address where i could sent my query |
| 16:28.00 | *** join/#asterisk DarkRift (n=dark@65.92.166.78) |
| 16:40.58 | *** join/#asterisk tarik (n=chatzill@41.250.213.129) |
| 16:54.41 | atis_work | ixremedy: he was here, nickname oej |
| 16:54.57 | atis_work | ixremedy: you can search he's email on asterisk-users or asterisk-dev ML |
| 16:55.15 | atis_work | ixremedy: or you can start by posting email to asterisk-users |
| 16:56.51 | atis_work | ixremedy: or just search web for edvina |
| 16:59.41 | drmessano | ixremedy: IMHO, the real answer is not to use SIP |
| 17:01.49 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 17:03.51 | drmessano | It would be nice if Asterisk had an XMPP daemon built in for messaging.. I just think SIP isn't the way to really go |
| 17:05.22 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:05.22 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:06.48 | *** join/#asterisk twanny796 (n=chatzill@195.158.124.102) |
| 17:07.45 | drmessano | kicks leifmadsen in the knee |
| 17:09.42 | leifmadsen | drmessano, don't do that. I almost punched a guy in the face at soccer last night for doing that |
| 17:10.10 | drmessano | lol |
| 17:10.35 | drmessano | I have put like 8 hours into trying to get the IMAP toolkit compiled successfully so I can play with IMAP.. I blame you |
| 17:10.59 | drmessano | So when I google search and see your name.. I throw things now |
| 17:11.44 | *** join/#asterisk jthurman42 (n=jthurman@c-67-169-218-181.hsd1.wa.comcast.net) |
| 17:13.43 | drmessano | lol |
| 17:13.57 | drmessano | throws things at leif as he's running away |
| 17:24.31 | *** join/#asterisk AlexTO (n=aacm_ale@190.25.208.136) |
| 17:24.43 | AlexTO | Hi Everyone.. |
| 17:26.22 | SlipperyChicken | good day :) |
| 17:27.08 | AlexTO | I need some help with a TDM410 |
| 17:27.18 | AlexTO | FXO and i have no audio, any ideas? |
| 17:28.52 | drmessano | No audio how? |
| 17:29.40 | rob0 | I map, you map, we all map for IMAP! |
| 17:29.42 | AlexTO | yes, no audio when i call in |
| 17:36.42 | [TK]D-Fender | drmessano: ... |
| 17:36.44 | [TK]D-Fender | ~iwmwb |
| 17:36.45 | infobot | I WANT MY WEEKEND BACK! |
| 17:37.50 | *** join/#asterisk orpheee (n=u-applet@jem75-10-88-182-221-20.fbx.proxad.net) |
| 17:43.52 | *** join/#asterisk matsk (n=matkar@c-ce8ee253.174-6-64736c10.cust.bredbandsbolaget.se) |
| 17:48.56 | *** join/#asterisk yjtmas (n=nux@host59-183-dynamic.6-79-r.retail.telecomitalia.it) |
| 17:49.20 | yjtmas | hi all |
| 17:49.21 | yjtmas | is asterisk gui svn 4797 compatible asterisk 1.6.1.0? |
| 17:50.50 | ixremedy | the thing is that i just don't want to have two separate systems for message exchange, phone calls and user voice talk |
| 17:51.02 | ixremedy | thats why i want to have just one single system that would be able handle it all |
| 17:52.07 | ixremedy | and if there is someone who wants to help getting RFC 3428/TCP in asterisk, he can send me a message to vinstance@gmail.com so that we can discuss the service and the payment options |
| 17:53.42 | ixremedy | i don't think that the RFC writers were so stupid to develop a recomendation on something that isn't gonna be used, considering that such bodies consist of people who are more often the users not the vendors |
| 17:54.18 | [TK]D-Fender | ixremedy: Who said anything about "isn't going to be used"? |
| 17:55.48 | ixremedy | i've spent about three years working at ericsson being an engineer and i have had a touch on IMS and things, thats where i've seen SIP being not just the means to transport INVITEs |
| 17:56.26 | [TK]D-Fender | ixremedy: oK, AND...? |
| 17:56.53 | *** join/#asterisk technot (n=technot@11.84-234-185.customer.lyse.net) |
| 17:57.19 | ixremedy | and if you take the SIP framework from day one you receive such goodies as MIKEY, etc. the price you're paying is the overhead and stuff, but the network capacity today isn't what it used to be |
| 17:57.28 | *** join/#asterisk PanicMan (i=Learner@122.102.33.80) |
| 17:57.38 | ixremedy | and i guess it is on the way to increase |
| 17:57.47 | PanicMan | hello, Looking for some help regarding Incoming SIP-Trunk issue |
| 17:58.01 | PanicMan | any helper care to help :( |
| 17:58.01 | technot | anyone happen to know of a SIP/AIX2 provider who still pass on the privacy bit to their costumers? appears flowroute has stopped doing it ;\ |
| 17:58.07 | technot | PanicMan: what's the problem ? |
| 17:58.40 | PanicMan | technot, i'm trying to configure a SIP trunk between VoIPSwitch & Asterisk server |
| 17:59.19 | PanicMan | its not working :( i can't see any request in my Asterisk server from VoIPSwitch |
| 17:59.34 | PanicMan | no hit @5060 UDP port |
| 17:59.49 | PanicMan | goal is to terminate calls with this Asterisk Gw |
| 17:59.55 | *** part/#asterisk ixremedy (n=ixremedy@client130-7.emplot.net.ua) |
| 18:00.07 | [TK]D-Fender | PanicMan: If no traffic is arriving at *, how is that *'s fault? |
| 18:00.07 | PanicMan | VoiPSwitch >> Asterisk Server >> SIP Client |
| 18:00.26 | PanicMan | is there any way to find out the panic ? |
| 18:00.52 | PanicMan | from that VoIPSwitch, I can use Xlite as SIP client |
| 18:01.11 | PanicMan | but need to route some IP via that VoIPswitch |
| 18:01.14 | [TK]D-Fender | PanicMan: Huh? |
| 18:01.37 | [TK]D-Fender | PanicMan: Again, you say traffic isn't arriving at your * server. What do yuo expect us to tell you about yuor soution? |
| 18:02.40 | PanicMan | if i use H323, i can see hit in Asterisk CLI |
| 18:03.05 | PanicMan | for trunk, Asterisk use 5060 UDP, right ? |
| 18:05.40 | jthurman42 | PanicMan: how are you checking for incoming traffic? (tcpdump?) |
| 18:09.52 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 18:10.09 | [TK]D-Fender | chan unload headlesschicken.so |
| 18:10.58 | *** join/#asterisk SebastianS (n=schu@adsl-dyn16.78-98-183.t-com.sk) |
| 18:13.30 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:38.31 | *** join/#asterisk KucukMubasir (n=Halliday@unaffiliated/kucukmubasir) |
| 18:45.29 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:45.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:50.38 | *** join/#asterisk stijnbe (n=stijnbe@78-21-61-204.access.telenet.be) |
| 18:51.56 | Aiatek | hi, i want to install dahdi and asterisk 1.6 without internet access |
| 18:52.04 | Aiatek | someone told me about tarball |
| 18:52.19 | Aiatek | how do i download the rights packages |
| 18:52.24 | Aiatek | ? |
| 18:56.39 | rob0 | Hard to download anything without Internet access. :) But perhaps your question will be answered by visiting the "Download" link at asterisk.org. |
| 18:57.31 | Aiatek | i have the package you smart boy but it wont install without internet access |
| 18:58.29 | Aiatek | <rob0> have you tried to install dahdi without internet access? |
| 18:58.29 | kaldemar | you don't need internet access to install it. |
| 18:58.38 | rob0 | Ignored. |
| 18:58.41 | Aiatek | the installation fails |
| 18:58.57 | Aiatek | make install try to download some packages from digium |
| 18:59.24 | rob0 | I have no need to put up with rude idiots who don't know how to properly phrase a question in IRC. |
| 18:59.45 | Aiatek | <rob0> im from dominican republic |
| 18:59.51 | Aiatek | im trying my best |
| 19:00.02 | Aiatek | and maybe your are the idiot |
| 19:00.45 | jthurman42 | Aiatek: I believe that by default the sound files are downloaded sometime during the build/install process of Asterisk 1.6 |
| 19:01.02 | Aiatek | ok |
| 19:01.25 | jthurman42 | You could try unchecking the core/extra sound packages, but I have not tried that |
| 19:01.39 | jthurman42 | and the MoH too |
| 19:01.43 | Aiatek | <jthurman42> thanks |
| 19:03.43 | jthurman42 | Aiatek: look at asterisk/sounds/Makefile it checks to see if the tiles exists in the sounds directory, and downloads them if they are not. |
| 19:04.04 | jthurman42 | You could download the ones you want and place them there so they still install. |
| 19:04.14 | jthurman42 | (because you will need sound files...) |
| 19:04.21 | Aiatek | hehehehehhee |
| 19:04.26 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 19:08.19 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 19:10.08 | *** part/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 19:10.43 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:10.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:12.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:15.52 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 19:18.33 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:20.50 | *** join/#asterisk \void\ (n=void@sub227-61.elpos.net) |
| 19:22.39 | *** join/#asterisk jthurman42 (n=jon@c-67-169-218-181.hsd1.wa.comcast.net) |
| 19:25.51 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:31.57 | Kevin` | can anyone recommend a softphone for osx? |
| 19:38.48 | wierdo | Kevin`, Xlite |
| 19:44.44 | *** join/#asterisk Kernel_Core (n=I@193.222.108.62) |
| 19:44.47 | Kernel_Core | hi all |
| 19:47.03 | Kernel_Core | I am running 1.4.24.1 ad dahdi latest version and I have te110p card , and configured it for E1 , the physical layer is up , but |
| 19:47.08 | Kernel_Core | when I issue : |
| 19:47.09 | Kernel_Core | X*CLI> pri show span 1 |
| 19:47.09 | Kernel_Core | Primary D-channel: 16 |
| 19:47.09 | Kernel_Core | Status: Provisioned, Down, Active |
| 19:47.10 | Kernel_Core | Switchtype: EuroISDN |
| 19:47.27 | Kernel_Core | it seems the signalling channel is down ! what can be wrong ?! |
| 19:50.19 | *** join/#asterisk yjtmas (n=nux@host59-183-dynamic.6-79-r.retail.telecomitalia.it) |
| 19:51.07 | *** join/#asterisk VoipForces (n=kvirc@199.243.95.42) |
| 19:51.35 | VoipForces | I have a call drop issue. I see "rtp.c: Don't know how to represent 'f'" then the call hangup. Any ideas? |
| 19:51.45 | VoipForces | SIP extensions with 2 PRI... |
| 20:07.36 | *** join/#asterisk orpheee (n=u-applet@jem75-10-88-182-221-20.fbx.proxad.net) |
| 20:14.15 | orpheee | hello, i've a problem with Meetme.conf i need to install package with it ? |
| 20:14.28 | *** join/#asterisk CapriCoRN^80 (n=int@207.176.6.77) |
| 20:15.04 | CapriCoRN^80 | hi ! my MV 370 gsm gateway un register from asterisk after few hours |
| 20:15.14 | kaldemar | orpheee: what problem? |
| 20:15.23 | CapriCoRN^80 | what can be the problem ? |
| 20:20.15 | [TK]D-Fender | CapriCoRN^80: Absolutely anything. |
| 20:21.03 | jaytee | anything from a busted franistat to a blown flux capacitor |
| 20:22.16 | [TK]D-Fender | stocks up on plasma relays and ODN junctions. |
| 20:22.39 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 20:22.52 | jaytee | don't forget the isolinear chips, they're bound to go up in price since we're in a recession |
| 20:23.16 | CapriCoRN^80 | [TK]D-Fender: well it worked fine and suddenly it is not register to asterisk. other sip users are working fine |
| 20:24.03 | CapriCoRN^80 | i guess its problem from gsm device |
| 20:24.22 | *** join/#asterisk jmacz (n=jmacz@201.244.168.237) |
| 20:25.36 | orpheee | well you can see the problem meetme there http://pastebin.com/d13d23a43 |
| 20:25.49 | carrar | jaytee, isolinear arent used much anymore, the new multiflexiso chips are replacing them |
| 20:26.04 | orpheee | maybe i need to install other thing to have "meeting" |
| 20:26.22 | jaytee | carrar, so much for using them with the modems in my garage to make an FXO channel bank :-) |
| 20:26.25 | carrar | It's usually the "other thing" |
| 20:26.44 | orpheee | but what thing.. |
| 20:26.52 | carrar | well isolinear chips work excellent with old sound cards |
| 20:26.54 | [TK]D-Fender | orpheee: [May 10 06:28:31] WARNING[5611]: pbx.c:1833 pbx_extension_helper: No application 'MeetMe' for extension (internal, 500, 2) <- you probably didn't compile and setup zaptel/DAHDI, which is required |
| 20:27.07 | orpheee | ok i try |
| 20:27.27 | jaytee | "There is no try, there is only do!" - Yoda |
| 20:27.36 | asteriskmonkey | is there an option in quese for call accept confirmation? |
| 20:27.50 | carrar | quese rock |
| 20:28.50 | carrar | is that spanish? |
| 20:29.10 | asteriskmonkey | queues |
| 20:29.55 | asteriskmonkey | [TK]D-Fender: still trying to get an M option to behave properly in dial, there has to be a better way... why the heck did digium drop the c option |
| 20:30.51 | [TK]D-Fender | asteriskmonkey: No idea, and don't care so much... work with what you've got |
| 20:31.09 | carrar | progess is what it's called |
| 20:31.20 | file | wonders what the 'c' option is |
| 20:31.25 | file | or was |
| 20:31.33 | asteriskmonkey | ie.. Zap/g1c/... |
| 20:31.46 | jaytee | wtf? |
| 20:31.52 | asteriskmonkey | would prompt for a # to accept message to complete the joining of the call |
| 20:31.52 | carrar | ftw |
| 20:32.27 | jaytee | I nevah hoid a such a ting! |
| 20:32.28 | carrar | Sounds like something from the cold war area |
| 20:32.49 | file | I have never heard of such a thing, and it would have been an option to Dial - not part of the technology and extension |
| 20:33.03 | jaytee | yeah, what he said! |
| 20:33.29 | carrar | asteriskmonkey, there are new better ways to do everything |
| 20:33.31 | jaytee | someone's been doing purp by the pound and it ain't me! |
| 20:33.52 | asteriskmonkey | im going from 1.2 to 1.6 so be gentle :) |
| 20:34.08 | carrar | YOU BET |
| 20:34.13 | file | I just looked in 1.2, it doesn't exist there |
| 20:34.17 | file | ^-^ |
| 20:35.29 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 20:35.37 | asteriskmonkey | let me englighten you in the old school > http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels |
| 20:35.45 | asteriskmonkey | read enjoy that c option :) |
| 20:36.18 | carrar | holy cow thats lame |
| 20:36.33 | file | bizarre |
| 20:36.46 | file | I just looked in 1.6.0, that is still there |
| 20:37.23 | asteriskmonkey | its not working in 1.6 |
| 20:37.46 | file | it's best to say the full version when talking about 1.6 |
| 20:38.05 | asteriskmonkey | Asterisk SVN-branch-1.6.2-r195590 |
| 20:38.10 | asteriskmonkey | to be more specific |
| 20:38.25 | file | it is also still there, nothing has been done on purpose to remove/break it |
| 20:38.35 | asteriskmonkey | mmmm |
| 20:38.42 | asteriskmonkey | ill retest |
| 20:38.52 | carrar | curious what you use it for |
| 20:39.00 | carrar | in what circumstance |
| 20:39.26 | asteriskmonkey | cell phones, you dial a bunch of people and you dont want to run the risk of hitting cell phone voicemail and still have the oppertinuity to get one of the others in the ring group |
| 20:39.37 | asteriskmonkey | if you know a better way :) im all eyes for learning the better way |
| 20:39.59 | CapriCoRN^80 | when i do the default settings of Mv 370 Gsm device and enter sip details .. it worked fine |
| 20:40.34 | asteriskmonkey | file: not working, dosnt prompt for nothing just bridges the call |
| 20:41.28 | asteriskmonkey | i think you just grepping as there is a c option in the last bit.. |
| 20:41.39 | file | I'm reading the code |
| 20:41.50 | asteriskmonkey | thanks |
| 20:42.21 | file | but *shrug* without labbing it up and trying it, can't say anything more |
| 20:43.49 | asteriskmonkey | file do you know another method for accomplishing the rining of multiple cellular devices and avoiding hitting vm/out of area messages and only bridging the call to the confirmed reciever? |
| 20:44.14 | file | Local channels and Dial with the macro execution option |
| 20:45.09 | asteriskmonkey | ive been trying that, its not behaving the same, also the other channels will still get the same macro played over even if the first on has acceped correct? |
| 20:45.38 | file | the others would get hung up |
| 20:45.43 | asteriskmonkey | k |
| 20:46.30 | asteriskmonkey | what result would you set if no confirmation? |
| 20:46.33 | carrar | asteriskmonkey, have you see followme.conf? |
| 20:46.41 | carrar | seen |
| 20:46.41 | file | I do not know. |
| 20:46.48 | asteriskmonkey | yes not working how i expect |
| 20:47.31 | asteriskmonkey | my issue is with the macro option if there is multiple people in a dialgorup there is no clean option to set for result status without accepting the call all others result in the call jumping or getting congestion etc.. major problem |
| 20:48.15 | file | that's why I said use a Local channel in front |
| 20:48.17 | asteriskmonkey | Im thinking queues might be the next best step to look at if queues had a confirmation before connection method.. perhaps queues coupled with the dial macro might be the only solutiopn |
| 20:48.52 | file | the Local channel wouldn't cause all the other calls to stop trying |
| 20:48.53 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
| 20:48.59 | wonderworld | hey |
| 20:49.09 | asteriskmonkey | file: but if your using local, say a user rejects it will still hit congestion and drop the others in the Local/ |
| 20:49.17 | asteriskmonkey | oh? |
| 20:49.19 | asteriskmonkey | really? |
| 20:49.38 | file | Dial(Local/100@need-confirm&Local/200@need-confirm) |
| 20:49.54 | file | inside of the need-confirm context you dial out using dial with the macro option |
| 20:50.04 | asteriskmonkey | yes but what result are you setting for the unanswerd extensions |
| 20:50.10 | carrar | or followme |
| 20:50.24 | file | I believe it is documented what to set |
| 20:50.42 | file | CONGESTION or BUSY |
| 20:50.44 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:50.55 | asteriskmonkey | ok thanks file, ill try it that way |
| 20:51.01 | file | but because you fronted that individual dialing attempt with Local, the others won't be stopped |
| 20:51.52 | asteriskmonkey | nice thats good to know.. thanks for that.. also if you or anyone you know at digium could look at that dahdi channel c option that would be great, i know of a ton of people that still use that.. or atleast comment in rel notes its going away |
| 20:52.22 | file | if you want it looked at please file an issue report at http://issues.asterisk.org/ |
| 20:52.23 | jaytee | how relevant is that WIKI article? I'd never seen half those options for zap |
| 20:52.43 | asteriskmonkey | thanks |
| 20:52.48 | file | Digium isn't the only one who looks at things. |
| 20:52.55 | file | (or does things) |
| 20:53.10 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
| 20:53.20 | jaytee | no, but Digium rocks! even more than quese! |
| 20:54.08 | carrar | quese, thats spanish for DIY FXO's |
| 20:54.35 | rob0 | Hmmm, there's a marketing idea ... they could sell Digium Rocks ! |
| 20:56.38 | jaytee | I just wish the color would change from that obnoxious orange to something a little less seizure inducing while at the same time not heading in a nauseous green shade like an * derivative whose name sounds like a breakfast cereal rabbits desperately try to steal. |
| 21:03.28 | *** join/#asterisk matt_keys (n=matt_key@208.71.232.42) |
| 21:04.56 | matt_keys | Does anyone know how to configure the second NIC on a AudioCodes Mediant 1000? I clicked Configuration, Network settings, IP settings, then added one and told it the address, prefix, gateway, vlanid, and name. I hit submit and the status page shows it green, but it's not pinging |
| 21:05.18 | carrar | Thankyou for joining AudioCodes support channel |
| 21:05.49 | matt_keys | if they had a channel, well hell... even a decent support page, i probably wouldn't be here asking |
| 21:05.49 | Micc | anyone know why when I load dahdi_dummy it doesn't create the /dev/dahdi/ devices? |
| 21:07.37 | Micc | does my kernel need rebuilt to use dahdi_dummy? |
| 21:07.55 | carrar | Micc, did you read the docs in the dahdi-tools driver package? |
| 21:08.17 | Micc | carrar, yes, I've done everything it says. |
| 21:08.42 | Micc | carrar, its also very confusing document. It doesn't say anything about how the /dev stuff is created. |
| 21:09.07 | Micc | carrar, I remember fixing this on another server with udev. |
| 21:10.04 | Micc | The docs for zaptel and dahdi are about the same. They both leave out the part I always need. |
| 21:16.05 | carrar | might look in daddi-linux |
| 21:16.21 | Micc | I'm not sure I gave dahdi the proper kernel source when it built. I went through all the steps to get the source from centos but then running make in dahdi-linux it says You do not appear to have the sources for the 2.6.18-128.1.6.el5PAE kernel installed. |
| 21:16.36 | Micc | But I did everything right to install the proper sources. |
| 21:16.40 | Micc | And put them in the right place. |
| 21:16.59 | Micc | I tweaked the makefile or something to get it to build. |
| 21:17.15 | Micc | I think it was just a naming thing. because of the PAE |
| 21:17.43 | carrar | make install-devices? |
| 21:18.51 | Micc | hmm... |
| 21:19.09 | carrar | (dahdi-linux) |
| 21:19.17 | Micc | yeah. |
| 21:19.39 | Micc | do I need to reload the module now after I've done that? |
| 21:19.43 | Micc | Maybe I missed that step before. |
| 21:20.11 | Micc | looks like it installed something new in /etc/udev/rules.d/ |
| 21:20.19 | Micc | So would I need to reload udev? |
| 21:21.16 | carrar | You don't have a /dev/dadhi |
| 21:21.24 | Micc | no. |
| 21:22.15 | carrar | did you do make install |
| 21:22.22 | Micc | of course |
| 21:22.48 | carrar | did you read the output |
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| 21:23.49 | Micc | I'm sure I did, but I'll do it again. |
| 21:24.18 | Micc | Now I remember what I had to do. I had to define KVERS to point to my kernel |
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| 21:25.13 | jthurman42 | Micc: Have you installed a newer kernel (+devel, headers) than the one you are currently building with? |
| 21:26.48 | Micc | I installed some rpms of kernel sources. |
| 21:26.58 | Micc | I haven't reboot since though. |
| 21:27.14 | Micc | Maybe I should check my boot drive to make sure its still the same kernel. |
| 21:27.51 | jthurman42 | Check if the version of the kernel you are running, and the kernel-headers, kernel-devel are the same. |
| 21:28.14 | Micc | jthurman42. they are. |
| 21:28.52 | jthurman42 | is dahdi_dummy loaded? (lsmod) |
| 21:29.15 | Micc | dummy is, but not dahdi_dummy |
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| 21:30.36 | Micc | the kernel source version I have is 2.6.18-128.1.6.el5-i686 |
| 21:30.51 | Micc | but uname -r shows 2.6.18-128.1.6.el5PAE |
| 21:31.08 | Micc | When I went through all the steps to get my kernel from centos, thats what it gave me. |
| 21:32.24 | jthurman42 | CentOS 5? |
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| 21:34.22 | Micc | yes |
| 21:34.32 | jthurman42 | I don't have any i386 test machines to test for PAE. (all 64bit) |
| 21:34.47 | jthurman42 | do you have kernel-PAE-devel installed? |
| 21:35.12 | jthurman42 | do an: rpm -qa | grep kernel |
| 21:37.31 | jthurman42 | You should have kernel-PAE-2.6.18-128.1.6, kernel-headers-2.6.18-128.1.6, kernel-PAE-devel-2.6.18-128.1.6 |
| 21:37.58 | Micc | kernel-headers-2.6.18-128.1.6.el5 |
| 21:37.59 | Micc | kernel-devel-2.6.18-128.1.6.el5 |
| 21:37.59 | Micc | kernel-PAE-2.6.18-128.1.6.el5 |
| 21:37.59 | Micc | kernel-PAE-2.6.18-128.el5 |
| 21:37.59 | Micc | kernel-PAE-devel-2.6.18-128.1.10.el5 |
| 21:37.59 | Micc | kernel-headers-2.6.18-128.1.6.el5 |
| 21:38.01 | Micc | kernel-devel-2.6.18-128.1.6.el5 |
| 21:38.03 | Micc | kernel-PAE-2.6.18-128.1.6.el5 |
| 21:38.07 | Micc | kernel-PAE-2.6.18-128.el5 |
| 21:38.07 | Micc | kernel-PAE-devel-2.6.18-128.1.10.el5 |
| 21:38.09 | Micc | doh! |
| 21:38.16 | mmlj4 | thanks for pasting |
| 21:38.23 | Micc | I didn't realize that was in the input buffer. |
| 21:38.25 | Micc | http://www.pastebin.ca/1434349 |
| 21:38.30 | Micc | I used pastebin. |
| 21:39.58 | jthurman42 | So, kernel-PAE-devel is a different version than the kernel you are running. (128.1.10, not 128.1.6) |
| 21:40.12 | Micc | yeah |
| 21:40.15 | Micc | I just noticed that. |
| 21:40.22 | jthurman42 | :) |
| 21:40.22 | Micc | I'm removing that one and installing the right one now. |
| 21:41.32 | Micc | ok, now we're cookin. |
| 21:41.50 | Micc | what a pain in the ass. I've been trying to get the right kernel source for weeks. |
| 21:42.15 | Micc | it was much easier than the pages of stuff on centos you have to do to get your kernel source. |
| 21:42.33 | jthurman42 | ? |
| 21:42.38 | Micc | maybe it wasn't in yum when I tried it before. I swear I tried that before. |
| 21:42.59 | jthurman42 | yum update kernel-PAE-devel kernel-headers kernel-PAE (reboot) |
| 21:43.08 | Micc | I just needed to yum install kernel-PAE-devel-2.6.18-128.1.6.el5 |
| 21:43.29 | jthurman42 | (until you update to kernel 128.1.10 :)) |
| 21:43.47 | Micc | when I did this a few weeks ago it said the package wasn't found. |
| 21:44.08 | Micc | is 128.1.10 just bug fix release? |
| 21:44.13 | jthurman42 | that happens depending on when it came out, what mirror... etc |
| 21:44.33 | jthurman42 | Haven't looked at the changes |
| 21:44.44 | Micc | I guess the early bird gets the problems. |
| 21:45.52 | jthurman42 | wouldn't say early: 128.1.10 has been out since May 7th |
| 21:47.09 | jthurman42 | (if you want to know the changes: http://www.pastebin.ca/1434362 ) |
| 21:47.24 | Micc | Damnit, Jim! I'm a programmer, not an admin. |
| 21:48.15 | Micc | look like I need to do it. |
| 21:49.37 | Micc | yeah! /dev/dahdi is alive! |
| 21:49.50 | jthurman42 | :) |
| 21:50.31 | Micc | I must say, administrating linux has come a long way since 0.98-pl12 when I first installed it on my 386 when I was 13. |
| 21:50.48 | Micc | Its gotten a lot easier. |
| 21:58.43 | Micc | jthurman42, thanks for your help. |
| 21:59.00 | jthurman42 | No problem, good luck! |
| 22:00.50 | Micc | jthurman42, when I ran that update for the kernel, will it run grub as part of the install? |
| 22:00.57 | Micc | I noticed it changed the grub.conf |
| 22:01.25 | Micc | does it need to update the boot sector or is that not necessary anymore? |
| 22:01.27 | jthurman42 | Yeah, it will, and it should only keep the last 2 kernels around |
| 22:01.33 | jthurman42 | nope. |
| 22:02.46 | Micc | ok |
| 22:03.37 | Micc | jthurman42, have you ever setup asterisk to use a T38 provider to send faxes? |
| 22:03.47 | Micc | Thats the last problem I'm having. |
| 22:04.08 | jthurman42 | Sorry, not T38 experience. yet... |
| 22:04.16 | jthurman42 | er.. no T38... |
| 22:04.22 | Micc | actually, I have another problem with asterisk-1.6.1.0 and the multiple parking lots. |
| 22:04.40 | Micc | It seems to be a bit of a niche |
| 22:05.21 | jthurman42 | Search through the asterisk-users list. T38 has been somewhat of a hot topic lately |
| 22:21.19 | Juggie | if VoiceMailMain() is called w/o a mailbox does it check the mailbox= parm on a given channel if set to see what mailbox to use? |
| 22:22.25 | Micc | Juggie, try it. |
| 22:22.42 | Micc | I'm curious to know the answer to that one myself. |
| 22:22.49 | Micc | It seems like it should. |
| 22:22.55 | Juggie | i tried and it did not seem to |
| 22:23.02 | Juggie | though i used an mailbox w/ alpha |
| 22:23.10 | Juggie | not that it should matter |
| 22:23.51 | Micc | the mailbox param in the channel config is for message waiting indicator I think. |
| 22:24.00 | Juggie | ya |
| 22:24.02 | Micc | maybe theres another parameter for voicemail |
| 22:24.11 | Juggie | seems silly that it cant try to use that for the mailbox though |
| 22:24.23 | Juggie | if VoiceMailMain() is called w/o a parm |
| 22:25.25 | jthurman42 | VoicemailMain(${CALLERID(num)}) |
| 22:26.20 | jthurman42 | Assuming that your extensions are your mailboxes. Haven't tried with alpha |
| 22:28.23 | Juggie | ya, i know that works, was just wondering, that model will break with multiple vm contexts etc |
| 22:29.13 | Juggie | you could just put a setvar in the channel config |
| 22:29.20 | Juggie | at least chan_sip supports that |
| 22:29.22 | Juggie | not sure what other channels |
| 22:29.31 | Juggie | but it seems a little redundant given its already there |
| 22:35.50 | jthurman42 | juggie: are your calls to voicemailmain in different dialplan contexts for different vm contexts? |
| 22:38.27 | NickRios05 | Am sorry, i have a little problem I cant seem to get calls out on asterisk |
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| 22:38.42 | NickRios05 | this is the output on CLI i get |
| 22:38.45 | NickRios05 | <PROTECTED> |
| 22:38.45 | NickRios05 | <PROTECTED> |
| 22:38.45 | NickRios05 | <PROTECTED> |
| 22:38.45 | NickRios05 | <PROTECTED> |
| 22:38.45 | NickRios05 | <PROTECTED> |
| 22:38.46 | NickRios05 | <PROTECTED> |
| 22:38.48 | NickRios05 | <PROTECTED> |
| 22:38.50 | NickRios05 | <PROTECTED> |
| 22:38.52 | NickRios05 | <PROTECTED> |
| 22:38.56 | NickRios05 | <PROTECTED> |
| 22:38.58 | NickRios05 | <PROTECTED> |
| 22:39.00 | NickRios05 | <PROTECTED> |
| 22:39.02 | NickRios05 | <PROTECTED> |
| 22:39.02 | xanderp | having problems with swift: |
| 22:39.04 | NickRios05 | <PROTECTED> |
| 22:39.06 | NickRios05 | <PROTECTED> |
| 22:39.08 | NickRios05 | <PROTECTED> |
| 22:39.10 | _ShrikE | ! |
| 22:39.10 | NickRios05 | <PROTECTED> |
| 22:39.12 | NickRios05 | <PROTECTED> |
| 22:39.27 | xanderp | NickRios05, how about using pastebin for that much text please? |
| 22:39.41 | _ShrikE | ~pb |
| 22:39.42 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 22:40.25 | NickRios05 | am sorry |
| 22:40.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 22:41.04 | xanderp | I'm having problems with voice selection for swift. exten => s,n,Swift(text) works with the correct voice, but exten => s,n,Playback(${TEMPWAV}) uses the wrong voice. I've changed it in swift.conf, PHPAGI, and environment variables... suggestions? |
| 22:42.59 | [TK]D-Fender | xanderp: Her's a suggestion, thats a playback of some random VARIABLE whose contents and origins are completely absent. |
| 22:44.02 | xanderp | [TK]D-Fender, the app that is generating it is the weatherzip nerdvittles app... but I get it with anything that uses the Playback command... |
| 22:44.10 | xanderp | I will test again to make sure. |
| 22:44.20 | [TK]D-Fender | xanderp: that playback means exactly NOTHING |
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| 22:44.33 | xanderp | ? |
| 22:44.55 | [TK]D-Fender | xanderp: What sets that variable? How? Why? Where is the option for the voice? |
| 22:45.06 | [TK]D-Fender | xanderp: that 1 line sets absolutely nothing |
| 22:45.18 | [TK]D-Fender | xanderp: And is something that happens after a ton of OTHER stuff |
| 22:45.24 | [TK]D-Fender | xanderp: once its too late |
| 22:46.23 | NickRios05 | am sorry any thoughs on this one http://pastebin.com/m63d8ee36 |
| 22:48.01 | xanderp | If I exten => s,1,Swift(Hello) it uses the default voice of the system. If I exten => s,1,Playback({Variable}) does it not use the defaults? |
| 22:49.07 | [TK]D-Fender | ricko73: @SIPtrunk <- looks redundent since you are already specifying "trunk_1" before as the alternat syntax |
| 22:49.28 | xanderp | [TK]D-Fender, I get you, I need to find where in the app the wav file is being created yes? |
| 22:49.29 | [TK]D-Fender | xanderp: defaults? Thats a stupid VARIABLE! Who the hell knows what's IN IT? |
| 22:49.50 | [TK]D-Fender | xanderp: And in the end it still only points to a file that was built by SOMETHING ELSE |
| 22:49.56 | xanderp | [TK]D-Fender, I'm asking questions to learn, please don't yell at me. |
| 22:50.34 | xanderp | So I need to dig into the application and see where they are generating the wav file. |
| 22:50.45 | [TK]D-Fender | xanderp: more keeping focus on the important part. when I go all-caps and start swearing, then you'll know :) |
| 22:50.48 | NickRios05 | so should i change that in extensions.conf?? |
| 22:50.56 | [TK]D-Fender | NickRios05: Clearly |
| 22:51.08 | NickRios05 | ok |
| 22:51.12 | NickRios05 | thnx let me try |
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| 22:51.52 | xanderp | the wav file is being generated by the nerdvittles weatherzip application... I bet they've hard coded a voice in there someplace. off to scan bad code... |
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| 23:13.16 | NickRios05 | thanks [TK]D-Fender its working now |
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