| 00:14.26 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:14.26 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:15.47 | Weezey | igcewieling: how many records you at? |
| 00:16.01 | igcewieling | 7.4 m |
| 00:16.31 | Weezey | almost half way! |
| 00:17.28 | igcewieling | made a few performance tweaks to the system and shaved 3 mins off the amount of time each batch takes |
| 00:21.09 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
| 00:36.52 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
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| 01:59.27 | mun24 | HI |
| 02:00.03 | mun24 | I want to install asterisk. Please guide me |
| 02:03.54 | igcewieling | ~book |
| 02:03.54 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:18.54 | ChannelZ | ./configure; make; make install; echo "YAY!" |
| 02:27.46 | Katty | raden: :> :> |
| 02:27.50 | Katty | hugskwishes on raden |
| 02:31.33 | raden | turns red |
| 02:31.44 | Katty | how'rechu |
| 02:32.05 | raden | good , and you my dear ? |
| 02:32.14 | raden | playing call of duty to relieve stress lol |
| 02:32.57 | Katty | going through fall decor on pinterest while the boy plays saints row 2 |
| 02:33.07 | Katty | why so stressed? |
| 02:38.22 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 02:45.14 | fling | [TK]D-Fender: Hello. :D |
| 02:46.07 | igcewieling | ChannelZ: ./configure && make && make install && echo "YAY!" |
| 02:46.17 | fling | igcewieling: :P |
| 02:46.22 | *** join/#asterisk vedic (~V@183.82.84.84) |
| 02:46.24 | igcewieling | 8-) |
| 02:49.20 | Katty | ice weasel! |
| 02:55.50 | raden | Katty, business is good but trying to get people to pay has sucked |
| 02:56.37 | ChannelZ | Do they have open carry in your state? |
| 03:01.28 | *** part/#asterisk vedic (~V@183.82.84.84) |
| 03:04.37 | *** part/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 03:09.14 | Katty | raden: yeah :< that's a bummer. |
| 03:17.57 | raden | Katty, it ok though :) |
| 03:18.29 | cusco | hi |
| 03:28.37 | Katty | mmm chai tea |
| 03:28.40 | Katty | sips |
| 03:28.47 | Katty | shares with raden |
| 03:28.51 | Katty | cusco: evening! |
| 03:30.12 | cusco | hello Katty |
| 03:30.27 | cusco | nice to see somelife at night |
| 03:31.36 | Katty | it's rare! |
| 03:33.33 | *** join/#asterisk Shaan (~quassel@unaffiliated/shaan) |
| 03:34.04 | *** join/#asterisk dimitry7 (~antonello@200.94.79.33) |
| 03:34.15 | cusco | ow |
| 03:34.26 | cusco | no squirrels at night?? |
| 03:34.53 | dimitry7 | Hello Guys, I am trying to connect my PBX to another PBX in another site.... but when I do: |
| 03:35.22 | dimitry7 | the PBX is not in the firewall |
| 03:35.32 | dimitry7 | when I do: tcpdump -i eth1 host 148.244.168.189 |
| 03:35.43 | dimitry7 | it sends this error: IP static-200-94-79-33.alestra.net.mx > gate.aaamerica.com.mx: ICMP static-200-94-79-33.alestra.net.mx udp port iax unreachable, length 66 |
| 03:35.54 | dimitry7 | why do you think it is? |
| 03:36.01 | dimitry7 | thank you in advance for any help! |
| 03:36.35 | Katty | cusco: lol, no. they're all asleep |
| 03:36.44 | Katty | cusco: and soon i will be too! |
| 03:37.28 | cusco | ;) |
| 03:37.44 | cusco | won't take long forme to join... err... |
| 03:37.49 | cusco | Imean .. to follow |
| 03:39.25 | Katty | lolll!! |
| 03:39.29 | Katty | i know what you mean ^_^ |
| 03:39.35 | dimitry7 | udp port iax unreachable |
| 03:39.38 | cusco | :) |
| 03:39.39 | dimitry7 | any ideas of thisĀ” |
| 03:39.46 | Katty | im a banket hog anyway. |
| 03:39.54 | Katty | blanket. |
| 03:39.55 | cusco | dimitry7: I never expirienced any of that... |
| 03:40.06 | Katty | dimitry7: i'm too tired to think. period. |
| 03:40.09 | cusco | you'reconnecting via iax? |
| 03:40.30 | dimitry7 | cusco, yes |
| 03:40.33 | Katty | after i finish my tea, i'm zonking out |
| 03:40.43 | Katty | maybe prior, never know! |
| 03:40.53 | dimitry7 | Katty, don't worry Katty |
| 03:41.06 | dimitry7 | cusco, ask me for all the information you need |
| 03:41.22 | cusco | she doesn't |
| 03:41.28 | cusco | else she would have too many wrinkles |
| 03:41.35 | dimitry7 | I think I am missing one part but been strugling |
| 03:41.41 | dimitry7 | cusco, I agree |
| 03:41.50 | dimitry7 | :) |
| 03:41.54 | cusco | dimitry7: I'm not really able to help you |
| 03:42.05 | cusco | perhaps I would try another time... |
| 03:42.13 | cusco | even tho I don't think I could help you |
| 03:42.22 | dimitry7 | cusco, someone you know here who might be willing to? |
| 03:42.24 | cusco | iax is udp? |
| 03:42.31 | cusco | I thought it was tcp |
| 03:42.35 | dimitry7 | cusco, yes |
| 03:42.43 | cusco | dimitry7: sure, come back in about 8 hours |
| 03:42.45 | dimitry7 | no sorry |
| 03:42.48 | dimitry7 | it is TCP |
| 03:46.00 | Weezey | We need a global fail2ban list. |
| 03:46.54 | Weezey | if 10+ of our servers are hammered by an IP trying to dial 90119725988113270 then guess what, dude's a fuck trying to steal our LD. |
| 03:47.27 | dimitry7 | Hello Guys, I am trying to connect my PBX to another PBX in another site.... but when I do: |
| 03:47.31 | dimitry7 | when I do: tcpdump -i eth1 host 148.244.168.189 |
| 03:47.39 | dimitry7 | it sends this error: IP static-200-94-79-33.alestra.net.mx > gate.aaamerica.com.mx: ICMP static-200-94-79-33.alestra.net.mx udp port iax unreachable, length 66 |
| 03:47.42 | Weezey | Did I just scroll up again? |
| 03:47.49 | dimitry7 | why do you think it is? |
| 03:48.46 | Weezey | port iax isn't permitted on gat.aaamerica.com.mx |
| 03:48.54 | Weezey | err the other one |
| 03:49.07 | Weezey | since it's the one sending the ICMP reject |
| 03:50.13 | Weezey | iptables -A INPUT --dport iax -j ACCEPT |
| 03:52.53 | dimitry7 | Weezey, IAX is permited on it |
| 03:52.53 | dimitry7 | what else should it be? |
| 03:53.10 | *** join/#asterisk AliRezaTaleghani (~Thunderbi@unaffiliated/AliRezaTaleghani) |
| 03:53.33 | dimitry7 | Weezey, I've checked my config... it is just enabled as you say |
| 03:53.56 | cusco | what's the LD? |
| 03:55.09 | dimitry7 | cusco, what is LD ? |
| 03:55.20 | cusco | I was asking Weezey |
| 03:55.44 | cusco | I don't know either |
| 03:56.33 | dimitry7 | cusco, oh ok... by the way... the error I am talking about is not documented on the web. |
| 03:56.55 | cusco | I don't know about your error |
| 03:56.57 | cusco | :( sorry |
| 03:57.53 | dimitry7 | cusco, :-( ... I do appreciate your willingness to help me my friend. I know this is weird. |
| 03:58.11 | dimitry7 | Weezey, if the firewall is Ok... what other config might be in the middle ? |
| 04:02.29 | igcewieling | IAX uses UDP. It always has, it always will. |
| 04:03.02 | cusco | ok my bad |
| 04:03.09 | dimitry7 | igcewieling, this is my output when I ping from one PBX to the other one: http://pastebin.com/HLEgDDNb |
| 04:03.19 | igcewieling | run: module show like iax to make sure it is loaded |
| 04:03.44 | igcewieling | dimitry7: so what is the problem? |
| 04:07.01 | igcewieling | on the pbx does netstat -an | grep 4569 show anything? |
| 04:39.05 | *** join/#asterisk c4t3l (~rcallicot@c-98-199-199-189.hsd1.tx.comcast.net) |
| 04:51.21 | *** join/#asterisk kannan (~chatzilla@123.238.229.201) |
| 04:56.17 | raden | Katty, delicious :) |
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| 06:11.22 | *** join/#asterisk dimitry7 (~antonello@200.94.79.33) |
| 06:11.25 | dimitry7 | hello |
| 06:11.33 | dimitry7 | why do you think a PBX is unreachable |
| 06:11.40 | dimitry7 | iaxvallejo/aere 148.244.168.189 (D) 255.255.255.255 4569 (T) (E) UNREACHABLE |
| 06:11.48 | dimitry7 | why might be that? |
| 06:12.33 | dimitry7 | also this: unable to create channel of type 'iax2' |
| 06:12.36 | ChannelZ | Because of a firewall, a network problem, a bad IP, it could be a lot of things. |
| 06:14.14 | dimitry7 | ChannelZ, how can I diagnose those? |
| 06:15.13 | dimitry7 | ChannelZ, so that I can dismiss one per one |
| 06:19.49 | *** join/#asterisk peetaur2 (~peter@x2f004bc.dyn.telefonica.de) |
| 06:22.57 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 06:25.19 | ChannelZ | Well that's up to you really. Is your Asterisk behind a firewall? Is the remote Asterisk as well? If so, do you have firewall rules setup to allow traffic on UDP port 4569 on either end, and/or port-forwarding rules as necessary? |
| 06:27.17 | dimitry7 | ChannelZ, yes it is accepting connections on that port |
| 06:27.59 | *** join/#asterisk dimitry7 (~antonello@200.94.79.33) |
| 06:28.37 | ChannelZ | Well apparently not, at least not entirely |
| 06:29.24 | dimitry7 | ChannelZ, -A INPUT -s 148.244.168.189 -p udp --dport 4569 -j ACCEPT |
| 06:30.20 | dimitry7 | I have 6 PBX online |
| 06:30.28 | dimitry7 | and I am having trouble only with one of them |
| 06:30.37 | dimitry7 | the only 5 say this_ |
| 06:30.38 | ChannelZ | on which server? That'd allow traffic in FROM 148.244.168.189 to whatever machine that rule is on.. but the 'unreachable' peer you showed above was 148.244.168.189 |
| 06:30.52 | dimitry7 | iaxbrownsville 12.7.147.18 (D) 255.255.255.255 1024 (T) (E) OK (50 ms) |
| 06:30.52 | dimitry7 | iaxpacifico 187.174.147.162 (D) 255.255.255.255 1024 (T) (E) OK (103 ms) |
| 06:30.53 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 06:31.00 | dimitry7 | and this |
| 06:31.01 | dimitry7 | iaxvallejo/aere 148.244.168.189 (S) 255.255.255.255 4569 (T) (E) UNREACHABLE |
| 06:31.35 | dimitry7 | 148.244.168.189 and 148.244.168.189 are the same |
| 06:32.00 | ChannelZ | is 148.244.168.189 supposed to register TO whatever server it is we're talking about? |
| 06:33.34 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 06:33.41 | dimitry7 | ChannelZ, exactly |
| 06:34.31 | ChannelZ | ok so is there a firewall possibly blocking ITS _outgoing_ traffic? |
| 06:34.52 | ChannelZ | Can that machine even ping your other? |
| 06:37.21 | dimitry7 | ChannelZ, One machine has ICMP disabled, but let me check |
| 06:47.21 | dimitry7 | ChannelZ, yes, I can ping them... and all other peers work good |
| 06:47.38 | dimitry7 | I can see each other with ping |
| 06:48.40 | ChannelZ | then I guess turn on iax debug and see if something else is going on, if the traffic is being sent and/or hitting the other asterisk and being ignored or what |
| 07:02.26 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 07:16.39 | dimitry7 | ChannelZ, sorry |
| 07:16.45 | dimitry7 | I was checking over and over again |
| 07:16.56 | dimitry7 | it ends up with this error |
| 07:16.57 | dimitry7 | http://pastebin.com/zFBTyh79 |
| 07:18.28 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.243) |
| 07:19.07 | dimitry7 | ChannelZ, and debug says: Unable to create channel of type 'IAX2' |
| 07:20.07 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 07:24.12 | dimitry7 | ChannelZ, do you have teamviewer? |
| 07:28.05 | *** join/#asterisk AliRezaTaleghani (~Thunderbi@unaffiliated/AliRezaTaleghani) |
| 07:30.39 | apb1963 | static-200-94-79-33.alestra.net.mx udp port iax unreachable, |
| 07:36.21 | dimitry7 | apb1963, allow it on you firewall |
| 07:45.07 | dimitry7 | ChannelZ, are you still there? |
| 07:47.38 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 07:51.26 | apb1963 | There are 3 possibilities.... the host sending the message is blocking the packet, the host meant to receive the packet is blocking, or it's being blocked somewhere in between. In other words, it could be the server, the client ... or the network. Steps to debug: check each one. If you're able to ping - in both directions - then check to see if there's anything listening on the port. |
| 07:51.37 | apb1963 | My dinner is ready... I'll be back in 30 minutes or so. |
| 07:51.47 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 08:09.04 | dimitry7 | apb1963, excellent help my friend... it helps me think of those little thins that sometimes we oversee, what is strange is that the other PBX are connecting right, but only two are not... btw the most importants :S |
| 08:09.08 | dimitry7 | * important |
| 08:10.34 | dimitry7 | apb1963, have a good meal :-) |
| 08:17.17 | *** join/#asterisk eject_ck (~Evgeniy@109.86.157.219) |
| 08:18.44 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.168) |
| 08:23.41 | eject_ck | H all, I have two external sip accounts at my VoIP providers where one is international (only outboutd calls, very cheap intl. calls) and other one local (cheap local calls + local DID). I have sip extensions working well, and I'm using prefix 777 for outbound calls via international_voip peer, all non-prefixed extensions goes to local voip provider. What I need is to make call from sip phone to international destination provider (with 777 prefix) and t |
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| 08:50.50 | ChannelZ | eject_ck: and .... ? |
| 08:57.15 | eject_ck | ChannelZ: what's wrong with my dialplan ? |
| 08:57.27 | eject_ck | I need to send call to local provider |
| 08:57.36 | eject_ck | during transfer |
| 08:57.44 | ChannelZ | I don't know your question wasn't even finished |
| 08:57.52 | eject_ck | sorry |
| 08:59.09 | eject_ck | First call from sip phone to 777XXXXXXXXX - got connection via international provider, then transfer #1 050XXXXXXX - I see that call goes through international provider again. |
| 08:59.35 | eject_ck | When I dial 050XXXXXXX it goes though local voip prov |
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| 09:10.36 | ChannelZ | well pastebin the verbose console output of one of these calls and transfers |
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| 10:10.26 | wrouesnel | is there a way from the Queue dialplan app, to disable periodic announcements for just that caller in a particular queue? |
| 10:11.54 | wrouesnel | urgh, announce override - doy! |
| 10:22.41 | eject_ck | ChannelZ: sec |
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| 10:36.13 | eject_ck | ChannelZ: http://pastebin.com/qvDQNiNw |
| 10:37.34 | eject_ck | I've pasted both calls 1) call via intl. outbound prefix 777096XXXXXXX then transfer to 097YYYYYYY 2) call to 097YYYYYYY |
| 10:38.09 | eject_ck | Just need know how can I make transfers possible via SIP/datagroup-office1_2/ |
| 10:38.57 | eject_ck | and why it goes through out-hotel |
| 10:39.00 | eject_ck | xotel |
| 11:07.40 | eject_ck | looks I need check local channel docs |
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| 14:07.19 | Grady2000 | hey again all, in the switch from asterisk 1.4 to 1.6 the command: "exec set channel(language)=fr" is not working, anyone knoe the 1.6 equivalent? |
| 14:07.27 | Grady2000 | this is for reading the digits in french |
| 14:07.36 | Grady2000 | of course all my recorded prompts are in french |
| 14:08.19 | *** join/#asterisk elmargol (~elmargol@host183-105-dynamic.20-79-r.retail.telecomitalia.it) |
| 14:08.40 | WIMPy | Looks correct to me. Might be the directory structure. digits/lang vs lang/digits. That changed some time and can also be configured IIRC. |
| 14:09.33 | Grady2000 | Set(CHANNEL(language)=hu) saw this online,, minor formatting change trying it now |
| 14:09.47 | Grady2000 | and then yes I'll confirm the digs are actually the french versions, good idea thx |
| 14:10.14 | WIMPy | Oh, right, why do you use the extra exec? |
| 14:10.33 | Grady2000 | because I'm calling it from a python script |
| 14:11.04 | WIMPy | setvar |
| 14:11.17 | Grady2000 | i dont follow you |
| 14:11.52 | *** join/#asterisk markit (~marco@88-149-177-66.v4.ngi.it) |
| 14:12.33 | WIMPy | AGI has a "set variable" command. |
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| 16:06.07 | Kobaz | so quiet |
| 16:13.21 | pabelanger | Long weekend in Canada |
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| 17:00.48 | pabelanger | ~itsp |
| 17:00.48 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 17:00.54 | pabelanger | ~itsplist-us |
| 17:00.55 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 17:05.09 | *** join/#asterisk Dovid (~Dovid@ool-1826d413.dyn.optonline.net) |
| 17:07.48 | igcewieling | Ugh. We have one customer with over 1,000 telephone numbers and nobody has an actual list of the numbers. 8-| |
| 17:08.41 | WIMPy | I'm sure their provider has. |
| 17:09.16 | igcewieling | WIMPy: we are their provider. The problem is they have 300+ accounts, with only one or two phone numbers under each account and the account names are all different |
| 17:09.53 | igcewieling | This is a towing company who has SO many telephone numbers, if you call 411 or look online at a ywllow pages, chances are no matter which telephone number you choose it rings at the same company |
| 17:10.06 | *** join/#asterisk vlad_starkov (~vlad_star@85.202.230.46) |
| 17:10.11 | igcewieling | An effective, if sleazy, business plan |
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| 17:17.51 | pabelanger | o.0 |
| 17:18.08 | pabelanger | not sure I'd want my customer know I think their business plan is sleazy |
| 17:18.11 | pabelanger | just saying |
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| 17:56.55 | igcewieling | 8-) |
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| 18:15.51 | *** join/#asterisk jkroon (~jkroon@196.37.70.13) |
| 18:16.42 | jkroon | hi guys, i'm having trouble where a Local/ originated channel transcodes via slin even though both sides of the conversation is using g729 |
| 18:17.29 | jkroon | is there anyone that might be able to assist at this point in time? |
| 18:18.37 | jkroon | I've reported this before (https://issues.asterisk.org/jira/browse/ASTERISK-20461) which was still on asterisk 1.8 and my patch there worked for me at that point in time (not generic fix, only for my use-case), but a similar hack for 11.X seems to be out of the question |
| 18:18.38 | LieutPants | [ASTERISK-20461] [Status: Open] channel originate Local/foo * forces translation via slin - https://issues.asterisk.org/jira/browse/ASTERISK-20461 |
| 18:23.57 | jkroon | core show channel output for both resulting channels: http://pastebin.com/EgxY1WiP |
| 18:27.03 | *** join/#asterisk peter (~peter@x2f004bc.dyn.telefonica.de) |
| 18:30.32 | igcewieling | jkroon: set transmit_silence=no in asterisk.conf |
| 18:30.54 | igcewieling | you also have to make sure Asterisk never tries sending ringback as audio |
| 18:31.15 | igcewieling | jkroon: your best option it to purchase some g729 license. Pass-thru almost never works as you want it to. |
| 18:31.33 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:31.51 | igcewieling | if those don't work, well, welcome to Asterisk. Deal with it. |
| 18:33.22 | igcewieling | Between transmitting silence and ringback, we decided to simply buy some transcoding cards and never worry about it again |
| 18:33.36 | igcewieling | And we never have worried about it again. |
| 18:34.33 | pabelanger | igcewieling, Digium transcoding hardware or something else? |
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| 18:38.20 | jkroon | Transmit silence during rec: Disabled <-- igcewieling i suspect this is what you're referring to? |
| 18:39.45 | jkroon | and i don't use premature media either ... |
| 18:40.03 | jkroon | the pass-thru works well for me except when I originate via a Local/ channel. |
| 18:45.30 | igcewieling | jkroon: asterisk.conf ;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of |
| 18:45.31 | igcewieling | <PROTECTED> |
| 18:46.36 | igcewieling | sorry, this is the correct one: ;transmit_silence = yes ; Transmit silence while a channel is in a ; waiting state, a recording only state, or ; when DTMF is being generated. Note that the silence internally is generated in raw signed linear format. This means that it must be transcoded into the native format of the ; channel before it can be sent to the device. ; It is for this reason that this is optional, as it may r |
| 18:47.08 | *** join/#asterisk jkroon (~jkroon@196.37.70.13) |
| 18:47.26 | igcewieling | sorry, this is the correct one: ;transmit_silence = yes ; Transmit silence while a channel is in a ; waiting state, a recording only state, or ; when DTMF is being generated. Note that the silence internally is generated in raw signed linear format. This means that it must be transcoded into the native format of the ; channel before it can be sent to the device. ; It is for this reason that this is optional, as it may r |
| 18:47.39 | jkroon | both set explicitly to no already |
| 18:48.09 | igcewieling | then you might have a legit bug |
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| 18:49.16 | jkroon | great, now that is almost more annoying that the forced slin transcoding ... |
| 18:50.22 | igcewieling | like I said, you can fight it and fight it, or you can get g729 licenses. We got a 192 channel Sangoma transcoding card for each of our 4 servers |
| 18:50.43 | igcewieling | wasn't cheap, but in the long run our customers are happier and we are less stressed. |
| 18:51.03 | jkroon | igcewieling, i've already set those options, and still no luck. it's only when using origination via Local/ that I've got a problem. |
| 18:51.22 | jkroon | so would prefer to sort it out if possible. |
| 18:51.33 | igcewieling | jkroon: sounds like an actual bug |
| 18:51.42 | igcewieling | oh! |
| 18:51.50 | igcewieling | have you tried the /n option to your Local/ channel? |
| 18:52.04 | igcewieling | (or if you are using /n have you tried not using /n) |
| 18:52.15 | jkroon | yes i have, exact same thing |
| 18:52.25 | jkroon | mind you, that was on asterisk 1.8 ... let's retry that. |
| 18:53.24 | jkroon | same result. |
| 18:53.54 | jkroon | https://issues.asterisk.org/jira/browse/ASTERISK-20461 contains the full report of what I've got so far. |
| 18:53.55 | LieutPants | [ASTERISK-20461] [Status: Open] channel originate Local/foo * forces translation via slin - https://issues.asterisk.org/jira/browse/ASTERISK-20461 |
| 18:56.11 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 18:56.25 | jkroon | actually, it looks like it *might* be possible to fix with ast 11.X |
| 18:56.39 | jkroon | origination code is MUCH simpler than in previous versions |
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| 19:04.08 | jkroon | does anybody know *why* app_originate only adds capabilities for AST_FORMAT_SLINEAR* and not ast_format_cap_all()? |
| 19:05.37 | igcewieling | that would be a question better asked on #asterisk-dev |
| 19:05.55 | WIMPy | Not sure how the chances for an answer are at theis time, but that's surely a question for -dev. |
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| 19:19.33 | gavimobile | im getting some errors when playing back some wav files. http://pastebin.com/A9V1fP09 I transferred these files from another working asterisk box to my new asteriskpbx box, I have a feeling that it has to do with dependencies. any thoughts guys? |
| 19:19.41 | *** part/#asterisk russum (~russum@94.139.140.97) |
| 19:20.22 | jkroon | ok, the origination code has spread out over three files now ... was looking at the wrong version ... separate files for cli based, application based and ami based. |
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| 19:21.47 | apb1963 | gavimobile: By what method did you transfer the files? FTP? |
| 19:21.47 | gavimobile | I was missing sox and sox-devel so I installed them both via rpm. than I recompiled asterisk just in case but same thing |
| 19:22.04 | gavimobile | apb1963: no, I mounted a flash drive |
| 19:22.16 | gavimobile | apb1963: do you think they got damadged? |
| 19:22.20 | jkroon | shouldn't need sox for playing wav, but sox is useful for transcoding files ... |
| 19:22.29 | apb1963 | How did you copy them to the flash drive? |
| 19:22.29 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 19:22.32 | gavimobile | jkroon: it won't hurt |
| 19:22.37 | gavimobile | apb1963: mv |
| 19:22.41 | gavimobile | no, cp |
| 19:22.46 | jkroon | gavimobile, try play them via some other player - and they do need to be in the exact right format iirc. |
| 19:23.09 | gavimobile | jkroon: well I know the originals are ok caues they were on my current/old pbx |
| 19:23.35 | gavimobile | ill tar ball them than transfer through http |
| 19:23.37 | gavimobile | via wget |
| 19:24.24 | jkroon | gavimobile, good luck on that one ... sounds like an almost just as strange one as I've got - except I think I've got a fix for mine now :) |
| 19:26.09 | apb1963 | what's the file format before and after copying? |
| 19:26.11 | gavimobile | jkroon: yea, the cp stoped in the middle a few times cause it was spazing about links |
| 19:26.22 | gavimobile | this should be a safer method |
| 19:26.28 | gavimobile | ill keep you guys updated |
| 19:26.33 | apb1963 | stopped in the middle? |
| 19:26.33 | jkroon | ooh, sounds wack |
| 19:29.18 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.198) |
| 19:29.25 | gavimobile | wow |
| 19:29.28 | gavimobile | worked flawless |
| 19:29.29 | gavimobile | !!! |
| 19:29.35 | gavimobile | thanks guys |
| 19:36.00 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 19:43.11 | jkroon | jay, my problem eradicated too |
| 19:43.29 | jkroon | exit |
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| 20:00.08 | Addisk | anyone have any idea how to fix this dahdi errors: |
| 20:00.09 | Addisk | dahdi: Warning: Span DYN/ethmf/eth1/00:50......CA/0 didn't specifiy a spantype. Please fix driver! |
| 20:00.19 | Addisk | i get this every time i do dahdi restart |
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| 20:18.40 | meatzie | hey guys, stupid question but I can't find any documentation on it |
| 20:19.04 | meatzie | whats the upper limit of extensions? |
| 20:19.26 | meatzie | that asterisk allows? |
| 20:20.34 | meatzie | to clarify, not trying to add 'X#' extensions. but what is the available range |
| 20:28.36 | ChannelZ | I imagine there is probably an index pointer or something that could cause a limit, but even if it's only a 16-bit int you're talking tens of thousands. |
| 20:28.51 | ChannelZ | I don't think I've ever heard of someone hitting a ceiling |
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| 20:46.53 | meatzie | Yeah I'm not worried about like, maxing out a box, i'm just thinking for me to be able to organize it better |
| 20:47.53 | meatzie | i guess i could look at a memory dump and see the address space for that value? |
| 20:48.23 | meatzie | or i'll just try numbers and square up unil I hit a limit |
| 20:48.54 | ChannelZ | I guess my question is 'why are you worried about it' |
| 20:49.04 | meatzie | not worried |
| 20:49.12 | ChannelZ | or care rather |
| 20:49.13 | ChannelZ | :) |
| 20:49.14 | meatzie | just wondering what my available range is |
| 20:49.22 | WIMPy | Are you asking about the number of extensions or about the possible value of them? |
| 20:49.33 | meatzie | so I can group off people |
| 20:49.38 | meatzie | possible value |
| 20:49.43 | ChannelZ | wait, are you talking about the number of digits? |
| 20:49.43 | WIMPy | Extensions are strings. |
| 20:49.52 | meatzie | waaaaaaah? |
| 20:49.56 | meatzie | they're strings? |
| 20:49.59 | meatzie | damn |
| 20:50.02 | WIMPy | There might be a length limit, bit none on contents. |
| 20:50.03 | meatzie | ok then |
| 20:50.11 | ChannelZ | yes they can contain letters |
| 20:50.22 | meatzie | I was unaware |
| 20:50.26 | meatzie | this is awesome |
| 20:50.35 | meatzie | I can't wait to abuse this new knowlegde |
| 20:50.43 | meatzie | brb i'm going to break things |
| 20:50.48 | WIMPy | :-) |
| 20:50.49 | meatzie | but seriously guys thanks |
| 20:50.50 | ChannelZ | more digits than you'd want people to have to remember to dial to reach the guy across the hall, for sure. |
| 20:51.58 | WIMPy | With the new official limits, you will need to be able to process at least 34 digit numbers anyway. |
| 20:52.29 | WIMPy | 37 |
| 20:52.38 | WIMPy | (I think) |
| 20:52.41 | meatzie | mkay |
| 20:53.00 | apb1963 | damit... I just got done assigning 42 digit extensions |
| 20:53.14 | Chainsaw | sets out to implement 01189998819991197253 |
| 20:53.23 | meatzie | now these limits are official from...? |
| 20:53.30 | *** join/#asterisk Aven (~user@195.22.106.52) |
| 20:53.56 | WIMPy | ITU. They are about to raise the possible length of phone numbers from 16 to 24 digits. |
| 20:54.10 | meatzie | ah ok |
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| 20:54.34 | WIMPy | Including access digits, feature codes and carrier selection, you can end up with 37 digits then. |
| 20:54.51 | ChannelZ | The phone number will never die |
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| 20:55.16 | apb1963 | thats what they said about the Roman Empire |
| 20:55.17 | WIMPy | *FSCK* |
| 20:55.36 | *** join/#asterisk peetaur2 (~peter@x2f004bc.dyn.telefonica.de) |
| 20:55.37 | Aven | whether it is possible to use dynamic realtime SIP outbound sip registration? |
| 20:55.38 | WIMPy | Why didn't I get my reminder. |
| 20:55.48 | WIMPy | Argh. |
| 20:57.40 | meatzie | hey also, you guys know any good SIP Billing engines? |
| 20:57.54 | meatzie | that work directly with asterisk? |
| 20:58.10 | WIMPy | D'oh. My clock is wrong :-( |
| 20:58.53 | apb1963 | computer clock? |
| 20:59.05 | apb1963 | or wall clock? |
| 20:59.29 | ChannelZ | his biological clock |
| 20:59.34 | ChannelZ | now he'll never have kids! |
| 20:59.47 | WIMPy | Computer. |
| 20:59.58 | apb1963 | that's why I use NTP |
| 21:00.10 | apb1963 | ChannelZ: That's funny |
| 21:00.16 | [TK]D-Fender | meatzie: there is no such thing as a "SIP billing engine". Billing is billing. It has nothing to do with protocol... |
| 21:00.40 | WIMPy | I'm still using the old radio clock, but that seems to have too bad a signal. |
| 21:00.47 | [TK]D-Fender | meatzie: How you flag your calls it up to you... this is just CDR |
| 21:01.01 | [TK]D-Fender | meatzie: mashed up with whatever pricing tables you have |
| 21:01.38 | apb1963 | I finally understand what mashable means |
| 21:01.45 | WIMPy | wonders if it's possible to do billing based on CDRs only. |
| 21:02.17 | [TK]D-Fender | WIMPy: Well.. you do have to have a price to multiply that by.... |
| 21:02.36 | meatzie | actually about CDR's, how are they stored? SQL? |
| 21:02.52 | WIMPy | The way you configured them. |
| 21:03.45 | [TK]D-Fender | indeed |
| 21:04.12 | apb1963 | indubitably |
| 21:22.47 | igcewieling | WIMPy: I feel CEL would be better for billing. More complicated, but better. |
| 21:23.11 | igcewieling | If needed you can use GenCELUserEvent in the dialplan to add additional information |
| 21:36.08 | *** join/#asterisk Aven (~user@195.22.106.52) |
| 21:36.22 | Aven | Whether it is possible to use dynamic realtime for SIP outbound registration? |
| 21:37.03 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 21:43.44 | igcewieling | Aven: why would you need that? |
| 21:44.35 | igcewieling | I suspect you can put your [general] options in a "peer name" of "general" in the database. |
| 21:47.35 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
| 21:50.52 | file | you can not. |
| 21:51.01 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 21:53.12 | igcewieling | file would know far better than I |
| 21:53.47 | igcewieling | file: how about the register in the peer? recent Asterisk's allow that in sip.conf, don't they? |
| 21:54.02 | file | I doubt that would work |
| 21:54.16 | file | you *might* have a chance with caching, but it certainly wasn't designed with it in mind |
| 21:54.36 | igcewieling | I gave up on realtime for now.8-| |
| 21:54.55 | file | outbound registration is a stateful action, realtime is stateless without caching |
| 21:55.11 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.198) |
| 22:09.55 | igcewieling | file: do you have any suggestions for getting peerstatus events into CEL? |
| 22:10.42 | Aven | igcewieling need for asterisk cluster and virtual pbx/centrex |
| 22:11.33 | Aven | multiple servers and tens of thousands of peers |
| 22:12.11 | igcewieling | Aven: Aven: shouldn't the peers be registering to the asterisk server (inbound reg) and not what you asked about |
| 22:13.24 | Aven | outgoing registration is necessary for that would register on third-party servers voip providers |
| 22:13.40 | igcewieling | Aven: how many 3rd party voip providers do you have? |
| 22:14.49 | igcewieling | I run a "clustered asterisk" as well, though our requirements are different |
| 22:15.07 | Aven | there will be a few hundred, every owner of a virtual PBX will be able to choose their own provider |
| 22:16.18 | igcewieling | It may be easiest to store that information in a database table and have a script which is called via #exec in sip.conf to generate the text config. If I understand file correctly, you cannot do what you want to do. |
| 22:17.15 | igcewieling | We have NO registrations to/from any of the 3 servers in our cluster. All devices are on static IPs. We have 5 or so "providers" |
| 22:17.49 | igcewieling | when you add registration into the mix it gets very complicated very fast. |
| 22:18.16 | Aven | Then you need to call "sip reload" on every sneeze, and it is a very heavy load at over 10,000 peers. |
| 22:18.30 | igcewieling | correct. But since you cannot do what you want to do..... |
| 22:18.53 | igcewieling | Ask on the Asterisk-Users mailing list (lists.digium.com) or search the archive |
| 22:19.12 | igcewieling | maybe someone can come up with a different way to accomplish what you want |
| 22:19.13 | WIMPy | Use a proxy? |
| 22:20.06 | igcewieling | WIMPy: "I have this problem with Asterisk which I'm going to solve with a proxy." *6 weeks later* "Now I have TWO problems" |
| 22:20.22 | igcewieling | Though with 10,000 peers it would be worth the extra hassle. |
| 22:20.53 | Aven | igcewieling I will try to combine realtime and # exec directive may not be a big load. |
| 22:21.14 | igcewieling | We have about 10,000 telephone numbers, but since each peer can handle 5 - 200 telephone numbers..... |
| 22:21.36 | Aven | WIMPy Yes, of course, to balance between nodes Asterisk |
| 22:22.18 | WIMPy | Maybe some proxy is better at handling outbound registrations. |
| 22:23.42 | *** join/#asterisk teff (~teff@client-86-31-246-93.oxfd.adsl.virginmedia.com) |
| 22:25.15 | igcewieling | next week I get to listen to hours of classical hold music over g729 |
| 22:39.27 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 22:39.51 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 22:40.16 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 22:48.57 | *** join/#asterisk corretico (~luis@186.96.91.236) |
| 22:51.22 | *** join/#asterisk corretico (~luis@186.96.91.236) |
| 22:51.31 | file | igcewieling, peerstatus really isn't a CEL thing |
| 22:51.50 | file | and you inevitably have to tell Asterisk to begin registering somewhere |
| 23:02.46 | *** join/#asterisk mun24 (~chatzilla@dsl-173-248-192-114.acanac.net) |
| 23:02.56 | mun24 | Hi |
| 23:04.03 | mun24 | I recently installed asterisk. When I use command sip show peers to see if asterisk is responding to commands. It gives me an error |
| 23:05.06 | mun24 | It gives me an error no such command |
| 23:05.32 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 23:05.34 | darkdrgn2k | hey all |
| 23:05.45 | darkdrgn2k | any one know where i can find provisionign files for an mp-202 |
| 23:07.22 | *** join/#asterisk DEMNVT (~Adium@rmsaus7.lnk.telstra.net) |
| 23:12.59 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 23:13.25 | *** join/#asterisk bazman (~army@220-245-36-238.static.tpgi.com.au) |
| 23:22.38 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 23:25.57 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 23:28.25 | *** join/#asterisk Nugget (nugget@rennsport.macnugget.org) |
| 23:30.27 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:44.38 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 23:50.38 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 23:55.11 | raden | hugs on Katty |
| 23:55.21 | Katty | :> |
| 23:55.24 | Katty | hugs raden |
| 23:55.30 | Katty | raden: just got back from my job! how're you? |
| 23:56.21 | raden | doing well |
| 23:56.28 | Katty | yay! |
| 23:56.29 | raden | just got done from running around like crazy |
| 23:56.44 | raden | now going to work on organizing my mess of a Wisp |
| 23:58.20 | raden | Whats Katty up to tonight ? |
| 23:59.40 | Katty | you know i really hadn't thought about it |
| 23:59.51 | Katty | my last plan was to run, if the nice weather held |