| 00:00.08 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:01.32 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-142-113.pools.arcor-ip.net) |
| 00:38.50 | SeRi | p3nguin: what app did you use for jailbraking? you said it was easier |
| 00:40.25 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 00:42.50 | *** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net) |
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| 01:01.04 | p3nguin | seri: For the iPhone, I used redsn0w, which I guess uses pwnage tool underneath. For my iPods, I've used greenpois0n and jailbreakme.com. |
| 01:03.16 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 01:05.38 | p3nguin | It really depends on what iOS version your device has as to what tool is best. |
| 01:12.35 | SeRi | just curious. I wanted to test yours since you said it was simpler |
| 01:12.43 | SeRi | I am working on one right now |
| 01:19.40 | p3nguin | Which iOS? |
| 01:21.55 | SeRi | 3.1.3 |
| 01:22.03 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 01:22.04 | SeRi | but looks like redsnow wont do 3.1.3 |
| 01:23.44 | SeRi | what iOS did you had? |
| 01:25.43 | SeRi | looks like I am fucked after a downgrade |
| 01:26.04 | p3nguin | 4.0.0 on the iPhone -- redsn0w does good on it. 4.2.1 on a couple iPods -- greenpois0n does it well. 4.3.3 on some other iPods -- jailbreakme.com takes care of this version. |
| 01:26.17 | p3nguin | How could you possibly downgrade? |
| 01:27.33 | ujjain | What is better for Android, CSipSimple or Sipdroid? |
| 01:27.41 | p3nguin | I can't get iTunes to change a version to anything other than the current newest version available. Most people have this same problem. |
| 01:30.45 | SeRi | Yes you can |
| 01:31.14 | SeRi | go in to restore mode and click shift + restore and you can browse to what ever ipsw you want to restore too |
| 01:31.25 | p3nguin | Good luck getting it to work. |
| 01:32.05 | p3nguin | iTunes will not restore/upgrade/downgrade to a version of iOS that they are not signing. You'll get an error. |
| 01:33.53 | raden | how do i get this to stop |
| 01:33.56 | raden | <PROTECTED> |
| 01:34.05 | raden | it keeps doing it non stop |
| 01:34.12 | p3nguin | raden: Turn your core verbose DOWN to a sane level. |
| 01:34.29 | p3nguin | At level 4 and above, it prints dnsmgr crap. |
| 01:34.30 | carrar | dump the iphone |
| 01:34.51 | raden | p3nguin, gotcha |
| 01:34.55 | raden | so thats normal ?> |
| 01:35.12 | p3nguin | If you're running at 4 or above, it is normal. |
| 01:35.16 | SeRi | p3nguin: wrong. |
| 01:35.23 | SeRi | p3nguin: I just did it |
| 01:35.24 | raden | why the hell it doing a lookup every 10 seconds ? |
| 01:35.38 | p3nguin | seri: Pics. |
| 01:35.46 | SeRi | pics to what? |
| 01:35.52 | SeRi | Is a process. Is done |
| 01:35.58 | p3nguin | Google it. You'll find out exactly what I just explained. |
| 01:36.03 | SeRi | went from 4.0 to 3.1.3 |
| 01:36.20 | p3nguin | Did you have a dump from the device when it was using 3.1.3? |
| 01:36.30 | *** join/#asterisk coppice (~coppice@m121-202-35-126.smartone.com) |
| 01:37.12 | p3nguin | I can't take this iPod I have with 4.3.5 on it and restore down to 4.3.3... iTunes will not allow it. |
| 01:37.26 | p3nguin | Apple is not signing 4.3.3 anymore, so I'll get an error code. |
| 01:38.35 | p3nguin | If you can do it today, they changed the signing policy within the last couple months. |
| 01:38.57 | p3nguin | They've been blocking version changes like this for quite a while now. |
| 01:39.05 | SeRi | one sec. I am trying to use the pwnage tool to try and build a custome restore file |
| 01:39.50 | raden | there we go much better |
| 01:41.34 | p3nguin | raden: Verbose level 3 is typically good enough for all call processing, because there is NOTHING that you will gain by going from verbose level 3 to level 4 besides the dnsmgr lookup crap. |
| 01:42.16 | p3nguin | At an even higher level, you'll gain CDR stuff, but most people don't often need to look at that. |
| 01:45.10 | *** join/#asterisk Devon_ (~chatzilla@adsl-65-8-196-66.mia.bellsouth.net) |
| 01:50.37 | SeRi | Mhhhh something is wrong with the custom image. |
| 01:52.16 | SeRi | I am going to try and do it on my vm |
| 01:54.19 | p3nguin | Why not upgrade to 4.0 and use redsn0w? |
| 01:55.56 | SeRi | 4.0 runs like shit on 3G |
| 01:56.07 | SeRi | almost unusable |
| 01:56.27 | carrar | yeah |
| 01:56.37 | carrar | keep your 3g at the origianl ios version |
| 01:57.04 | carrar | unfortunately there is no going back |
| 01:58.01 | SeRi | well I am down to 3.1.3 but cant get it to jailbrake |
| 01:58.22 | carrar | are you able to make and receive calls? |
| 01:58.25 | p3nguin | Oh, come on... you mean I can't randomly pick an old version and downgrade/restore to that version by supplying the file? |
| 01:59.29 | SeRi | carrar: I dont have an ATT SIM so it wont come up. I am stuck at unsupported sim |
| 02:00.03 | p3nguin | Why not use that one you never sent to me. |
| 02:00.30 | phix | hmmmm |
| 02:00.31 | carrar | install http://www.cyanogenmod.com/ |
| 02:00.41 | phix | I recommend you get an Android :) |
| 02:01.18 | p3nguin | But do you get to use all the iApps on Android? |
| 02:02.29 | SeRi | p3nguin: because it did went out and I dont have it |
| 02:03.06 | p3nguin | I wasn't referring to the first one you said got lost. |
| 02:03.18 | SeRi | oh. |
| 02:21.04 | phix | p3nguin: why would you want to do that |
| 02:21.13 | p3nguin | Do what? |
| 02:21.23 | phix | use any iApps on an Android |
| 02:21.39 | phix | most apps have been ported to ANdroid |
| 02:21.59 | p3nguin | I didn't know that. |
| 02:22.04 | phix | I mean you can get an emulator but that would be slooow |
| 02:26.27 | SeRi | well I am fucked. the phone can not get pass unlocking |
| 02:26.34 | SeRi | back to 4.0 |
| 02:34.52 | raden | why does polycom have like 200 files to config there phones ? |
| 02:41.04 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net) |
| 02:41.13 | KingDavidNYC | hello everybody |
| 02:42.34 | p3nguin | How's D.C. this time of year? |
| 02:42.47 | KingDavidNYC | DC is nice man |
| 02:42.57 | KingDavidNYC | how do you know I am in DC man? |
| 02:43.11 | SeRi | ~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net |
| 02:43.13 | p3nguin | washdc.fios.verizon.net |
| 02:44.09 | KingDavidNYC | oops |
| 02:44.34 | KingDavidNYC | I was checking where you guys get that info :) |
| 02:45.28 | SeRi | it displays when you login to the chan and log off |
| 02:45.31 | SeRi | is normal |
| 02:45.50 | KingDavidNYC | Guys, something strange just haappened to me... I found out that if I start a line with ;-- it freaking invalidates the rest of the extensions.conf file!! |
| 02:46.49 | raden | ok this readme dont tell me crap about configing the phone with these files :( |
| 02:47.36 | KingDavidNYC | that freaking ;-- in the vi file made me lose at least 3 weeks of work |
| 02:51.16 | KingDavidNYC | Anybody want to talk about the weather? |
| 02:54.41 | p3nguin | I'm going to test your theory with the ;-- thing. Based on what you're asserting, if I put that on the first line all by itself, the entire file will be worthless and I will have no extensions. Is my understanding correct? |
| 02:55.41 | p3nguin | raden: You need the admin guide. |
| 02:57.18 | p3nguin | [Jan 3 20:57:02] WARNING[30893]: config.c:1604 config_text_file_load: Unterminated comment detected beginning on line 1 |
| 02:57.47 | p3nguin | At least there was a warning. |
| 03:02.53 | KingDavidNYC | p3nguin: yes Sir |
| 03:03.37 | raden | p3nguin, only 15 phones ill just do them manually I think |
| 03:03.39 | raden | lol |
| 03:04.29 | KingDavidNYC | I am using Centos |
| 03:04.35 | SeRi | raden: good luck. I thought the same thing but you are better of using the provisioning service. |
| 03:04.55 | SeRi | just download the split files and delete what is not need it |
| 03:05.03 | raden | split files ? |
| 03:05.07 | raden | f**** |
| 03:05.09 | raden | bla |
| 03:05.15 | raden | i wish they just had a nice flat file |
| 03:05.29 | SeRi | like who? |
| 03:09.10 | *** join/#asterisk god-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 03:09.37 | raden | i mean just something like SERVER = 192.168.1.1; USER = john ; pass = secret ; |
| 03:09.39 | raden | simple |
| 03:12.54 | SeRi | than you dont need a polycom |
| 03:13.13 | SeRi | looks like you do not need any futures or options just a plain and simple phone |
| 03:26.30 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:27.41 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:27.42 | raden | SeRi, i just want simplicity ! |
| 03:28.01 | SeRi | raden: dont we all :) |
| 03:28.33 | SeRi | I wish but fucking an iphone was simpler. Thanks to corporate fuckers the US locks every damn phone to carriers. |
| 03:28.50 | SeRi | s/but/butt/ |
| 03:29.19 | SeRi | so. I am back to ios 4 and the phone is garbage. |
| 03:29.34 | SeRi | Ill have to buy an android phone for my son. |
| 04:05.44 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
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| 04:50.37 | joobie | hey guys.. anyone know how i can subscribe my polycom 650 to watch a particular extension? i have got the hint part setup properly, i can see the state coming up in 'show hints' and i have enabled <bw> in my directory.xml also.. but it always says watchers 0 in the 'show hints' and doesnt work |
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| 06:49.43 | *** part/#asterisk irroot (~gregory@197.111.1.164) |
| 06:50.38 | jaxon11247 | I have pasted my issue in http://pastebin.com/zBGVmdcY |
| 06:50.38 | jaxon11247 | I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to '"Anonymous" <sip:test02@anonymous.invalid>;tag=as57d3a806' |
| 06:50.38 | jaxon11247 | i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me. |
| 06:52.41 | jaxon11247 | please find sip.conf file in http://pastebin.com/zBGVmdcY |
| 06:56.38 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:00.34 | kaldemar | jaxon11247: you should also pastebin a failed call with verbosity and sip debug enabled. |
| 07:01.37 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
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| 07:08.50 | jaxon11247 | Hello Kaldemar, I have pasted sip debug with verbosity of failed call http://pastebin.com/jL2ki0s8 |
| 07:14.24 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:17.42 | *** join/#asterisk prasanthprabalan (~root@218.248.24.19) |
| 07:18.23 | prabalan | hi all |
| 07:18.38 | prabalan | how to run linux command from extension |
| 07:18.53 | prabalan | i tried System() |
| 07:19.28 | prabalan | 5000=> 1,System(/usr/bin/iceweasel) |
| 07:19.55 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:19.56 | prabalan | its not opening the browser using this extension |
| 07:20.11 | prabalan | is there any problem in the script |
| 07:20.21 | fenrus | does your asterisk user have permissions to speak to the xserver |
| 07:20.35 | fenrus | try with somehing easier, that dont require X |
| 07:20.45 | prabalan | am running asterisk as root |
| 07:20.55 | fenrus | like echo "tmp" /tmp/test |
| 07:21.22 | fenrus | err, > between |
| 07:21.22 | fenrus | :D |
| 07:23.02 | prabalan | i already tested with some other commands. |
| 07:23.08 | prabalan | like below |
| 07:23.18 | prabalan | ;exten => s,1,System(/bin/echo "'${CALLERID(num)}'" > /root/Desktop/testfile.txt) |
| 07:23.24 | prabalan | to get callerid |
| 07:23.31 | prabalan | it is working |
| 07:24.25 | prabalan | how to set permissiom to xserver |
| 07:27.49 | kaldemar | jaxon11247: your credentials are wrong, nothing more than that. check that fromuser/defaultuser/secret are correct. |
| 07:28.05 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-nldnoejfvofuxesw) |
| 07:29.49 | prabalan | thank u |
| 07:30.38 | jaxon11247 | Kaldemar: I have checked the same many times ..But i am checking again |
| 07:32.43 | jaxon11247 | Kaldemar: I have checked. fromuser, defaultuser, secret fields..this fields are correct |
| 07:33.30 | jaxon11247 | kaldemar: also sip show registry show that sip trunk is registered.. |
| 07:34.30 | jaxon11247 | prabalan: are u able to solve the issue? |
| 07:34.36 | prabalan | no.. |
| 07:34.52 | prabalan | actually i got confused with some other message. |
| 07:35.15 | prabalan | am running asterisk as root. |
| 07:35.34 | kaldemar | jaxon11247: not looking at those pastes alone. take a sip trace from a call with the x-lite if it works, so we'll have something more to look at. |
| 07:35.40 | prabalan | but how to check the permission to speak to xserver |
| 07:35.52 | fenrus | prabalan, with the xhost command as i recall |
| 07:36.27 | prabalan | ok i will check it and revert |
| 07:37.11 | jaxon11247 | kaldemar: I have tested call from Xlite with this trunk. works well!! |
| 07:37.19 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 07:37.44 | jaxon11247 | only problem is that i am not able to make calls from asterisk when registered this trunk in sip.conf |
| 07:38.53 | kaldemar | jaxon11247: what does "sip show registry" in CLI say? is the registration successful? registering is really separate from outbound calls, they are not related. |
| 07:42.21 | jaxon11247 | kaldemar: 192.168.1.55:5060 N test02 105 Registered.. |
| 07:42.21 | jaxon11247 | .. |
| 07:42.21 | jaxon11247 | is there any problem with my SIP URI..its showing "From: "Anonymous" <sip:test02@anonymous.invalid>;" |
| 07:42.31 | kaldemar | jaxon11247: you might need to change type=friend to type=peer |
| 07:44.12 | jaxon11247 | Kaldemar: I have chenged type=peer.. But no luck |
| 07:45.56 | kaldemar | jaxon11247: it might be that the "Anonymous" part is the problem. depends on how the other end behaves. |
| 07:47.07 | jaxon11247 | Kaldemar: How do i Change that URI? |
| 07:47.37 | jaxon11247 | Kaldemar: Can u throw some light on this...plz |
| 07:48.30 | *** join/#asterisk asterisk-Tester (~Ramyt@193.227.170.247) |
| 07:48.59 | kaldemar | the URI is sip:test02@anonymous.invalid, the display name is something that comes from the caller id name part. would be unlikely if that's the problem. i'd suggest you take a sip trace from a call with x-lite so it can be compared. |
| 07:49.36 | jaxon11247 | trunk works great with Xlite at same time its not working with Asterisk.. I will take sip trace from Xlite and let u know :) |
| 07:53.32 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:53.34 | schmidts | good morning |
| 07:54.58 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:57.06 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 07:59.03 | asterisk-Tester | helo |
| 08:02.55 | *** join/#asterisk jacco_7564 (~Jacco@195-241-134-176.ip.telfort.nl) |
| 08:03.06 | jacco_7564 | good morning |
| 08:04.51 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 08:05.17 | *** part/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
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| 08:13.03 | jacco_7564 | Just wondering, for a newcomer to asterisk, what do people recommend as the best PBX for newbs like me? |
| 08:16.49 | ChannelZ | uhmm.. asterisk? |
| 08:20.18 | *** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net) |
| 08:20.44 | schmidts | jacco_7564 asterisk is the PBX itself, i guess what you meant is which gui you should use and the answer depends on what you like but IMHO you will learn most about it when you do it by hand. |
| 08:21.20 | schmidts | ~book |
| 08:21.20 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 08:21.33 | schmidts | jacco_7564 maybe you should start with this ^^ |
| 08:27.54 | jacco_7564 | thats one hell of a book |
| 08:32.05 | *** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net) |
| 08:35.08 | schmidts | jacco_7564 its not "A" book, its "THE" book ;) |
| 08:37.16 | jacco_7564 | Ye turns out ive read quite a bit of it already, on another site though. I am using the elastix distro now |
| 08:48.06 | *** join/#asterisk Nasga (~Nasga@112.4.118.78.rev.sfr.net) |
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| 08:50.36 | *** part/#asterisk irroot (~gregory@197.170.179.42) |
| 08:51.12 | jacco_7564 | "Asterisk can appear quite daunting and complex to a new user" - very true |
| 08:56.25 | *** part/#asterisk manji (~manjiki@penicuik.physics.upatras.gr) |
| 09:00.34 | jaxon11247 | Kaldemar: I traced the issue. While Dialing call fro Xlite send following Sip header F=sip:test02@192.168.1.55. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me |
| 09:07.23 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 09:09.04 | kaldemar | jaxon11247: you must have fromdomain=anonymous.invalid in your sip.conf. remove it. |
| 09:10.15 | *** join/#asterisk siaulius (~chatzilla@83.171.9.142) |
| 09:10.19 | prabalan | hi.. |
| 09:10.34 | prabalan | how to open one application from asterisk extension |
| 09:11.21 | jaxon11247 | kaldemar: I dont have fromdomain=anonymous.invalid |
| 09:11.26 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:11.57 | jaxon11247 | kaldemar: even i just tried to set fromdomain=192.168.1.55 But no luck |
| 09:15.47 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 09:17.16 | prabalan | <PROTECTED> |
| 09:17.35 | prabalan | asterisk is opened as root |
| 09:17.51 | schmidts | prabalan core show application system |
| 09:18.50 | prabalan | 5000=> 1,System(/usr/bin/iceweasel) |
| 09:18.53 | kaldemar | jaxon11247: the anonymous.invalid must come from somewhere. pastebin your whole sip.conf, masking secrets. |
| 09:19.00 | prabalan | i already tried System. |
| 09:19.08 | prabalan | that command not working |
| 09:19.20 | schmidts | prabalan what do you want to do? |
| 09:19.52 | kaldemar | he wants to open a GUI application for a user that asterisk does not run as. |
| 09:20.14 | prabalan | i want to open iceweasel browser with specific link when call lands on asterisk |
| 09:21.48 | prabalan | so if call made to number 500 it should open browser with a link say an example google.com |
| 09:22.21 | prabalan | iceweasel google.com will work in command line. But not working with asterisk. |
| 09:22.29 | schmidts | prabalan do you need the http request or do you really want to have the browser? |
| 09:22.42 | kaldemar | prabalan: you need to set display before /usr/bin/iceweasel. like System(export DISPLAY=:0.0 ; /usr/bin/iceweasel) |
| 09:22.56 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 09:23.25 | kaldemar | prabalan: to find the correct content for DISPLAY, echo the variable as the user you want the browser to be opened for. |
| 09:24.17 | jaxon11247 | kaldemer: I pastebin my whole sip.conf. http://pastebin.com/22wJcfpW |
| 09:24.28 | prabalan | ok i will check it |
| 09:24.49 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 09:25.53 | kaldemar | jaxon11247: ; must be escaped inside system, so System(export DISPLAY=:0.0 \; /usr/bin/iceweasel) |
| 09:26.25 | *** join/#asterisk irroot (~gregory@197.170.229.97) |
| 09:26.41 | jaxon11247 | Hey Prabalan above message is for u :) |
| 09:27.00 | kaldemar | oops.. |
| 09:27.07 | *** part/#asterisk irroot (~gregory@197.170.229.97) |
| 09:28.11 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 09:28.54 | prabalan | Thanks jaxon11247 and kaldemar. I will check it now |
| 09:32.08 | jaxon11247 | kaldemar: Have u checked my pastebin? |
| 09:35.05 | *** join/#asterisk d0mi (~domi@mail.tas.de) |
| 09:36.34 | prabalan | System(export DISPLAY=:0.0 \; /usr/bin/iceweasel) is not working |
| 09:37.21 | kaldemar | prabalan: is DISPLAY right? |
| 09:38.06 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:38.36 | prabalan | am able to run " export DISPLAY=:0.0;/usr/bin/iceweasel " in terminal. |
| 09:38.46 | prabalan | so i think display is right |
| 09:39.07 | Tuju | is it possible to add caller id so that there is a sip url instead of number? |
| 09:39.12 | Tuju | for a line |
| 09:41.42 | prabalan | kaldemar>: echo $DISPLAY is giving same output :0.0 |
| 09:42.59 | *** join/#asterisk jmls (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 09:43.05 | jmls | morning all |
| 09:43.32 | jmls | I'm having some audio problems / nat issues (never come across these before) and need some help |
| 09:44.18 | jmls | trying to run asterisk 1.4 head on amazon ec2. I have the server up and running, and a softphone (3cx / xlite) connected and registered |
| 09:44.45 | jmls | each phone can place a call, but I get no audio from the ec2 if I call voicemail or play some music on hold |
| 09:45.14 | jmls | I have nat=route, the externip=my elastic ip address, and localnet=10.0.0.0/255.0.0.0 |
| 09:45.26 | jmls | what am I missing ? |
| 09:46.37 | prabalan | Hi any help ???? |
| 09:47.32 | kaldemar | prabalan: well, works here. |
| 09:48.14 | prabalan | kaldemar: Is the same command working there |
| 09:48.25 | prabalan | can to give the extension |
| 09:48.52 | schmidts | jmls have you tried sip debug and see what ips are told from asterisk to your soft client in the sdp body? |
| 09:49.04 | kaldemar | i'm tried with firefox. |
| 09:51.38 | prabalan | Ya i am able to open gedit with that. I shall try it for some other browser |
| 09:52.07 | jaxon11247 | Any Help ..I Just want to replace my sip header part from anonymous.invalid to my domain ip. |
| 09:52.37 | kaldemar | works fine remotely too. remotely as in asterisk on host A opens a browser for a user on host B. |
| 09:52.41 | jmls | schmidts: will do. what am I looking for ? |
| 09:54.07 | prabalan | kaldemar: Its working with google-chrome |
| 09:54.21 | prabalan | kaldemar: Thanks for the help |
| 09:54.44 | prabalan | kaldemar: I think i can implement my application with google-chrome |
| 09:54.47 | kaldemar | it will probably work with iceweasel too, using the right iceweasel command. |
| 09:55.15 | prabalan | Ok if u r getting plz tell me how to.. |
| 09:55.53 | prabalan | Mean while i will also try |
| 09:56.18 | schmidts | jmls just enable sip debug on your softclient and then try to call some extension where you should hear audio. The INVITE and the 200 OK response is interesting |
| 09:57.38 | *** join/#asterisk brdude (~brdude@c-24-130-20-142.hsd1.ca.comcast.net) |
| 10:05.01 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 10:09.36 | jmls | hrrmph. No idea what I'm looking ar / for. I've pb'd the sip debug from asterisk here : http://pastebin.com/tsr0uR6N -> there seem to be three addresses in play 1) SoftphoneIP, routerIP and ec2 ElasticIP |
| 10:10.11 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:12.35 | jaxon11247 | kaldemar: I am getting Proxy Authentication failed when i tried outbound call. I think its bcoz of this sip header problem. isnt it/ |
| 10:12.37 | jaxon11247 | ? |
| 10:21.29 | *** join/#asterisk timahvo1 (~rogue@197.178.198.227) |
| 10:23.26 | *** join/#asterisk master_of_master (~master_of@p57B559BA.dip.t-dialin.net) |
| 10:23.57 | schmidts | jmls is this ec2 elasticIP a public ip or private? |
| 10:24.43 | jmls | schmidts: it's a public ip |
| 10:25.34 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 10:27.59 | siaulius | Asterisk doesn’t detect that the caller has hung up the call. Called phone is still ringing for a while. After some time asterisk generates call to the same phone and these calls continues in periodic times.. Who knows where could be the problem? |
| 10:28.58 | schmidts | jmls and this MySoftphoneIP is a private one i guess? |
| 10:29.13 | jmls | yup. a 192.168.xx.yy |
| 10:29.37 | jmls | it's the ip of the pc that's running the softphone. All ip's are dished out by dhcp on the router |
| 10:29.54 | schmidts | jmls so just normal nat setup ;) |
| 10:30.00 | jmls | uh huh |
| 10:30.18 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 10:30.34 | jmls | I don't have to add the 192.168 network to localnet, do I ... ? |
| 10:30.41 | schmidts | jmls no |
| 10:30.57 | schmidts | jmls what is your nat setting in sip.conf? |
| 10:31.07 | jmls | nat=route |
| 10:31.11 | kaldemar | jaxon11247: looks like it, but don't know how to solve it. maybe it does a DNS resolution which screws it up. |
| 10:31.20 | jmls | tried nat=yes, couldn't register the phone |
| 10:32.20 | jaxon11247 | kaldemar: Now reading this http://www.faqs.org/rfcs/rfc5379.html |
| 10:32.35 | *** join/#asterisk irroot (~gregory@41.51.104.173) |
| 10:32.45 | schmidts | jmls maybe you can also do a rtp set debug on and look if you can see if asterisk tries to send out packets and where |
| 10:32.49 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
| 10:33.01 | *** part/#asterisk irroot (~gregory@41.51.104.173) |
| 10:33.07 | Yourname` | I know wrong place to ask, but does anyone have experience setting up a GSM gateway as a SIP trunk on Elastix? |
| 10:33.12 | jaxon11247 | may be it can help me with this issue ;) |
| 10:33.18 | kaldemar | siaulius: you need disconnect supervision on the incoming line. that's something that the telco offers with a technology. to use is in asterisk, you must configure the signaling to be kewlstart and configure hanguponpolarityswitch=yes in chan_dahdi.conf if the disconnects are signaled with a polarity switch. |
| 10:35.06 | kaldemar | siaulius: if you can't get disconnect supervision, you can enable busydetect=yes on the line in chan_dahdi.conf (and tune it with busycount) but that is not reliable and may cause unwanted hangups. |
| 10:37.56 | jmls | schmidts: http://pastebin.com/mNns3aLh |
| 10:39.09 | schmidts | jmls i dont see any rtp packets |
| 10:39.25 | jmls | urgh |
| 10:40.48 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:41.02 | jmls | maybe that's why there's no audio :) |
| 10:41.14 | jmls | seriously, I typed |
| 10:41.17 | jmls | rtp debug |
| 10:41.18 | jmls | and got |
| 10:41.30 | jmls | RTP Debugging Enabled |
| 10:41.34 | schmidts | jmls ok |
| 10:41.39 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:f27b:cbff:fe9d:f3e4) |
| 10:42.00 | siaulius | kaldemar: just before your post I tried busy detect=yes and it worked. Now I will try hanguponpolarityswitch=yes. One more question: when the call comes from outbound it tooks for about 5 seconds when the phone starts ringing in Asterisk system. Is there anything to do to reduce this time? |
| 10:42.48 | *** join/#asterisk mikal (~mikal@ppp59-167-142-8.static.internode.on.net) |
| 10:46.14 | jmls | ohhh, schmidts. This is a 1.4 asterisk, and I didn't install zaptel |
| 10:46.18 | kaldemar | siaulius: if you don't see a channel going off-hook before the 5 seconds, no. |
| 10:46.29 | jmls | does that make a difference to timing / sounds / rtp ? |
| 10:47.13 | schmidts | jmls normally music on hold should work even without zaptel |
| 10:47.25 | schmidts | jmls imho this looks to me like a typical double nat problem |
| 10:48.04 | jmls | oh. |
| 10:48.18 | jmls | bangs his head on the table |
| 10:49.06 | jmls | do I have to specify the ip on rtp debug ? If so, which one ? ;) |
| 10:49.31 | schmidts | jmls i dont know for 1.4 but normally its global if you dont specify an ip |
| 10:50.02 | jmls | that's what I thought. So why no rtp in the output ? |
| 10:50.54 | schmidts | cause asterisk recognize that he should send rtp to a private ip and wait for the other side to send it but the other side maybe doesnt reached the asterisk through nat cause the port is not open |
| 10:51.04 | schmidts | jmls you said asterisk also is behind nat right? |
| 10:51.16 | jmls | it has to be |
| 10:51.19 | jmls | so, yes |
| 10:51.46 | jmls | 192.168.xx.yy->publicrouterip->elasticip->ec2 |
| 10:52.37 | schmidts | so you have a double nat in between and when asterisk handling rtp over nat, it normally waits for the other side to send first, to just send the rtp back to the now open port but if both sides are behind nat, the needed ports will never be open |
| 10:53.32 | jmls | right - but how does anyone then get asterisk to work on an ec2 ? |
| 10:54.17 | schmidts | jmls first it would make sense to switch to 1.8, imho nat handling is much better there |
| 10:54.38 | jmls | yeah, was thinking that. Although I was going to go to 10 ;) |
| 10:54.53 | jmls | let me do that, and I'll let you know what happens |
| 10:55.08 | jmls | is there any difference in the sip.conf options ? |
| 10:58.21 | jaxon11247 | any update? |
| 10:59.34 | schmidts | jmls most of it should just work, but your extensions.conf will be broken |
| 10:59.35 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 10:59.48 | schmidts | but i guess its not that big when you just started ;) |
| 10:59.54 | siaulius | kaldemar: hanguponpolarityswitch=yes didn't work, so I will stay with busydetect=yes option. But time when phone starts ringing and hungs up in asterisk system is more then 8 secs, so it is a little bit too long i think.. |
| 11:01.29 | kaldemar | siaulius: busydetect listens to the audio on the line and tries to detect the busy tone. busycount setting defines how many detected tones are considered a hangup. if you lower that count, you'll get a faster hangup but the probability of false hangups will increase. |
| 11:01.41 | *** part/#asterisk asterisk-Tester (~Ramyt@193.227.170.247) |
| 11:03.18 | *** join/#asterisk prospektor (4c576337@gateway/web/freenode/ip.76.87.99.55) |
| 11:03.59 | kaldemar | jaxon11247: btw, is http://pastebin.com/22wJcfpW really supposed to be your _whole_ sip.conf? |
| 11:04.30 | jaxon11247 | kaldemar: yes |
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| 11:06.29 | *** part/#asterisk irroot (~gregory@197.173.243.40) |
| 11:14.29 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 11:16.40 | jaxon11247 | Kaldemar>>: I had put this issue in asterisk bug tracker (JIRA) system. But they closed my ticket saying that this is nut a Bug :) |
| 11:17.18 | kaldemar | jaxon11247: yep, it is more of a problem on the proxy side. |
| 11:20.24 | kaldemar | jaxon11247: on the other hand, you didn't provide enough information for them to have any reason to assume that it would be a bug. |
| 11:21.09 | kaldemar | jaxon11247: just wondering where the anonymous.invalid comes from. |
| 11:23.00 | *** join/#asterisk jkroon (~jkroon@196.25.195.42) |
| 11:23.17 | jkroon | is there a way to (from within asterisk dialplan CLI) query the spooldir that asterisk is using? |
| 11:23.25 | schmidts | kaldemar maybe is from here: channels/sip/include/sip.h:#define FROMDOMAIN_INVALID "anonymous.invalid" |
| 11:23.48 | schmidts | jkroon system application |
| 11:24.23 | jkroon | schmidts, how would the System() application help me? It allows to execute external commands, but how does it find the spooldir for me? |
| 11:25.06 | schmidts | jkroon aaahh ok i thought you know this allready ;) i use it like this: http://pastebin.com/AMKHYz32 but imho this doesnt helps you |
| 11:25.13 | kaldemar | schmidts: but how, that's the question. it's usually an ip address. |
| 11:25.53 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 11:26.22 | schmidts | kaldemar privacy settings ;) |
| 11:26.24 | kaldemar | jkroon: ${AST_CONFIG(asterisk.conf,directories,astspooldir)} |
| 11:26.39 | jkroon | kaldemar, thanks! that looks perfect. |
| 11:26.58 | schmidts | row 11825 in chan_sip.c /* if we are not sending RPID and user wants his callerid restricted */ |
| 11:27.26 | jkroon | kaldemar, func_config.c:168 config_function_read: 'astspooldir' not found in [directories] of 'asterisk.conf' |
| 11:27.36 | jkroon | so if astspooldir isn't explicitly set it won't help. |
| 11:28.22 | jmls | ok. |
| 11:28.27 | jmls | 10 installed and running |
| 11:28.38 | jmls | schmidts: set rtp debug gives me loads of the following |
| 11:28.57 | jmls | [Jan 4 11:28:04] Sent RTP packet to <<MySoftphoneIP>>:40018 (type 00, seq 057610, ts 044160, len 000160 |
| 11:29.01 | kaldemar | jkroon: maybe you could IF it somehow. |
| 11:29.28 | schmidts | jmls what i thought, asterisk tries to send it to your 192.168.x.x ip |
| 11:29.49 | jkroon | need to write a piece of dialplan code that goes onto different OS's, and on one it's /usr/local/var/spool/asterisk, on the other it's /var/spool/asterisk ... |
| 11:30.24 | jmls | right. So how do we tell * to use the routerip address instead ? |
| 11:30.58 | schmidts | jmls try externaddr |
| 11:31.39 | jmls | externaddr=publicip of ec2 ? |
| 11:31.45 | schmidts | jmls yes |
| 11:31.49 | jmls | done that :( |
| 11:32.55 | schmidts | jmls try setting media_address=ip of ec2 |
| 11:33.34 | schmidts | but i guess you will still have a problem if this ec2 ip doesnt forward all incoming packages to your box |
| 11:33.51 | jmls | no joy |
| 11:34.15 | schmidts | jmls then maybe you need stun |
| 11:34.22 | jmls | urgh |
| 11:34.31 | jmls | do you recomend any one ? |
| 11:34.37 | schmidts | or just get a real public ip for your asterisk |
| 11:34.42 | jkroon | So I'm guessing I could use kaldemar SYSINFO ... almost. |
| 11:34.59 | jmls | schmidts: it is a *real* public ip |
| 11:35.18 | schmidts | jkroon do you have access to the asterisk.conf file for your application? cause you could set it there |
| 11:35.34 | jkroon | schmidts, that's the point, we're trying to not vary the config files at all. |
| 11:35.42 | jkroon | so we're trying to figure out what the compiled in defaults are |
| 11:35.44 | schmidts | jmls but your asterisk is behind nat so it doesnt have a public ip |
| 11:36.19 | schmidts | jkroon i am not sure, maybe kaldemar knows this but imho you could execute a script at asterisk startup to check this values and just write it into the astdb |
| 11:36.32 | jkroon | hmm, not a bad idea either. |
| 11:37.11 | jmls | schmidts: the point I was trying to make is that you can't get any other type of ip for amazon ec2 |
| 11:37.16 | schmidts | jkroon or just be lazy and do the system check on all possible directories, or atleast do this once on a call and then store the information you find |
| 11:37.31 | kaldemar | jkroon: Noop(${IF($["${AST_CONFIG(asterisk.conf,directories,astspooldir)}x" = "x"]?/var/spool/asterisk:${AST_CONFIG(asterisk.conf,directories,astspooldir)})}) |
| 11:37.55 | jkroon | kaldemar, and if ./configure was run with something other that --prefix=/usr ? |
| 11:38.07 | jkroon | i'll figure something out ... thanks for all the help. |
| 11:39.30 | kaldemar | you could always execute a script but that might not be wanted on a high call volume. |
| 11:40.01 | kaldemar | the last Noop i posted works, it prints errors then astspooldir is not configured in the file though. |
| 11:40.14 | schmidts | kaldemar what i was thinking of is launching a script on the first call after asterisk started and just store the information into astdb, if this information is available dont start the script |
| 11:41.33 | *** join/#asterisk mikal (~mikal@ppp59-167-142-8.static.internode.on.net) |
| 11:41.42 | kaldemar | schmidts: yep, maybe "#exec /path/to/script" in extensions.conf under [globals], where script outputs a global variable with the spool dir as the value. |
| 11:42.12 | schmidts | something like this, yes ;) |
| 11:45.25 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-228-151-20.inter.net.il) |
| 11:45.38 | jmls | schmidts: ooohhhh |
| 11:45.50 | jmls | has some music coming out of the speakers ... |
| 11:46.00 | schmidts | cool ;) |
| 11:46.11 | schmidts | what have you changed jmls? |
| 11:46.48 | jmls | xlite, changed settings in topology to "autodetect firewall traversal method (using ICE)", with server address of stun.counterpath.net |
| 11:48.06 | schmidts | jmls then xlite sends your public ip for rtp out |
| 11:52.06 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 11:55.54 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 11:56.07 | IsUp | hi |
| 11:57.14 | jaxon11247 | anyway thanks for the help..:D |
| 11:59.05 | jaxon11247 | kaldemar: Thanks for spending u r valuable time for us.. |
| 12:00.54 | *** join/#asterisk Nasga (~Nasga@239.212.10.93.rev.sfr.net) |
| 12:04.06 | jmls | weeeeeeeeee ! |
| 12:04.14 | jmls | got the 3cx softphone working as well |
| 12:04.23 | jmls | I had to untick the "register with proxy" option |
| 12:04.47 | jmls | schmidts: thanks so much for all your help this morning. Very much appreciated |
| 12:05.03 | schmidts | jmls your welcome ;) |
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| 12:46.55 | siaulius | kaldemar: do you know why asterisk generates calls for itself to internal sip phones. It makes theese calls continuously in some time intervals? |
| 12:47.58 | kaldemar | siaulius: it doesn't. something else triggers them. |
| 12:49.07 | siaulius | kaldemar: do you know how to find theese triggers? |
| 12:49.46 | schmidts | siaulius you should see these calls in the cli or log, then you could find out what kind of channel starts them (sip or local or something else) |
| 12:54.45 | siaulius | schmidts: theese calls come from DAHDI/29 channel, It is FXO channel, the only one I am connected at the moment. |
| 12:55.37 | kaldemar | siaulius: maybe someone calls you then. |
| 12:57.01 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 12:57.02 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 12:58.01 | siaulius | kaldemar: when I answer I get clear signal, nothing else. I tested this line when I call from my mobile it works and I can comunnicate. |
| 13:01.04 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:01.11 | luckman212 | WARNING[27622] loader.c: Module 'app_flite.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 13:01.14 | luckman212 | argh |
| 13:01.46 | luckman212 | i built a new * using 1.8.9.0-rc1 source, and did the usual cd /usr/src/asterisk-flite && make && make install |
| 13:02.10 | luckman212 | but still getting that warning & flite app will not load -- any ideas on where to go from here? |
| 13:05.26 | *** join/#asterisk tiuman (~ya-online@95.167.25.38) |
| 13:05.29 | tiuman | hi all |
| 13:06.04 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:06.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:06.07 | fenrus | hi |
| 13:06.49 | tiuman | how to determine why the trunk is not registered? |
| 13:07.11 | tiuman | how do you know why the trunk is not registered? |
| 13:07.12 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:08.02 | jacc0 | Tiuman: use 'tshark -R sip' and restart asterisk |
| 13:08.05 | jacco_7564 | what would be a correct way to apply authentication to exten => 1,n,Goto(timeconditions,1,1) ? |
| 13:08.20 | jacc0 | hi jacco :) |
| 13:08.40 | jacco_7564 | Haha, hi |
| 13:09.05 | tiuman | jacc0 |
| 13:09.11 | *** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net) |
| 13:09.17 | tiuman | i restart many try |
| 13:09.39 | jacc0 | what did you see with wireshark? |
| 13:09.47 | [TK]D-Fender | enable SIP DEBUG ast * CLI and do "sip reload" and watch your registration attempt |
| 13:10.01 | jacc0 | that is the asterisk way of doing it i guess ;) |
| 13:10.29 | [TK]D-Fender | Yes, the wasy that tells you what is matched and gives you actual debug information, not just raw packets. |
| 13:10.37 | [TK]D-Fender | And requiring external tools |
| 13:11.00 | tiuman | i type sip debug? |
| 13:11.04 | tiuman | in CLI |
| 13:11.12 | [TK]D-Fender | "sip set debug on" |
| 13:11.14 | tiuman | not found this command |
| 13:11.19 | tiuman | yes |
| 13:11.20 | tiuman | ok |
| 13:14.37 | kaldemar | luckman212: "make clean" |
| 13:14.56 | luckman212 | make clean in /usr/src/asterisk-flite ? |
| 13:15.43 | tiuman | http://pastebin.com/YCcL2wKp |
| 13:15.56 | tiuman | what this problem? |
| 13:16.35 | [TK]D-Fender | tiuman, Contact: <sip:4112391248@192.168.0.10:5060> |
| 13:16.41 | [TK]D-Fender | tiuman, improper NAT setup. |
| 13:16.43 | tiuman | yes |
| 13:16.43 | kaldemar | luckman212: yes |
| 13:16.58 | [TK]D-Fender | tiuman, And as you're using FreePBX, please go to their channel for continued support <- |
| 13:17.00 | [TK]D-Fender | ~freepbx |
| 13:17.01 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:17.02 | tiuman | [TK]D-Fender nat=yes |
| 13:17.08 | [TK]D-Fender | tiuman, Not enough <- |
| 13:19.02 | luckman212 | kaldemar: holy sh** that worked |
| 13:19.59 | tiuman | [TK]D-Fender nat=no |
| 13:20.04 | *** join/#asterisk like_a_horse (~like_a_ho@firect.saao.ac.za) |
| 13:20.12 | tiuman | result no |
| 13:20.13 | [TK]D-Fender | tiuman, #freepbx |
| 13:20.25 | like_a_horse | hey all |
| 13:20.54 | like_a_horse | any know if there is something specific you u need to add to asterisk to get faxing working? |
| 13:21.14 | [TK]D-Fender | like_a_horse, What isn't working for you? |
| 13:21.41 | jacco_7564 | in time conditions, how can i put authentication before the 7006 extension is dialed? just a pincode like 1234? I know its Authenticate(1234) but im not quite sure how its done correctly |
| 13:21.49 | *** join/#asterisk Cadey (5a980315@gateway/web/freenode/ip.90.152.3.21) |
| 13:21.56 | jacco_7564 | exten => 3,1,GotoIfTime(08:30-21:00,*,*,*?from-did-direct,7006,1) |
| 13:22.21 | *** join/#asterisk irroot (~gregory@197.168.236.43) |
| 13:22.49 | Cadey | Hi guys, is there any type of flow dialgrams to show how modules such as voicemail work, which dessisions it has to make etc like is the mail box full blah blah |
| 13:22.50 | *** part/#asterisk irroot (~gregory@197.168.236.43) |
| 13:23.20 | like_a_horse | [TK]D-Fender, i had a normal fax line connected to a printer which now I've just replaced by shoving the cable into our patton smartnode which connects to our asterisk box. I have a fax2email script setup to catch incoming faxes and email them to a single email addy |
| 13:23.28 | [TK]D-Fender | jacco_7564, Authenticate(1234) |
| 13:23.42 | Cadey | im trying to map out the flow of a call you see and when it goes into a module like voice mail I loose what is actualy happening |
| 13:23.51 | jacco_7564 | yes but where should u put that bit of code? |
| 13:24.25 | [TK]D-Fender | jacco_7564, As step 1 <- |
| 13:24.47 | jacco_7564 | so before the gotoiftime? |
| 13:24.59 | ollii | *CLI> core show application Authenticate |
| 13:25.18 | like_a_horse | which doesn't seem to be working.. i know my config is generally correct as the call comes in and goes through the patton and does hit my asterisk box. I see the "ReceiveFAX" application kick in and save the file... hmm.. wait a sec |
| 13:25.27 | [TK]D-Fender | jacco_7564, Depending if you want to PW before checking the time or not. |
| 13:25.40 | jacco_7564 | no |
| 13:26.12 | jacco_7564 | but isnt the destionation if the time matches defined in the gotoiftime? |
| 13:26.38 | [TK]D-Fender | jacco_7564, Then change your GotoIf to jump over a Hangup and onto a priority with your Authenticate before doing a Goto yourself |
| 13:27.34 | jacco_7564 | And how would one do that? |
| 13:27.41 | [TK]D-Fender | ~book |
| 13:27.41 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:27.44 | jacco_7564 | :) |
| 13:28.34 | *** join/#asterisk Nasga (~Nasga@239.212.10.93.rev.sfr.net) |
| 13:32.16 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 13:33.19 | *** part/#asterisk tiuman (~ya-online@95.167.25.38) |
| 13:37.02 | jacco_7564 | Can't i do it like exten => 3,1,GotoIfTime(08:30-21:00,*,*,*?from-did-direct,authenticate(1234),7006,1) so with the authenticate in the time condition? |
| 13:38.26 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 13:39.22 | leifmadsen | why do you have multiple ( ) inside GotoIfTime() |
| 13:39.25 | kaldemar | jacco_7564: you can't make up new syntax as you go. |
| 13:39.30 | leifmadsen | that ^^^ |
| 13:40.01 | jacco_7564 | Well I am very new to this, as you obviously have noticed |
| 13:40.17 | leifmadsen | look at the documentation and sample for GotoIfTime() again |
| 13:40.32 | leifmadsen | true and false conditions are separated with a colon (:) |
| 13:41.39 | leifmadsen | after the ? is: true-context,true-extension,true-label:false-context,false-extension,false-label |
| 13:42.42 | jacco_7564 | ah ok so I am now using the authenticate where the extension should be defined |
| 13:44.04 | [TK]D-Fender | jacoYou will have to have multiple sets to this extension which I already told you. |
| 13:44.20 | prospektor | Hi all, wondering if anyone can give some advice for a voip provider... |
| 13:44.35 | [TK]D-Fender | jacco_7564, :"core show application {appnme]" to read the instructions for each application. |
| 13:45.04 | [TK]D-Fender | prospektor, For calling in what region(s) primarily? |
| 13:45.24 | prospektor | Primarily within the United States. |
| 13:45.54 | jacco_7564 | How do you mean multiple sets? |
| 13:46.02 | prospektor | Specifically I am looking for a provider that has unlimited calling and will also allow me to set my own caller id information. |
| 13:46.06 | [TK]D-Fender | prospektor, voip.ms , vitelity.net , les.net |
| 13:46.36 | [TK]D-Fender | prospektor, "unlimited" pretty much always has limits, and may not be worth the price |
| 13:46.50 | [TK]D-Fender | prospektor, Make sure to properly analyse your projected usage |
| 13:47.18 | prospektor | Ideally I'd like to beable to put sip clients on my android phones and do wifi calling while at home. |
| 13:47.45 | *** join/#asterisk Guest39479 (ionic@ionic.de) |
| 13:49.02 | leifmadsen | IMSdroid |
| 13:49.33 | leifmadsen | it's more likely to be cheaper to not be an unlimited plan |
| 13:49.50 | leifmadsen | per minute I find is much cheaper on the whole |
| 13:50.13 | prospektor | it's been about 5 years since I played around with asterisk, back then I used it primarily for cid spoofing for fun... |
| 13:50.17 | leifmadsen | I use my phone every day as a consultant, calls all over north america, and my bill is usually about $10 a month (and that includes the $3.50 for the DID) |
| 13:50.21 | *** join/#asterisk luke-jr_ (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 13:50.52 | prospektor | but I am thinking I might be able to save enough minutes to cut back on my cell plan |
| 13:51.39 | leifmadsen | possibly |
| 13:51.51 | leifmadsen | don't bother with an unlimited plan though -- you always come out behind |
| 13:52.01 | prospektor | is 1.5 cents/minute pretty standard these days? |
| 13:52.14 | [TK]D-Fender | Or less |
| 13:52.17 | leifmadsen | usually less |
| 13:52.22 | leifmadsen | closer to 1c/min is normal |
| 13:52.27 | leifmadsen | that's what I pay with Unlimitel |
| 13:52.41 | leifmadsen | voip.ms is similar rate (cheaper DIDs though) |
| 13:52.45 | leifmadsen | I think voip.ms is 1.1/min |
| 13:54.00 | prospektor | which providers offer setting cid? or is it pretty much all of them these days? |
| 13:54.36 | leifmadsen | most allow it, some don't |
| 13:55.57 | prospektor | because ideally I'd like to be able to have me, the wife and 2 kids calling through the box while at home and beable to have the outbound cid set accordingly. |
| 13:56.36 | leifmadsen | for me, that just means multiple DIDs, and I place the call out the account for that DID |
| 14:05.32 | *** join/#asterisk serafie (~erin@nat/digium/x-rsanufpnldlftlla) |
| 14:08.53 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 14:10.57 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qsylhwqilhveybah) |
| 14:10.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:16.30 | *** join/#asterisk irroot (~gregory@41.49.200.8) |
| 14:16.30 | luckman212 | anybody using Polycom in here now? having an impossible time getting blind transfers to work Asterisk 1.8.9.0-rc1 + Polycom 3.3.3 fw |
| 14:17.48 | *** part/#asterisk irroot (~gregory@41.49.200.8) |
| 14:17.49 | luckman212 | A---calls--->B, B---blindXfer--->C, B hangs up and "C" never rings, and "A" just has dead air while Asterisk console show 0(Zero) active calls |
| 14:19.10 | [TK]D-Fender | luckman212, Show us |
| 14:20.23 | luckman212 | k |
| 14:23.13 | *** join/#asterisk ilk (~ilk@unaffiliated/ilk) |
| 14:23.23 | luckman212 | i just redacted my cellphone# to 5555555555 but here you go: http://pastebin.com/NqEHZxpd |
| 14:23.29 | garymc | when I drop a call filel into the system, if it goes to an answer phone is this classed as answered? |
| 14:24.42 | kaldemar | garymc: answered as in cdr disposition? |
| 14:24.44 | luckman212 | prospektor: I know for sure that Callcentric does *NOT* allow setting arbitrary CNUM |
| 14:26.32 | prospektor | luckman212: thanks, any idea which providers do? and in the last 5 years did IAX just disappear? |
| 14:26.45 | ilk | wonders why les.net have become so crap? |
| 14:26.56 | luckman212 | prospektor: voipms does |
| 14:27.38 | luckman212 | prospektor: not sure about IAX I never really had the opportunity to use it. voipms claims IAX support as well although when I tested it about a year ago I had some problems and was told to switch to SIP :-/ |
| 14:27.56 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 14:27.57 | prospektor | any restrictions on concurrent connections? I'd like to be able to do conference calling as well for work. |
| 14:28.02 | *** join/#asterisk bowzak (~bowzak@50-76-224-105-static.hfc.comcastbusiness.net) |
| 14:28.37 | ilk | shouldn't les.net be removed from recommended list of canadian ITSPs? |
| 14:28.39 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-vjiuetrgdurmbpbc) |
| 14:28.59 | bowzak | anyone ever have issues running SIP over a VPN? my phones lose registration until i change the port.. then they work for a while.. then drop reg again |
| 14:30.09 | luckman212 | [TK]D-Fender: it's really weird because I am sure this was working a couple of weeks ago. (transfers) |
| 14:30.38 | [TK]D-Fender | luckman212, There is no SIP debug there to show a transfer attempt. I saw 703 "answer" apparently, not thing else is evident |
| 14:30.49 | prospektor | man I feel old and out of it these days... a far cry from when I'd reverse cid scan entire nxx ranges |
| 14:30.52 | luckman212 | I am fairly sure all I've changed is updated asterisk from 1.8.8.0-rc5 --> 1.8.9.0-rc1 |
| 14:31.26 | luckman212 | [TK]D-Fender: ah, sorry I didn't enable SIP debug will do that |
| 14:32.08 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 14:33.26 | garymc | kaldemar : I think so |
| 14:33.51 | *** part/#asterisk ilk (~ilk@unaffiliated/ilk) |
| 14:33.54 | kaldemar | garymc: you think so? |
| 14:34.04 | garymc | i dont know |
| 14:34.56 | *** join/#asterisk mattsqz (~luser@67-61-162-124.cpe.cableone.net) |
| 14:36.53 | garymc | in a clal file you set max retries |
| 14:36.56 | garymc | *call |
| 14:37.26 | garymc | if an answer phone picks up the call, will the call file retry that call? |
| 14:37.50 | IsUp | no |
| 14:37.55 | kaldemar | if the given channel answers, there won't be retries. |
| 14:37.59 | bowzak | <-------------> |
| 14:38.00 | bowzak | --- (12 headers 0 lines) --- |
| 14:38.00 | bowzak | Sending to 192.168.20.63 : 5063 (NAT) |
| 14:38.00 | bowzak | trixbox1*CLI> |
| 14:38.00 | bowzak | <--- Transmitting (NAT) to 192.168.20.63:5063 ---> |
| 14:38.00 | bowzak | SIP/2.0 401 Unauthorized |
| 14:38.01 | bowzak | Via: SIP/2.0/UDP 192.168.20.63:5063;branch=z9hG4bK-88a2db51;received=192.168.20.63 |
| 14:38.01 | bowzak | From: "ST2 Security" <sip:239@192.168.10.10>;tag=a5ec9f9271c5cc0do0 |
| 14:38.02 | bowzak | To: "ST2 Security" <sip:239@192.168.10.10>;tag=as753bdbc5 |
| 14:38.02 | bowzak | Call-ID: 33345a9a-e2860ba9@192.168.20.63 |
| 14:38.03 | bowzak | CSeq: 10461 REGISTER |
| 14:38.03 | bowzak | User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78 |
| 14:38.04 | bowzak | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO |
| 14:38.04 | bowzak | Supported: replaces, timer |
| 14:38.05 | bowzak | WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="46681024" |
| 14:38.05 | bowzak | Content-Length: 0 |
| 14:38.07 | IsUp | BEEEEEEEEEEH |
| 14:38.13 | acidfoo | head shot |
| 14:38.15 | kaldemar | bowzak: use pastebin please. |
| 14:38.18 | kaldemar | ~pb |
| 14:38.18 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:38.31 | bowzak | sorry.. noted |
| 14:38.53 | mattsqz | now bowzak, change your nick to ballzak for the remainder of your visit :P |
| 14:39.54 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 14:45.56 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:50.07 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 14:52.32 | garymc | kaldemar so how do call centres get around that issue? |
| 14:55.35 | [TK]D-Fender | garymc, What issue? There is nothing to "retry" if it answers |
| 14:57.41 | *** join/#asterisk CyfordTechnologi (~allen@184-77-45-149.gar.clearwire-wmx.net) |
| 15:01.09 | *** join/#asterisk funpunx (daemon@173-15-183-209-BusName-Philadelphia.hfc.comcastbusiness.net) |
| 15:05.44 | garymc | i mean if answer phone answers, as thats uselesss we need to try them again |
| 15:06.19 | garymc | unless we leave voice message, but if it answers and we place them in queue to be spoken too then the answer message will just contain our hold music etc |
| 15:06.27 | [TK]D-Fender | <garymc> i mean if answer phone answers, as thats uselesss we need to try them again <- huh? |
| 15:06.30 | garymc | or hold messgae |
| 15:06.48 | *** join/#asterisk cyford (~allen@184-77-45-149.gar.clearwire-wmx.net) |
| 15:06.48 | garymc | yeah we cant sell to an answer phone |
| 15:07.01 | garymc | the answer phone wont habd your card details over :P |
| 15:07.06 | garymc | *hand |
| 15:07.40 | [TK]D-Fender | garymc, And how does it know this has happened? |
| 15:08.06 | mattsqz | could you use the message tone as an indicator |
| 15:08.26 | garymc | How many call files can be dropped into the OUTGOING folder first without breaking the system? |
| 15:08.45 | [TK]D-Fender | garymc, Depends on your system and what it will be doing with those |
| 15:10.05 | garymc | how does the system know that its an answer phone???? That is what im asking, that must just appear to be an answered call no? |
| 15:10.11 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 15:10.39 | kaldemar | "answer phone" - what? |
| 15:11.01 | [TK]D-Fender | garymc, "answering machine". Get it right |
| 15:11.19 | [TK]D-Fender | garymc, garymcand that's what AMD() is for. This is YOUR job in the dialplan |
| 15:11.41 | [TK]D-Fender | garymc, Dial + M() + app_amd |
| 15:11.59 | mattsqz | voicemail, the artist formerly known as electronic microcassette telephone answering system |
| 15:12.11 | garymc | right |
| 15:14.54 | *** join/#asterisk CyfordTechnologi (~allen@184-77-45-149.gar.clearwire-wmx.net) |
| 15:19.53 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 15:26.31 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:26.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:27.34 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 15:33.55 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 15:35.15 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 15:36.02 | funpunx | i'm having what appear to be timing slips when connected to another piece of equipment via T1 PRI. i've tried modifying my signaling back and forth acting as CPE any NET with no change in behavior |
| 15:36.20 | funpunx | suggestions on where next? |
| 15:37.32 | kaldemar | is clocking properly set in dahdi? |
| 15:37.49 | [TK]D-Fender | funpunx, suggest you fill in all the details as to your server hardware OS & environment, * versions, etc. |
| 15:38.07 | WIMPy | Is exactely one side providing timing? |
| 15:38.18 | [TK]D-Fender | funpunx, Show card configs, give us proper details of what you're connecting to.... |
| 15:40.15 | funpunx | centOS, asterisk 1.6.1 (IIRC) not sure the dahdi version off the top of my head |
| 15:40.32 | funpunx | connecting to Dialogic DMV1200BTEP cards on the other side |
| 15:40.37 | funpunx | and 1 PSTN trunk |
| 15:40.45 | funpunx | the connections to dialogic are what slip consistently |
| 15:40.47 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:40.52 | funpunx | chan_dahdi.conf: http://pastebin.com/TgsKz88q |
| 15:41.20 | funpunx | was originally connected to an older set of Dialogic cards and worked fine, they're trying to upgrade the hardware on the other side |
| 15:41.53 | funpunx | i have a call into Dialogic about this as well but i want to rule anything out on the PBX side while i wait to hear back |
| 15:42.03 | kaldemar | pastebin system.conf for DAHDI aswell |
| 15:42.11 | [TK]D-Fender | funpunx, system.conf , card details, server specs, "cat /proc/interrupts",etc.... |
| 15:43.25 | funpunx | will have shortly |
| 15:45.09 | mattsqz | can i use the 's' extension to initiate call recording via 'monitor' ? |
| 15:45.47 | mattsqz | like s,n,Monitor(wav,m) |
| 15:47.13 | kaldemar | if your call lands in the extension |
| 15:47.24 | ollii | _X.,1,Monitor(wav,m) ; 4123123,2,Dial(SIP/foo) ; 49499494,2,Dial(SIP/bar) |
| 15:47.29 | ollii | that might work |
| 15:48.14 | funpunx | dahdi system.conf: http://pastebin.com/dFV3zLQn |
| 15:48.54 | funpunx | output of dahdi_hardware: http://pastebin.com/iF02JbWi |
| 15:48.54 | [TK]D-Fender | mattsqz, "s' doesn't just happen. |
| 15:49.09 | [TK]D-Fender | mattsqz, and your sample call to monitor didn't include the filename |
| 15:49.11 | mattsqz | hmm that looks like jibberish to me still, sorry. im still learning the basics here. im trying to record every call that goes through, and i read here that 's' is used when starting a call |
| 15:49.34 | funpunx | output of uname -a: 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386 GNU/Linux |
| 15:49.46 | kaldemar | funpunx: your span 2 should be providing timing to the dialogic side if you're the one who's connected to PSTN. so span=2,0,0... |
| 15:50.00 | funpunx | any other specs on the box? it's a dual core xeon, not sure of the generation, 2 or 4 gigs of RAM |
| 15:50.15 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 15:50.39 | mattsqz | do i set a seperate monitor line for each extension then? |
| 15:50.54 | funpunx | we tried doing that and we still had slips. and of course, the dialogic side is also connected to the PSTN so that introduces some possibilities |
| 15:51.02 | WIMPy | funpunx: Do you have a timing cable installed? |
| 15:51.20 | [TK]D-Fender | mattsqz, Depends how you made your dialplan. |
| 15:51.49 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 15:51.57 | funpunx | WIMpy: i'm not on site so i can't confim. is there any way to confirm from the command line? |
| 15:52.05 | mattsqz | well, the previous guy has a subfolder with about 25 .conf files in it, and these are pointed to by extensions.conf |
| 15:52.36 | mattsqz | so this isnt as easy as it could be, with everything split up like that |
| 15:52.55 | WIMPy | funpunx: I'm not sure, but dmesg after loading the drivers could give a clue. |
| 15:53.14 | mattsqz | this is a small customer service center, less than 10 employees. 99% of our calls are incoming. |
| 15:53.42 | [TK]D-Fender | mattsqz, Guess you'd better go figure out what you've got. |
| 15:54.09 | [TK]D-Fender | <mattsqz> this is a small customer service center, less than 10 employees. 99% of our calls are incoming. <- because these details don't tell us where to put stuff in your dialplan |
| 15:54.13 | mattsqz | in the process. just looking for a little help with deciphering these exten => lines. |
| 15:54.28 | funpunx | could this be a problem? timing source auto card 0! timing source auto card 0! timing source auto card 0! timing source auto card 1! |
| 15:54.41 | WIMPy | funpunx: Are your lines all from the same telco? |
| 15:54.42 | [TK]D-Fender | mattsqz, Dialplan is very simplistic as "programming" goes |
| 15:54.55 | mattsqz | i see many that have the extension number first, many that start with s, and some that have underscores and . instead |
| 15:55.24 | [TK]D-Fender | ~book |
| 15:55.24 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:55.26 | [TK]D-Fender | ^^^ |
| 15:55.26 | funpunx | WIMPy: negative on that. the trunks on the Dialogic side are a different provider than the trunk on the * side |
| 15:55.32 | *** join/#asterisk navaismo (~navaismo@187.170.1.107) |
| 15:55.33 | WIMPy | I'm not even sure what timing combinations dahdi is capable of doing. |
| 15:55.50 | mattsqz | i have that open right now and its not explaining it very well. |
| 15:55.53 | WIMPy | So they might be asynchromous. |
| 15:56.51 | [TK]D-Fender | mattsqz, calls come in from a device into a context. Typically those are dialed numbers. It will match in that context and where it goes from there is up to you. |
| 15:57.28 | [TK]D-Fender | mattsqz, You can jump to other extens including "s" which has some historical named purposes, etc |
| 15:57.31 | [TK]D-Fender | ~stdextens |
| 15:57.31 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 15:58.04 | mattsqz | ok. THAT is what i needed. |
| 15:58.04 | [TK]D-Fender | mattsqz, Typically aclls from ITSP's, PRI, phones, etc all have actual target #'s to start with. |
| 15:58.27 | [TK]D-Fender | mattsqz, "_" at the start of an exten says that what follows is a pattern, not "literal". |
| 15:58.49 | [TK]D-Fender | mattsqz, hence things like "exten => _NXXNXXXXXX,1,abcdefg()" |
| 15:58.59 | mattsqz | so what is _X.? |
| 15:59.13 | [TK]D-Fender | mattsqz, so that those aren't littler letters of the alphabet your system is expected to be seeing coming in. |
| 15:59.44 | [TK]D-Fender | mattsqz, _X. = Is a pattern. Starts with a numeric digit. Has one or more charaters of any type following. |
| 16:00.05 | [TK]D-Fender | ".' = 1+ of anything |
| 16:00.09 | mattsqz | so X is numeric? |
| 16:00.13 | [TK]D-Fender | mattsqz, Read up on your pattern basics |
| 16:00.16 | [TK]D-Fender | X = any digit |
| 16:00.24 | mattsqz | N = ? |
| 16:00.46 | [TK]D-Fender | non-0 |
| 16:01.20 | [TK]D-Fender | mattsqz, Go read up on the patterns section again. |
| 16:01.25 | mattsqz | ok. |
| 16:01.31 | mattsqz | thanks |
| 16:01.41 | WIMPy | Or the sample dialplan. |
| 16:02.35 | mattsqz | is that the extensions.conf.sample file? |
| 16:02.45 | WIMPy | yes |
| 16:03.04 | mattsqz | i had to download a tarball to find it, this machine has had all that stuff removed |
| 16:03.54 | *** part/#asterisk stix (~stix@193.89.191.209) |
| 16:06.09 | *** join/#asterisk cyford (~allen@184-77-45-149.gar.clearwire-wmx.net) |
| 16:06.48 | leifmadsen | X = 0-9, N = 2-9, Z = 1-9 |
| 16:07.01 | leifmadsen | . = 1 or more character, ! = 0 or more characters |
| 16:07.45 | WIMPy | And ! will match immediately if unique. |
| 16:07.53 | mattsqz | and extensions can be called by names, emails, mac addresses, numbers, etc? |
| 16:08.12 | WIMPy | Whatever you want. |
| 16:08.17 | mattsqz | cool |
| 16:08.38 | WIMPy | But it you want to be able to dial them from a phone you should stick with what you find on the keypad. |
| 16:11.50 | [TK]D-Fender | mattsqz, an extension is what is dialed. |
| 16:12.14 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 16:13.00 | elliot98 | what is the reliability of SNOM phones? Some dropped calls are occurring and wondering if the SIP phones may be contributing to it |
| 16:13.23 | ollii | elliot98: which firmware version are u using? |
| 16:13.33 | IsUp | what you guys think about Linksys SPA* phones? |
| 16:13.52 | WIMPy | The only phone that has ever randomly dropped calles on me was the Allnet 9760. |
| 16:14.09 | elliot98 | where do you check firmware version? |
| 16:14.17 | WIMPy | IsUp: Fits perfectyl. I heve them drop calls at session timer. |
| 16:14.28 | WIMPy | elliot98: Press ? |
| 16:15.40 | IsUp | WIMPy: okay, if u remember i was talking about a setup in an office. i know that grandstream "sucks" but, they'll buy 15x grandstream GXP280 phones for employees |
| 16:15.58 | IsUp | WIMPy: and they will buy 3x Linksys SPA962 phones for management |
| 16:16.04 | mattsqz | ok, so it looks like i need to muck about in the dialplan and find the portion where calls are sent out from the ivr/aa to the individual extensions, and have Monitor() begin recording at that point |
| 16:16.26 | WIMPy | IsUp: The 962 is great as a desk lamp. |
| 16:16.38 | WIMPy | Make sure you supply sunglasses with them. |
| 16:17.08 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:17.34 | IsUp | WIMPy: haha. they are asking me about "conference". just to be sure, conference can be handled by PHONE or by Asterisk (MeetMe), is that right? |
| 16:17.34 | mattsqz | grandstream isnt the highest quality phone. the PoE port on the gxp2000's will stop functioning if you sneeze on it. ive had to replace that daughterboard in 6 phones here. |
| 16:17.48 | WIMPy | I've never used that thing. A friend asked if he ycould try it and sonn realized why I said, that he probably didn;t want to. |
| 16:18.10 | IsUp | mattsqz: we have 24x gxp2000 in our office. works well. not great but its ok. |
| 16:18.14 | WIMPy | IsUp: yes. But I'd let Asterisk do it. |
| 16:18.41 | mattsqz | IsUp: how long have you had them? when they are working, they work fine. |
| 16:18.55 | IsUp | WIMPy: okay and whats the proper way to do it? should i tell them "dial 123 for conference room"? |
| 16:19.10 | ollii | e.g. |
| 16:19.23 | IsUp | mattsqz: almost 3 years. also we have 5x gxv3000 |
| 16:19.41 | WIMPy | IsUp: Yes, you have to transfer the caller to a conference room. |
| 16:19.53 | WIMPy | Bad side effect: You can;t get them back. |
| 16:20.09 | WIMPy | Not from the phone at least. |
| 16:20.20 | [TK]D-Fender | IsUp, Where are you located? |
| 16:20.27 | IsUp | WIMPy: yeah. honestly we are running Asterisk in our office too. but we are just making internal calls. i dont have much experience with "real" setup. |
| 16:20.41 | IsUp | [TK]D-Fender: im in Turkey if thats what you asking :p |
| 16:21.04 | IsUp | *office setup actually. |
| 16:21.11 | [TK]D-Fender | IsUp, Then Linksys SPA phones are probably the best dollar value phones in your area |
| 16:21.19 | [TK]D-Fender | IsUp, With quality in mind |
| 16:21.20 | mattsqz | the only grandstreams we have are the gxp2000. perhaps we got a bad batch, we got a box of around 20 phones with incremental mac addresses. so far just under half are dead. and we only have a need for 10 phones at a time. |
| 16:22.04 | WIMPy | For me Snom has been the least shitty ones. |
| 16:22.04 | IsUp | [TK]D-Fender: Linksys SPA962 330$ with taxes |
| 16:22.15 | [TK]D-Fender | IsUp, Polycom > All, but the ROI is different depending where you are. In North America there is no general quality reason to pick anything else. Some specialized needs warrant other specific models, but no so much any general use |
| 16:22.33 | [TK]D-Fender | IsUp, And what are you going to do with this phone? That's a lot of money... |
| 16:23.12 | IsUp | [TK]D-Fender: well they are just asking me for "executive" phones. i think its about ego. so i dont mind :> |
| 16:23.45 | *** join/#asterisk bmg505 (~leon@196-209-171-251.dynamic.isadsl.co.za) |
| 16:23.54 | [TK]D-Fender | IsUp, Ok, the need is "ego". Sure, coloour it up and all... get them a highly unnecessary side-car as well |
| 16:24.13 | [TK]D-Fender | IsUp, Then you can tell them .. this is what we gie underlings like your RECEPTIONIST :p |
| 16:24.14 | mattsqz | well if its about ego, pick up some of those new avaya android desktop sip devices. |
| 16:24.40 | [TK]D-Fender | mattsqz, Avaya = crap |
| 16:24.43 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 16:24.57 | WIMPy | For ego I'd go for the completely senseless 8xx Snoms with touch screen ot the Siemens OpenStage things with the giant displays. |
| 16:25.11 | [TK]D-Fender | mattsqz, top-end Cisco GigaBit colour videophone.. then we're talking... |
| 16:25.18 | [TK]D-Fender | VIDEO <- |
| 16:25.49 | IsUp | [TK]D-Fender: :D |
| 16:26.26 | IsUp | i was talking to local resellers of Digium, i cant find fxo/fxs modules. im going to use Sangoma A400 |
| 16:26.42 | IsUp | they are all out-of-stock |
| 16:26.47 | elliot98 | any way to see which firmare is being in the GUI? |
| 16:26.48 | [TK]D-Fender | IsUp, What port mix are you looking for? |
| 16:27.02 | IsUp | [TK]D-Fender: 4 FXS + 4 FXO |
| 16:27.05 | WIMPy | elliot98: I told you: Press ? |
| 16:27.17 | elliot98 | WIMPy: phone is remote |
| 16:27.33 | [TK]D-Fender | IsUp, Whats the FXS for? |
| 16:27.38 | WIMPy | Then use the web I/F |
| 16:27.44 | elliot98 | ok |
| 16:28.04 | WIMPy | 'd rater ask whe the FXO is for. You're not in a 3rd world country. |
| 16:28.06 | IsUp | [TK]D-Fender: just for future actually. ill just use 3 FXO |
| 16:28.14 | WIMPy | s/whe/what/ |
| 16:28.24 | [TK]D-Fender | IsUp, Serious, what are you planning on putting on those? |
| 16:28.35 | elliot98 | also, is necessary to run an Answer command in dialplan if the dialplan is dialing an extension? |
| 16:28.56 | elliot98 | does * need to answer the call before dialing to another UA? |
| 16:28.59 | [TK]D-Fender | elliot98, Answer is required if you want * to answer the calling side first. |
| 16:29.00 | WIMPy | elliot98: Yoy usually don't want that. |
| 16:29.09 | [TK]D-Fender | elliot98, and you dial a device, not an "extension" |
| 16:29.26 | [TK]D-Fender | " does * need to answer the call before dialing to another UA?" <- no |
| 16:29.33 | [TK]D-Fender | Will you want to? Maybe |
| 16:29.34 | IsUp | [TK]D-Fender: well like i said, just for future. GSM gateway/FCT maybe. |
| 16:29.52 | mattsqz | avaya may be crap.. really any android tablet with a front facing camera is the same thing. |
| 16:29.59 | elliot98 | because was having an issue that RTP packets were not being sent until progressinband was set |
| 16:30.00 | IsUp | also we dont have much option. our provider is unable to send calls with SIP. i can use PRI but it costs 1300$ per month |
| 16:30.02 | [TK]D-Fender | IsUp, If you're thinking audio, then don't. Just get an ATA when the time comes. FAR more flexible and cost-effective. |
| 16:30.12 | [TK]D-Fender | IsUp, Get a B600d for the card <- |
| 16:30.31 | WIMPy | IsUp: Don't you have BRIs? |
| 16:31.02 | IsUp | WIMPy: no. we dont have BRI in Turkey. also i dont have any option between "providers" because we just have 1 provider :) |
| 16:31.05 | elliot98 | even though SIP should proper INVITE/200/ACK negotiation, the UA was sending RTP packets until some sort of inband connection was set up |
| 16:31.24 | elliot98 | s/should/has |
| 16:31.25 | IsUp | [TK]D-Fender: you are probably right but card is ordered already :/ |
| 16:31.26 | WIMPy | IsUp: Oh, I thought the situation was better there. |
| 16:31.42 | IsUp | i'll go for setup tomorrow |
| 16:31.50 | [TK]D-Fender | IsUp, Might want to canvas for idas a bit more first :) |
| 16:31.52 | [TK]D-Fender | ideas* |
| 16:32.18 | IsUp | [TK]D-Fender: right |
| 16:33.12 | IsUp | i cant decide about * version. also GUI. because ill just go for setup they will give me IVR chart, i'll write dialplan |
| 16:34.37 | IsUp | but i am mostly scared about analog lines. i hope i can do it without any problem |
| 16:36.36 | elliot98 | the firmware is snom320-SIP 7.3.30 6059 |
| 16:36.41 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:37.01 | elliot98 | any issues with this version of firmware? |
| 16:37.27 | mattsqz | geez [TK]D-Fender, thanks for the quick rundown earlier. i learned more from the asterisk manual in the last 10min than i did all day yesterday, after just having some things simplified for me :) |
| 16:37.34 | IsUp | [TK]D-Fender: what you can recommend for * version? im thinking about 1.4 |
| 16:38.13 | [TK]D-Fender | IsUp, 1.8 until * 10 has just a little more time to see that it's stable |
| 16:38.25 | [TK]D-Fender | IsUp, 1.8 = LTS, everything else = disco |
| 16:41.03 | IsUp | [TK]D-Fender: okay. well ill do more research for tomorrow. they'll send me IVR flow, ill write dialplan today. i have some time |
| 16:43.13 | [TK]D-Fender | IsUp, I am referring to everyone's time, not just yours... It's still less than 1 month old. I'd say wait till opinions immediately following the first fix-release for it before considering the branch. |
| 16:52.19 | IsUp | okay |
| 16:52.28 | IsUp | thanks for help WIMPy and [TK]D-Fender |
| 16:52.32 | IsUp | i gotta go, take care! bbl |
| 16:52.54 | [TK]D-Fender | IsUp, But I would probably want to make sure that code & configs being written are as upwardly compatible as possible and to know what is waiting in * !) to prepare for. |
| 16:53.04 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-202.mobile.uci.edu) |
| 16:57.29 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-202.mobile.uci.edu) |
| 17:06.44 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
| 17:17.10 | p3nguin | |
| 17:17.29 | r0m|u | waz up p3nguin |
| 17:17.35 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 17:17.49 | r0m|u | the damn 501 went belly up.... |
| 17:18.09 | r0m|u | all I get is the redlight when booting up.... |
| 17:18.10 | edge | How do I select a specific DAHDI port to dial to? I want to select the 1 DAHDI card and the 4th port to dial out of. |
| 17:18.38 | *** join/#asterisk bmg505 (~leon@196-209-10-111.dynamic.isadsl.co.za) |
| 17:19.14 | r0m|u | I am trying to recover it but I get nothing all I get is the damn red light to lit up.... |
| 17:21.06 | [TK]D-Fender | edge, Dial(DAHDi/channelnumber/numbertodial) |
| 17:21.10 | p3nguin | That's a pretty shitty deal. |
| 17:23.00 | r0m|u | I was having issues with it..... but I didnt think it will just die... |
| 17:23.15 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 17:23.43 | r0m|u | I would make calls and not hear any body but they could hear me.... If I would pick up the hand set than I could hear them... |
| 17:23.59 | r0m|u | but the speaker was shoot.... |
| 17:24.05 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 17:24.42 | edge | [TK]D-Fender channelnumber is just 4 then right? no - (dash) |
| 17:24.52 | *** join/#asterisk batfastad (~benb@93-97-161-95.zone5.bethere.co.uk) |
| 17:25.04 | [TK]D-Fender | edge, If you have 1 card then likely, yes |
| 17:25.26 | [TK]D-Fender | edge, Dial(DAHDI/1/5551212) |
| 17:32.34 | *** join/#asterisk Micc (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 17:32.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:32.54 | Micc | can I reload chan_sip while there are calls using it? |
| 17:33.29 | [TK]D-Fender | yes |
| 17:35.55 | Micc | wow thats amazing. |
| 17:36.19 | leifmadsen | changes won't affect existing calls |
| 17:36.29 | leifmadsen | just don't unload then load |
| 17:37.00 | p3nguin | "sip reload" amazes me sometimes, too. |
| 17:39.21 | Micc | I don't think it took my changes. |
| 17:39.32 | p3nguin | What change are you looking for? |
| 17:39.45 | carrar | less gov |
| 17:40.05 | p3nguin | Asterisk doesn't have that feature, yet. |
| 17:40.36 | carrar | maybe Russel can get that on the list |
| 17:40.50 | Micc | I made a change in initreqprep |
| 17:41.26 | p3nguin | Never heard of it. |
| 17:42.04 | p3nguin | There's no mention of it in the sample sip.conf. |
| 17:44.16 | carrar | changing the source code related to callerid? |
| 17:45.48 | leifmadsen | carrar: which Russell? Russell Bryant doesn't do Asterisk anymore |
| 17:45.57 | carrar | oh |
| 17:45.59 | Micc | changing how it uses the reg contact. It should be trusting fullcontact in this case. I must have made a typo in the code. |
| 17:46.04 | carrar | Where did he go? |
| 17:46.09 | leifmadsen | Red Hat |
| 17:46.15 | leifmadsen | you didn't read his blog I guess |
| 17:46.17 | carrar | wow |
| 17:46.19 | leifmadsen | russellbryant.net |
| 17:46.22 | *** part/#asterisk batfastad (~benb@93-97-161-95.zone5.bethere.co.uk) |
| 17:46.23 | carrar | blags |
| 17:46.23 | leifmadsen | this was like 2+ months ago |
| 17:46.24 | carrar | haha |
| 17:46.26 | carrar | naw |
| 17:46.36 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:47.27 | carrar | life gets in the way of reading blogs |
| 17:47.53 | carrar | who took his place? |
| 17:48.20 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-374596.home.otenet.gr) |
| 17:48.28 | p3nguin | If you changed the chan_sip.c, sip reload is going to do nothing for it. You have to recompile chan_sip.so, unload the old one, install the new one, then load the new one. |
| 17:53.37 | Micc | well that explains it then. |
| 17:54.03 | Micc | so really I have to do a restart |
| 17:54.08 | p3nguin | no |
| 17:54.12 | Micc | I'm gonna loose my calls. |
| 17:54.17 | Micc | so its the same thing. |
| 17:54.18 | p3nguin | You just have to do what I said. |
| 17:54.59 | p3nguin | But yes, you will not be able to make or keep calls during the period where you unload the old one to the time you load the new one. |
| 17:55.15 | p3nguin | If you're quick, you can do it in a few seconds. |
| 17:55.57 | Micc | so it is possible to unload while calls are happening? |
| 17:56.40 | p3nguin | You could install the new module and run "core restart gracefully". |
| 17:56.54 | Micc | it has use count of 34 |
| 17:57.27 | p3nguin | Putting the new module into place should not affect anything because the module is already loaded and in RAM. |
| 17:58.15 | p3nguin | I would copy the new module into the modules directory and run "core restart gracefully" or "core restart when convenient" and when there are no more calls, asterisk should restart and it will be using the new module. |
| 17:58.20 | carrar | unless there is a typo |
| 17:58.25 | carrar | in the source |
| 17:58.45 | p3nguin | Unless there is a typo. Then you're screwed. |
| 17:58.45 | carrar | always best to make changes in the middle of the day when it's busy |
| 17:59.21 | Micc | yeah I know. I was supposed to do this last night, but I forgot, so I'm kinda screwed. |
| 17:59.44 | p3nguin | And I guess you don't have a test system to make sure your module works. |
| 17:59.56 | Micc | I do. |
| 18:00.07 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:00.11 | p3nguin | Problem solved. |
| 18:00.49 | Micc | with a little down time. |
| 18:00.51 | carrar | as long as you only have no more then 1 call from each company active you can restart the system :) |
| 18:01.18 | p3nguin | Fix up your chan_sip.c, recompile chan_sip.so, install it, test it. If it is good, install it in the production system. |
| 18:01.22 | Micc | carrar, thats funny because its almost true. |
| 18:01.38 | carrar | heh |
| 18:02.40 | Micc | 10am is peak. But noon is slow, so thats probably my window. |
| 18:03.04 | p3nguin | If you've tested it in your test system, what is the problem? |
| 18:03.20 | p3nguin | Test it. If it is good, put it on the production system and restart when convenient. |
| 18:03.59 | Micc | because restart when convenient still takes time. |
| 18:04.13 | Micc | and there will always be a few calls going on during the day. |
| 18:04.15 | p3nguin | restart gracefully was my first suggestion. |
| 18:04.44 | Micc | calls will fail over to the backup if the primary stops responding. |
| 18:04.51 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:04.56 | p3nguin | That's even better. |
| 18:05.15 | p3nguin | So now there is really no downtime at all. |
| 18:05.18 | Micc | but it will take time for customers to reregister on the backup, then time to swap back when its up again. |
| 18:05.53 | p3nguin | How long do you really anticipate it will take for a graceful restart to take? |
| 18:05.59 | Micc | theres only no downtime if I had the backup server take over the ip address. |
| 18:06.14 | carrar | or bleed them over |
| 18:06.42 | Micc | carrar, what do you mean by bleed them over? |
| 18:06.51 | carrar | don't accept new calls |
| 18:06.56 | Micc | max call length right now is about 15 minutes. |
| 18:06.57 | carrar | force them over |
| 18:07.15 | Micc | right, thats what gracefuly does, right? |
| 18:07.29 | p3nguin | graceful does stop new calls, yes. |
| 18:08.29 | p3nguin | Or if you use when convenient, asterisk will wait for a time when there are no calls. |
| 18:09.05 | p3nguin | And the down time will be a few seconds while asterisk starts back up. |
| 18:09.33 | Micc | If I wait till noon theres won't be much going on, I could do it then without many people noticing. |
| 18:11.26 | Micc | I hate telling customers the bad news. Its all fun when you get to solve there problem with a few edits and a dialplan reload. |
| 18:12.19 | p3nguin | If it were me, I would probably use gracefully during the lull of the day. |
| 18:12.30 | p3nguin | New callers can wait a few minutes. |
| 18:12.41 | coppice | the should be a "sod 'em" option |
| 18:14.59 | [TK]D-Fender | coppice, feel's "sod 'em"-y to me ;) |
| 18:15.53 | coppice | [TK]D-Fender: every building needs a telecoms shaft |
| 18:16.38 | p3nguin | coppice: There is. It is the "core restart now" command. |
| 18:21.59 | Micc | I could change the dns to the backup server, but the ttl is 15 minutes. |
| 18:25.21 | p3nguin | Or you could let new calls wait while old calls die off. |
| 18:26.15 | p3nguin | When they get a congestion tone, they'll try again in a few minutes. |
| 18:27.16 | Micc | I have too many fail over servers that handle traffic for other servers. |
| 18:27.32 | Micc | it would be too much work to do that, but that would be a good idea if I could do that just for the DIDs on this one server. |
| 18:28.12 | Micc | actually, thats what I'll do once it dies down. should only take a few minutes. |
| 18:28.53 | Micc | if I kill the backup server too, it should fail on most. some have fail over numbers in the database that it will forward to, but most should just be busy. |
| 18:29.48 | *** join/#asterisk JTLove (~JTLove@129.112.244.66.dsi.smithvilledigital.net) |
| 18:30.44 | JTLove | hi |
| 18:30.51 | JTLove | all |
| 18:33.10 | JTLove | Needing some help on Freebpx installation |
| 18:33.22 | leifmadsen | you'll need to see the #freepbx channel |
| 18:33.26 | WIMPy | ~freepbx |
| 18:33.26 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:33.31 | leifmadsen | this channel is for vanilla Asterisk help |
| 18:33.48 | *** join/#asterisk mito (~mito@186.59.36.160) |
| 18:33.53 | JTLove | ok thanks |
| 18:34.06 | JTLove | Just trying to find my way in this |
| 18:34.42 | JTLove | I have already gotten the vanilla Asterisk going but I thought I would try to incorporate the web interface |
| 18:34.42 | coppice | leifmadsen: its not vanilla only. many people find a rocky road in here |
| 18:34.43 | mito | hi, i can send sms via sip.voicetrading.com whit my account from linux bash script? sorry my english |
| 18:34.49 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 18:34.51 | Qwell | JTLove: Why try installing FreePBX yourself? There are plenty of Asterisk distros that handle all of that for you. |
| 18:34.59 | leifmadsen | coppice: well played :) |
| 18:35.01 | Qwell | coppice: Those people are wrong. |
| 18:35.09 | Qwell | oh, I see now. |
| 18:35.12 | Qwell | NEVERMIND |
| 18:35.14 | [TK]D-Fender | JTLove, You don't "incorporate" a GUI like FreePBX. It's more like selling your soul to the lowest bidder. |
| 18:35.34 | nny | quick Q, which application should I be looking at if I want to create a star code that sends the current channel to a recording/disconnects callee? |
| 18:35.50 | nny | Bridge? |
| 18:35.52 | Qwell | vim |
| 18:36.37 | JTLove | [TK]D-Fender, what would you propose for GUI for Asterisk |
| 18:36.57 | [TK]D-Fender | JTLove, FreePBX until it proves insufficient. |
| 18:38.08 | Micc | TKD, hasn't it proved insufficient already? |
| 18:38.21 | JTLove | [TK]D-Fender, how insufficient is FreePBX? Or should I first try it first |
| 18:38.23 | [TK]D-Fender | Micc, For whom? |
| 18:38.30 | JTLove | I am a first trial |
| 18:38.45 | [TK]D-Fender | JTLove, Criminal or civil? |
| 18:38.53 | *** part/#asterisk mito (~mito@186.59.36.160) |
| 18:39.07 | leifmadsen | nny: look at features.conf |
| 18:39.17 | leifmadsen | transfer to an exntesion that does that |
| 18:39.35 | Micc | We took over another company's freepbx server a couple weeks ago. I really hate it. |
| 18:39.41 | leifmadsen | or just use the transfer key on your SIP phone to transfer to some extension that handles that -- you could also BLACKLIST() or whatever if you wanted |
| 18:39.56 | leifmadsen | I refuse to work with companies that use FreePBX -- I tell them to go somewhere else |
| 18:40.20 | leifmadsen | making changes to the system typically take 3-4 times as long if the system can't handle it (and it usually doesn't if they are calling a consultant) |
| 18:40.37 | leifmadsen | 3-4x as long versus standard configuration file changes |
| 18:40.58 | leifmadsen | FreePBX works fine up to the point you want it to do something it doesn't |
| 18:42.46 | Micc | leifmadson, that sounds about right. We'll be moving any customers that want changes to our regular servers. |
| 18:43.36 | nny | leifmadsen: I like the simplicity, but assuming I can't hard code the extension. I have created custom features before, would using a custom context with Transfer to the extension be smart still? |
| 18:53.58 | nny | hmm |
| 18:53.59 | nny | close |
| 18:54.24 | nny | http://pastebin.com/UpyiNGFD |
| 18:54.41 | nny | if i use Dial instead of Transfer it works, but Dial keeps both channels open |
| 18:55.08 | nny | the feature code (and the proper channel (peer and user (self)) work great |
| 18:56.27 | nny | leifmadsen: ha I have a client who INSISTS on using Fpbx for 200 concurrent call system. You won't believe the amount of crap that one generates. My vanilla clients make me feel like the maytag repair man |
| 18:56.38 | tzanger | Don't remember if I asked this or not, but is asterisk's NOTIFY support capable of notifying about transfers and such, or is it strictly device state |
| 18:56.54 | nny | it's not fpbx's fault though, it clearly says not suggested for anything above soho etc |
| 18:57.52 | *** join/#asterisk jkroon (~jkroon@dsl-241-231-233.telkomadsl.co.za) |
| 18:59.31 | r0m|u | p3nguin: you do any type of video conf? |
| 19:02.20 | nny | can someone take a look at http://pastebin.com/UpyiNGFD and tell me why it doesn't behave the same when Dial is used instead of Transfer? |
| 19:02.34 | nny | er doesn't work* of course two apps won't behave the same ;) |
| 19:03.31 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 19:05.34 | Gugge | nny: why would you transfer to a Local channel? |
| 19:05.41 | Gugge | just Goto :) |
| 19:05.49 | Gugge | or gosub |
| 19:06.12 | nny | Gugge trying to create a function that sends the callee to the channel and hangsup the caller |
| 19:07.08 | Gugge | hmm, i dont really have any idea how to do that :) |
| 19:07.20 | nny | Gugge: ok thanks! |
| 19:07.24 | Micc | kinda like how park works? |
| 19:08.01 | nny | Micc: I am sending the callee to a recording, parking would be putting them into a parking lot I thought? |
| 19:08.39 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 19:08.46 | Micc | I'm just trying to relate it to something I know. Is your callee the transferee or the transferer? |
| 19:08.55 | nny | transferee |
| 19:09.14 | nny | Micc Dial works, but the caller's channel is still open |
| 19:09.16 | luckman212 | Micc: what do you use instead of FreePBX? |
| 19:09.22 | Micc | well then wouldn't you just transfer to a Playback? |
| 19:09.38 | Micc | luckman212, just plain asterisk. |
| 19:10.20 | nny | Micc well I assumed (my macro right now is simple for testing) that transferring to Local/6@macro-playback would send the callee to the exten => 6 inside the macro. |
| 19:10.23 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 19:10.49 | nny | Micc in theory the second half could be a seperate context but it should work either way afaik? |
| 19:11.04 | Micc | nny, yes, that sounds correct. |
| 19:11.14 | luckman212 | Micc: you are using a single asterisk instance to serve multiple customers? |
| 19:11.25 | Micc | nny, I use a lot of Dial(Local/#@context) |
| 19:11.39 | Micc | nny, how are you doing the transfer from the phone or something else? |
| 19:11.54 | nny | Micc using a feature code |
| 19:11.57 | nny | Micc: custom |
| 19:12.05 | nny | Micc here is the console output http://pastebin.com/ExSgTXa0 |
| 19:12.14 | Micc | luckman212, yes, thats called multi-tenant. Really I use multiple servers to serve lots of customers. |
| 19:12.30 | nny | Micc: here is the feature code sendplayback => *6,peer/caller,Macro,playback |
| 19:12.45 | nny | Transfer just doesn't behave like i would expect it to |
| 19:13.07 | nny | er one sec that output is missing some stuff |
| 19:13.09 | Gugge | nny: AFAIK Transfer() only works on the same Tech |
| 19:13.17 | Gugge | so you can only transfer a sip channel to SIP/ |
| 19:14.00 | Micc | nny, I don't know much about using feature codes. |
| 19:14.20 | nny | Micc: http://pastebin.com/TGgVw8iU yeah that feature code works, it's Transfer that's not quite workin |
| 19:14.35 | nny | Gugge: ahh interesting |
| 19:14.37 | luckman212 | Micc: nice, so the Asterisk servers are in some kind of cluster? I've not seen any good guides on HA or multi-tenant Asterisk setups |
| 19:14.45 | Micc | nny, does the normal transfer work like #6? |
| 19:14.47 | Gugge | even if transfer worked, it would not cause a hangup on the caller |
| 19:15.25 | nny | Gugge: hmm any more elegant methods to perform this? I could just use Dial and tell the caller to hangup I guess, but trying to find the most promising solution |
| 19:15.34 | Micc | luckman212, me either. There are a few people that do them here, but there aren't a lot of information on it. You kind of have to architect it yourself. |
| 19:15.48 | Gugge | nny: what are you trying to do? |
| 19:15.51 | Gugge | and why :) |
| 19:16.07 | *** join/#asterisk BuenGenio (~Gene@177-207-202-46.pool.ukrtel.net) |
| 19:16.11 | nny | Gugge: send the callee to a recording for a client |
| 19:16.20 | Gugge | I cant really imagine when a callee needs a featurecode that hangs up the caller :) |
| 19:16.29 | nny | Gugge: the feature code is for the caller |
| 19:16.39 | nny | the caller dials *6 and it sends the callee to the recording |
| 19:16.40 | Gugge | so you need the caller to send the callee (client) to a recording? |
| 19:16.49 | nny | Gugge: that is how it is currently setup yes |
| 19:17.33 | luckman212 | Micc: that's not something I would attempt to do myself. I am having enough trouble just keeping my single server stable and functioning well |
| 19:18.04 | Gugge | nny: i have no idea how, sorry :( |
| 19:18.13 | Micc | nny, try using the normal transfer feature code with an extension, like #6, and put your extension 6 in the sip context of the caller. |
| 19:18.38 | nny | Gugge: thanks either way |
| 19:18.58 | nny | Micc: wouldn't that require the transferer to enter the extension? This kind of has to be a 1 click solution |
| 19:19.32 | Micc | nny, yeah but the extension is 6, Its the same number of keys as *6 |
| 19:20.13 | nny | Micc: yeah the idea is to just have the caller hit *6 and then hangup or have the system hang them up. Having them do *6,6 is not quite what I am looking for |
| 19:20.24 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:20.31 | Micc | your tranferer is the callee, so just make sure you use the correct case of T in the dial command and set the TRANSFER_CONTEXT correctly before dialing. |
| 19:20.59 | Micc | wait who hits the *6, the caller or callee? |
| 19:21.02 | nny | Micc: the transferer is the caller |
| 19:21.33 | Micc | ok, so thats normal. if # is the transfer key and 6 is the extension, then all you need is to hit #6. Not #6,6 |
| 19:21.51 | nny | Micc interesting, i'll look at that |
| 19:22.15 | [TK]D-Fender | nny, there is a channelvar with the bridged channelname IIRC. Call a script to do an AMI redirect on that channel to your playback extent and it will drop the caller like a rock. |
| 19:23.10 | nny | [TK]D-Fender: i see, I could use TrySystem or System too i assume? |
| 19:23.39 | [TK]D-Fender | Don't recall if there is a CLI redirect command or not. If so, yes |
| 19:24.28 | [TK]D-Fender | Actually, yes you are calling an external script. Could be an AGI or System() TrySystem(), etc |
| 19:24.56 | [TK]D-Fender | If you use a call to asterisk -rx (if there really is an option), o9r AMI (which there is) is up to you |
| 19:25.08 | nny | s,n,TrySystem(asterisk -rx "channel originate SOME_BRIDGE_VARIABLE application Bridge Local/6@macro-transfer") |
| 19:25.11 | Gugge | app ChannelRedirect |
| 19:25.14 | Gugge | wont that do it? |
| 19:26.10 | nny | i use a variant of that now for a seamless cell phone switch |
| 19:26.37 | Gugge | ChannelRedirect(channel,macro-playback,6) |
| 19:27.14 | nny | Gugge: yeah i'll need to find the variable for the callee's channel. I think [TK]D-Fender's solution is also to drop the caller channel |
| 19:27.39 | nny | this is how my cell phone switch variable works, it kills the caller channel immediately and sends the callee to my bridged channel |
| 19:28.12 | nny | just need to find that variable and i can play with it more |
| 19:28.48 | nny | there is a command from CLI to show all channel variables available on a given channel, anyone know it off the top of their head? |
| 19:28.56 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 19:29.02 | [TK]D-Fender | nny, core show channel [x] |
| 19:29.55 | nny | ahh |
| 19:30.13 | nny | s,n,TrySystem(asterisk -rx "channel originate ${BRIDGEPEER} application Bridge Local/6@macro-playback") |
| 19:30.20 | nny | should work let me test |
| 19:31.18 | [TK]D-Fender | nny, No, not bridge.. that latches on, but doesn't disconnect the caller IIRC |
| 19:31.42 | [TK]D-Fender | I was thinking "bridgepeer", but was unsure if that was intdeed the full channelname |
| 19:33.19 | nny | actually |
| 19:33.30 | nny | this is being peformed on the callee, so BRIDGEPEER would be the caller |
| 19:33.40 | Micc | ${CHANNEL} ? |
| 19:33.57 | nny | Micc: aye lol |
| 19:34.09 | pabelanger | ~sipnat |
| 19:34.09 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 19:35.28 | nny | [TK]D-Fender: yeah i don't think bridge is right exten => s,1,TrySystem(asterisk -rx "channel originate ${CHANNEL} application Bridge Local/6@playback") is what I have now |
| 19:35.44 | *** join/#asterisk PoWeRKiLL (~powerkill@87.68.246.10.adsl.012.net.il) |
| 19:36.06 | [TK]D-Fender | nny, That is certain failure. that single app gets called and then has nothing more to do and dies |
| 19:36.35 | [TK]D-Fender | nny, You need 2 ends. 1 to do the hijack, the other to do the playback. So Local/ for the hijack-side |
| 19:37.57 | nny | [TK]D-Fender: assumed that it originated the channel and then used bridge to connect it to Local/6@playback, (where local/6@playback plays the message).. not the case I see. |
| 19:44.50 | nny | hmm |
| 19:44.54 | nny | ok that worked, thanks Gugge |
| 19:45.51 | nny | http://pastebin.com/12uDtfqD |
| 19:45.58 | nny | hangs up the caller, plays monkeys to the callee |
| 19:46.22 | nny | now I my plan to conquer the world by robo dialing and sending monkeys to everyone can begin |
| 19:46.47 | [TK]D-Fender | Ahh, ChannelRedirect.... yup, that looks legit |
| 19:46.48 | nny | laughs maniacally |
| 19:47.14 | [TK]D-Fender | And should eb able to do direct from the featurecode if it parses channelvars right (or called via MAcro() |
| 19:47.57 | nny | yeah thanks, I was keen on both methods, i use this http://pastebin.com/MJz9Q7v3 to send calls to my cellphone from my desk phone without the caller knowing |
| 19:49.59 | nny | [TK]D-Fender: i see what you mean, sounds like a good refinement to just call the app from the feature code thanks |
| 20:03.20 | *** join/#asterisk timahvo1 (~rogue@197.176.180.75) |
| 20:09.13 | *** join/#asterisk afink (~afink@207.106.66.194) |
| 20:11.20 | luckman212 | if I want to output the value of a CDR variable using Verbose($some_var) do I need to set a variable first e.g. Set(foo=${CDR(dcontext)}) and then Verbose(foo=${foo}) ? or is there a more efficient way? |
| 20:11.45 | p3nguin | Verbose(${CDR(dcontext)}) |
| 20:13.33 | luckman212 | p3nguin: thanks |
| 20:18.26 | luckman212 | how can I get Verbose() to output a newline (i.e. "\n") |
| 20:18.45 | [TK]D-Fender | luckman212, You don't |
| 20:19.01 | *** join/#asterisk luke-jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 20:19.21 | p3nguin | You could run an empty Verbose() to print a blank line, I'd imagine. |
| 20:19.29 | [TK]D-Fender | luckman212, Multiple verbose is it. |
| 20:19.30 | luckman212 | [TK]D-Fender: I'm sure I've done it before somehow just can't remember... it involved setting a variable in globals.conf which contained e.g. $NL=(newline char) |
| 20:20.03 | luckman212 | then Verbose(blah blah${NL}foo foo bar baz) |
| 20:20.09 | [TK]D-Fender | luckman212, Don't go trying to treat * CLI as a raw terminal to display whatever you'd like |
| 20:20.17 | p3nguin | I've never even heard of globals.conf before. |
| 20:21.09 | luckman212 | globals_custom.conf it's a freepbx include |
| 20:21.29 | p3nguin | Oh. We don't do FreePBX here, so it's no wonder I haven't heard of it. |
| 20:21.34 | luckman212 | yep sorry my bad |
| 20:21.50 | luckman212 | either way it wasn't related... was just wondering about outputting newlines. but it seems like a "no no" |
| 20:22.13 | p3nguin | Just add an empty Verbose() and see what you get. |
| 20:23.14 | luckman212 | p3nguin: empty Verbose() doesn't do it |
| 20:23.38 | p3nguin | I'm sure I've done it that way. Let me look where/how. |
| 20:24.10 | p3nguin | That's how I did it. same => n,Verbose(); |
| 20:24.43 | luckman212 | p3nguin: hmm that's what I typed as well, for me it output nothing (no blank line either) |
| 20:25.02 | p3nguin | How about Verbose(0, )? |
| 20:25.45 | luckman212 | p3nguin: ah yes that works |
| 20:26.00 | p3nguin | How about Verbose(0,)? |
| 20:26.26 | luckman212 | yes that also works |
| 20:26.57 | p3nguin | I suppose I could test these things myself, but since you're already in it knee-deep, I figured I'd let you do the dirty work. |
| 20:27.41 | luckman212 | no worries, thank you |
| 20:29.42 | *** join/#asterisk singler (~singler@84.15.187.182) |
| 20:29.59 | luckman212 | I didn't even know Verbose() took arguments, that's actually pretty useful, thanks p3nguin |
| 20:30.28 | luckman212 | now I can put my debug stuff at loglevel 1 to keep it quiet unless needed |
| 20:30.30 | p3nguin | core show application Verbose |
| 20:48.54 | *** join/#asterisk banditti (~banditti@unaffiliated/banditti) |
| 20:49.04 | banditti | question. If i ping from my client to my asterisk server I show 42ms of latency, but SIP show peers is a lot higher. Thoughts? |
| 20:49.39 | mattsqz | hmm wouldnt peers be twice the ping? |
| 20:50.47 | p3nguin | The qualify time is not directly related to ping time. |
| 20:51.19 | p3nguin | Qualify time is the time it takes for the peer to respond to an OPTIONS packet. |
| 20:51.29 | p3nguin | Ping time is ICMP echo. |
| 20:52.08 | p3nguin | s/echo/echo and reply/ |
| 20:53.25 | banditti | what is a decent qualify time? I ping at 36ms and traceroute of 10 hops. Qualify time is 150 to 250 depending on extension |
| 20:53.50 | p3nguin | Extensions are also not related to qualify times. |
| 20:54.16 | p3nguin | A good qualify time depends on your reason for using qualify at all. |
| 20:55.22 | p3nguin | Are you using it because you think it's what you should do, or because it serves a purpose in your configuration? |
| 20:56.13 | banditti | I think both. I have a asterisk server in a datacenter. Clients behind one IP (NAT) to the server over a public internet connection |
| 20:56.25 | banditti | I would think qualify makes sense in that case? Thoughts? |
| 20:56.46 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 20:56.53 | p3nguin | In that case, if a NAT keepalive is necessary, using qualify makes sense to me. |
| 20:57.11 | p3nguin | If a NAT keepalive isn't needed, there's not much point to a qualify. |
| 20:57.43 | *** join/#asterisk logicwrath (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 20:57.44 | banditti | I installed FOP2 and occasionally the status goes from green to grey and back to green. The forum admin said it was due to a LAGGED state. I don't think so as I grep'd the logs and nothing |
| 20:57.53 | banditti | so, down the rabbit whole I go |
| 20:57.58 | banditti | *hole |
| 20:58.05 | *** join/#asterisk eth00 (~r00t@unaffiliated/eth00) |
| 20:58.14 | banditti | Is there a qualify time that I should worry about? is 220 high? |
| 20:58.30 | p3nguin | 220 is not high in most cases. |
| 20:58.47 | p3nguin | 2000 can even be acceptable. |
| 20:59.25 | banditti | Thank you for your feedback, it is appreciated. Any other thoughts? |
| 20:59.35 | p3nguin | I like to see a lower time, but sometimes network gear and other crap is slow. |
| 21:00.37 | [TK]D-Fender | I think you should stop thinking about it |
| 21:01.07 | p3nguin | At least until an actual problem arises. |
| 21:01.44 | p3nguin | And if you don't need a NAT keepalive, stop qualifying and using up bandwidth that could potentially be used for something else. |
| 21:02.48 | banditti | Thank you again. Now if I can just figure out why FOP goes Green / Grey / Green |
| 21:03.05 | p3nguin | Does it base the color on ping time or qualify time? |
| 21:03.11 | [TK]D-Fender | bandBecause it takes a number and thinks it's meaningful. It is mistaken |
| 21:03.18 | [TK]D-Fender | Qualify. |
| 21:03.21 | [TK]D-Fender | Thre is no "ping" |
| 21:03.30 | banditti | bandbecause |
| 21:03.35 | banditti | ? |
| 21:03.46 | [TK]D-Fender | autocomplete fail. Get a more unique name |
| 21:03.54 | [TK]D-Fender | banditti, Because it takes a number and thinks it's meaningful. It is mistaken |
| 21:03.57 | p3nguin | There is if the application pings something. I don't use FOP2, so I didn't know if it actually performs a ping. |
| 21:04.05 | [TK]D-Fender | it doesn't |
| 21:04.11 | [TK]D-Fender | It just pulls info from AMI |
| 21:05.37 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 21:05.47 | banditti | This is what I was told: |
| 21:05.49 | banditti | Your sip peers are probably becoming LAGGED and then back online again. FOP2 does not invent those states, they are reported by asterisk. I think you are actually seeing something you did not notice before. |
| 21:05.49 | banditti | Be careful as your extensions are not going to receive calls if they become Lagged (they can dial out, but usually inbound calls fail). |
| 21:06.24 | p3nguin | By extensions, he meant phones. |
| 21:06.32 | p3nguin | Extensions are something else in asterisk. |
| 21:06.52 | [TK]D-Fender | banditti, Increase your qualify time or disable it altogether if it's for a trunk |
| 21:06.53 | banditti | I just can't find any indication that he is correct. |
| 21:07.22 | p3nguin | If a phone is too lagged, there is a possibility that it will not be able to get a call. |
| 21:07.35 | p3nguin | It isn't certain, but it is possible. |
| 21:07.50 | p3nguin | If it is unreachable, then it is certain that it will not receive a call. |
| 21:08.03 | banditti | Understood, but wouldn't that be logged? I have never seen any symptoms or can't find a trace of it. |
| 21:08.12 | p3nguin | With qualify, you have reachable, lagged, and unreachable. |
| 21:08.32 | p3nguin | Lagged simply means it is taking longer to respond than was expected. |
| 21:08.55 | p3nguin | So unless you actually have a problem, STOP WORRYING. |
| 21:09.11 | banditti | My problem is green / grey / green. |
| 21:09.22 | p3nguin | (1506.52) <[TK]D-Fender> banditti, Increase your qualify time or disable it altogether if it's for a trunk |
| 21:09.30 | banditti | The program won't do transfers when grey |
| 21:09.37 | banditti | I will up my qualify time as a test |
| 21:09.46 | [TK]D-Fender | <p3nguin> (1506.52) <[TK]D-Fender> banditti, Increase your qualify time or disable it altogether if it's for a trunk |
| 21:09.52 | p3nguin | :/ |
| 21:09.52 | banditti | if default is 2 seconds, should I bump to 3? |
| 21:10.01 | [TK]D-Fender | or whatever |
| 21:10.03 | p3nguin | maybe 6000? |
| 21:10.24 | p3nguin | If it doesn't respond in 6 seconds, there could be a problem. |
| 21:10.32 | banditti | Sounds reasonable. |
| 21:10.40 | banditti | Thank you all again for your insight. |
| 21:10.52 | mattsqz | banditti: 585ohm? |
| 21:11.49 | banditti | mattsqz, I don't understand |
| 21:12.07 | mattsqz | green, grey, green |
| 21:12.20 | p3nguin | heh |
| 21:20.12 | banditti | Clever. |
| 21:23.30 | *** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net) |
| 21:24.07 | *** join/#asterisk willbradley (~wbradley@unaffiliated/willbradley) |
| 21:24.19 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 21:24.49 | willbradley | any idea why two SIP clients on the same Asterisk server (and same subnet) would have trouble hearing each other? did a packet capture from each client, Asterisk doesn't seem to be sending them on correctly. |
| 21:25.38 | IsUp | willbradley: sounds like a NAT issue |
| 21:25.47 | willbradley | points at same subnet |
| 21:26.03 | lauris | is asterisk on the same subnet ? |
| 21:26.06 | lauris | as clients |
| 21:26.08 | willbradley | they all are |
| 21:26.35 | luckman212 | what are these SIP clients? |
| 21:26.38 | lauris | could be codec issue? |
| 21:27.36 | willbradley | 3cx on laptop and on phone |
| 21:27.47 | willbradley | i think they're both only using ulaw |
| 21:28.12 | luckman212 | 3cx on laptop-->check firewall not blocking RTP ports |
| 21:28.37 | willbradley | laptop: 10.1.21.21 asterisk: 10.1.20.190 phone: 10.1.21.25 (the subnet is .254, they can all ping each other directly no problem) |
| 21:29.23 | p3nguin | What do you mean by "the subnet is .254" in your description? |
| 21:29.25 | luckman212 | ping doesn't mean they will be able to exchange RTP, ping uses ICMP and RTP is UDP at random port ranges |
| 21:29.30 | willbradley | sure but that doesn't really answer why the phone (android) wouldn't be able to hear stuff from other sip clients (polycom phones) |
| 21:29.48 | willbradley | i.e. no firewalls there, laptop is troubleshooting (as i check windows firewall) |
| 21:29.54 | luckman212 | can 2 polycom phones hear each other OK? |
| 21:30.11 | willbradley | nope, which is why i'm troubleshooting |
| 21:30.35 | willbradley | i.e. is there a situation where asterisk would be like "nah, i'll send those packets elsewhere!" |
| 21:30.52 | Qwell | ~nat |
| 21:30.52 | infobot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 21:30.53 | willbradley | of note: laptop and phone work fine if they call each other's PSTN number which bounces off the T1 and comes back in |
| 21:31.04 | luckman212 | do you have any of: directrtpsetup=yes , canreinvite=yes , directmedia=yes in your sip.conf ? |
| 21:31.15 | willbradley | points at "same subnet" -- there is literally no NAT possible between these devices |
| 21:31.25 | willbradley | chceks |
| 21:32.09 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 21:32.20 | luckman212 | willbradley: asterisk should ideally not be proxying media for 2 UA's that are on the same subnet. (unless you are recording calls etc) |
| 21:32.46 | luckman212 | or transcoding |
| 21:33.19 | willbradley | good point. what does it define as "same subnet"? cuz 10.1.21.x and 10.1.20.x are on the same subnet (a /23) but look different |
| 21:33.49 | willbradley | though that's just my troubleshooting-- in production, very different subnets are trying to communicate and failing -- so it looks like maybe proxying isn't working? |
| 21:33.58 | luckman212 | you need the correct localnet= entries |
| 21:34.25 | willbradley | luckman212: grep 'directrtp' and 'directmedia' return nothing in /etc/asterisk but i have lots of canreinvite=no entries |
| 21:34.40 | IsUp | <luckman212> you need the correct localnet= entries |
| 21:35.04 | luckman212 | any SIP peer that you assign 'canreinvite=no' to will force Asterisk to insert itself into the media path |
| 21:35.25 | willbradley | just anywhere in the sip.conf? |
| 21:35.46 | luckman212 | so for devices that ARE on the same subnet and are fixed (e.g. desk phones) you should set canreinvite=yes for those |
| 21:36.25 | IsUp | willbradley: under [general] |
| 21:36.37 | luckman212 | in the peer definitions, usually they are in sip.conf but they could be elsewhere depending on your situation |
| 21:36.51 | p3nguin | localnet is a general setting. |
| 21:37.49 | *** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net) |
| 21:38.21 | luckman212 | willbradley: this may be a silly question but are you sure you've correctly defined your IPv4 settings (ifconfig) for your /23 subnet |
| 21:38.43 | willbradley | it's a good question and i checked; also i did ping check |
| 21:39.56 | luckman212 | what is your rtpstart=, rtpend= set to in /etc/asterisk/rtp.conf ? how about iptables (if you're using it) are you allowing that port range?? |
| 21:40.27 | willbradley | start 10000 end 20000 |
| 21:40.52 | willbradley | is rtp a distinct thing that would be used in sip-asterisk-sip comms? cuz any other combo works fine |
| 21:41.13 | willbradley | also i thought this worked previously so i'm not sure what degraded |
| 21:41.33 | luckman212 | RTP is used everywhere that 2 UAs want to send media |
| 21:41.52 | p3nguin | RTP is used when 1 UA wants to send/receive media. |
| 21:42.00 | p3nguin | RTP is where a SIP phone's audio is. |
| 21:42.02 | luckman212 | whether asterisk is in the middle depends on a lot of things, but RTP is not anything "special" its just UDP data |
| 21:42.49 | *** part/#asterisk JTLove (~JTLove@129.112.244.66.dsi.smithvilledigital.net) |
| 21:42.56 | willbradley | iptables is empty |
| 21:43.05 | luckman212 | willbradley: I suggest you turn on SIP debug and look at where your devices are asking/expecting to receive audio (what ports, codec etc) |
| 21:43.37 | willbradley | ok i'll dig deeper when there isn't so much call traffic |
| 21:43.40 | IsUp | willbradley: did you set localnet |
| 21:43.40 | willbradley | thanks guys |
| 21:43.53 | willbradley | yep but i can't tell if it made a difference |
| 21:44.10 | willbradley | i'm also gonna do a packet capture on the asterisk server itself |
| 21:45.16 | *** part/#asterisk willbradley (~wbradley@unaffiliated/willbradley) |
| 21:46.57 | luckman212 | if you turn on SIP debugging, you will see some lines in the INVITE like "c=IN IP4 a.b.c.d" as well as "m=audio NNNN RTP/AVP x x" where a.b.c.d is the IP that your phone wants the other side to contact and NNNN is the UDP port it expects to receive audio |
| 21:47.13 | luckman212 | those should give you a clue as to what could be wrong |
| 22:03.17 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 22:04.33 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
| 22:04.38 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 22:11.04 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:11.18 | luckman212 | is there a way to output "(" or ")" with a Verbose() command? sorry I am obsessed with logging today |
| 22:11.55 | IsUp | luckman212: \( and \) maybe? |
| 22:12.03 | luckman212 | maybe...testing |
| 22:12.32 | luckman212 | no |
| 22:12.45 | luckman212 | that prints exactly \( and \) |
| 22:13.10 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 22:13.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:14.05 | IsUp | Verbose((test)) gives me (test) |
| 22:14.37 | luckman212 | IsUp: well whaddya know, so it does |
| 22:15.59 | luckman212 | thanks |
| 22:16.35 | carrar | -=>TEST<=- |
| 22:18.59 | luckman212 | <=- FAIL -=> |
| 22:19.06 | p3nguin | .:/(|-=>TEST<=-|)\:. |
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