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| 01:38.38 | igcewieling | sigh Packets: Sent = 2374, Received = 2237, Lost = 137 (5% loss), |
| 01:39.16 | Katty | hugs igcewieling |
| 01:39.19 | Katty | igcewieling: this calls for beer :< |
| 01:52.59 | Kobaz | mmm foods |
| 01:53.20 | Kobaz | Mes Reves |
| 01:55.09 | igcewieling | It calls for something, that is for sure. |
| 01:57.11 | Katty | someone come play xbox with me |
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| 03:02.39 | krapper | is mpg123 required for format_mp3. module to compile? |
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| 03:41.30 | dlynes | Does disposition or hangupcause expose themselves to agi? |
| 03:41.58 | dlynes | krapper, yes, or 123mpg |
| 03:42.23 | dlynes | krapper, if you're building it from the command line, you'll see that you need to go into the mpg123 subdirectory to build it and install it |
| 03:42.35 | dlynes | krapper, so you don't have to worry about which version you need |
| 03:50.29 | Kobaz | http://t.qkme.me/3t1ogl.jpg |
| 03:57.36 | dlynes | Kobaz, Murphy's new law? |
| 03:58.22 | igcewieling | dlynes: you can get the value of those variables from an AGI yes. |
| 04:06.32 | Kobaz | hah yeah |
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| 08:39.35 | jaxon007_ | I am playing an audio file in asterisk using playback , it was playing very well but now there are some issues in voice ..while playing its occur some silence and some break on audio ..I installed asterisk on amazon servers ..what will be the issues |
| 08:40.35 | jaxon007_ | asterisk version -1.8.17.0 |
| 08:42.58 | ectospasm | could be many things |
| 08:43.31 | ectospasm | ...codec selection, available bandwidth/CPU, proper QoS not set up, etc. |
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| 08:48.49 | jaxon007_ | @ectopasm What are the thing we need to look on Qos? |
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| 08:54.34 | ectospasm | jaxon007_: I'm not a QoS expert, but you need to give priority to the RTP traffic |
| 08:54.55 | ectospasm | that may require cooperation with your ISP |
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| 09:23.32 | atan | Yay, 10.8 -> 11.3 went just beautifully! Asterisk FTW |
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| 10:09.22 | ectospasm | atan: not much difference between those, IIRC |
| 10:09.52 | ectospasm | I mean, anything that was deprecated in 10 is not in 11, etc., etc.... |
| 10:20.02 | emk | How do I add modules _AFTER_ I've already built asterisk and run `make install`? I need to add format_mp3 etc to my "Asterisk 11.3.0" install so that I can install FreePBX. |
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| 10:28.18 | WIMPy | Do it again. |
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| 13:37.05 | Dovid | hello everyone |
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| 15:27.28 | igcewieling | emk: usually you don't, you rebuild asterisk. harmless thing to do. make; make install don't do any other makes |
| 15:29.51 | Kobaz | harmless you say |
| 15:30.04 | Kobaz | ...assuming you didn't add any bugs in between the last build and the new one |
| 15:30.30 | Kobaz | busily adds some bugs |
| 15:32.51 | igcewieling | bicycling is harmless unless you crash |
| 15:54.22 | Kobaz | harmless you say? |
| 15:54.34 | Kobaz | :P |
| 15:54.47 | Kobaz | what about the bugs that get squished as you ride |
| 15:54.57 | emk | igcewieling: thanks Ive just gone through the rebuild |
| 15:55.29 | Kobaz | and to think that this all revolves around bugs... oh the humanity |
| 15:55.54 | emk | but I can't build chan_dongle now |
| 15:56.03 | Kobaz | pastebin |
| 15:56.34 | emk | Anyone know if AsteriskNOW (latest) is compatible with the latest chan_dongle (on google code)? I'm gettin errors building it. |
| 16:06.10 | emk | Here's the pastebin: http://pastebin.com/C9CnftkD... I'm trying to install/build chan_dongle on AsteriskNOW (Asterisk version 11) |
| 16:06.40 | igcewieling | emk: so few people use chan_dongle you might have better luck searching the mailing lists |
| 16:07.29 | igcewieling | looks to me like chan_dongle may not be compatible with Asterisk 11 |
| 16:07.35 | igcewieling | or Asterisk 12 8-) |
| 16:07.57 | emk | igcewieling: hmm, so is there an alternative for connecting USB modems? |
| 16:08.38 | emk | igcewieling: I've actually seen a lot of dead-ends in the mailing list. |
| 16:08.41 | igcewieling | emk: only crazy people do that, so no. |
| 16:09.08 | igcewieling | emk: what exactly are you trying to accomplish? |
| 16:10.15 | emk | I'm trying to get a Huawei Modem to behave as a trunk, make voice calls. |
| 16:11.06 | emk | And the Huawei Modems are the only hardware I have (no option to try building out anything else, with alternative hardware) |
| 16:11.46 | igcewieling | good luck. |
| 16:12.10 | igcewieling | your best bet is to downgrade to 1.8 and see if it works on that version |
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| 16:33.23 | stanreg | jmetro: hm |
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| 16:37.44 | racho | is it appropriate to create a catch-all context in extensions.conf and use an agi script to dispatch the calls to different locations from inside the script? |
| 16:39.14 | igcewieling | racho: only if your goal is an overly complex system with the likelyhood of a bug somewhere which allows toll fraud. |
| 16:40.00 | igcewieling | at a minimum you want trusted devices to go into one context and untrusted devices (like your ITSP) going to another context |
| 16:42.41 | racho | igcewieling, that asterisk box will play the role of a router to another asterisk box in our VPN. by definition all of the outgoing calls will originate from a trusted source. my sip.conf has the appropriate contactpermit/contactdeny lines |
| 16:43.05 | igcewieling | racho: then you should be ok with one context |
| 16:44.12 | racho | igcewieling, thank you for the clarification |
| 16:45.47 | igcewieling | racho: we do something similar with our core call routing boxes. Call comes in, an AGI is run to look up from a database how the call should be routed, the agi sets dialplan variables and exits, the dialplan then does the actual dialing, after the call ends an agi is run to clean some stuff up and log. |
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| 16:47.14 | racho | igcewieling, that's what we had in mind when started the cleaning of a crufty old extension.conf |
| 16:48.20 | igcewieling | racho: we handle routing for around 6,000 telephone numbers with a fairly small extensions.conf/extensions.ael. -= 43 extensions (421 priorities) in 17 contexts. =- |
| 16:51.35 | igcewieling | I bet you can clean out most of your extensions.conf by going to a database based setup |
| 16:53.36 | racho | igcewieling, actually the server in our VPN is full realtime. we just want outgoing calls to move through another box with a public ip |
| 17:04.29 | *** join/#asterisk ang-st (~pg@rsbac/developer/ang-st) |
| 17:05.48 | ang-st | hello |
| 17:07.29 | WIMPy | eek: Google for "chan_dongle Asterisk 11" and you will find a version that works. |
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| 17:22.23 | ang-st | i'm trying to dev a simple module that send tone via sip/rtp |
| 17:22.58 | ang-st | i've setuped like this : http://pastebin.com/eSYavjvT |
| 17:23.15 | WIMPy | Why do you need a module to do that? |
| 17:23.50 | ang-st | in fact i want to modulate v23 |
| 17:24.05 | ang-st | but the audio is scrambled when it goes over rtp |
| 17:24.12 | ang-st | http://imgur.com/EW26I54 |
| 17:24.40 | ang-st | the first signal is dumped from wireshark ... et |
| 17:24.53 | ang-st | the second from the monitor app |
| 17:25.13 | WIMPy | hopes you want to use it locally. |
| 17:25.29 | ang-st | yep :) |
| 17:27.34 | ang-st | anyway i have no clue on where looking to fix this audio thing |
| 17:34.41 | ang-st | http://pastebin.com/3YUKj5Ri sip debug trace |
| 17:35.24 | ang-st | thanks in advance :) |
| 17:35.33 | WIMPy | I guess you will have more luck to find somebody to comment on that stuff tomorrow. |
| 17:36.54 | ang-st | WIMPy: ok, i will hang around :) |
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| 17:41.21 | igcewieling | ang-st: ask on #asterisk-dev or the asterisk-dev mailing lists. |
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| 17:52.21 | ang-st | igcewieling: thanks will do |
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| 18:14.43 | danfromuk | Hi. I'm looking for a good oneway encryption solution and someone recommended blowfish. Has anyone got experience of this? Is it any good? |
| 18:16.44 | danfromuk | Oops. Wrong channel. |
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| 20:55.54 | hex`` | I'm just starting to use Asterisk. Which terms should I search/study in order to be able to get data (such as pressed button, etc) from a call? |
| 20:56.21 | WIMPy | To do a menu? |
| 20:57.14 | hex`` | yes, that too, but mainly I want to monitor which phone is chosing certain option in the menu |
| 20:57.31 | hex`` | I'm using a python module for asterisk called PyCall |
| 20:57.40 | WIMPy | That would happen locally on the phone. |
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| 20:57.58 | igcewieling | hex``: read The Book, the output from "core show applications" and "core show functions", if you are reading other documentation check the UPGRADE*.txt files to find the differences between what the old outdated docs say (like from voip-info.org) and YOUR version of Asterisk. |
| 20:58.01 | igcewieling | ~book |
| 20:58.02 | 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 |
| 20:58.28 | igcewieling | hex``: The phone does not send digits to Asterisk until after the you are done dialing. |
| 20:58.43 | WIMPy | Depends on the phone. |
| 20:58.52 | hex`` | Thanks for the guidance, igcewieling! |
| 20:58.55 | hex`` | I'll read it |
| 20:59.04 | WIMPy | But to get info abot menus sounds more like havong to hack a phone. |
| 20:59.10 | *** join/#asterisk tftech (~tftech@108-206-106-39.lightspeed.miamfl.sbcglobal.net) |
| 20:59.18 | igcewieling | hex``: if you are wanting to monitor what digits are received by asterisk during an IVR you would use AMI (asterisk manager). |
| 20:59.32 | hex`` | yes, that is exactly what I'm looking for |
| 20:59.42 | hex`` | ok, asterisk manager |
| 20:59.47 | igcewieling | hex``: The Book should have information on AMI / Asterisk Manager |
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| 21:00.35 | igcewieling | you may have to add UserEvents to your IVR "UserEvent: Send an arbitrary event to the manager interface" |
| 21:00.35 | hex`` | right, thanks a lot igcewieling! :) |
| 21:00.37 | tftech | Hello, I have an H323 question. I was able to get asterisk communicating both ways with an Avaya with H323. My issues is that I can only have one call at a time |
| 21:00.47 | tftech | I cannot find how to increase this |
| 21:01.04 | tftech | Any help is greatly appreciated |
| 21:01.42 | igcewieling | tftech: H323 with Asterisk is very unusual, it is unlikely anyone here uses it. You might want to ask on the mailing lists or search the mailing list archives (add site:lists.digium.com to your google search) |
| 21:02.38 | WIMPy | Yes, but we can be reasonably sure there's no limit to the number of calls. So that probably meants the Avaya manual is the right place. |
| 21:02.40 | tftech | thank you igcewieling, I understand that it is very rare. I was able to find information about the basic configuration. I am not however able to find out anything about channel limits |
| 21:03.06 | tftech | Of course the Avaya technician says that they are set for 10 calls and it must be on the Asterisk side |
| 21:03.09 | tftech | :) |
| 21:03.46 | tftech | I agree WIMPy, I do not see where to limit the channels for H323 |
| 21:03.48 | WIMPy | Increase verbose and possibly debug and try. You should be able to see what's going on. |
| 21:04.12 | tftech | WIMPy, the system is remote, I will try that as soon as I have access again. |
| 21:04.15 | tftech | Thank you all |
| 21:04.21 | tftech | I really appreciate all help |
| 21:04.49 | tftech | Have to go now |
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| 21:48.07 | _abc_ | Hello. I have trouble with newer cisco skinny phones and asterisk, the protocol seems to have changed. I get log error messages about phones sending out of order packets without registering, and registration fails. |
| 21:48.26 | iprouteth0 | curios. Why no SIP on the cisco phones? |
| 21:48.30 | _abc_ | Has anyone seen something like this? Pointers? Links? |
| 21:48.38 | _abc_ | The phones are sccp and are to stay sccp |
| 21:49.01 | WIMPy | Do you use a current Asterisk? |
| 21:49.06 | iprouteth0 | never run SCCP with asterisk. |
| 21:49.15 | _abc_ | the common point is they are 794x and 796x newer phones with very new firmware (post 2007) |
| 21:49.18 | iprouteth0 | You could use SCCP with call manager express and trunk via SIP to asterisk |
| 21:49.19 | _abc_ | older ones run great |
| 21:49.30 | _abc_ | iprouteth0: never say never. Why do you say that? |
| 21:49.35 | _abc_ | Please? |
| 21:49.55 | _abc_ | This is an old hacked astrisk but newer ones also have a problem there (I tried) |
| 21:49.58 | iprouteth0 | SCCP and call manager express are very easy to setup I suppose |
| 21:50.06 | _abc_ | I also read that phones bought off of ebay sometimes have such trouble |
| 21:50.08 | WIMPy | How new? |
| 21:50.32 | WIMPy | There has been quite some work on sccp recently. |
| 21:50.34 | iprouteth0 | I always just messed around with GNS3 and a 3745 IOS image that I installed call manager express on |
| 21:51.02 | _abc_ | The one whose wire level ethernet logs I am looking at now has fw SCCP41.9-3-1SR2-1S which is very weird and does not appear on the cisco list of firmware matrices |
| 21:51.06 | iprouteth0 | never done CME on real hardware since I didnt have the money for reall hardware. I have had very little trouble using SIP firmware on 79xx phones with asterisk however |
| 21:51.35 | iprouteth0 | what is the reason you need to keep them SCCP firmware? |
| 21:51.45 | _abc_ | in this case asterisk is an in between station in a lab, helping to sort out phones before they go out for real work |
| 21:51.49 | _abc_ | sometimes with asterisk sometimes with ccm |
| 21:52.03 | _abc_ | so no more upgrade talk please :) |
| 21:52.28 | _abc_ | what I need to know it, has someone seen serious trouble of this kind with cisco phones? |
| 21:52.38 | WIMPy | Then it's a case of bad luck, I guess. |
| 21:52.42 | iprouteth0 | I see. Afraid I can't offer much help at the moment as I've never run SCCP with asterisk. Suppose I could load one of my cisco phones with SCCP firmware, but it'd take a good bit of time |
| 21:52.43 | _abc_ | There are threads on the web about such trouble with ccm (no registration!) |
| 21:52.58 | _abc_ | iprouteth0: eh yes don't bother |
| 21:53.04 | iprouteth0 | I try to stay away from the cisco 79xx phones now. Too many problems |
| 21:53.10 | _abc_ | Indeed |
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| 21:53.19 | WIMPy | You should find the topic on jira as well, I think. |
| 21:53.22 | _abc_ | Yet some people buy them and are stuck with them, too big an investment |
| 21:53.30 | _abc_ | jira? |
| 21:53.33 | iprouteth0 | sip support on them is almost laughable in the way it handles it vs other manufacturers |
| 21:53.47 | _abc_ | laughable bad or good? |
| 21:53.48 | _abc_ | *y |
| 21:54.29 | _abc_ | cisco has this crazy versioning policy where there are enterprise versions of stuff they make which are custom builds outside the mainstream |
| 21:54.34 | iprouteth0 | I agree. Wish I could help you further. I have a lot of experience with those phones and SIP, and even a good bit of call manager express experience. Just very very little SCCP and asterisk |
| 21:54.41 | iprouteth0 | you could try MGCP which asterisk supports as well |
| 21:54.44 | _abc_ | you can't find info on those anywhere. The fw number I have above is such a custom one afaik |
| 21:55.07 | _abc_ | The 'SE' at the end is a telltale sign for such trouble |
| 21:55.18 | _abc_ | And I have no idea if the problem is the phone or asterisk |
| 21:55.20 | _abc_ | Bummer |
| 21:55.37 | _abc_ | Also there are several phones in the same situation but they come from the same installation |
| 21:55.49 | iprouteth0 | Wish I could find work around here doing asterisk consulting. I mainly just play around with my own gear. My endpoints are now android handsets running Csipsimple with SIP/TLS + sRTP |
| 21:56.32 | _abc_ | no idea what that is |
| 21:56.35 | iprouteth0 | you could try GNS3 and Call Manager Express just to test the phones |
| 21:56.40 | _abc_ | Won't touch goopile stuff :) |
| 21:56.56 | _abc_ | iprouteth0: yes, but it is more complex than that |
| 21:57.04 | iprouteth0 | csipsimple is a sip softphone on android. I run SIP with TLS encryption with certificates as well as an encrypted media stream |
| 21:57.10 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 21:57.15 | _abc_ | I assume someone else aready tried that before they landed in my jurisdiction |
| 21:57.33 | _abc_ | runs sip over ssh tunnels and vpns ... |
| 21:57.56 | _abc_ | anyway thanks for the tips, if anyone has anything to add, feel free |
| 21:58.17 | _abc_ | I'll go ask in #cisco too. The people there are umm not so open source friendly? |
| 21:58.20 | _abc_ | For some reason. |
| 21:58.24 | iprouteth0 | I used to used OpenVPN to run it over, but with SIP/TLS its not needed anymore. |
| 21:58.50 | iprouteth0 | lol. Yeah I don't know that I've ever really closely examined a packet stream of SCCP registration |
| 21:59.47 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 21:59.54 | iprouteth0 | I mainly used SCCP and MGCP on Call Manager 6 way back in 2008 and then SCCP again on GNS3 emulated router in 2010 I think. Been doing SIP on the 79xx's consistently since 2011 |
| 22:00.21 | iprouteth0 | could you reflash your phones with a different SCCP firmware? |
| 22:00.46 | _abc_ | probably, but after consulting with the real user(s) |
| 22:01.07 | _abc_ | #cisco is not so lively now and I am very tired, it's 1AM here |
| 22:01.15 | _abc_ | I'll come back tomorrow |
| 22:01.17 | _abc_ | thanks |
| 22:01.24 | igcewieling | _abc_: Cisco doesn't want you to use their phones with anything except CCM. |
| 22:01.27 | iprouteth0 | I think that might be your best route to get them working. Keeps them on the firmware you need |
| 22:01.44 | igcewieling | i.e. they don't want your business so I don't thnk it is fair to give them your business |
| 22:01.47 | _abc_ | igcewieling: yes but #cisco should help here on freenode, or be booted asap off freenode |
| 22:01.53 | _abc_ | It's called freenode for a reason. |
| 22:02.01 | _abc_ | They can pay if they are an official support channel |
| 22:02.07 | igcewieling | I'm preferring to the company not #cisco |
| 22:02.32 | _abc_ | It's like that in many telecom settings. |
| 22:02.39 | _abc_ | Most companies will have that sort of trouble |
| 22:03.00 | igcewieling | I like Cisco routers, but their attempt at forcing people who use their phones into using their PBX horrible. |
| 22:03.30 | _abc_ | is a tcl/wish programmer among other things and hates ios cli. It could be much better. |
| 22:04.26 | iprouteth0 | I like cisco routers and switches. Their unified communications solutions are excellent, but there are so many hidden costs, and majority of the features can be provided by open source anyway |
| 22:04.55 | iprouteth0 | working with the open source also helps keep skills broader as there are so many vendors out there |
| 22:05.23 | _abc_ | I am okay with the hardware so far, the docs are nightmarish, and the ios cli looks like 3 drunk high school student's attempt at writing their 1st shell in tcl in their spare time |
| 22:05.26 | igcewieling | The nice thing about Asterisk as compared to CCM is that with Asterisk you don't need to use scientific notation to quote the price. |
| 22:05.53 | _abc_ | where the students were not on talking terms with each other when they wrote it |
| 22:05.59 | WIMPy | ... if your time comes for free. |
| 22:06.08 | igcewieling | _abc_: Cisco phones have many useful features which are only available with CCM/SCCP and not SIP. |
| 22:06.19 | _abc_ | I think I know a few |
| 22:06.27 | _abc_ | I am not debating the hw quality. Very good usually. |
| 22:06.39 | _abc_ | It's the marketing and firmware versioning which kill the product. |
| 22:06.40 | igcewieling | WIMPy: even when your time is not free CCM is still more expensive. |
| 22:06.54 | _abc_ | I had to do with Nortel in a similar context some time ago before they died/were reborn |
| 22:07.03 | WIMPy | igcewieling: Your price is too low. |
| 22:07.11 | _abc_ | Same thing, great hw, idiots in suits architecting the company into the ground |
| 22:07.32 | WIMPy | Or you are in to mass production mode. |
| 22:08.46 | iprouteth0 | asterisk is my favorite so far. There are some cool softwares out there. Ever look at sipXecs? |
| 22:08.48 | _abc_ | Pity that most Western companies with a good product are dying out like that |
| 22:08.49 | igcewieling | last mediaum CCM system I saw a quote for was $250K and it was not a large system |
| 22:08.59 | _abc_ | Sun, Nortel, not Cisco seems to be limping |
| 22:09.38 | _abc_ | igcewieling: look, there is no point in being ecstatic. I had to do with Meridians and they sure cost a lot of money, but they are built to last and do last |
| 22:09.43 | WIMPy | There are a lot of really good commercial systems around for a fraction of the cost. |
| 22:09.51 | _abc_ | 10,000 lines in a cupboard is do-able |
| 22:10.04 | _abc_ | There are now, there were none than |
| 22:10.19 | igcewieling | WIMPy: indeed there are, but not Cisco |
| 22:10.23 | _abc_ | It's sad they destroyed these companies with idiot suits hatching ever new money making schemes. |
| 22:10.47 | WIMPy | Obviousely not. But SCCP still seems to be of the better tech. |
| 22:10.54 | iprouteth0 | you guys ever worked with metaswitch at all? |
| 22:12.04 | iprouteth0 | thats an interesting platform. More carrier grade really, but does some cool stuff. They are also making SBCs now that I've just started seeing |
| 22:12.22 | iprouteth0 | Haven't heard much lately in regards to the asterisk SCF stuff.... |
| 22:12.47 | WIMPy | Seems to have gone out of fashion. |
| 22:15.50 | iprouteth0 | I would love to see better support for redundancy in asterisk |
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| 22:58.27 | igcewieling | If you limit your needs you can have reasonable redundantcy on a per call basis |
| 22:58.54 | igcewieling | though it could be much improved to support in call redundancy |
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| 23:21.39 | iprouteth0 | i'm likely going to start playing around with configuring high availability asterisk systems once my hardware allows |
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| 23:25.39 | igcewieling | iprouteth0: we have found that using SRV records and not using registration works well. All clients can connect to all servers and all servers can connect to all clients. |
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| 23:34.57 | iprouteth0 | that makes sense. I've used nonregistering peers before in the past for certain providers or between odd devices |
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| 23:47.50 | igcewieling | none of our peers register, which makes things a bit easier |
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