| 00:04.22 | ChannelZ | of your own making? |
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| 00:14.57 | igcewieling1 | if anyone wants to play around with it http://140.239.234.172/transcode/index.php |
| 00:18.48 | igcewieling1 | if people like it, I'll add g729 transcoding too |
| 00:19.22 | qmr | huh |
| 00:19.27 | qmr | I have mplayer vlc and ffmpeg already |
| 00:19.39 | *** join/#asterisk DEMNVT (~Adium@220-245-156-82.static.tpgi.com.au) |
| 00:20.18 | igcewieling1 | qmr: lots of peopel don't know how to use those to transcode into a format Asterisk likes |
| 00:20.32 | qmr | o |
| 00:20.34 | igcewieling1 | and lots of people don't know how to use ffmpeg or sox |
| 00:20.50 | igcewieling1 | ..er..I mean Lots of people don't have ffmpeg or sox |
| 00:21.13 | igcewieling1 | (like your customer service people 8-) |
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| 01:24.43 | ChannelZ | ffmpeg: easy to download, hard to use |
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| 01:32.36 | dijib | bitches |
| 01:32.41 | dijib | how is everyone in here |
| 01:32.47 | dijib | i hope you are all doing well |
| 01:33.05 | dijib | is anybody running an astering server monitored by an MSP application? |
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| 01:43.53 | dijib | are you guys still partying from the 4th of july?? j'esus |
| 01:51.52 | ChannelZ | hells yeah! |
| 01:51.59 | ChannelZ | continues firing into the air |
| 01:55.06 | sawgood | fourth of july is over, man |
| 01:56.06 | WIMPy | and the fifth |
| 01:56.13 | WIMPy | and the sixth |
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| 03:06.20 | jmordica | are you guys still working on the asterisk distributed platform? |
| 03:06.38 | jmordica | I think it was a multi tenant distributed system using asterisk |
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| 03:30.20 | blc | has anyone seen asterisk do a tcp-level disconnect of a client while using TLS? |
| 03:30.36 | blc | it doesn't happen when using tcp with no tls |
| 03:30.56 | igcewieling1 | jmordica: We use PMXware, which does multi-tenant and "clustering" |
| 03:31.04 | igcewieling1 | sorry, PBXware |
| 03:31.46 | blc | the client will register, and can receive calls for a few seconds…maybe 10-15, then a FIN gets sent, and the TCP connection is closed. So, the client can no longer receive calls |
| 03:32.23 | jmordica | igcewieling1: cool is this a hosted platform or something I can run myself? |
| 03:32.36 | blc | the strange part is that the client still thinks it is registered |
| 03:32.37 | igcewieling1 | jmordica: a hosted platform you run yourself |
| 03:32.48 | igcewieling1 | It is commercial |
| 03:32.56 | jmordica | so how is it multi tenant if it uses asterisk? |
| 03:33.04 | jmordica | it would have to use freeswitch right? |
| 03:33.35 | igcewieling1 | jmordica: Asterisk can do multi-tenant is you jump through enough hoops. We pay them to do the hoop jumping for us. |
| 03:34.10 | jmordica | Cool. So how does load balancing work in a clustered environment? |
| 03:34.13 | igcewieling1 | We prefer totally open source solutions when possible, but in the case of multi-tenant we went with a commercial Asterisk offering. |
| 03:34.20 | jmordica | do i setup my own open sips? |
| 03:34.26 | jmordica | in front of a cluster? |
| 03:34.28 | igcewieling1 | jmordica: I have no idea. We don't use it in a cluster, only in standalone. |
| 03:34.59 | igcewieling1 | It uses Asterisk and has a GUI designed for multi-tenant, |
| 03:36.49 | jmordica | it seems there is a call center edition and multi tenant edition |
| 03:37.04 | jmordica | what if I run the multi tenant edition but need call center features for business customers |
| 03:37.06 | jmordica | ? |
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| 03:50.25 | igcewieling1 | That sounds like a question for sales |
| 03:50.47 | igcewieling1 | be warned though, you can't customize it. |
| 03:51.49 | jmordica | got ya. ok thanks |
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| 08:30.19 | AndyN_ | Hello, I'm looking to set up incoming fax-to-email with asterisk 1.8, using iaxmodem + hylafax |
| 08:30.41 | AndyN_ | Centos doesn't have that many tutorial on google. Any suggestion ? |
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| 12:09.19 | mr_claus | hi, i try to start a call and it takes a long time to get the call signaling on the other end of the line, if i take the call it takes a long time again to establish the connection, it worked a few days ago so i think it's a provider problem, but how i could be sure thats not an issue with the local asterisk installation? |
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| 13:52.11 | Jymmm | Is it possible to connect two ATA's together without using an asterisk server as a point-to-point thing? |
| 13:52.49 | Jymmm | a glorified intercom of sorts. |
| 13:54.12 | Jymmm | I have a couple of "Grand Central" (?) ATAs |
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| 14:05.41 | WIMPy | Sure |
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| 14:20.59 | Jymmm | Ok, care to elaborate a bit? |
| 14:21.20 | WIMPy | Depends on the model. |
| 14:21.30 | WIMPy | Some will support direct IP dialling. |
| 14:21.43 | Jymmm | hang on... |
| 14:22.02 | WIMPy | Maybe you can use a phonebook or as a last resort you can make one the "provider" of the other. |
| 14:23.42 | Jymmm | I think this is it... http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht701 |
| 14:23.56 | Jymmm | let me see if I can find it |
| 14:26.45 | Jymmm | I couldn't find a pair of elephants humping in this place |
| 14:28.08 | Jymmm | WIMPy: Thanks for the ideas. |
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| 17:44.08 | transfinite | is there a way to evaluate expressions in the CLI? I want to do some simple tests before modifying my actual config |
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| 17:48.08 | [TK]D-Fender | transfinite: No |
| 17:48.40 | [TK]D-Fender | that also requires varibes etc to actually be able to evaluate which means they need contents |
| 17:48.48 | [TK]D-Fender | testing means doing. |
| 17:53.41 | igcewieling | there is a utility included with asterisk to test expressions outside the dialplan |
| 17:53.53 | igcewieling | ast-exprcheck2 or something. check in menuconfig |
| 17:55.26 | transfinite | i'm trying to do some massaging of the VM_* variables in voicemail.conf, but it doesn't look like that does anything except plain variable substitution, is that right? Is there any way to define other info in the dialplan that gets passed as variables to the voicemail function? |
| 17:55.47 | igcewieling | [TK]D-Fender: experimental, for people who have trouble converting sound files into an Asteirsk format. http://140.239.234.172/transcode/ |
| 17:56.15 | igcewieling | transfinite: dialplan example code needed. put it on a pastebin |
| 17:56.17 | igcewieling | ~pb |
| 17:56.17 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:56.35 | igcewieling | voicemail.conf does not do expressions |
| 17:58.01 | transfinite | I don't have anything written yet; I wanted voicemail.conf's pagersubject variable to have different values based on the callerid |
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| 17:59.14 | igcewieling | transfinite: that is unlikely to work. |
| 18:06.03 | transfinite | The example voicemail.conf uses dialplan functions in its definition of "emailbody" |
| 18:11.53 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/11/configs/voicemail.conf.sample?revision=371121&view=markup |
| 18:12.21 | [TK]D-Fender | transfinite: That seems to say you can use full variable & function logic in there... so that defines what you can do |
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| 19:49.46 | igcewieling | I'm skeptical, but if the docs say.... |
| 19:50.43 | [TK]D-Fender | It does say it.... |
| 19:51.32 | [TK]D-Fender | They show a sample using a standard dialplan function... which of course opens us a question of just how much you could do in there given the kind of blocking calls you're potentially capable of using |
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| 20:28.25 | hesco | I'm running Asterisk 10.8.0, which has the CALLERID() function but no SetCallerID() application. I've pored over multiple web searches well past the first page and have been unable to find any guidance for how to effectively pass along the callerID from an inbound call to the outbound recipient. Can anyone here please provide guidance? |
| 20:35.50 | [TK]D-Fender | because you SET the function. |
| 20:35.56 | [TK]D-Fender | ^ |
| 20:36.00 | [TK]D-Fender | welcome to functions 101 |
| 20:36.14 | [TK]D-Fender | That application died over half a DECADE ago |
| 20:36.34 | [TK]D-Fender | "how to effectively pass along the callerID from an inbound call to the outbound recipient" <- this is automatic |
| 20:36.44 | [TK]D-Fender | Thre is no speicifc step to take |
| 20:36.54 | [TK]D-Fender | the only question is where it is getting OVERRIDEN in your scenario |
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| 20:37.51 | carrar | Set(CALLERID(number)=2068675309) |
| 20:39.16 | flightnut | Anyone heard of a bug with " ' " in the CNAM causing incoming fax's to fail? |
| 20:40.34 | [TK]D-Fender | flightnut: It doesn't |
| 20:40.58 | [TK]D-Fender | flightnut: What you posted eariler was an improper escaping in that bash script call |
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| 20:41.14 | [TK]D-Fender | flightnut: You already ID'd the problem and the cause is exactly that obvious and should get fixed |
| 20:41.47 | [TK]D-Fender | flightnut: This is not an Asterisk problem, it is a FreePBX problem and needs to be fixed there |
| 20:42.49 | flightnut | Sorry, that was my first bug that I found and it's been a while and In the mean time. I am trying to figure out a workaround. |
| 20:43.39 | [TK]D-Fender | Go change the code that calls that script then |
| 20:48.51 | flightnut | Okay, here is the code that calls the script exten => h,n,System(${ASTVARLIBDIR}/bin/fax2mail.php --to "${FAX_RX_EMAIL}" --dest "${FROM_DID}" --callerid '${CALLERID(all)}' --file ${ASTSPOOLDIR}/fax/${UNIQUEID}.tif --exten "${FAX_FOR}" --delete "${DELETE_AFTER_SEND}") |
| 20:49.06 | flightnut | Figuring out how to make it ignore the ' is the tricky part |
| 20:50.36 | [TK]D-Fender | flightnut: "core show function STRREPLACE" <- |
| 20:51.07 | [TK]D-Fender | flightnut: Pretty easy. |
| 20:52.40 | flightnut | Gotcha, thank you. |
| 20:55.31 | [TK]D-Fender | flightnut: To have this fixed more permanently post another update on your report to bump it, and make a post on the message board for it to get attention |
| 20:57.37 | flightnut | WIll do, which message board are you referring to? (Their forums?) |
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| 21:01.17 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:01.43 | [TK]D-Fender | Yes. |
| 21:09.10 | hesco | Thanks [TK]D-Fender: I'd guess my callerID is being over-ridden by my outbound trunk provider, but I can find nothing in their configuration options which exposes to me control over what it might be. Guess its time to create a ticket in their support queue. Thanks for clarifying that. |
| 21:09.34 | ghost75 | ChannelZ ? |
| 21:09.44 | [TK]D-Fender | hesco: So far .. you are simply guessing. I don't see you looking at any debug to see what's actually happening... |
| 21:12.12 | ChannelZ | eh? |
| 21:23.49 | *** join/#asterisk guitar_Hester (~guitarHes@mobile-166-147-126-191.mycingular.net) |
| 21:26.47 | *** join/#asterisk floren (~floren@unaffiliated/floren) |
| 21:27.35 | floren | [TK]D-Fender: are you busy? |
| 21:28.38 | floren | guys, in the last 3 days i lost endless hours trying to configure my cosco 8961 with asterisk and voip.ms |
| 21:28.59 | [TK]D-Fender | yes. I am currently breathing. This is likely to pove a very time consuming ordeal.... |
| 21:29.21 | floren | i really need your advice with my configurations, i simply cannot figure it |
| 21:29.33 | floren | http://forums.asterisk.org/viewtopic.php?f=1&t=87047&p=189315#p189315 |
| 21:29.42 | [TK]D-Fender | I presume that is 7961.... |
| 21:29.50 | floren | no, 8961 |
| 21:30.04 | Katty | hi |
| 21:30.14 | floren | the phone registers, is just the asterisk configs i'm fighting with |
| 21:30.19 | [TK]D-Fender | Hrm new model... |
| 21:30.36 | [TK]D-Fender | floren: You are spending too long staring at configs.. and not enough at what is actually happening. |
| 21:30.40 | floren | i was able to make outbound calls from the cisco phone |
| 21:30.48 | floren | but i could not make inbound calls |
| 21:31.11 | floren | [TK]D-Fender: trust me, i stared at the logs too :) |
| 21:31.30 | floren | hi Katty |
| 21:31.48 | Katty | smiles at floren |
| 21:32.12 | floren | [TK]D-Fender: if you have time, please read my post, you will see that i actually tried to to solve the damn thing |
| 21:32.30 | [TK]D-Fender | Show new debug for the first thing you're looking to fix here... |
| 21:32.50 | Katty | fender bender, come reorganize my basement |
| 21:33.01 | Katty | it's an overwhelming mess. |
| 21:33.38 | floren | [TK]D-Fender: i'm trying to make working inbound calls, see this debug log: http://codepad.org/uZPxZhKf |
| 21:33.49 | [TK]D-Fender | floren: -- Got SIP response 480 "Temporarily Unavailable (Call limit)" back from 67.205.74.184:5060 <--- you are sending your incoming call from VoiP.ms in a LOOP. |
| 21:34.01 | floren | the problem is: i know my configs are wrong, related to the devices |
| 21:34.10 | [TK]D-Fender | INVITE sip:5143602121@192.168.1.8:5060 SIP/2.0 |
| 21:34.14 | ChannelZ | floren: well your phone isn't registering so Asterisk doesn't know its IP (and thus can't call it, or match it as a peer for incoming) at least according to your sip show peers |
| 21:34.23 | [TK]D-Fender | Came in targeting :5143602121 |
| 21:34.41 | floren | 5143602121, that is the voip.ms did |
| 21:34.41 | [TK]D-Fender | <PROTECTED> |
| 21:34.57 | [TK]D-Fender | yes, and it comes in.. and you dial OUT to YOUR OWN PHONE NUMBER |
| 21:35.04 | floren | :) |
| 21:35.05 | [TK]D-Fender | you are LOOPING |
| 21:35.18 | [TK]D-Fender | You are clearly not paying attention to where you are sending your calls |
| 21:35.21 | floren | i'm smiling but is not funny. ya i noticed |
| 21:35.31 | floren | the problem is, i dont know what to chnage into config |
| 21:35.37 | [TK]D-Fender | your DIALPLAN |
| 21:35.49 | [TK]D-Fender | you pointed your incomcing call into a context that just sends it back out |
| 21:35.53 | [TK]D-Fender | change where it goes <- |
| 21:35.57 | floren | [101-cisco] |
| 21:35.58 | floren | exten => 5143602121,1,Answer() |
| 21:35.58 | ChannelZ | the stuff in your 101-voipms should probably really be in 101-cisco and vice-versa |
| 21:35.59 | floren | include => 101-voipms |
| 21:36.03 | floren | this si wrong |
| 21:36.05 | floren | i know it |
| 21:36.08 | [TK]D-Fender | CHANGE IT |
| 21:36.15 | [TK]D-Fender | you provider should go to it's own context |
| 21:36.16 | Penguin | [TK]D-Fender: its |
| 21:36.19 | floren | ya, i tried that |
| 21:36.26 | ChannelZ | though you need to not just Answer but have it dial your local device. |
| 21:36.28 | floren | let me do it again and post the debug |
| 21:36.39 | [TK]D-Fender | Looking for 5143602121 in 101-voipms (domain 192.168.1.8) <- |
| 21:36.51 | [TK]D-Fender | that context allows it to go out and you are not patying attention to its contents |
| 21:37.56 | floren | [TK]D-Fender: if i would have on hand a real working config, i would understand a lot faster. as the verbal language is confusing even more as i'm not used to asterisk terms |
| 21:38.34 | floren | let me switch the contexts |
| 21:38.44 | [TK]D-Fender | that is a bucket that hold the things that match what is dialed |
| 21:38.54 | [TK]D-Fender | in there you have something that matches the call coming IN from your provider |
| 21:39.07 | [TK]D-Fender | and the thing you have in there that matches it... is told to DIAL OUT |
| 21:39.26 | [TK]D-Fender | You need toseparate what a PHONE can dial, and where calls from your PROVIDER go |
| 21:39.30 | [TK]D-Fender | this is 2 DIFFERENT places |
| 21:39.45 | [TK]D-Fender | You put stuff that dials OUT in the IN BOX |
| 21:41.11 | floren | i have no idea how to do this [TK]D-Fender |
| 21:41.27 | [TK]D-Fender | [makeanothersection] <---- |
| 21:41.34 | floren | new extensions.conf: http://codepad.org/TYfSnGVG |
| 21:42.03 | [TK]D-Fender | include => 101-voipms <- your context ./... should not INCLUDE ITSELF |
| 21:42.07 | ChannelZ | remove your include |
| 21:42.10 | [TK]D-Fender | You keep trying to make everything talk to itself |
| 21:42.20 | floren | heh :) |
| 21:42.21 | [TK]D-Fender | POhonnes point to ONE context, your procvider entries point to NAOTHER |
| 21:42.26 | [TK]D-Fender | make 2 contexts |
| 21:42.30 | [TK]D-Fender | they should look NOTHING alike |
| 21:42.34 | floren | ChannelZ: ok let me post the new config |
| 21:42.50 | ChannelZ | and add exten => 5143602121,n,Dial(SIP/101) after the Answer or nothing is going to happen |
| 21:43.37 | [TK]D-Fender | show us a new call and new dialplan |
| 21:43.48 | floren | one sec :) |
| 21:44.14 | floren | first i want to make sure i got right what you guys told me to do, let me post the new config |
| 21:44.30 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 21:44.53 | [TK]D-Fender | [17:43][TK]D-Fendershow us a new call and new dialplan |
| 21:45.39 | floren | is that right? http://codepad.org/CT2ixHcA |
| 21:45.59 | floren | if is right, i'll post the call log |
| 21:46.04 | ChannelZ | no |
| 21:46.12 | ChannelZ | it's completely upside down |
| 21:46.54 | [TK]D-Fender | [101-voipms] <-- this is where you pointied your PROVIDER to. It has stuff going OUT in there. This is BAD |
| 21:47.09 | floren | ChannelZ: can you edit the code? this is what you mean? http://codepad.org/AMaIRkLH |
| 21:47.36 | [TK]D-Fender | floren: exten => 5143602121,1,Answer() <-- blengs in the OTHER context |
| 21:47.42 | [TK]D-Fender | You are breaking your extension patterns up here... |
| 21:47.48 | [TK]D-Fender | belongs* |
| 21:48.03 | floren | [TK]D-Fender or ChannelZ: at the bottom of the page you can edit the code. please make me happy :) |
| 21:48.07 | ChannelZ | http://codepad.org/v1BWlmMy |
| 21:48.26 | [TK]D-Fender | floren: MOVE LINE 14 |
| 21:48.32 | [TK]D-Fender | \Guess where... |
| 21:48.36 | floren | ChannelZ: ok I'm adding this into config pand post a call trace |
| 21:49.07 | [TK]D-Fender | floren: Get rid of [default] entriely, and remove all this "101" junk |
| 21:49.18 | ChannelZ | calls from your provider wind up in [101-voipms] so the extension there handles the incoming call and dials your SIP/101 device. Calls from your device wind up in [101-cisco] so the extensions THERE are for making outgoing calls. |
| 21:49.28 | [TK]D-Fender | floren: EVERYTHING is "101"-something the way you keep wording it and it isn't helping you mentally separate things |
| 21:49.30 | ChannelZ | And yes I didn't even deal with the default section, it should just go |
| 21:50.03 | floren | one sec, let me crate a separate entry code |
| 21:51.42 | ChannelZ | I think this is about to get worse |
| 21:52.07 | floren | sip.conf: http://codepad.org/vFpnfBqT extensions.conf: http://codepad.org/40AqjJrP |
| 21:52.24 | floren | can you guys please chnage the names to whatever you like so i understand the logic? |
| 21:52.53 | ChannelZ | it doesn't matter if we understand it |
| 21:52.54 | floren | btw, i really apreciate your help |
| 21:52.54 | ChannelZ | you need to |
| 21:53.27 | floren | agreed, but if you do it i will get it rightaway. using a descriptive title will help me, like voip-incoming etc |
| 21:54.39 | [TK]D-Fender | floren: you are shoving "101" in EVERY name you have. IN your templates for a phone, in every context name including the one used by your provider, etc. |
| 21:54.41 | ChannelZ | what you have will work (I think), it just makes no particular sense to call your context "101-voipms" because 101 has nothing to do with anything. |
| 21:54.51 | [TK]D-Fender | floren: ^ |
| 21:56.14 | floren | thank you guys. now let post the way i see things go |
| 21:57.03 | floren | the process is simple: cisco 8961 registers 101 then 101 registers to voip.ms, that's how i see the communication going |
| 21:57.21 | ChannelZ | nope. |
| 21:57.35 | ChannelZ | The phone on your desk has nothing to do with voip.ms |
| 21:57.48 | floren | the phone talks to asterisk only, right? |
| 21:57.55 | floren | then asterisk talks to teh provider |
| 21:58.04 | ChannelZ | yes |
| 21:58.11 | floren | great, at least this i see it right |
| 21:58.16 | ChannelZ | and connects calls between the two (if you make it) |
| 21:59.03 | floren | now, [cisco] is a startup point, as I will have 3 lines 101, 102 and 103 |
| 21:59.26 | [TK]D-Fender | don't call those "lines" |
| 21:59.38 | [TK]D-Fender | you may have 3 DEVICES .... |
| 21:59.48 | floren | [101](cisco) is used to handle calls from SIP phone |
| 21:59.57 | [TK]D-Fender | sip phone = device |
| 22:00.04 | floren | aha great |
| 22:00.13 | [TK]D-Fender | lets get your terminology straightened out right now |
| 22:00.35 | floren | thank you guys, much apreciated :) |
| 22:01.02 | [TK]D-Fender | There is no reason to have a template for your voip.ms entry as well. |
| 22:01.10 | floren | now, the [101](cisco) context should be used for calls made FROM voip.ms |
| 22:01.15 | floren | do i get it right? |
| 22:01.20 | [TK]D-Fender | It has nothing to do with your phone(s) and should have anything reusable in there anyway |
| 22:01.22 | ChannelZ | nope |
| 22:01.35 | [TK]D-Fender | [18:00]florennow, the [101](cisco) context should be used for calls made FROM voip.ms ---> TO Voip.ms |
| 22:01.51 | floren | i see that is what i was missing |
| 22:01.58 | [TK]D-Fender | You dial TO them. You get calls FROM them |
| 22:02.13 | [TK]D-Fender | Not understanding in vs out is a bad start |
| 22:02.17 | *** join/#asterisk guitarHester (~guitarHes@mobile-166-147-126-191.mycingular.net) |
| 22:02.24 | floren | fantastic, this is so much clearer in my head |
| 22:03.26 | floren | [101](cisco): incomming calls from voip.ms, outgoing calls to voip.ms |
| 22:03.40 | [TK]D-Fender | no |
| 22:03.51 | [TK]D-Fender | that is your PHONE |
| 22:04.12 | [TK]D-Fender | that is ONLY to say where to send calls FROM your phone |
| 22:04.24 | floren | i see |
| 22:04.48 | floren | btw, after al this is done, please give me your paypal so i make a donation, teh beer is on me tonight |
| 22:06.12 | [TK]D-Fender | I still recommend getting rid of your use of templates. |
| 22:06.24 | ChannelZ | I hate beer and paypal |
| 22:06.30 | floren | heh |
| 22:06.33 | [TK]D-Fender | It isn't worth the price of confusion to be using those at all. |
| 22:06.41 | floren | you can buy whatever drinks you enjoy ChannelZ |
| 22:06.53 | floren | i dont mind making a donation, you both deserve it |
| 22:07.03 | floren | i really apreciate it guys |
| 22:07.15 | ChannelZ | thanks but I'm fine |
| 22:07.58 | floren | now, i will remove the template usage as you requested |
| 22:13.43 | floren | sip.conf: http://codepad.org/FT7mkWPt extensions.conf: http://codepad.org/40AqjJrP |
| 22:14.12 | floren | sorry wrong link |
| 22:14.15 | floren | one sec |
| 22:14.32 | floren | extensions.conf http://codepad.org/vQcGeglU |
| 22:14.47 | floren | i think i finally got the logic |
| 22:14.53 | floren | please say yes :) |
| 22:15.08 | ChannelZ | looking better |
| 22:15.10 | floren | i renamed the contexts to reflect that |
| 22:15.27 | floren | thanks you :) |
| 22:15.40 | ChannelZ | Looks like you are understanding the flow more |
| 22:15.53 | floren | ya is very clear now, finally |
| 22:16.04 | floren | i understand a lot faster when i see a quick example |
| 22:16.29 | floren | i just chnaged the names to show that i'm not a lunatic heh |
| 22:17.25 | floren | now, i have another question related to var/log/asterisk/messages entry |
| 22:17.41 | floren | [Jul 7 17:41:26] WARNING[3798] chan_sip.c: Subscription failed for MWI. The remote side said that our dialog did not exist. |
| 22:18.06 | floren | mwi => 123456:voipmspassword@montreal.voip.ms:5060/456021 |
| 22:18.31 | floren | 456021 is the voip.ms voicemail id |
| 22:18.36 | ChannelZ | are you wanting to use their voicemail and not your own? |
| 22:18.42 | floren | theirs |
| 22:19.04 | floren | for now |
| 22:19.21 | floren | i just need to get the damn line working i'm without the phone for 4 days |
| 22:19.34 | floren | then i will dig into asterisk slowly and learn everything |
| 22:20.13 | floren | let me apply the new configs and reload asterisk |
| 22:20.53 | ChannelZ | well I guess I don't know specifically with voip.ms if they allow that remote subscription or what. They might do it automatically and send you notices without having to do that, dunno. |
| 22:21.22 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 22:29.26 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 22:33.42 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 22:36.40 | floren | i still get issues when i dial from my mobile to the DID number. i get the voicemail |
| 22:36.43 | floren | http://codepad.org/ILmfjMrE |
| 22:37.45 | floren | calling from 101 line to outside world does not work also :( |
| 22:38.00 | floren | hours of pleasure with asterisk :) |
| 22:38.12 | floren | at least i got the logic clear into my head |
| 22:38.29 | ChannelZ | I think your device is still not registering |
| 22:38.34 | ChannelZ | what does 'sip show peers' show |
| 22:38.41 | floren | is registered |
| 22:39.15 | floren | montreal.voip.ms:5060 N 123456 105 Registered Sun, 07 Jul 2013 18:38:44 |
| 22:39.21 | ChannelZ | that's not your phone |
| 22:39.23 | floren | and the peers... |
| 22:39.26 | *** join/#asterisk bazman (~army@220-245-36-238.static.tpgi.com.au) |
| 22:39.42 | floren | 101/101 (Unspecified) D 0 Unmonitored |
| 22:39.45 | floren | voipms 67.205.74.184 a 5060 Unmonitored |
| 22:39.49 | ChannelZ | see (Unspecified) |
| 22:39.55 | floren | aha |
| 22:40.03 | ChannelZ | Asterisk can't call SIP/101 because it doesn't know how. |
| 22:40.07 | floren | should say 192.168.1.8 i presume |
| 22:40.12 | ChannelZ | yeah |
| 22:40.31 | ChannelZ | you can either assign it that IP statically in sip.conf or config the phone to register properly |
| 22:40.58 | floren | well, the phone is shows as registered |
| 22:41.08 | ChannelZ | it's confused |
| 22:41.18 | floren | i see all the proper info onphone screen |
| 22:41.27 | ChannelZ | reboot it and see if it registers |
| 22:41.54 | floren | i can set a static ip into [101]? |
| 22:42.01 | floren | ok |
| 22:43.12 | floren | i restarted asterisk service and rebooted the phone... the phone shows as registered |
| 22:43.29 | floren | still unspecified |
| 22:43.33 | ChannelZ | then it didn't |
| 22:43.35 | floren | i will look into it |
| 22:44.04 | floren | the way i know the phone is registered is: the screen turns black and the dim light (ON) shows |
| 22:44.19 | floren | if the phone config is wrong, it will say on screen: phone not registered |
| 22:44.31 | floren | it does not, i see my 101 line |
| 22:44.54 | floren | i'll fiddle with the config until i get an ip into peer |
| 22:45.05 | floren | at least i know what to work with |
| 22:45.46 | ChannelZ | what do you have in the phone config for the server or proxy? |
| 22:46.39 | floren | CALLMANAGER is the way you need to make it work. cannot define actual ip or fqdn |
| 22:46.44 | floren | i'm trying something now |
| 22:47.42 | ChannelZ | ugh CM is a totally different thing |
| 22:47.49 | floren | yeah |
| 22:48.04 | floren | if you use an ip, it will not register |
| 22:48.20 | floren | can i define an ip into sip.conf, under [101]? |
| 22:48.43 | ChannelZ | yes, host= |
| 22:49.55 | floren | the host is dynamic |
| 22:50.12 | ChannelZ | which is why Asterisk doesn't know the IP. |
| 22:50.23 | floren | i can put there an actual ip? |
| 22:50.30 | ChannelZ | because your (call manager I guess) doesn't register to Asterisk |
| 22:50.34 | ChannelZ | yes, that's what I said. |
| 22:51.09 | floren | ok now i get an ip |
| 22:51.23 | floren | 101/101 192.168.1.9 5060 Unmonitored |
| 22:51.26 | floren | voipms 67.205.74.184 a 5060 Unmonitored |
| 22:51.36 | floren | 192.168.1.9 is the actual phone ip, static |
| 22:52.13 | floren | asterisk runs on 192.168.1.8 |
| 22:52.20 | ChannelZ | but your phone doesn't actually talk SIP apparently? It's connecting through call manager? |
| 22:52.26 | ChannelZ | I don't know how that shit works |
| 22:52.37 | floren | ya |
| 22:52.44 | floren | it is using the call manager |
| 22:53.09 | *** join/#asterisk guitarHester (~guitarHes@71-90-253-28.dhcp.leds.al.charter.com) |
| 22:53.16 | floren | i logged intoa ctual phone to watch the logs, there is very nice info there showing the communication between the phone and asterisk |
| 22:53.34 | floren | cisco did a good job in this area, for debugging purposes |
| 22:53.49 | floren | ok teh phone shows as registered and i get an ip |
| 22:55.06 | floren | i can make inbound calls now, calling from my mobile to cisco phone works :D |
| 22:55.21 | floren | but calling from cisco to mobile does not :( |
| 22:56.23 | [TK]D-Fender | Next... don't say "mobile". That is not a "thing" here. |
| 22:56.54 | [TK]D-Fender | So far I see 2 SIP entries.... which represent different things... the word "mobile" is not unique or specifically meaningful in either. |
| 22:57.21 | floren | well it works partially, for inbound calls. when i call from my mobile phone to cisco phone, i get this: |
| 22:57.32 | *** join/#asterisk igcewieling1 (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 22:58.40 | floren | http://codepad.org/8Cclnbh8 |
| 22:59.17 | floren | let me disable ALG in my router |
| 22:59.22 | [TK]D-Fender | #1 thing to do <- |
| 22:59.23 | ChannelZ | wait that makes no sense |
| 23:01.15 | floren | disabled ALG, restarting asterisk service and resetting the phone |
| 23:01.18 | ChannelZ | My understanding is that you phone (device) does NOT speak SIP. It talks to Call Manager. Call Manager speaks SIP and is the bridge between your phone and Asterisk. |
| 23:01.46 | floren | it does speak SIP but call manager deals with proxy registration |
| 23:02.03 | ChannelZ | that makes no sense |
| 23:02.20 | [TK]D-Fender | this is network issue anyway... |
| 23:02.23 | floren | there is a patch made by asterisk devs that solve the compatibility issues there are many people now who use this model properly with the patch |
| 23:02.27 | ChannelZ | Is Call Manager a separate thing? (A piece of software running on some other machine, or a black box, or what?) |
| 23:02.28 | [TK]D-Fender | "sip set debug on" <--------- |
| 23:02.57 | floren | [TK]D-Fender: asterisk -rvvvvvvvvvv |
| 23:03.05 | floren | and sip set debug on |
| 23:03.17 | floren | i always run it like that to help see more |
| 23:03.41 | ChannelZ | In any case it seems like whatever is at 192.168.1.9 isn't listening and doesn't give a shit |
| 23:03.56 | floren | peers look good, registry also |
| 23:04.07 | floren | i'm placing a call to my DID now, let's see |
| 23:04.42 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.145) |
| 23:07.38 | floren | this is the call trace: http://codepad.org/WE4PaTKa |
| 23:08.22 | *** join/#asterisk DEMNVT (~Adium@rmsaus7.lnk.telstra.net) |
| 23:08.40 | ChannelZ | etransmitting #1 (no NAT) to 192.168.1.9 |
| 23:08.52 | floren | when i call my DID, it rings once on my 101 device and the call dies. |
| 23:08.55 | ChannelZ | Again: whatever 192.168.1.9 is, isn't listening or doesn't care. |
| 23:09.20 | ChannelZ | or your networking is otherwise hosed such that it's not communicating freely. |
| 23:09.25 | floren | ChannelZ told me to set the ip, i set the actual cisco ip, not asterisk'd |
| 23:09.47 | floren | cisco phone 192.168.1.9, asterisk 192.168.1.8 |
| 23:10.27 | ChannelZ | I don't know what to say with this Call Manager involved because I don't know anything about it. |
| 23:10.47 | floren | there is something funny, i managed to make outbound calls without issues, using my old config |
| 23:11.04 | ChannelZ | My understanding is that your actual device DOES NOT SPEAK SIP, otherwise CM wouldn't even be in this vocabulary |
| 23:11.27 | igcewieling1 | ChannelZ: Call Manager is the Cisco enterprise PBX |
| 23:11.29 | [TK]D-Fender | floren: You have spoken about the phone... you have spoken about Asterisk. You have spoken about voi.ms. |
| 23:11.30 | floren | let me do this, put in place the old config that allows outbound calls |
| 23:11.45 | [TK]D-Fender | floren: where the hell is CISCO CALL MANAGER in this equation? |
| 23:12.12 | floren | CALLMANAGER is a program built into actual phone |
| 23:12.13 | ChannelZ | igcewieling1: right.. and it's being a bridge between the device that are presumably speaking Skinny or whatever the hell it's called, and Call Manager is what is actually talking SIP to Asterisk, yes? |
| 23:12.15 | [TK]D-Fender | floren: You have not properly described the entire chain here.... |
| 23:12.30 | [TK]D-Fender | ChannelZ: Sit back and lets just wait for the explanation |
| 23:12.43 | floren | [TK]D-Fender: let me say it again, the proper way: |
| 23:13.43 | floren | cisco phones are designed to communicate with CUCM severs ONLY through CM, in other words, CM what makes possible the comm between the CUCM and actual phone |
| 23:13.54 | floren | so the flow is: |
| 23:14.07 | floren | phome > CM > CUCM > CM > phone |
| 23:14.25 | floren | now, the asterisk devs made a patch making possible the flow like that: |
| 23:14.32 | [TK]D-Fender | ... |
| 23:14.33 | floren | phome > CM > Asterisk > CM > phone |
| 23:14.33 | [TK]D-Fender | no |
| 23:14.46 | ChannelZ | what is this patch now!? |
| 23:15.04 | floren | https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 23:15.05 | LieutPants | [ASTERISK-13145] [Status: Open] [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML - https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 23:15.24 | [TK]D-Fender | that is one stupid presence patch |
| 23:15.30 | [TK]D-Fender | that is no t "can't call without this patch". |
| 23:15.45 | floren | https://issues.asterisk.org/jira/browse/ASTERISK-13145?focusedCommentId=204848&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-204848 |
| 23:15.46 | LieutPants | [ASTERISK-13145] [Status: Open] [patch] Presence subscription on Cisco SIP phone needs special Cisco-styled XML - https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 23:15.53 | [TK]D-Fender | Who cares about presence? |
| 23:15.56 | [TK]D-Fender | you are placing CALLS |
| 23:16.02 | [TK]D-Fender | NOT RELATED |
| 23:16.05 | ChannelZ | yeah this has nothing to do with anything |
| 23:16.50 | floren | well, if you read through the notes the only way to get the cisco 89xx and 99xx work and register with asterisk is to apply the patch |
| 23:17.00 | [TK]D-Fender | I don't care how many times it says "cisco" all over it.... you have NO presence set up and there is NO SUBSCRIPTION even going on here. |
| 23:17.02 | floren | i tried first without the patch |
| 23:17.13 | floren | it simply did nto work, the phone was never registering |
| 23:17.21 | [TK]D-Fender | We can't voice for that. |
| 23:17.24 | [TK]D-Fender | Anyway, lets move on. |
| 23:17.29 | floren | ya :) |
| 23:17.34 | [TK]D-Fender | I asked for your CURRENT chain <- |
| 23:17.47 | [TK]D-Fender | [19:14]florenphome > CM > Asterisk > CM > phone <- this is not it |
| 23:18.03 | [TK]D-Fender | You have a PROVIDER.... and you didn't meantion a SECON CM |
| 23:18.24 | [TK]D-Fender | So you are giving us hypothetical junk possibilties and not reality |
| 23:18.42 | floren | phone and asterisk server connects to cisco switch/router |
| 23:18.57 | floren | locally |
| 23:19.05 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 23:19.10 | [TK]D-Fender | Try drawing that in a proper chain.... |
| 23:19.59 | floren | phone > cisco router > asterisk server > cisco router > phone |
| 23:20.44 | floren | cisco router assigns static ip's to each hardware |
| 23:21.25 | floren | phone 192.168.1.9 asterisk 192.168.1.8 cisco gateway 192.168.1.1 |
| 23:21.26 | [TK]D-Fender | yNow you aren't being clear on which one(s) are running CCM and which is just being a DUMB ROUTER. |
| 23:21.39 | [TK]D-Fender | Do you understand the word "thorough"? |
| 23:22.01 | floren | yes |
| 23:22.06 | [TK]D-Fender | You are using the word "cisco" for EVERYTHING |
| 23:22.13 | [TK]D-Fender | Be specific about what they are RUNNING |
| 23:22.25 | [TK]D-Fender | And how it affects the chain |
| 23:22.49 | floren | is a cisco SA520W security pro router, i cannot be more specific, sorry |
| 23:22.54 | [TK]D-Fender | WHICH ONE? |
| 23:23.02 | [TK]D-Fender | you said TWO Cisco routers. |
| 23:23.08 | [TK]D-Fender | I can;'t tell WHICH ONE you are even talking about\ |
| 23:23.22 | floren | no, just one router |
| 23:23.38 | [TK]D-Fender | [19:19]florenphone > cisco router > asterisk server > cisco router > phone <-- why do I see the same device TWICE here? |
| 23:23.39 | floren | is the main gateway connecting all my devices |
| 23:23.46 | [TK]D-Fender | I asked for a chain and you show me a LOOP |
| 23:24.09 | [TK]D-Fender | I shouldn't see the same device TWICE |
| 23:24.49 | floren | i could draw a picture if that helps you. the setup is very simple. a gateway has connected all my devices (asterisk server, ip phone, 3 dev servers, etc.) |
| 23:25.11 | [TK]D-Fender | Polycom SIP phone -> dumb switch -> Asterisk -> shitty D-Link Router -> internet -> Voip Provider. |
| 23:25.13 | [TK]D-Fender | THAT ias a chain |
| 23:25.25 | [TK]D-Fender | straight lien, NO repeats |
| 23:25.27 | [TK]D-Fender | line* |
| 23:26.46 | floren | well, both sip phone and asterisk server connect to router |
| 23:26.49 | ChannelZ | What version of the Crisco firmware is your phone running? |
| 23:28.01 | floren | sip8961.9-3-2SR1-1 |
| 23:29.52 | [TK]D-Fender | What is CCm actually doing for you? |
| 23:30.32 | ChannelZ | Those phones are apparently SIP trainwrecks from everything I'm reading |
| 23:30.44 | floren | registers the device? |
| 23:30.48 | [TK]D-Fender | .... |
| 23:30.54 | [TK]D-Fender | How is this a QUESTION? |
| 23:30.59 | floren | deals with proxy registration |
| 23:31.20 | [TK]D-Fender | Is it some useless piece of junk that is just sitting inthe middle JUST BECAUSE? |
| 23:31.22 | ChannelZ | they apparently only did SIP over TCP until 2011 |
| 23:31.39 | floren | ya, they switched to upd recently |
| 23:31.52 | floren | i setup the phone with forced udp |
| 23:32.23 | ChannelZ | well apparently it's still not listening correctly, or has a wierd port number, or something.. |
| 23:32.49 | [TK]D-Fender | Nothing is apparent. I don't see debug |
| 23:33.44 | [TK]D-Fender | Or do I... |
| 23:33.45 | ChannelZ | you missed it then.. Asterisk tries to INVITE 192.168.1.9, nothing happens, 5 retransmissions, blah. |
| 23:33.45 | floren | if it helps, i'm going to setup the old config that allowed palcing outbound calls properly |
| 23:33.48 | [TK]D-Fender | searches back |
| 23:34.18 | ChannelZ | http://codepad.org/ILmfjMrE I think |
| 23:34.28 | floren | at least it shows that the phone actually works |
| 23:34.38 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 23:34.50 | floren | if you have time, let me know so i reset everything the way it was before |
| 23:35.07 | floren | so i can post a sip debug |
| 23:35.15 | floren | for the working part |
| 23:36.07 | ChannelZ | well just pastebin the configs to start, you don't have to actually install them - I'm curious to see what the hell was even there |
| 23:36.38 | floren | i have to redo the configs to match the logic you teached me |
| 23:36.45 | floren | in thsi way we see thinsg clear |
| 23:36.53 | floren | not on 101, 102. etc |
| 23:37.23 | ChannelZ | just sip.conf is all I want to see. |
| 23:37.48 | ChannelZ | I don't care what things were called then, you said it worked.. |
| 23:38.18 | ChannelZ | but something about the peer config is different |
| 23:38.23 | floren | http://codepad.org/WP1Ch5aU |
| 23:40.19 | floren | and extensions.conf was http://codepad.org/fTk02Lmb |
| 23:40.48 | floren | this works to place outbound calls only |
| 23:41.43 | ChannelZ | I don't see how. |
| 23:42.24 | floren | let me post a debug so you see better |
| 23:42.31 | [TK]D-Fender | [19:38]florenhttp://codepad.org/WP1Ch5aU <- what is this? Looks like you undid all the wrok we did |
| 23:42.47 | ChannelZ | [TK]D-Fender: those were his old configs that supposedly worked |
| 23:42.49 | floren | no, i still have the configs you asked in place |
| 23:43.01 | [TK]D-Fender | Then why am I looking at old crap? |
| 23:43.06 | floren | i pasted my previous config that was working only for outbound calls |
| 23:43.07 | ChannelZ | I asked to see it |
| 23:43.21 | floren | ya ChannelZ asked for |
| 23:43.41 | [TK]D-Fender | Yes well that las failure was an INBOUND call from provider |
| 23:43.43 | ChannelZ | and I think "worked" is a stretch. It might have done *something* but I don't see how it could possibly work. |
| 23:44.11 | floren | i see |
| 23:44.13 | ChannelZ | Your two peers have the same hostname/IP, so [voipms] and [101] are the same thing. |
| 23:44.19 | [TK]D-Fender | provider -> * -> Cisco |
| 23:44.26 | ChannelZ | So I don't know how your actual phone ever did ANYTHING. |
| 23:44.28 | floren | i understand |
| 23:44.32 | [TK]D-Fender | that was not a call OUT from cisco, that was IN towards Cisco |
| 23:44.54 | [TK]D-Fender | ChannelZ: that PB you asked for was while he was splitting templates up |
| 23:45.34 | floren | can i redo teh config based on what you tought me and see if it works for outbound calls? |
| 23:45.47 | floren | i really like to show you the debug log |
| 23:45.51 | ChannelZ | Well I asked for the config that supposedly "worked before" but I guess that was a dumb thing to ask for, because if it EVER worked, we wouldn't have been having this conversation for the last 2 hours :) |
| 23:46.03 | floren | :) |
| 23:47.38 | ChannelZ | When you say it "worked for outbound calls", you're saying you actually picked up the handset on this Cisco device sitting on your desk, dialed a number, and Astierk lit up and placed the call out through your voip.ms provider, and the call connected and functioned?? |
| 23:47.49 | floren | yes |
| 23:48.02 | floren | i was calling different land lines and talked for over 15min to test it |
| 23:50.04 | ChannelZ | I see now. |
| 23:50.42 | ChannelZ | Your phone sent a totally unauthenticated call to Asterisk which it accepted, sent into the [default] context, which in your old dialplan included all your other extensions, which allowed it to dial back out. |
| 23:51.56 | ChannelZ | Your phone wasn't ever _really_ connected to Asterisk in a proper way. Now that we have your configs sorted out and making what is normal sense, your device is crippled because it's never really ever been right. |
| 23:52.12 | floren | i see |
| 23:53.05 | ChannelZ | What happens if you try to dial OUT from that phone right now? |
| 23:53.17 | floren | nothing it sits there doing nothing |
| 23:53.43 | floren | i hear a dial tone after the number is composed and it sits there |
| 23:53.43 | ChannelZ | you see *nothing* on the asterisk console? |
| 23:53.47 | floren | no |
| 23:53.51 | floren | nada |
| 23:53.59 | floren | i could log into actual phone |
| 23:54.01 | ChannelZ | then the phone is screwed up |
| 23:54.15 | floren | so i see what it shows there, they have output similar to asterisk console |
| 23:54.24 | ChannelZ | it's either sending its traffic to completely the wrong place, or nowhere at all, or not SIP, or... who knows |