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| 00:32.11 | nny | hi. I am working with '"sip show channel stats" and seeing some odd things. http://pastebin.com/AxdFgUF7 are two back to back queries on the channel. (within 2 seconds). The Recv/Lost packet count is way off. Is there something I can do to correct this? |
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| 01:40.55 | volga629 | Is res ldap support encrypted connection to ldap server, just can't find some info about |
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| 02:57.23 | nickgaw | Hi, I am trying to setup my asterisk system here at home so when my extension is called as I have a sip extention elsewhere that when my local asterisk system receives a call my local system will answer the call and transfer it to the extention 1000 or what ever is the quickest way to send the caller directly to the demo process? I am totally blind and trying to read threw these very large configuration files is kind of over w |
| 02:57.59 | nickgaw | I have entered the sip information in sip.conf but not sure what to put in extensions.conf |
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| 03:46.04 | dandate2 | so my DID provider will send calls as g729 to my pbx, who will then talk to a softphone using g729. do i need 2 licenses or 1 for this? |
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| 03:48.32 | dandate2 | i know thats 1 concurrent call but apparantly 2 channels |
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| 03:55.59 | drmessano | dandate2, that's passthru.. doesn't use a license. Licenses are for transcoding |
| 03:56.39 | dandate2 | interesting, so if i have my DID provider send all calls as g729 I don't need any licenses at all? |
| 03:56.56 | drmessano | If you don't need to ever transcode.. no.. |
| 03:57.49 | dandate2 | do you think my DID gateway would have better hardware than my pbx and better results? |
| 03:58.16 | drmessano | huh? |
| 03:58.35 | dandate2 | proprietary equipment for compression |
| 03:59.01 | drmessano | Not sure how that fits into the equation here |
| 03:59.44 | dandate2 | i'm thinking mabye the gateway has faster computers and can mabye create less delay |
| 04:00.41 | drmessano | Less delay than what.. ? You're not speaking in complete sentences |
| 04:01.13 | dandate2 | oh than if the gateway sends all calls as ulaw and the compression is done on my pbx |
| 04:01.53 | drmessano | Not sure why you would want to do that.. if you're going to use G729 on the endpoints, why force transcoding? |
| 04:02.20 | dandate2 | got it |
| 04:03.15 | drmessano | G711 to the provider from the PBX and then G729 to the endpoints means your calls are still going to be limited to the most compressed part of the link.. so no point |
| 04:04.12 | drmessano | Different if you had maybe a few mobile endpoints and WANTED g711 quality for the other 98% of the calls.. but I assume you're going to use G729 on the ALL the endpoints, as you've expressed before |
| 04:05.30 | dandate2 | it does make the music on hold sound pretty ghetto heh |
| 04:05.40 | drmessano | Yes it does |
| 04:06.14 | drmessano | I force everything to G729 here.. have a lot of mobile endpoints. Its a tradeoff.. |
| 04:06.36 | drmessano | But I think I only have 4 G729 licenses, and they are rarely needed |
| 04:07.42 | dandate2 | with the music kids listen to these days they might just appreciate the degraded moh quality |
| 04:07.45 | drmessano | dandate2, this may help ----> http://www.voip-info.org/wiki/view/Asterisk+G.729+pass-thru |
| 04:08.27 | drmessano | Couple good tips on there, especially for G729 voicemail |
| 04:10.20 | drmessano | My bollywood streaming moh doesn't sound half bad transcoded |
| 04:10.49 | drmessano | Though I only use it for my home automation menus, and I dont pay much attention |
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| 04:48.25 | volga629 | sorry just DSL jumps |
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| 05:56.18 | mealstrom | good morning |
| 05:57.58 | CommaCrazy | :( morning |
| 05:58.26 | CommaCrazy | wish I haven't woken up yet |
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| 07:27.13 | pokus | hi all...i have asterisk in two subnet. e.g. user 1 is connected from 192.168.10.10 and user two is connected from 192.168.20.10...asterisk machine have ip 192.168.10.2 and alias 192.168.20.2....how is simplest solution for make call between user 1 and user 2 through asterisk (when canreinvite=no -> all media go through asterisk) i not found any solution when one server (asterisk) know work with two interfaces |
| 07:28.07 | pokus | only create two servers (one asterisk for subnet 20.10 and second for 10.10 ) and then marge them together |
| 07:28.34 | jacc0 | it should work |
| 07:28.46 | pokus | yes but exist some other solution? |
| 07:28.52 | bulkorok | one server should handle that too |
| 07:28.54 | jacc0 | solution for what? |
| 07:29.08 | pokus | for this proiblem...two subnet one asterisk |
| 07:29.23 | jacc0 | it is no problem; it should work |
| 07:29.26 | pokus | bulkorok: and how? |
| 07:29.47 | pokus | jacc0: yes but maintaint two server is little more difficult |
| 07:29.57 | jacc0 | You need only one |
| 07:30.05 | jacc0 | it should work with only one server\ |
| 07:30.10 | bulkorok | in sip.conf there is udpbindaddr |
| 07:30.19 | pokus | and how...can me give hint ? |
| 07:30.20 | jacc0 | make it 0.0.0.0 |
| 07:30.29 | jacc0 | <PROTECTED> |
| 07:30.31 | bulkorok | set this to 0.0.0.0 and asterisk listens to every ip of that host |
| 07:30.42 | jacc0 | there is nothing more to it |
| 07:31.00 | bulkorok | it's commented in the sample sip.conf |
| 07:31.05 | jacc0 | do you see incomming calles from both subnets in asterisk? |
| 07:31.10 | jacc0 | *calls |
| 07:31.50 | jacc0 | what asterisk version are you using? |
| 07:32.30 | jacc0 | becasue 'canreinvite' is renamed to 'directmedia' |
| 07:33.16 | pokus | jacc0: but i have no provlem for listening...i have probem with subnet IP...asterisk use default interface 10.2. when user want register to asterisk it works because asterisk send back IP 10.2 what is right but when user 2 want register or call to asterisk...asterisk send back ip (in sip header) 10.2 not 20.2 |
| 07:35.02 | pokus | it's trixbox Asterisk 1.6.0.26-FONCORE-r78 |
| 07:35.13 | jacc0 | then you are in the wrong channel for asking |
| 07:35.27 | pokus | nut its not trixbox problem |
| 07:35.45 | pokus | its asterisk |
| 07:35.58 | jacc0 | <PROTECTED> |
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| 07:39.12 | jacc0 | how can I convert byte to char in dialplan? |
| 07:40.28 | pokus | okdifferent..is possible set peer groubs which will be use some IP? i think this: when user1 want make call or register..asterisk will be work with him with IP1 address i mean to sip header will be set IP1 or in SDP for rtp will be set IP1 too.. when user 2 want do same asterisk will be use IP2 in SIP or SDP protocol |
| 07:41.06 | jacc0 | trixbox is not asterisk |
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| 07:42.18 | kaldemar | pokus: you do not want to use a single peer for more than one client. when the peer gets a call, only one will get it. |
| 07:42.25 | elliot98 | on the ReceiveFAX application, is there any way to force T.38? |
| 07:46.59 | pokus | kaldemar: yes i have only one client per peer....i just want use another server IP for peer. my server have 2 diferent IP and i want select IP which should be used by peer |
| 07:48.49 | pokus | jacc0: ok forget for trixobx..we can talk only about asterisk |
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| 07:52.38 | bitwize | pokus: host=[ip]? |
| 07:53.03 | pokus | yes |
| 07:53.48 | jacc0 | you should realy join #trixbox |
| 07:53.52 | bitwize | Have you defined that attribute in your peer configuration? |
| 07:53.54 | jacc0 | and ask there |
| 07:55.02 | pokus | bitwize: how attribute? |
| 07:55.45 | pokus | jacc0: whan i use asterisk it will be same problem |
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| 07:56.48 | bitwize | you said "i want select IP which should be used by peer", you can do that by specifying the ip in your peer config... |
| 07:58.51 | pokus | bitwize: we dont understand...i dont want set IP for peer...i wnt use different asterisk ip for peer |
| 07:58.51 | kaldemar | pokus: you can't select it. OS takes care of that. |
| 07:59.07 | pokus | kaldemar: yes i know asterisk use defualt interface |
| 07:59.59 | pokus | i just want ask how can i solve this problem |
| 08:00.16 | pokus | asterisk on two subnet |
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| 08:00.49 | jacc0 | he is using trixbox |
| 08:01.05 | pokus | we can talk about pure asterisk |
| 08:01.17 | jacc0 | okay, you need to do nothing; it works |
| 08:01.44 | jacc0 | just set udpbindaddr=0.0.0.0 in your /etc/asterisk/sip.conf |
| 08:01.56 | jacc0 | and it will work |
| 08:03.10 | jacc0 | i've done this a lots of times |
| 08:03.15 | jacc0 | it always works |
| 08:04.24 | jacc0 | nothing else is needed |
| 08:05.33 | pokus | peer(192.168.10.10) <----> ( 192.168.10.2) asterisk (192.168.20.2) <---->peer(192.168.20.10) ..this is my situation....BUT when you do this in SDP and SIP messages you will wee this: |
| 08:05.47 | pokus | peer(192.168.10.10) <----> ( 192.168.10.2) asterisk (192.168.10.2) <---->peer(192.168.20.10) |
| 08:06.16 | jacc0 | in trixbox maybe.........did you try asterisk? |
| 08:08.32 | pokus | i will and result will bew same because trixbox use asterisk :) |
| 08:08.44 | pokus | ok thx for you time |
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| 09:11.58 | weinerk | Please help saying numbers in arabic or other languages: |
| 09:11.58 | weinerk | Seems like there is a couple ways to do it: |
| 09:11.58 | weinerk | 1) SayNumber dialplan cmd, but seems like there is no implementation for arabic ast_say_number_full_ar |
| 09:11.58 | weinerk | 2) Playback("num:XXXf|say") - with custom say.conf, but where can I get say.conf with definitions for other langages? |
| 09:20.31 | tzafrir_laptop | weinerk, hmmm... |
| 09:21.38 | tzafrir_laptop | weinerk, I guess not. Patches are welcomed |
| 09:21.50 | tzafrir_laptop | Should be pretty trivial programming task |
| 09:22.19 | tzafrir_laptop | Note that say.conf is probably useless as it has an issue with male/female variants |
| 09:23.40 | tzafrir_laptop | look for other languages in main/say.c (functions called ast_say_number_full_<something>) |
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| 09:36.08 | weinerk | tzafrir_laptop: as always - thanks for helping! |
| 09:36.09 | weinerk | re: patching main/say.c - I have never done that, but will try to muster the currage, but that seems a longer term. |
| 09:36.09 | weinerk | for short term - re: say.conf - what is the issue with male/female ? is it not implemented at all? |
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| 09:37.55 | bulkorok | hi... somebody knows phones from jablocom!? |
| 09:37.58 | tzafrir_laptop | My Arabic is quite poor, but I assume that for certain uses you need a male form of the number and for others you need the female form (for various counting and such) |
| 09:38.52 | tzafrir_laptop | e.g. in Hebrwe "month" and "day" are male and "year" is female, and hence a date string can easily involve mixed male/female forms |
| 09:42.20 | weinerk | Yes I know (same here - very little arabic, but from hebrew and russian - I know about male/female) |
| 09:42.20 | weinerk | But in SayNumber(XXX|f) - you pass the gender as params. |
| 09:42.20 | weinerk | I thought Playback("num:XXXf|say") also supports that? |
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| 09:57.59 | kami | Hello |
| 09:58.28 | wdoekes2 | bon giorno |
| 09:58.31 | kami | I would like to connect my HFC based USB ISDN dongle to asterisk 1.8. |
| 09:58.40 | *** join/#asterisk nickfennell (~nick@unaffiliated/nickfennell) |
| 09:58.46 | kami | I did that with an older version of asterisk already |
| 09:59.11 | wdoekes2 | chan_dongle? |
| 09:59.25 | kami | But the new box has asterisk 1.8 as default and I read that ISDN support is now better in asterisk. |
| 09:59.29 | kami | chan_lcr |
| 10:00.00 | kami | wdoekes2: is there a 'chan_dongle' in asterisk 1.8? |
| 10:00.18 | kami | wdoekes2: or are you pulling my leg? |
| 10:00.39 | wdoekes2 | not "in", but there is a chan_dongle.. I don't know if it works with your hfc, nor if it does what you want |
| 10:01.13 | kami | wdoekes2: I see. Thank you. |
| 10:02.23 | kami | The combination of linux-call-router and mISDN2 was a good combo. Is there anybody who uses this combination with ast 1.8? |
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| 10:28.47 | jurij1 | i'm on esx 4.1. i've noticed that migrating datastore causes voip issues. we run asteris as vm. anyone? |
| 10:45.55 | *** join/#asterisk din3sh (~din3sh@41.136.244.231) |
| 10:46.01 | din3sh | hi all |
| 10:48.24 | nickfennell | jurij1: that's a downside of running asterisk on VM |
| 10:48.28 | nickfennell | IO constraint |
| 10:48.38 | nickfennell | moving a datastore will be using all possible IO |
| 10:48.57 | nickfennell | Asterisk can't do anything about that and your audio will suffer. |
| 10:49.17 | nickfennell | Don't migrate datastores whilst calls are in progress :) |
| 10:51.33 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 10:53.29 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 10:56.16 | WIMPy | kamiL I used to. |
| 10:56.33 | WIMPy | And chan_dongle is for 3G, not for ISDN. |
| 10:56.49 | WIMPy | Now on 10. |
| 11:02.35 | din3sh | polycom IP 7000 v/s Cisco 7937g |
| 11:02.44 | din3sh | which is better? |
| 11:07.57 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 11:09.55 | jacc0 | polycom |
| 11:10.04 | jacc0 | if you ask me :P |
| 11:16.39 | *** join/#asterisk qjb (~qjb@strange.xs4all.nl) |
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| 11:30.43 | dandate2 | should an end point softphone using g729 require 24kb or 50kb available bandwidth? |
| 11:31.51 | dandate2 | its interesting we have a vpn tunnel to the pbx on our cisco phone tftp server, the stated bandwidth use of each g729 extension is 24kb, however the actual usage is 2x that amount |
| 11:33.39 | dandate2 | heres a screenshot from our iftop readings on the tftp server http://tinypic.com/r/25tzxxl/6 |
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| 11:41.00 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:50.33 | angryuser | Hello i have two same calls, but sometimes asterisk do not send rtp when second reinvite comes from the remote party |
| 11:50.48 | angryuser | Have someone encountered that kind of issue ? |
| 11:51.15 | angryuser | This is driving me crazy, i have two dialogs, same exchange, but in the secon case it does not send rtp |
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| 12:47.21 | MrTelephone | What are the chances a CDR record could be wrong |
| 12:47.41 | MrTelephone | someone called 911 and they claim they didn't, I highly doubt it was an error with the system |
| 12:49.18 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 12:52.40 | wdoekes2 | depends on the complexity of your dialplan.. with local dials and call forwards, things can get messy |
| 12:54.06 | TriJetScud | Naw cdr doesn't lie - typically |
| 12:57.09 | MrTelephone | My dialplan isn't too complex. A lot of macros that return odbc defining if a user is suspended or what not |
| 12:57.42 | MrTelephone | The only thing I can see happen is if 2 endpoints are configured with the same credentials which I know didn't happen |
| 12:58.45 | MrTelephone | peerip needs to be added to my cdr. |
| 13:01.30 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 13:08.24 | Jasnejac | hi all. I have a problem with upgrading asterisk 1.8.3.3, seemingly to any other later asterisk version. a capture of the console output during boot doesn't show any problems that I can see yet pbx functionality doesn't fully load, there is no ael and any attempt to reload pbx_ael.so causes it to hang. anybody got any suggestions how to debug, please? |
| 13:10.48 | [TK]D-Fender | Jasnejac, pastebin the attempt to start it manually, "ln -ls /etc/asterisk" etc |
| 13:13.37 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
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| 13:14.36 | Jasnejac | [TK]D-Fender: the entire log of the startup, all 105k of it? |
| 13:15.09 | [TK]D-Fender | Should be that big. |
| 13:15.13 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 13:16.00 | *** join/#asterisk G00r (~G00r@123.254.116.187) |
| 13:16.38 | G00r | cracks his knuckles and prepares to pump out a whole lotta newbie questions :D |
| 13:16.52 | Jasnejac | http://pastebin.com/5FxpFHWC |
| 13:19.53 | [TK]D-Fender | doesn't look like all of it |
| 13:19.54 | [TK]D-Fender | -- [0mRemote UNIX connection |
| 13:20.01 | [TK]D-Fender | and why is that connection ahppening? |
| 13:21.02 | Jasnejac | that was me seeing if it was working from a second putty session but that's as far as it gets. I think there may be more in the buffer but the "core stop now" or any other core stop or restart options are not there. I have to kill it to stop it |
| 13:24.09 | MrTelephone | How do you log the noops and activity messages to a log file? |
| 13:24.32 | [TK]D-Fender | get a bigger buffer |
| 13:25.46 | Jasnejac | lol. that log was generated with asterisk -f -vvvvvvvvvvg -c > /tmp/1.8.12.0-asterisk-start.log. is that right? |
| 13:26.14 | kaldemar | or use script or tee |
| 13:26.48 | leifmadsen | I like using tee |
| 13:27.02 | leifmadsen | asterisk -rvvvv | tee /tmp/myasterisk.log |
| 13:27.31 | leifmadsen | or you can just enable verbose logging to a file in logger.conf |
| 13:27.38 | leifmadsen | which sounds more like what you want |
| 13:28.16 | MrTelephone | I just want to log dialing activity. What is the best way to do that? Dialplan stuff |
| 13:28.38 | leifmadsen | depends.... is this just debugging temporarily or do you want it all the time? |
| 13:28.41 | leifmadsen | latter == logger.conf |
| 13:29.22 | MrTelephone | all the time. when you goto console and see Executing... and Noop messages |
| 13:30.12 | MrTelephone | I have the default logger.conf but it doesn't log the console messages |
| 13:30.55 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:30.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:30.56 | Jasnejac | gives me exactly the same. hmmm# |
| 13:31.57 | Jasnejac | gives me exactly the same. I use quite a bit of ael. aelparse finds no errors yet pbx_ael doesn't load |
| 13:35.48 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
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| 13:48.57 | *** join/#asterisk Ralcala (~demo@201.127.251.207) |
| 13:49.17 | Ralcala | morning |
| 13:49.24 | Ralcala | to everyone here |
| 13:50.09 | Ralcala | somebody here is awake? |
| 13:50.29 | bulkorok | sure |
| 13:50.39 | Ralcala | hey |
| 13:51.16 | Ralcala | iam gonna install my first asterisk and |
| 13:51.43 | bulkorok | great :-) |
| 13:52.32 | Ralcala | i dont know what version put on my pc? |
| 13:52.58 | Ralcala | its a celeron 2.6 2 gb ram and 320 Gb HD... |
| 13:53.06 | bulkorok | ok |
| 13:53.15 | Ralcala | and of course centos 5.4 |
| 13:53.26 | Ralcala | can tou guide me please |
| 13:53.34 | bulkorok | linux, great ;-) |
| 13:53.43 | Ralcala | yeap |
| 13:53.47 | bulkorok | look here for lifetime: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 13:54.08 | Ralcala | ok |
| 13:54.20 | Ralcala | then i put on last version |
| 13:54.25 | Ralcala | its is right? |
| 13:54.37 | wdoekes2 | that's a good choice |
| 13:54.38 | *** join/#asterisk serafie (~erin@nat/digium/x-zpodzmesnptpfkfi) |
| 13:54.47 | bulkorok | asterisk 1.8 gets updates til end of 2015 |
| 13:54.48 | Ralcala | or there is a version to match my hardware? |
| 13:54.54 | volga629 | Is any special reason why res_srtp is not by default on asterisknow ? |
| 13:54.54 | volga629 | How I can build one, I got lsrtp library in place with share option, and when try load module complain that is not same version |
| 13:54.54 | volga629 | Module 'res_srtp.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 13:54.54 | volga629 | any help thank you in advanc |
| 13:55.10 | bulkorok | asterisk 10 to the end of 2013 |
| 13:55.26 | wdoekes2 | bulkorok: yes.. but the upgrade from 10 to 12 is always easier than from 1.8 to 12 |
| 13:55.32 | *** join/#asterisk LiuYan1 (~liu.yan@211.154.128.135) |
| 13:55.32 | bulkorok | there is no special hardware for any version required |
| 13:55.53 | Ralcala | ok thank you |
| 13:56.12 | bulkorok | wdoekes2: right... but I prefer he LTS versions... |
| 13:56.56 | wdoekes2 | yes.. but this is his first asterisk.. better be up to date from the start |
| 13:57.03 | bulkorok | thats right |
| 14:00.29 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
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| 14:01.44 | Ralcala | one more question |
| 14:02.10 | Ralcala | i browse asterisk.org and i see standard version and long term support version |
| 14:02.44 | Ralcala | then standard > 10 or lts > 1.8? |
| 14:02.59 | Ralcala | which one you suggest me... |
| 14:03.42 | leifmadsen | MrTelephone: as I stated earlier: <leifmadsen> or you can just enable verbose logging to a file in logger.conf |
| 14:03.52 | leifmadsen | MrTelephone: the default logger.conf does not do verbose logging |
| 14:04.34 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 14:06.20 | MrTelephone | I'll check it out |
| 14:20.38 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
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| 14:25.39 | MrTelephone | leif, verbose is working I gather |
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| 14:34.19 | MrTelephone | I think I'm pretty much ready to quit this and work for homocide because I'm good at catching liars |
| 14:34.28 | *** join/#asterisk scgm11 (~Sebastian@r186-50-87-94.dialup.adsl.anteldata.net.uy) |
| 14:34.50 | MrTelephone | I want to work with caroline mason |
| 14:34.56 | MrTelephone | *whistles* |
| 14:37.09 | scgm11 | I´m having a problem with version 1.6.2.X, that I would like to know if someone had this problem, through E1 ISDN PRI I make a call and the other end there is an IVR when I hit a number my channel is put on hold by asterisk, in the console I can see starting moh on my channel, is very strange, this doesn't happen if I connect the same ivr through analog FXO lines. Debug didn't help, I just see passing through dtmf because is |
| 14:37.10 | scgm11 | a feature and then starting music on hold |
| 14:37.26 | scgm11 | could you point me in the code where this could be? on features perhaps?? |
| 14:37.34 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 14:37.50 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 14:39.19 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 14:44.17 | MrTelephone | pastebin your dialplan I guess. Hard to tell scgm11 |
| 14:44.27 | MrTelephone | Are you just using the basic Dial function? |
| 14:44.45 | scgm11 | yes just the dial function, I isolated the context to just that |
| 14:44.50 | scgm11 | is like a feature thing |
| 14:44.56 | MrTelephone | you reloaded after changing it? |
| 14:45.30 | MrTelephone | how much traffic on your box? can you provide full debugging ? |
| 14:45.30 | scgm11 | yes, I also check the debug, and doesn't give me a hint to why is starting moh |
| 14:45.36 | MrTelephone | ~debug |
| 14:45.36 | infobot | ACTION DeBuggers $1 |
| 14:45.42 | MrTelephone | hmm |
| 14:46.08 | scgm11 | after saying DMTF is not a feature letting it pass trough it starts moh on my channel |
| 14:46.28 | scgm11 | is not like starts it to the other end as a normal hold would be |
| 14:46.41 | scgm11 | and this happens against some IVRs not all |
| 14:46.51 | scgm11 | so I guess maybe is something that the other end |
| 14:46.54 | scgm11 | is sending |
| 14:47.03 | scgm11 | that makes my channel go on hold |
| 14:47.28 | *** join/#asterisk pbxMan (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 14:47.29 | leifmadsen | are you allowing features to be triggered via DTMF? |
| 14:47.32 | scgm11 | for that I would like to know where in the code is listening for far end data that could trigger a feature |
| 14:47.43 | leifmadsen | features.conf |
| 14:48.08 | scgm11 | I know the conf that is ok, I have the basic features, no application map |
| 14:48.27 | scgm11 | just ## for blind transfer and #0 for attended transfer |
| 14:48.30 | scgm11 | no other feature |
| 14:48.57 | scgm11 | so I guess there something that is on the code about this channel putting himself on hold |
| 14:49.17 | leifmadsen | it might not be DTMF that is the problem at all.... |
| 14:49.17 | MrTelephone | you need to enable intense debugging. It might help |
| 14:49.33 | leifmadsen | could be the other end is sending something that causes Asterisk to start MOH |
| 14:49.40 | leifmadsen | look at the PRI debug too |
| 14:49.48 | scgm11 | leifmadsen: yes that is what I think |
| 14:50.11 | scgm11 | the other end is sending something that is making this channel on hold |
| 14:50.26 | scgm11 | I will look at PRI debug then |
| 14:50.36 | scgm11 | theres is something weird too |
| 14:50.51 | scgm11 | In fxo lines I can't reproduce |
| 14:50.54 | scgm11 | on PRI yes |
| 14:51.10 | MrTelephone | because fxo channel doesn't know what to do with signal probably |
| 14:51.18 | leifmadsen | because on FXO it's analog and can't send information to set a channel on hold |
| 14:51.22 | MrTelephone | pri is digital and there may be a packet indicating hold |
| 14:51.24 | leifmadsen | PRI is digital and has signalling |
| 14:51.27 | scgm11 | on a SIP trunk connected to other asterisk box that goes out with fxo I can reproduce it as wll |
| 14:51.47 | leifmadsen | again, same thing -- SIP is a signalling protocol that can place calls on hold via a signal |
| 14:52.27 | scgm11 | ok but the far end is connected to FXO is the FXO didn´t get the signal, how could be passed through sip?? |
| 14:52.32 | MrTelephone | If you dialing out and the line is put on hold why would asterisk play moh? isn't that a task for the remote side? |
| 14:53.33 | MrTelephone | is your local fxo the same company as the remote fxo? |
| 14:53.48 | [TK]D-Fender | PSTN connections don't have a "hold" signalling. If they end up there then you gave then that ability. Show us the call |
| 14:53.50 | scgm11 | no |
| 14:54.14 | MrTelephone | tone signalling is the only signals through analog interfaces |
| 14:54.33 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:54.41 | MrTelephone | fsk codes |
| 14:54.56 | scgm11 | I know, but, how could a remote channel put my originating channel on hold?? it doesn't make sense |
| 14:55.13 | scgm11 | the flow is: |
| 14:55.53 | scgm11 | I dial--> the call is answered by the other end --> IVR --> I press 5 -> I hear music on hold and I can see in the console |
| 14:56.01 | scgm11 | that music on hold started for my channel |
| 14:56.03 | scgm11 | for example |
| 14:56.21 | leifmadsen | it makes perfect sense |
| 14:56.29 | leifmadsen | the other end is sending a signal and your asterisk is listening to it |
| 14:56.30 | scgm11 | if Im SIP/1000….. and the other end is DAHDI/4 then staring mho on SIP/1000 |
| 14:56.31 | leifmadsen | simple as that |
| 14:56.45 | leifmadsen | the channel is getting a signal to start MOH |
| 14:56.53 | [TK]D-Fender | scgm11, Show us |
| 14:57.07 | scgm11 | ok, tell me what kind of debug you want |
| 14:57.10 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 14:57.12 | leifmadsen | we already did |
| 14:57.15 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 14:57.28 | scgm11 | pri debug? |
| 14:57.55 | leifmadsen | you're trying to track down the signal that is causing the MOH to be triggered -- that does seem like a logical place to start |
| 14:58.41 | scgm11 | ok, I will get the debug in some minutes and I put it here on pastebin |
| 14:58.43 | MrTelephone | what is the other end of your e1 hooked up to? a pbx? |
| 14:58.53 | scgm11 | I don't know about the other end |
| 14:58.58 | scgm11 | but is not just for one |
| 14:58.59 | MrTelephone | through your telco? |
| 14:59.02 | scgm11 | I have more than one |
| 14:59.08 | scgm11 | places that this happens |
| 14:59.16 | MrTelephone | is there any other options other than 5? |
| 14:59.34 | scgm11 | yes I tested other numbers same result, 5 was just ilustrative |
| 14:59.45 | MrTelephone | what do those numbers do? |
| 14:59.51 | MrTelephone | extensions? |
| 14:59.54 | scgm11 | navigate an ivr |
| 15:00.19 | scgm11 | but I get the PRI debug to see if there is some signal send from the far end |
| 15:01.44 | *** part/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 15:02.01 | MrTelephone | if you find out what signal it is then you can search that and there is most likely a way to turn it off |
| 15:02.46 | MrTelephone | Hey Leif why do some registrations show up in verbose logging as > Saved useragent blah blah blah |
| 15:03.34 | leifmadsen | because that's what was printed to the console at the verbose log level |
| 15:04.24 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
| 15:08.33 | MrTelephone | Yeah but there are a 100 endpoints registering |
| 15:08.43 | MrTelephone | Any only a few show up |
| 15:08.44 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 15:09.39 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:09.47 | scgm11 | leifmadsen: ok seems a remote hold take a look at the debug http://pastebin.com/eNrZWFxa |
| 15:14.00 | leifmadsen | scgm11: sorry I can only help guide, not help solve the problem (work comes first) |
| 15:14.41 | *** join/#asterisk Synapse29 (~Sean_Nien@41-133-165-146.dsl.mweb.co.za) |
| 15:14.54 | scgm11 | yes thanks, I will take a look at libpri, it seems that the other end is sending a remote hold, so I maybe can disable it in libpri on my end right? |
| 15:15.13 | leifmadsen | maybe |
| 15:16.25 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:16.58 | [TK]D-Fender | scgm11, [May 23 12:09:21] VERBOSE[24009] pbx.c: -- Executing [929150054@default:13] Dial("IAX2/888-13837", "DAHDI/g1/29150054,60,TtKk,") in new stack |
| 15:17.07 | [TK]D-Fender | scgm11, You are allowing BOTH sides to TRANSFER the call |
| 15:17.15 | [TK]D-Fender | scgm11, Normally a .... |
| 15:17.20 | [TK]D-Fender | ~soso |
| 15:17.20 | infobot | [~soso] Shoot-On-Sight Offense |
| 15:17.47 | Synapse29 | Hi All, I have an issue where our PBX (Elastix, Asterisk 1.8.7, FreePBX 2.8.1) where if you call a user who's phone is offline, we receive a message saying "An error has occurred", normally this isn't an issue but the user is in a ring group and since this picks up the call, I'm sure you can fee the issue |
| 15:17.50 | scgm11 | yes ok, but I don't think is the transfer that has the problem is the remote hold |
| 15:17.58 | [TK]D-Fender | Transfer triggers a hold |
| 15:18.00 | [TK]D-Fender | stop that now. |
| 15:18.02 | scgm11 | ok |
| 15:18.03 | Synapse29 | Can anyone shed any light on this or point me in the right direction? |
| 15:18.07 | scgm11 | I´ll try it |
| 15:18.24 | [TK]D-Fender | Synapse29, #freepbx <- |
| 15:18.46 | Synapse29 | right, thanks |
| 15:19.01 | *** part/#asterisk Synapse29 (~Sean_Nien@41-133-165-146.dsl.mweb.co.za) |
| 15:19.03 | [TK]D-Fender | Synapse29, And you are 2 branches out of date and nowhere near the latests within that branch... of FreePBX alone. |
| 15:21.59 | scgm11 | [TK]D-Fender: taking all the dial options out, is the same result |
| 15:22.13 | scgm11 | [TK]D-Fender: is a remote hold |
| 15:22.33 | [TK]D-Fender | Kk <- |
| 15:22.42 | scgm11 | I take all out |
| 15:22.45 | scgm11 | no options |
| 15:22.46 | scgm11 | at all |
| 15:22.49 | scgm11 | same result |
| 15:23.24 | [TK]D-Fender | [May 23 12:09:30] VERBOSE[21182] chan_dahdi.c: < Notification indicator (len= 3): Ext: 1 Remote hold (121) |
| 15:23.28 | [TK]D-Fender | This is looking odd... |
| 15:23.45 | scgm11 | this is send by the far end |
| 15:24.09 | scgm11 | I´m looking if in chan_dahdi.conf there is some parameter that can disable this feature |
| 15:24.51 | [TK]D-Fender | I don't I've ever heard any reference to such an action in the first place... |
| 15:24.58 | [TK]D-Fender | Seriously Google-worthy |
| 15:25.10 | scgm11 | someone did http://lists.digium.com/pipermail/asterisk-users/2008-January/203638.html |
| 15:25.24 | scgm11 | a long time ago but sill apply same thing |
| 15:27.40 | scgm11 | what about using this?? mohinterpret=passthrough |
| 15:29.58 | agisamentum | hates polycom phones... slow moving pieces of garbage |
| 15:32.02 | MrTelephone | I hear your pain agisamentum, you need to see mr miagi |
| 15:32.08 | MrTelephone | breath in, breath out buddy |
| 15:32.24 | MrTelephone | paint a fence and come back when you have cooled down |
| 15:32.26 | agisamentum | MrTelephone: haha, just too verbose and SLOWWW |
| 15:32.33 | MrTelephone | what model? |
| 15:32.50 | agisamentum | soundpoint ip 321 |
| 15:32.57 | MrTelephone | I'm tired of the subscribe messages my phone is sending out |
| 15:33.06 | MrTelephone | I beleive I have a bunch of the same phones |
| 15:34.03 | agisamentum | yeah their "splash sound" is absolutely unnecessary and the screen displaying the network configuration is just another time waster, if I want to know the network stats I'll use network tools from the MAC |
| 15:34.26 | agisamentum | </rant> |
| 15:34.30 | MrTelephone | Splash sound is definitely removable |
| 15:34.39 | MrTelephone | it's loading in the background |
| 15:34.54 | MrTelephone | Not much you can do |
| 15:35.22 | MrTelephone | They must be using atmega 4mhz processors |
| 15:35.56 | agisamentum | yeah, I moved to this phone because for some reason I could not get my Inn-Phone D-5905 to establish an RTP stream once a call was answered |
| 15:35.58 | MrTelephone | They rarely reboot though, are you trying to config them and needing to restart a lot? |
| 15:36.40 | MrTelephone | agisamentum, nat or no nat? |
| 15:37.04 | agisamentum | when the call was picked up, it would drop unexpectedly, went through my logs in /var/log/messages and didn't see much; MrTelephone: NAT |
| 15:37.15 | MrTelephone | probably rtp timeout |
| 15:37.43 | MrTelephone | you have to check your sip messages to see what the phone offers for rtpmap in the sdp part of the invite |
| 15:37.55 | MrTelephone | tcpdump on asterisk to see if packets are being sent there |
| 15:38.09 | MrTelephone | put your phone in the dmz to see if it works? |
| 15:39.06 | agisamentum | hmm maybe that would explain why it worked earlier in the day and stopped randomly... I did run tcpdump to check RTP traffic, not sure why it would time out from the blue |
| 15:39.23 | agisamentum | the polycom despite my rant seems to be running fine |
| 15:39.43 | MrTelephone | tcpdump showed rtp in both directions? |
| 15:39.51 | agisamentum | indeed |
| 15:40.14 | agisamentum | oh well, I'll look into the Inn-Phone later |
| 15:40.17 | MrTelephone | async nat router? |
| 15:40.54 | MrTelephone | some routers try and open the same outside port as the local phone port. some routers pick a different port so if asterisk sends to the phone port it won't get through the nat |
| 15:41.36 | agisamentum | hmm that's good to know |
| 15:41.58 | *** join/#asterisk nix8n82 (~none@65.161.180.230) |
| 15:42.38 | MrTelephone | those spa2102's work like a charm though |
| 15:47.09 | agisamentum | MrTelephone: price isn't too bad on them either |
| 15:48.14 | agisamentum | I don't see PoE on the SPA 2102 however |
| 15:49.16 | leifmadsen | I don't think the SPA2102 has POE |
| 15:49.16 | [TK]D-Fender | Linksys doesn't make PoE ATA's |
| 15:49.27 | [TK]D-Fender | And I've never seen ANY |
| 15:49.28 | agisamentum | ah ok |
| 15:49.32 | [TK]D-Fender | from anyone |
| 15:49.37 | leifmadsen | it's an ATA so.... |
| 15:49.41 | agisamentum | right |
| 15:49.42 | leifmadsen | it doesn't really make a lot of sense |
| 15:50.19 | agisamentum | puts on the dunce cap |
| 15:57.54 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 16:01.45 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
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| 16:57.50 | *** part/#asterisk scgm11 (~Sebastian@r186-50-87-94.dialup.adsl.anteldata.net.uy) |
| 16:58.18 | navaismo | Hello, anybody have experience with video IVR using Asterisk? |
| 16:59.03 | MrTelephone | That's a neat idea |
| 16:59.17 | MrTelephone | I want to see the tech support people who make the messages |
| 16:59.43 | MrTelephone | mouser.com is expensive |
| 17:05.27 | angryuser | hello people, i have installed latest 10.4 asterisk and noticed that if her is reinvited couple of times sometimes he does not send rtp, have you encountered such behaviour ? |
| 17:05.52 | angryuser | if i replace it with freeswitch with same base config, there is no problem |
| 17:08.42 | [TK]D-Fender | freeswith can't have "same base config". |
| 17:08.47 | [TK]D-Fender | They are completely different. |
| 17:09.21 | [TK]D-Fender | reinvites where NAT is concerned = fail of course |
| 17:09.35 | [TK]D-Fender | As wrong IP's, forwardings, etc all get in the way |
| 17:17.05 | navaismo | ¬¬ ¬¬' |
| 17:19.31 | kami | >WIMPy: sorry, was afk. You use lcr/misdn2? |
| 17:19.59 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:20.28 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:20.29 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:23.28 | MrTelephone | angryuser, what version did you upgrade from? |
| 17:23.51 | angryuser | MrTelephone, 1.8.7 |
| 17:24.22 | angryuser | [TK]D-Fender, there is no nat involved, everybody are local |
| 17:25.12 | angryuser | [TK]D-Fender, i got same calls to the same DID with same dialogs, but sometimes aserisk do not send rtp at the second reinvite |
| 17:26.09 | angryuser | [TK]D-Fender, i've passed hours finding any sip difference, there is just nothing ;( |
| 17:26.38 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:27.22 | angryuser | [TK]D-Fender, well when i say the base config, i mean the following scenario, 1: Receive call 2: match DID : Forward to uri |
| 17:27.43 | angryuser | In simple they are just a media servers forwarding to proxy |
| 17:27.57 | [TK]D-Fender | angryuser, Well * is a B2BUa. Not sure if FS's "forward" is something else (SIP REFER, etc) |
| 17:28.42 | angryuser | [TK]D-Fender, well i do not forward i bridge and Dial withing asterisk, media goes through them |
| 17:28.50 | angryuser | within* |
| 17:28.52 | *** join/#asterisk Tim_Toady (~fuzzy@62.1.174.66.dsl.dyn.forthnet.gr) |
| 17:30.53 | angryuser | something like this in asterisk :exten => 123456789,1,Dial(SIP/123456789@222.222.222.222,120,Ttr) |
| 17:31.19 | [TK]D-Fender | angryuser, that should prevent a reinvite completely. |
| 17:31.25 | [TK]D-Fender | Tt <--------- |
| 17:31.30 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 17:32.03 | angryuser | hm, Tt allows the transfer by caller, and called party, right ? |
| 17:32.16 | [TK]D-Fender | yes |
| 17:32.25 | [TK]D-Fender | and forces * to remain in the RTP stream |
| 17:32.33 | [TK]D-Fender | prevents reinvites |
| 17:32.37 | angryuser | Why reinvite is considered ? |
| 17:32.44 | angryuser | i see |
| 17:32.44 | [TK]D-Fender | it isn't considered. |
| 17:34.01 | angryuser | [TK]D-Fender, the weirdest thing that its on second reinvite |
| 17:34.30 | angryuser | [TK]D-Fender, thanks for hint |
| 17:36.37 | angryuser | [TK]D-Fender, t: Allow the called party to transfer the calling party by sending |
| 17:36.38 | angryuser | <PROTECTED> |
| 17:36.38 | angryuser | <PROTECTED> |
| 17:37.12 | angryuser | damn, why there is no info on that in app description |
| 17:39.35 | [TK]D-Fender | Dunno.. it's been daw-of-time info.... |
| 17:39.38 | [TK]D-Fender | dawn* |
| 17:40.06 | angryuser | looks like taht dawn of time info i missed |
| 17:40.26 | *** join/#asterisk Tim_Toady (~fuzzy@62.1.174.66.dsl.dyn.forthnet.gr) |
| 17:40.27 | *** join/#asterisk qjb (~qjb@strange.xs4all.nl) |
| 17:40.42 | angryuser | :( very unfortunate |
| 17:46.14 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:48.21 | citywok | has anybody tested how many concurrent calls they can get on a Supermicro 1U D525? |
| 17:49.29 | [TK]D-Fender | citywok, Incredibly vague |
| 17:49.52 | citywok | [TK]D-Fender: yea, depends on transcoding, recording, etc. I was gonig to leave it vague on purpose to see what people have done / how many calls they got with their config. |
| 17:52.02 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 17:57.08 | [TK]D-Fender | no mention of RAM, bandwidth, tech/protocols, recording, transcoding, or anything |
| 17:57.38 | citywok | yep, i was hoping somebody had tested on a D525, and could tell me what they got with their setup |
| 17:57.57 | [TK]D-Fender | too specific a model to bet on. |
| 17:58.03 | [TK]D-Fender | What do you NEED it to be capable of? |
| 17:58.06 | citywok | i highly doubt somebody has done my exact scenario, so i figured if anybody had they could tell me what they did. best i've found is an old review of an atom 230 |
| 17:58.44 | citywok | 30-40 concurrent calls, everything being MixMonitor, most calls in GSM codec, going out a PRI (so transcode from GSM) |
| 17:58.44 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 17:59.20 | citywok | I'm hoping to use something small / low power (literally, not a 400w server) |
| 18:00.27 | [TK]D-Fender | What card? SWEC adds load too.. |
| 18:00.44 | [TK]D-Fender | What are your endpoints? |
| 18:01.38 | citywok | 30% softphone, 70% hard phone |
| 18:02.01 | citywok | digium quad port, t400 or something maybe? cant' remember exactly |
| 18:02.04 | [TK]D-Fender | Soft explains the GSM |
| 18:02.09 | citywok | yep |
| 18:02.41 | [TK]D-Fender | Again, SWEC can impact this. Verify your card. And how many phones / PRI's? |
| 18:02.50 | [TK]D-Fender | and are you going to be recording everything, etc? |
| 18:03.07 | citywok | 95% recorded |
| 18:03.13 | [TK]D-Fender | SSD / record-to-ramdrive will certanly help |
| 18:03.52 | citywok | yea, would be SSD |
| 18:04.33 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 18:05.08 | citywok | [TK]D-Fender: Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) |
| 18:05.21 | [TK]D-Fender | No HWEC. |
| 18:05.33 | [TK]D-Fender | That'll lower your capacity potentially... |
| 18:05.48 | [TK]D-Fender | (assuming other factors don't do worse to you already. |
| 18:05.51 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 18:07.14 | citywok | yea, we only ahve a single PRI, we just happened to have a quad port card lol |
| 18:07.43 | [TK]D-Fender | I would think that you should be able to support 50 phones on that easily enough.. |
| 18:07.58 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 18:08.33 | citywok | yea, my guess is we will probably be okay with ~100 phones, and our typical 20 concurrent calls (and be okay up to 30 or 40) |
| 18:08.49 | citywok | the bench i read had a 1.6ghz single atom transcoding 30 concurrent from g729 to ulaw |
| 18:09.46 | [TK]D-Fender | GSM is much lower load |
| 18:09.52 | [TK]D-Fender | That should mean you're just fine |
| 18:10.08 | citywok | yea i was guessing that. our hardphones just use ulaw (we have plenty of bandwidth) |
| 18:11.02 | [TK]D-Fender | if you're looking at mixed int/ext concurrency I really can't see this as being a big load for your system |
| 18:11.35 | [TK]D-Fender | record to SSD/RD is recommended though. Ram is the cheapest and easiest really. |
| 18:11.46 | [TK]D-Fender | well... maybe SSD is easier, but the wear increses. |
| 18:11.52 | citywok | yea currnetly we record to old 10k 73gb scsi drives. lol. |
| 18:12.23 | [TK]D-Fender | I'm referring to recording as you go. Then 1-shot copy when finished. This eliminates multi-call thrash |
| 18:12.42 | citywok | i'd probably record to RD, i imagine the write cycle would be brutal |
| 18:12.45 | [TK]D-Fender | So you don't have you heads jumping up and down like a Jack Russell Terrier :p |
| 18:13.06 | citywok | call storage is stored on an NFS mount somewhere else, not on the local PBX |
| 18:13.48 | [TK]D-Fender | I might still do it local them pump it over 1-shot. |
| 18:13.58 | [TK]D-Fender | just so you can cache on NFS fail/etc |
| 18:14.07 | [TK]D-Fender | food for thought |
| 18:17.31 | citywok | oh, store local for sure, then we transcode them to MP3 and ship them to the NFS mount |
| 18:17.43 | citywok | nice the transcoding process so it can happen whenever the server has "free" time |
| 18:18.04 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 18:19.00 | citywok | Our PBX is currently on a DL360 G4 which is huge, power hungry, and I might have to come up with a replacement solution here shortly. |
| 18:20.23 | [TK]D-Fender | citywok, Reviews on the SM system though says it runs hot and there have been issues. Check more reviews before settling on a specific build |
| 18:21.05 | citywok | [TK]D-Fender: i've actually used 3 of those so far for small offices (20person or less) and haven't had any issues, but we just stuck a single memory chip, and an SSD in each one |
| 18:21.27 | citywok | they may run hot, idk, i never checked, they are both in AC'd rooms that keep a nice low temp |
| 18:21.49 | citywok | the 3rd one is actually sitting somewhere, we bought it to test stuff and then we didn't end up using it lol |
| 18:22.08 | citywok | but i'm not sold on those, it could be something else small / efficient |
| 18:24.25 | [TK]D-Fender | citywok, Well some of the reviews looked funny. Do your research... |
| 18:24.52 | citywok | good to know, thanks. i may look for something slightly more powerful just so i enver have to deal with it. lol |
| 18:27.21 | *** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net) |
| 18:32.51 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:32.51 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:36.36 | pabelanger | sruffell: remember the clock slip issue from the other week? |
| 18:37.13 | *** join/#asterisk paladinn (~paladin@186.204.178.8) |
| 18:37.14 | pabelanger | Finally heard back from the customer, it is an issue with the Nortel BCM. |
| 18:37.17 | sruffell | yes |
| 18:37.24 | sruffell | makes sense |
| 18:37.24 | pabelanger | dahdi and asterisk working like a champ |
| 18:37.29 | pabelanger | yup |
| 18:37.35 | *** part/#asterisk paladinn (~paladin@186.204.178.8) |
| 18:38.02 | pabelanger | I always thought it was the problem however it only took 6 months of dropped calls and 4 bell techs to finally tell the customer what the problem was |
| 18:38.19 | sruffell | so…if you know, what was the problem with the Nortel BCM? |
| 18:39.14 | pabelanger | I think a problem with the back-plane, but nobody really knows. The plan is just to remove it |
| 18:39.45 | sruffell | hmm….if they haven't removed it yet, how do they know for certain it's the problem? Another tech came out and hooked up a simulator? |
| 18:40.53 | pabelanger | they removed the asterisk gateway last week, still having clock sync issues. They did more troubleshooting and can see an issue with the BCM. However, it is EOL and the customer does not want to spend any $$$ to fix it |
| 18:42.01 | sruffell | ahh..ok |
| 18:47.14 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 18:48.16 | leifmadsen | pabelanger: \o/ |
| 18:48.57 | rgsteele | I just upgraded (or am in the process of doing so) an old 1.4 installation, and saw that 'queue-lessthan' is no longer a valid option. Anybody know what the replacement is? |
| 18:51.29 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 18:52.59 | bent_screwdriver | is there a way to have menu options in Voicemail() besides * and 0? I want to use 1 for something but don't readily see a way to do this... |
| 18:53.20 | [TK]D-Fender | bent_screwdriver, app_minivm |
| 18:53.21 | [TK]D-Fender | <- |
| 18:53.22 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 18:53.29 | [TK]D-Fender | not with app_voicemail |
| 18:54.00 | bent_screwdriver | [TK]D-Fender: thanks |
| 18:55.00 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-pyuwqlbsdcdtcnxf) |
| 18:59.49 | WIMPy | kami: Now I've been away. Yes, I use it on multiple installations. |
| 19:01.21 | *** join/#asterisk rossand (~aross@12.53.158.2) |
| 19:01.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:01.34 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:01.57 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 19:03.42 | timholum | hello I am running asterisk 1.8.3, I am trying to get a time based ivr set up, but when I add exten => s,n,,GotoIfTime(9:00-17:00,mon-fri,*,*?101-day) I get an error "No application ',GotoIfTime', is there a replacement for this? |
| 19:03.58 | timholum | nevermind |
| 19:04.05 | timholum | I just noticed the double , |
| 19:04.07 | [TK]D-Fender | timNo, yousimply don't put a double comma there |
| 19:11.02 | agisamentum | anyone have any experience with the yealink SIP-T26 phones? |
| 19:11.18 | agisamentum | for some reason I can't get this thing to even pick up network |
| 19:11.52 | agisamentum | takes the power from PoE, but won't seem to attempt to talk or take a lease |
| 19:12.07 | GeoGeek | agisamentum: We have a few Yealinks...no T26's but the ones we have we are happy with. |
| 19:12.21 | kami | WIMPy: do you compile mISDN and LCR from the git repos? |
| 19:13.20 | WIMPy | kami: Usually only mISDNuser and LCR. On theproduction boxes I use mISDN from the kernel. |
| 19:13.31 | agisamentum | GeoGeek: anything special to make the yealinks talk? saying network unavailable and I'm not it's MAC on the LAN at all, bit strange |
| 19:13.52 | WIMPy | But as we learned some weeks ago, you shouldn;t do that if you've got a RedHat kernel. |
| 19:14.22 | GeoGeek | agisamentum: You have tried more than one I assume? |
| 19:15.39 | kami | WIMPy: I have just compiled misdn, misdnuser, lcr from git repo. the Makefile of lcr needed some uncommenting to build chan_lcr.so. Now, my HFC USB interface is correctly identified and calls arrive at LCR. |
| 19:15.40 | agisamentum | nah, I'll do some more troubleshooting, switch has been acting up on me, just surprised it doesn't want to talk out of the box |
| 19:16.10 | kami | WIMPy: although i load chan_lcr in asterisk (1.8), I don't see a log message in LCR's log that asterisk has connected. |
| 19:16.31 | GeoGeek | agisamentum: We didn't have any trouble in that area....I would suspect switch or DHCP or bad cable etc...they say Network Unavailable when they are booting... |
| 19:16.37 | WIMPy | kami: You should see it in lcradmin status. |
| 19:17.45 | WIMPy | lcradmin state, sorry. |
| 19:19.55 | kami | WIMPy: I see 'Ext(port 0: HFC-S_USB.1) TE ptmp use:0 L2 UP L1 UP' |
| 19:20.02 | kami | no mention of asterisk |
| 19:20.37 | agisamentum | GeoGeek: yeah, excuse me for being a bad tech, my PoE switch is DOA, just swapped the switch and it's working |
| 19:21.18 | WIMPy | kami: Do you have lcr and Asterisk running as the same user? Otherwise you may have to tweak permissions on lcrs socket. |
| 19:21.39 | kami | ahhh |
| 19:21.45 | WIMPy | See 'socketrights' in options. |
| 19:21.53 | kami | WIMPy: I probably started lcr as root! |
| 19:22.38 | *** join/#asterisk mKn0wt (~mtv@200.30.170.218) |
| 19:28.10 | GeoGeek | agisamentum: Cool...they are great phones for the money. |
| 19:29.28 | GeoGeek | Anybody here using Festival? I keep getting "festival_exec: festival_client: connect to server failed". Found lots of references to that in google but no answers so far... |
| 19:29.40 | agisamentum | GeoGeek: yeah they appear to be, one of the perks to this model is being able to connect it to multiple PBX's offsite to coordinate multiple organizations under one hat |
| 19:30.21 | GeoGeek | agis: We might be going to that level at some point....just trying to get our first * the way we want it for now... |
| 19:30.51 | kami | WIMPy: thank you. After restarting lcr as user 'asterisk', I have 'Remote: asterisk' in lcradmin state. |
| 19:31.28 | agisamentum | GeoGeek: doing the same here, good luck on your journey :) |
| 19:31.42 | WIMPy | kami: Ok, so you can use it now. |
| 19:34.40 | *** join/#asterisk wonko (~wonko@wiggum.4amlunch.net) |
| 19:35.53 | wonko | so, trying to get Zoiper talking to asterisk. i click the register button but it doesn't appear to actually register. neither things give any sort of error so how do i find that? |
| 19:37.23 | kami | WIMPy: have to correct something in routing.conf ... |
| 19:38.07 | WIMPy | kami: One line ": remote application=asterisk" is enough. |
| 19:38.28 | WIMPy | But ann additional contex=whateva might be a good idea. |
| 19:40.00 | kami | WIMPy: you configure the MSN's in asterisk in the whateva context? |
| 19:40.14 | WIMPy | Exactely |
| 19:40.43 | kami | WIMPy: OK. |
| 19:41.09 | kami | I had the different MSN in routing.conf, but will change that. |
| 19:41.33 | WIMPy | No need to do so if you want to send them all to Asterisk. |
| 19:41.43 | WIMPy | And to the same context. |
| 19:43.28 | newtonr | wonko: take a look at https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information you can use sip debug from the CLI to see whats happening when you attempt to register, and at least gather info so you can ask others to help if you can't figure it out on your own |
| 19:44.11 | *** join/#asterisk Sicelo (sicelo@unaffiliated/sicelo) |
| 19:45.10 | wonko | newtonr: ok, i'll give that a look, thanks |
| 19:45.16 | wonko | i'm assuming the same thing works for iax2 |
| 19:45.24 | wonko | i just setup a sip channel for zoiper and that works fine |
| 19:45.27 | wonko | bah |
| 19:45.37 | wonko | i might just use SIP instead. :) |
| 19:45.42 | newtonr | yah |
| 19:45.56 | wonko | no firewall to deal with, so simple |
| 19:46.12 | wonko | need to do IAX to another asterisk box though so didn't want to use sip and iax but, whatever. :) |
| 19:48.11 | volga629 | Is have any way add res srtp to asterisk now ? |
| 19:48.13 | *** join/#asterisk rossand (~aross@12.53.158.2) |
| 19:50.26 | kami | WIMPy, it works as follows: incoming call is forwarded to the context in asterisk where 'exten => 33.....,1,Dial(SIP/201,20,tr)' is configured (copied that from some tutorial or sth). My sip phone rings. I can hear the caller on the sip phone, but the caller cannot hear what I speak. |
| 19:52.02 | WIMPy | kami: That sounds like the bug in git a few versions ago. Are you using the drivers from a broken Kernel? What exact version? |
| 19:52.45 | wonko | oh |
| 19:53.56 | wonko | the username and what's in [] need to match |
| 19:54.07 | WIMPy | Or it's just a normal RTP isse.. |
| 19:54.22 | kami | WIMPy: modinfo gives me /lib/modules/2.6.32-41-generic/extra/mISDN_core.ko which means that the new git version is used |
| 19:55.22 | Sicelo | hmm, i fear that i'll ask a dumb question -- why would i install asterisk? what would i do with it? would i need any special hardware as well? unfortunately the asterix site and wiki don't give me those clues. noob here :-$ |
| 19:56.07 | WIMPy | kami: Do you have other modules under kernel/drivers as well? They would take preceedence. |
| 19:56.14 | [TK]D-Fender | Sicelo, * is a PBX & telephony toolkit |
| 19:56.28 | kami | They are still present. But doesn't modinfo report the loaded one? |
| 19:56.51 | [TK]D-Fender | Sicelo, You can set up systems like you see companies use to support multiple phones and types of lines. confrence, voicemail and more |
| 19:57.14 | WIMPy | kami: No, modinfo just gives info on some module, it doesn;t have to be loaded. |
| 19:57.28 | [TK]D-Fender | Sicelo, You can may a system to auto-call-out people for announcements, making polling systems, call centers, you could set it up as a boring voicemail box, etc. |
| 19:57.42 | [TK]D-Fender | Sicelo, I use mine as a jukebox and coffee-maker. |
| 19:57.56 | Sicelo | :) |
| 19:57.57 | kami | WIMPy: OK. Do you usually delete the original modules completely or is it enough to rename the extension? |
| 19:58.02 | WIMPy | kami: Is that that RedHat Kernel I mentioned? |
| 19:58.12 | kami | WIMPy: ubuntu lucid |
| 19:58.19 | WIMPy | kami: You can just delete them. |
| 19:58.29 | Sicelo | so do i need anything more that my pc? i mean .. how will i get calls to and from landlines? |
| 19:58.31 | WIMPy | Great. |
| 19:59.02 | WIMPy | Why do they all use broken versions? The kernels from kernel.org didn't have that issue. |
| 20:00.25 | kami | WIMPy: because they feel that handing out a stock kernel would lower their value as distro providers? |
| 20:01.02 | WIMPy | FAIL |
| 20:01.38 | WIMPy | They should do some testing if they feel like using GIT versions instead of the official ones. |
| 20:01.55 | sruffell | Ubuntu among other things, adds AppArmor into the kernel, which isn't carried in the kernel.org kernels. |
| 20:03.24 | WIMPy | I've been told at the LUG camp that RH kernels had MD impementations that were incomatible to other (normal) kernels. |
| 20:06.26 | [TK]D-Fender | Sicelo, Depends what you want to use |
| 20:06.36 | [TK]D-Fender | Sicelo, If you have physical lines then you will need a hardware interface. |
| 20:06.45 | [TK]D-Fender | Sicelo, Or you could use an ITSP |
| 20:06.47 | [TK]D-Fender | ~itsp |
| 20:06.47 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:07.39 | Sicelo | thanks [TK]D-Fender. now i have a good idea. :) |
| 20:08.41 | kami | WIMPy: I removed the original modules, reloaded everything and now the situation has changed: when I pick up the SIP phone, the caller continues to hear the ring tone |
| 20:10.02 | [TK]D-Fender | Sicelo, Ask us when you'd like to confirm what you think your end scenario should look like and we'll let you know if your choices may have complications. |
| 20:11.15 | WIMPy | kami: What does lcradmin state display? What about the log? |
| 20:12.07 | *** join/#asterisk mpe (~mpe@109.70.54.56) |
| 20:13.07 | kami | WIMPy: the former 'Ext(port 0: HFC-S_USB.1) TE ptmp use:0 L2 UP L1 UP' |
| 20:15.08 | WIMPy | ... while calling. |
| 20:17.58 | Sicelo | thanks [TK]D-Fender. i'll do so |
| 20:18.07 | kami | MT_ALERTING REQUEST is the last log line while the sip phone is ringing |
| 20:18.21 | kami | and it remains so, even when I pick up the SIP phone |
| 20:19.19 | kami | the SIP phone shows the incoming caller id and when I hang up, the caller hears a busy tone |
| 20:19.44 | kami | this is where the lcr log continues with 'TONE to CH(4) directory default name cause_14' |
| 20:19.54 | kami | and 'DISCONNECT to CH(4) cause value=20 location=1-Local-PBX' |
| 20:20.12 | WIMPy | Doesn't look look like it gets any information about you trying to pick up then. |
| 20:21.05 | WIMPy | 20? Hmm. That could come from Asterisk. |
| 20:21.59 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:22.10 | WIMPy | Standard Asterisk debuggin procedure... |
| 20:22.11 | kami | yes, asterisk log says 'SIP/201-00000007 is ringing' |
| 20:22.20 | kami | 'Nobody picked up in 20000 ms' |
| 20:22.27 | WIMPy | You could try something like Echo() first to eliminate other factors. |
| 20:22.34 | WIMPy | BBIAB |
| 20:22.39 | WIMPy | ok |
| 20:25.40 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 20:26.17 | [TK]D-Fender | checkout time, later all |
| 20:32.25 | WIMPy | Ok, so maybe that was a RTP issue earlier on as well and ubuntu is not bad? |
| 20:34.12 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 20:35.05 | kami | WIMPy: I had two sip phones in the Dial(...) directive although one of them was not registered. This seems to be the reason for that error code 20. |
| 20:36.06 | WIMPy | As long as the other one is working, that doesn't matter. |
| 20:37.21 | kami | When I remove that one, I reliably get 'SIP/201-00000003 answered lcr/3' in asterisk's log. |
| 20:38.23 | kami | Only occasionally, I can hear the sip phone speaking, with random length pauses. Is that an RTP issue? |
| 20:38.46 | WIMPy | Probably. |
| 20:39.39 | WIMPy | Additional phones in the Dial() that are unreachable shouldn't make a difference, either. So that smells like some other issue, like networking. |
| 20:40.18 | kami | WIMPy: yes, that was not the real issue. I put it back and it still works as described. |
| 20:40.29 | volga629 | In asterisk spec file have this mv %{name}-sources-%{version}-%{release}.make.err /var/log/ Is any specif reason to move under /var/log because without sudo will permissions deny |
| 20:40.58 | WIMPy | kami: Try MusicOnHold() or Echo() to test both parts seperately. |
| 20:41.36 | kami | WIMPy: will do. Thanks. |
| 20:49.26 | *** join/#asterisk gusto (~gusto@88.128.141.249) |
| 20:54.49 | kami | WIMPy: Echo and SayUnixTime work flawlessly with the sip phone. |
| 20:55.25 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:59.23 | WIMPy | And from lcr? How doesd that behave? |
| 21:01.05 | WIMPy | Could there be an IRQ issue? |
| 21:04.20 | kami | WIMPy: from lcr, it's still the same. During some calls, I can hear audio for some seconds. During others, no sound at all. |
| 21:04.53 | kami | So, it must be related to LCR, when sip phone + echo works. |
| 21:05.14 | kami | Would the module debugging options help? |
| 21:05.46 | GeoGeek | Anybody here using Festival? I am installing it on Ubuntu 11.10 server. Asterisk is calling it but I am getting: SIOD error: unbound variable : " |
| 21:05.57 | WIMPy | I've got no idea. I never had an issue like that. |
| 21:06.08 | GeoGeek | Asterisk 1.8.12 |
| 21:07.22 | kami | WIMPy: thank you sincerely for your help! I've taken a huge step with your help. |
| 21:15.42 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:15.42 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:17.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:22.47 | *** join/#asterisk schultza (~schultza_@rc1.rcherbals.com) |
| 21:22.58 | schultza | how do i get a asterisk server to talk to itself to test the ports? |
| 21:25.33 | [TK]D-Fender | poor test. Set up a softphone or something on another PC |
| 21:26.34 | *** join/#asterisk navaismo (~navaismo@201.123.94.189) |
| 21:26.50 | schultza | using a sangoma wanpipe device? |
| 21:27.56 | [TK]D-Fender | First you said "ports" without saying what type. |
| 21:28.05 | [TK]D-Fender | Now you mention sangoma hardware? |
| 21:28.20 | [TK]D-Fender | You should be a lot clearer about what you have and what it is you are trying to actually "test" |
| 21:30.12 | schultza | i have a sangoma tech A104d quad t1 aft card.. i want to test those ports |
| 21:31.28 | [TK]D-Fender | then go place calls in/out |
| 21:31.35 | [TK]D-Fender | trying = using |
| 21:33.37 | schultza | i'm sorry. i am extremely new to asterisk. do i need to set up only dialplans for t1 ports or are there further things i need to setup for this test? ive been working on a lab for softphones, but it doesnt get me the t1 ports i need to work on |
| 21:34.33 | [TK]D-Fender | well you need to DIAL them... so yes, that's clearly dialplan |
| 21:34.38 | WIMPy | You can build a cross over cable and connect one port to another. But I'm not sure that makes much sense. |
| 21:34.54 | schultza | i have that... im on the software end for a wall right now |
| 21:35.09 | schultza | software / config |
| 21:36.30 | [TK]D-Fender | I would like to be able to assume you did your system.conf, chan_dahdi.conf, wanpipe configs, etc..... |
| 21:36.33 | [TK]D-Fender | but then again... |
| 21:39.39 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 21:44.08 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:47.48 | kami | WIMPy: I found this thread 'One-Way Audio': http://thread.gmane.org/gmane.linux.isdn.i4l.user/5929/focus=5936 |
| 21:48.47 | Chainsaw | One-way audio is nearly *always* a NAT problem with the SIP (RTP really) leg of the call. |
| 21:48.48 | WIMPy | Yes, I told you there has been a bug in the GIT version. |
| 21:49.01 | kami | Reverted to mISDN git commit a1898f3 and now it works. |
| 21:49.18 | Chainsaw | So I will read that one, because that's outside the line of expectations. Thanks. |
| 21:50.03 | WIMPy | Hmm. So you're saying this issue is present in the current git version (again)? |
| 21:50.44 | kami | Yes, although the very last commit which I have here is 'Fix oneway audio with LCR' :) |
| 21:51.59 | kami | Just one last question: how exactly do you configure outbound calls from SIP phones to Dial(LCR...)? |
| 21:52.03 | WIMPy | Interesting. I thought I used the git version on my test box last week. |
| 21:52.15 | *** part/#asterisk Sicelo (sicelo@unaffiliated/sicelo) |
| 21:52.22 | WIMPy | LCR/interface/number/options |
| 21:52.31 | WIMPy | interface is the section name in interface.conf. |
| 22:02.21 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 22:04.08 | kami | WIMPy: thanks. Now I have everything working. Can I send you flowers by Fleurop? |
| 22:04.46 | WIMPy | Ha, I'd rather get rid of some :-) |
| 22:04.54 | kami | :) |
| 22:05.22 | *** join/#asterisk jsjc (~Adium@173.Red-2-136-104.dynamicIP.rima-tde.net) |
| 22:09.28 | kami | Good night. |
| 22:09.52 | *** part/#asterisk kami (~user@unaffiliated/kami-) |
| 22:14.18 | *** join/#asterisk uskerine (~uske@237.Red-95-122-134.staticIP.rima-tde.net) |
| 22:15.06 | uskerine | hi, i would like to have recommendations on which board should I use for a Primary ISDN access, operator is telling me that they will install G.703 (pair of Tx/Rx Coax) |
| 22:16.01 | WIMPy | You will need a balun to get to TP. |
| 22:16.14 | uskerine | what is TP? |
| 22:16.25 | WIMPy | Twisted Pair. |
| 22:16.38 | uskerine | so asterisk boards are twisted pair (rj45)? |
| 22:16.46 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 22:16.59 | [TK]D-Fender | uskerine: http://www.alibaba.com/product-gs/553811689/g_703_bnc_to_rj45.html |
| 22:17.03 | WIMPy | I haven't seen a card with BNC connectors. |
| 22:17.27 | WIMPy | I've seen them on routers, but not on PC cards. |
| 22:20.08 | uskerine | thanks |
| 22:20.19 | uskerine | more questions |
| 22:20.38 | uskerine | I see as per the alibaba product that there are actually several types of 75ohm coax, right? |
| 22:21.02 | uskerine | so there is Siemenz, BT43 and SMB |
| 22:21.32 | uskerine | should I ask to the provider which coax connector are they going to use? |
| 22:22.11 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
| 22:22.42 | WIMPy | Yes, it's just the type of connector. |
| 22:23.18 | uskerine | and regarding the 120ohm TP, is there more than one type? I see also several types for this adapter |
| 22:24.23 | uskerine | 120 ohm option 1: Krone IDC120 ohm option 2: 3 PIN 120 ohm option 3: 2 PIN |
| 22:24.25 | *** join/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 22:24.45 | philipp64|laptop | if "sendrpid=yes" is set, what value is used to populate that header field? |
| 22:25.06 | WIMPy | You surely want the version where you can plug in a standard patch cable. |
| 22:25.17 | uskerine | so u mean RJ45 |
| 22:25.20 | uskerine | right? |
| 22:25.37 | WIMPy | Apart from the misnomer, yes. |
| 22:26.21 | uskerine | why is RJ45 misnomer? |
| 22:26.59 | WIMPy | Because RJ45 is a two wire data line. The plug is a modular or western 8P8C. |
| 22:27.54 | WIMPy | RJ48 is indeed a ISDN line, but I don't know if it's BRI or PRI and at which reference point. |
| 22:28.15 | WIMPy | So RJ48 might be correct. |
| 22:28.22 | uskerine | ok |
| 22:28.43 | uskerine | which board would you recommend? (just one PRI) |
| 22:29.13 | WIMPy | If you want hardware echo cancellation, Digium or Sangoma. |
| 22:29.39 | WIMPy | If not I'd gof for anythign HFC-E1 as that can be used with multiple drivers. |
| 22:29.58 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:30.29 | uskerine | I guess hardware echo cancellation is always a good idea, right? |
| 22:30.47 | WIMPy | Depends on what you plan to do. |
| 22:31.28 | uskerine | it is going to be a small call center |
| 22:31.35 | WIMPy | If you want to forward calls via VOIP, it's a good idea. |
| 22:31.59 | WIMPy | If you do IVR, it's not neccessary. |
| 22:32.26 | uskerine | i do not plan to mix TDM and VoIP |
| 22:33.13 | WIMPy | If you say call certer, how do you plan to make the agents speak? |
| 22:33.20 | uskerine | errr |
| 22:33.21 | uskerine | sorry |
| 22:33.25 | uskerine | you are totally right :) |
| 22:33.35 | uskerine | there will be VoIP on the customer agents side |
| 22:34.02 | uskerine | so echo cancellation is recommended then, right? |
| 22:34.06 | WIMPy | So some eco cancellation is a good idea. |
| 22:34.12 | uskerine | good |
| 22:34.15 | uskerine | thanks i will go for it |
| 22:34.28 | WIMPy | But it may be cheaper to get just a devent PC and do it in software. |
| 22:34.35 | uskerine | what about the server? i had these servers available: http://www.dell.com/uk/business/p/poweredge-r210-2/fs?~ck= |
| 22:36.18 | WIMPy | That should definitely be enough to do the EC. |
| 22:37.34 | uskerine | ok, anyway, if it is not very expensive, it is not bad idea to buy the hardware EC, as in future i could need to do some integrations with other systems with the IVR |
| 22:37.40 | uskerine | so the more free the cpu is the better |
| 22:38.23 | WIMPy | My old PIII could do about 20 channels. |
| 22:38.44 | uskerine | ok |
| 22:39.58 | uskerine | if the server has this: |
| 22:39.59 | uskerine | I/O Slots 1 PCIe x16 G2 slot |
| 22:40.10 | uskerine | then i guess this is my card |
| 22:40.12 | uskerine | http://store.digium.com/productview.php?product_code=1TE121BF |
| 22:40.13 | uskerine | right? |
| 22:40.44 | uskerine | (sorry for so many questions but this is my first asterisk installation and I do not want to screw up with the hardware selection) :) |
| 22:41.02 | WIMPy | Yes, that fits. |
| 22:41.53 | uskerine | anyone could recommend where to buy it in europe? |
| 22:44.12 | WIMPy | http://www.mapleleaf-technologies.com/store/index.php |
| 22:44.19 | WIMPy | http://voipango.de/ |
| 22:48.24 | uskerine | thanks |
| 22:48.27 | uskerine | do they ship fast? |
| 22:48.41 | uskerine | i see there is TE121BF and TE122BF, but can not detect any difference |
| 22:48.46 | uskerine | why two models? |
| 22:49.33 | uskerine | forget it |
| 22:49.35 | uskerine | i see now |
| 22:49.38 | uskerine | it is PCIe and PCI |
| 22:51.02 | uskerine | as PRI lead times from operator are really slow ones |
| 22:51.07 | uskerine | i would launch with 4 RTB lines |
| 22:51.18 | uskerine | so i would also buy a 4 RTB port board |
| 22:51.25 | WIMPy | RTB? |
| 22:51.27 | uskerine | sorry |
| 22:51.28 | uskerine | POTS |
| 22:51.34 | WIMPy | What about BRI? |
| 22:51.50 | uskerine | it will be as painful as PRI in terms of lead time |
| 22:51.54 | uskerine | which country are u from? |
| 22:52.02 | WIMPy | de |
| 22:52.09 | uskerine | de is the only country with decent ISDN |
| 22:52.17 | WIMPy | was |
| 22:52.21 | uskerine | i am in spain, ISDN is really uncommon |
| 22:52.21 | *** join/#asterisk Bullmoose (~Bullmoose@71-37-168-220.bois.qwest.net) |
| 22:52.44 | WIMPy | It's all NGN here now. |
| 22:53.55 | uskerine | i though it was still common to have ISDN in de |
| 22:54.21 | uskerine | so what would u recommend for 4 POTS board? (PCIe) |
| 22:55.02 | WIMPy | Well, you can have ISDN on NGN, but with very limited functionality, off course. |
| 22:55.13 | WIMPy | Can't comment on POTS. Analog is evil. |
| 22:55.50 | uskerine | i plan to live only for 1 or 2 months with it |
| 22:56.04 | uskerine | i see digium PCIe board is really expensive (PCI is cheap) |
| 23:01.48 | uskerine | WIMPy |
| 23:01.56 | uskerine | should this fit for the temporary 4 POTS? http://www.voipango.es/es/ISDN-Interfaces/Sangoma/Sangoma-Analog/Sangoma-A200BRME-Basecard-PCIe.html |
| 23:02.08 | uskerine | (even when I know you do not deal with evil analog) |
| 23:02.09 | uskerine | :) |
| 23:08.25 | phix | :D |
| 23:14.26 | WIMPy | That should fit. It's a bit large, but I'm not sure if you find smaller ones in PCI-e. |
| 23:14.28 | *** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net) |
| 23:15.08 | WIMPy | But you need the modules for the 4 ports. |
| 23:16.13 | *** join/#asterisk budster (~brhatigan@cpe-069-134-227-054.nc.res.rr.com) |
| 23:17.34 | *** part/#asterisk mpe (~mpe@109.70.54.56) |
| 23:19.11 | uskerine | WIMPy you mean that I need something else apart from that board? |
| 23:19.26 | uskerine | i thought i came with 4 FXS/FXO port by default |
| 23:19.49 | WIMPy | Oh, ok, didn;t notice that part. |
| 23:21.20 | uskerine | i am just asking |
| 23:21.31 | uskerine | i think it might be actually need the analog module |
| 23:22.16 | uskerine | what do I need FXO or FXS? (agents are VoIP, and calls are received through 4 POTS lines) |
| 23:23.02 | jpsharp | You need FXO ports to connect to phone lines. |
| 23:23.33 | uskerine | it is very expensive then |
| 23:23.48 | uskerine | any recommendation for a 4FXO port PCIe card? |
| 23:24.40 | WIMPy | Better use the money to bribe your telco to get teh PRI faster :-) |
| 23:25.38 | uskerine | they don't care |
| 23:26.04 | uskerine | telefonica (O2) didn't even answer |
| 23:26.24 | jpsharp | Or get a couple of ATAs with FXO ports on them as a stopgap measure. |
| 23:26.33 | WIMPy | is on telefonica as well. |
| 23:26.51 | uskerine | jpsharp i am listening: any recommendation for cheap ATAs? |
| 23:30.39 | jpsharp | I can't "recommend" one, but I can suggest the Grandstream HT-503. Single FXO port, $60USD each. |
| 23:32.29 | uskerine | did u use it? |
| 23:32.49 | jpsharp | Not that particular one, but I've had good luck with Grandstream products in general. |
| 23:33.01 | uskerine | that's good |
| 23:33.13 | uskerine | i can buy it for 45Eu |
| 23:33.13 | WIMPy | So you disagree with infobot? |
| 23:33.55 | jpsharp | What does infobot say? |
| 23:34.07 | WIMPy | ~gs |
| 23:34.07 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:34.19 | uskerine | :-D |
| 23:34.33 | jpsharp | Then, yes. |
| 23:35.11 | WIMPy | Anyway... analog is never going to be reliable, so I'm not sure how much impact on the inevitable headache that will make. |
| 23:35.26 | jpsharp | Apart from them tending to catch fire when you connect the wrong power supply to them... |
| 23:35.45 | WIMPy | Multimedia? |
| 23:35.49 | uskerine | 4 GrandSuck ATAs saves 100Eu less than having a proper sangoma card |
| 23:36.02 | WIMPy | If you can;t find your lighter than may come handy. |
| 23:36.13 | uskerine | why analog is not reliable? |
| 23:36.36 | jpsharp | Echo & Call progress detection |
| 23:36.37 | uskerine | (it has been there since 19th century) |
| 23:36.45 | WIMPy | You can't find out what's happening at the other end, except for listening to tones. |
| 23:37.05 | uskerine | why should I find out what is happening at the other end? |
| 23:37.09 | WIMPy | Yes, but it wasn't made for anything half way automatic. |
| 23:37.34 | uskerine | could you please give me an example on issues i could face? |
| 23:37.45 | WIMPy | The most basic thing would be to free the line when the other end has hung up or isn't reachable. |
| 23:37.47 | uskerine | customers will call here, what can happen and how often? |
| 23:38.45 | WIMPy | You can't have DDI. And *if* DR is available, it will take time and might not be totally reliable. |
| 23:38.55 | uskerine | i will only receive calls |
| 23:38.59 | uskerine | i won't make calls |
| 23:39.21 | WIMPy | Yes, but do you need more than one number? |
| 23:39.42 | uskerine | i plan to have a single header number (group call i think it is called) |
| 23:39.50 | jpsharp | If someone calls in, you put them on hold, and they hang up, Asterisk may not detect the hangup. |
| 23:39.59 | uskerine | why not? |
| 23:40.28 | WIMPy | It can only "listen" for a busy tone. |
| 23:40.45 | jpsharp | Or "reverse battery disconnect", if your telco supports it. |
| 23:40.55 | uskerine | but if the caller is a mobile |
| 23:40.57 | WIMPy | And if that part is too sensitive, it may disconnect calls that are still alive. |
| 23:40.59 | uskerine | there will be a busy tone, right? |
| 23:41.30 | uskerine | that would only happen with land lines |
| 23:41.33 | WIMPy | Yes, polarity reversal cerainly helps, but I'm not sure that's very common outside of NA. |
| 23:41.47 | uskerine | where the caller can hook off and the call is not ended |
| 23:41.52 | uskerine | is that correct? |
| 23:42.14 | WIMPy | It's your end that matters. |
| 23:42.31 | uskerine | i do not follow you |
| 23:42.37 | uskerine | someone calls |
| 23:42.38 | jpsharp | It doesn't matter who's calling. If they hang up, Asterisk may not see the disconnect. |
| 23:42.40 | WIMPy | BTW: If you can start with just 4xPOTS, are you sure you need a PRI? |
| 23:42.50 | uskerine | WIMPy yes i need it |
| 23:42.52 | WIMPy | It doesn;t matter what technology the caller uses. |
| 23:43.06 | uskerine | so i am trying to understand it |
| 23:43.09 | WIMPy | Ok. |
| 23:43.11 | uskerine | someone will call |
| 23:43.20 | uskerine | asterisk will answer on one of the 4 FXO ports |
| 23:43.28 | uskerine | the customer will be put on hold |
| 23:43.30 | WIMPy | It;s just that I've seen enough installations that never needed a PRI. |
| 23:43.44 | *** join/#asterisk Freeaqingme (~Freeaqing@91.214.168.110) |
| 23:43.52 | uskerine | then we will be out of business and there will be no need for asterisk wimpy, but that's a different history |
| 23:43.53 | uskerine | :) |
| 23:44.00 | *** join/#asterisk rossand (~aross@12.53.158.2) |
| 23:44.23 | uskerine | so the customer is put on hold |
| 23:44.34 | uskerine | if the customer waits until he is attended by an agent |
| 23:45.23 | WIMPy | If you didn;t use Asterisk, you might not even need channels for the customers on hold. |
| 23:45.31 | uskerine | then it is transferred, the operation takes place and either the agent or the customer hangs up |
| 23:46.06 | uskerine | WIMPy i do not want centrex because i will have software integrations in future |
| 23:46.40 | WIMPy | I'm not talking centrex. |
| 23:46.42 | uskerine | i am trying to understand when and why it could fail |
| 23:46.52 | WIMPy | But how many concurrent calls do you expect? |
| 23:47.57 | WIMPy | When the customer hangs up. When the agent hears that and also hangs up, ok. If it happens while the customer is on hold, your line may stay engaged until an agent picks the call and then hangs up. |
| 23:47.58 | uskerine | WIMPy, i need asterisk and i need to be in full control of every incoming call through IVR |
| 23:48.26 | WIMPy | What are you arguing about? |
| 23:48.33 | WIMPy | I never questioned that. |
| 23:49.06 | uskerine | i thought you were suggesting not using asterisk |
| 23:49.24 | uskerine | and i was trying to explain why i decided that asterisk was needed |
| 23:49.26 | WIMPy | No, my question is just, how many channels you need. |
| 23:49.55 | uskerine | my best estimation (which is still an estimation) is that in the near months i will have 10 agents and less than 30 concurrent calls |
| 23:50.09 | WIMPy | Yes, I often get annoyed by some of Asterisks misfeatures, so I tend to point them out. |
| 23:50.51 | uskerine | do you think that is there any other way to avoid the PRI? |
| 23:51.28 | WIMPy | Yes, a bunch of BRIs. Depending on the amount of concurrent calls, they may be quite a bit cheaper. |
| 23:52.54 | uskerine | PRI is very cheap |
| 23:52.58 | uskerine | it is 200Eu/month |
| 23:53.11 | WIMPy | That's not what I call cheap :-) |
| 23:53.27 | uskerine | each BRI will cost 14Eu/month |
| 23:53.47 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 23:53.48 | WIMPy | That's quite a difference. |
| 23:53.51 | uskerine | so it is roughly the same |
| 23:54.02 | uskerine | no, because with a PRI i have 30 lines |
| 23:54.06 | uskerine | with BRI i have 2 lines |
| 23:54.11 | WIMPy | Only if you use all channels. |
| 23:54.39 | uskerine | i might use them, i can not have busy tone or unknown number of queded customers |
| 23:54.47 | WIMPy | But 14 for a BRI is a really good price. |
| 23:54.48 | uskerine | i have to pass every customer through the IVR |
| 23:55.20 | uskerine | i think it is 14 (I am not sure, that's the cost of a RTB and I assume it is the same ammount) |
| 23:55.28 | uskerine | sorry RTB=analog |
| 23:55.33 | WIMPy | There isn't really a difference between BRI and PRI, except for the physical side. |
| 23:55.59 | WIMPy | The good thing about BRIs is that you only need to pay for as many as you really need. |
| 23:56.09 | uskerine | i am fine paying a PRI |
| 23:56.15 | uskerine | and i can even pay half a PRI (15 channels) |
| 23:56.28 | uskerine | it is enough granularity for me |
| 23:56.42 | WIMPy | Fractional PRIs don;t exist here any more AFAIK. |
| 23:56.49 | uskerine | in spain they do |
| 23:56.56 | uskerine | you can order 15 and 30 channels |
| 23:57.11 | WIMPy | But ok, sounds like they are more flexible in your area. |
| 23:57.49 | uskerine | i have even a quote |
| 23:57.53 | uskerine | so PRI is ok to simplify |
| 23:57.59 | uskerine | and i can waive telefonica by ordering PRI |
| 23:58.05 | uskerine | BRI are only served by telefonica |
| 23:59.02 | uskerine | telefonica is scored as lowest customer service among all operators |
| 23:59.11 | WIMPy | Yes, it's always good if you have choice. |
| 23:59.32 | uskerine | but please help me to understand the limitations of the temporary analog solution |
| 23:59.45 | uskerine | as i would like to undertstand what i going to face the first month of call center |