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| 00:28.05 | the_nation | hi, Im trying to get an asterisk install running on my router, I have toll free going but when making a call, through gtalk, I get an error in the CLI and not sure where to start. |
| 00:28.26 | the_nation | app_dial.c:2039 dial_exec_full: Unable to create channel of type 'gtalk' (cause 66 - Channel not implemented) what is missing (: |
| 00:34.08 | the_nation | anyone alive? |
| 00:34.14 | carrar | no |
| 00:34.48 | the_nation | thx |
| 00:36.55 | carrar | http://forums.asterisk.org/viewtopic.php?f=1&t=75929 |
| 00:37.29 | carrar | http://tinyurl.com/4ym52h7 |
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| 02:10.50 | gruvfunk | the_nation: your gtalk channel is not implemented |
| 02:12.12 | Freeaqingme | think a make menu would suffice here. |
| 02:15.01 | gruvfunk | depends on installation, if compiled from source, then yes... provided all other prerequisites are installed |
| 02:16.11 | gruvfunk | if using packages/repos, likely missed a prerequisite before installing asterisk |
| 02:16.23 | gruvfunk | namely, iksemel has bitten me once or twice |
| 02:17.41 | gruvfunk | and then I'd question what version he's using to install on a router, is it 1.8? |
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| 02:27.52 | astruasdh | please, guys. I'm using jain-sip to talk to asterisk. When I use my app against a voip phone, it works: i sent invite, receive the forbidden msg, then send the invite with auth parameters, and then the app receives 200 ok. When I run my app against asterisk eerything works, except the 200 ok message: this message is sent by asterisk (I can see in wireshark), but processResponse() in my app is not executed for the 200 ok message. |
| 02:28.55 | astruasdh | has anyone some idea about this problem? |
| 02:29.17 | leifmadsen | sounds like a problem in your application... |
| 02:30.15 | astruasdh | leifmadsen, yeah... probably some parameter to tell jain-sip "hey, start handling ok messages". do u know some parameter t o do this? |
| 02:30.58 | leifmadsen | zero idea |
| 02:31.03 | leifmadsen | never heard of jain-sip |
| 02:31.20 | leifmadsen | and I imagine no one here is really going to be able to help you -- you should find people familiar with jain-sip |
| 02:31.49 | coppice | jain-sip wouldn't harm a fly |
| 02:33.42 | astruasdh | yeah... I asked here because is the closest place to so this. tyvm! |
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| 03:40.24 | `paul | how do i know if a call leaves a queue because its empty ? |
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| 04:32.28 | kdas | does anyone know why my sipphone is regestered to my asterisk box but when i do "sip show peers" in asterisk it shows host as "unspecified" |
| 04:35.41 | kaldemar | pastebin a cli output of a registration and the output of sip shoe peers. |
| 04:35.58 | kaldemar | ~pb |
| 04:35.58 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 04:36.03 | p3nguin | My guess would be that it isn't really registered. |
| 04:37.19 | kdas | p3nguin: i can make outbound calls with it |
| 04:37.26 | kdas | kaldemar: http://pastebin.com/ETn1wmMY |
| 04:37.29 | p3nguin | So? |
| 04:37.43 | p3nguin | That simply means you can authenticate. |
| 04:37.49 | p3nguin | Try making a call TO it. |
| 04:38.01 | p3nguin | Asterisk won't know where to send the call. It will fail. |
| 04:38.10 | p3nguin | And that's the purpose of registration. |
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| 04:42.27 | kdas | p3nguin: yes you are right i cant recieve calls only make them because asterisk dont know where to send them |
| 04:42.43 | kdas | is there some config thing going on or what? |
| 04:43.16 | p3nguin | The phone isn't registered. Find out why and fix it. |
| 04:43.55 | p3nguin | kaldemar seemed willing to help you find out why. |
| 04:45.24 | kdas | p3nguin: on another subject. do you know if i can get sip (simple) messaging working in asterisk? |
| 04:45.52 | kdas | s/if i can/ how i can/ |
| 04:47.17 | p3nguin | I wouldn't have any idea. |
| 04:50.51 | kdas | ok thanks |
| 05:02.42 | kaldemar | he was hasty. |
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| 05:40.55 | p3nguin | typical |
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| 06:07.55 | the_nation | hi I am having a porblem with asterisk not loading from my modules.conf file I have defined about 30 modules to load, but asterisk is loading up 184 and not the ones I want/need (: |
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| 06:13.52 | SiNGLer | the_nation: check if modules can be loaded via cli, maybe some errors happen so modules are not loaded |
| 06:15.07 | imcdona | Anyone here using BLF on Polycom phones with Asterisk? |
| 06:15.13 | the_nation | [May 5 06:15:00] WARNING[1109]: loader.c:400 load_dynamic_module: Module 'chan_gtalk.so' did not register itself during load |
| 06:15.13 | the_nation | [May 5 06:15:00] WARNING[1109]: loader.c:839 load_resource: Module 'chan_gtalk.so' could not be loaded. |
| 06:15.58 | the_nation | I'm also trying to run the stop now command and its showing as an unknown command |
| 06:16.30 | the_nation | reload appears to reload, but not from the modules.conf :/ |
| 06:18.32 | the_nation | back in a sec |
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| 06:26.05 | nosbig | So, I am having some troubles with a migration from my ITSP. Anyone here use Teliax? |
| 06:26.29 | the_nation | progress (: I had to actually restart the freaking router before things took hold. I now have outgoing working, looking to fix the incoming woo |
| 06:26.48 | the_nation | btw, installing it on an asus 520gu |
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| 06:51.42 | schmidts | good morning |
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| 07:02.14 | porche | Hi all |
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| 07:02.49 | porche | is there a way to check if a number is working? |
| 07:03.02 | porche | without completing the call or from some other way? |
| 07:04.24 | kaldemar | you'd have to define what working means and what technology would be used to reach whatever is behind the number. |
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| 07:08.24 | porche | kaldemer, working means it's valid and a call can be connected to it if it's called |
| 07:09.49 | kaldemar | valid as in the first hop asterisk box has a matching extension or valid as in some device in the network, be it in the first asterisk or behind interface X, is behind the number? |
| 07:11.19 | porche | yeah, for example |
| 07:11.27 | porche | 44 is UK, 4420 is london |
| 07:11.36 | porche | and after 4420 there must be 7 digits |
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| 07:12.05 | porche | issue here, 11 digits with area code checks is a step on checking validity |
| 07:12.20 | porche | but how can we know that the number is really valid, ie it can be called? |
| 07:14.30 | kaldemar | ask your provider if they have a service for that. what's the real value though? the caller will probably hear an announcement from the telco for an invalid number anyway. |
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| 07:35.46 | *** join/#asterisk oquidave_ (~oquidave@41.190.129.127) |
| 07:35.53 | oquidave_ | hello people |
| 07:36.27 | oquidave_ | are configuring a sip trunk, however, am when i sip debug, i get the following error |
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| 07:36.46 | oquidave_ | SIP/2.0 500 Server Internal Error |
| 07:36.47 | oquidave_ | Via: SIP/2.0/UDP 172.18.0.34:5060; |
| 07:37.12 | oquidave_ | from the remote sip server(SIP read from UDP:10.1.16.72:5060) |
| 07:37.22 | oquidave_ | does anyone know how to go about this |
| 07:38.12 | ChannelZ | probably have to ask your provider |
| 07:38.44 | ChannelZ | Are you behind NAT? |
| 07:39.02 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 07:39.07 | jacc0 | good morning :0 |
| 07:39.10 | ChannelZ | I assume so with the 10.1.16.72 IP |
| 07:39.10 | kaldemar | oquidave_: what else is there? retry-after header? |
| 07:39.48 | oquidave_ | kaldemar: |
| 07:39.50 | oquidave_ | Call-ID: 0d8a26e95a2676aa579a540866d5ebc5@172.18.0.34 |
| 07:39.51 | oquidave_ | From: "asterisk"<sip:asterisk@172.18.0.34>;tag=as278c4fad |
| 07:39.52 | ChannelZ | Looks like maybe they don't know you're behind NAT and are trying to send something to a bogus IP though it's interesting you're getting a SIP response at all in that case |
| 07:39.53 | kaldemar | ~pb |
| 07:39.53 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 07:39.53 | oquidave_ | To: <sip:$(EXTEN)@10.1.16.72:5060>;tag=5453b430 |
| 07:39.54 | oquidave_ | CSeq: 102 INVITE |
| 07:39.56 | oquidave_ | Reason: Q.850;cause=0;text="unknown" |
| 07:39.57 | oquidave_ | Content-Length: 0 |
| 07:40.27 | kaldemar | don't paste here, use a pastebin. and paste the whole dialog. |
| 07:41.04 | oquidave_ | kaldemar: thanks, am new to irc |
| 07:41.13 | oquidave_ | kaldemar: let me do that |
| 07:41.24 | oquidave_ | infobot: thanks for the links...my bad |
| 07:41.24 | infobot | gern geschehen, oquidave_ |
| 07:42.03 | kaldemar | infobot is a bot, not a person. seems to appreciate the thanks though. :) |
| 07:42.15 | ChannelZ | Gesundheit. |
| 07:42.42 | jacc0 | I was setting up fail-over asterisk yesterday using flip1405 |
| 07:42.52 | ChannelZ | ohh.. I see one problem |
| 07:42.53 | *** join/#asterisk k3asd` (~k3asd`@static-217-133-87-244.clienti.tiscali.it) |
| 07:43.02 | jacc0 | 2 cloned servers with asteris 1.8.4-rc3 |
| 07:43.06 | ChannelZ | your To: line says $(EXTEN)@10.1.16.72 |
| 07:43.06 | oquidave_ | kaldemar: here is the link ....http://pastebin.com/ERENiDEc |
| 07:43.28 | kaldemar | ChannelZ: +1, that's probably it. |
| 07:43.34 | ChannelZ | something is messed up for your sip peer in sip.conf |
| 07:43.34 | oquidave_ | ChannelZ: here is the link ....ChannelZ |
| 07:43.57 | k3asd` | hi all |
| 07:44.00 | jacc0 | when I turned of one everything seemd fine exapt for the the audio with the cisco callmanager: no audio in both directions (phones do ring) |
| 07:44.08 | ChannelZ | either that or your Dial() in extensions is crazy |
| 07:44.19 | kaldemar | or in dialplan it says $(EXTEN) where is should say ${EXTEN}. |
| 07:44.24 | kaldemar | oquidave_: ^ |
| 07:44.35 | jacc0 | no clue with the problem was; maybe some arp-poisening protection in the cisco call manager? |
| 07:45.02 | oquidave_ | ChannelZ: let me send you my sip.conf and extension.conf file for the trunk.....you guys are being very helpful...thanks |
| 07:45.21 | oquidave_ | kaldemar: let me send you my sip.conf and extension.conf file for the trunk.....you guys are being very helpful...thanks |
| 07:45.37 | jacc0 | the cisco call manager was behind a router; router arp-list seemed to be updated okay |
| 07:45.42 | ChannelZ | not necessary probably, as we said, look at your extensions.conf for whatever extension is being used to dial out, you've got a syntax error |
| 07:46.26 | ChannelZ | you need ${EXTEN} instead of $(EXTEN) |
| 07:46.44 | oquidave_ | ChannelZ: okay let me change that...thanks |
| 07:47.23 | ChannelZ | Your ITSP is being asked to dial $(EXTEN) instead of the actual number and it's basically saying "I am confused" |
| 07:48.00 | oquidave_ | ChannelZ: i have changed it to exten => 0800707020,1,Dial(SIP/${EXTEN}@trunk-smsmedia-warid, 60) |
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| 07:48.29 | ChannelZ | looks right |
| 07:50.16 | oquidave_ | ChannelZ: let me try again |
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| 07:52.09 | oquidave_ | ChannelZ: now it's giving me the error....app_dial.c:1759 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 07:52.27 | jacc0 | @kaldemar: any experiance with flip1405 in combination with a cisco callmanager behind a router? |
| 07:53.50 | jacc0 | it's a national holiday over here; liberty day |
| 07:54.01 | jacc0 | is going to watch the wailers tonight :) |
| 07:56.30 | oquidave_ | kaldemar: i haev made the neccessary changes as instructed, however, i get the error....app_dial.c:1759 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 07:56.53 | oquidave_ | kaldemar: do you any idea what could be wrong...thanks? |
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| 07:57.28 | kaldemar | oquidave_: did you dial the same number as before? cause 20 is because asterisk does not know an address for the destination. which would be likely in the case of SIP/${EXTEN}@trunk-smsmedia-warid since asterisk believes that trunk-smsmedia-warid is a host name. |
| 07:58.05 | kaldemar | oquidave_: if you want to dial using a device define in sip.conf, like [trunk-smsmedia-warid], use SIP/trunk-smsmedia-warid/${EXTEN}. |
| 07:58.59 | kaldemar | jacc0: no experience in flip1405. some interesting experiences with call manager. |
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| 08:00.46 | oquidave_ | kaldemar: thanks for the reply.....[trunk-smsmedia-warid] is a trunk i have defined in my sip.conf file |
| 08:00.49 | jacc0 | I now leftout the cisco totaly and managed to make a fail-over construction connecting to a private GSM server directly (it is asterisk 1.6 based) |
| 08:01.29 | jacc0 | dial(sip/trunk-smsmedia-warid/${EXTEN}) |
| 08:01.45 | jacc0 | @oquidave: ial(sip/trunk-smsmedia-warid/${EXTEN}) |
| 08:02.38 | oquidave_ | kaldemar: thanks for the reply.....[trunk-smsmedia-warid] is a trunk i have defined in my sip.conf file |
| 08:02.49 | oquidave_ | kaldemar: thanks for the reply.....i have changed it to exten => 0800707020,1,Dial(SIP/trunk-smsmedia-warid/${EXTEN}, 60) |
| 08:03.07 | oquidave_ | jacc0: i have changed it to ....exten => 0800707020,1,Dial(SIP/trunk-smsmedia-warid/${EXTEN}, 60) |
| 08:03.56 | jacc0 | oquidave: NL ? |
| 08:04.04 | jacc0 | looks like a dutch number |
| 08:04.20 | jacc0 | :) |
| 08:04.41 | *** part/#asterisk trijezdci (~trijezdci@192.100.123.77) |
| 08:05.39 | oquidave_ | kaldemar: thanks for the help.....now am getting ...SIP/2.0 504 Server Time-out, Via: SIP/2.0/UDP 172.18.0.34:5060 |
| 08:05.55 | oquidave_ | jacc0: thanks for the help.....now am getting ...SIP/2.0 504 Server Time-out, Via: SIP/2.0/UDP 172.18.0.34:5060 |
| 08:07.23 | oquidave_ | kaldemar: and Unable to create channel of type 'SIP' (cause 20 - Unknown) when i dial the number 0800707020 |
| 08:07.49 | oquidave_ | kaldemar: thanks for the help, i truely appreciaate |
| 08:08.11 | kaldemar | oquidave_: does [trunk-smsmedia-warid] has an address in the host option or is it "dynamic"? |
| 08:08.31 | *** join/#asterisk coppice (~chatzilla@m121-202-40-53.smartone-vodafone.com) |
| 08:09.50 | oquidave_ | kaldemar: it has a static address 10.1.16.72 |
| 08:12.57 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 08:14.01 | oquidave_ | kaldemar: here is my trunk configs in the sip.conf file if u don't mind....http://pastebin.com/SX3eaY29a |
| 08:15.06 | kaldemar | oquidave_: "Unknown Paste ID!" |
| 08:15.43 | kaldemar | oquidave_: is 10.1.16.72 correct? |
| 08:17.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:18.06 | oquidave_ | kaldemar: what do you mean?please clarify |
| 08:18.58 | oquidave_ | kaldemar: incase you mean whether it's the right ip address of the remote sip server, then yes it is correct. thanks |
| 08:22.09 | kaldemar | it's a private address. |
| 08:23.14 | oquidave_ | kaldemar: yes it is |
| 08:24.40 | oquidave_ | kaldemar: here is the trunk config....http://pastebin.com/EEfMUdW5 |
| 08:28.04 | kaldemar | and a CLI output of a call with sip debug enabled? |
| 08:29.03 | oquidave_ | kaldemar: yes...should i get you the output? |
| 08:30.24 | jacc0 | @oquiadve_ : yes, and could you also getus ip/subnetmask/gateway of both serevers |
| 08:30.48 | jacc0 | I susspect subnet configuration to be wrong |
| 08:31.42 | jacc0 | are they in the same subnet? or routed? |
| 08:32.48 | oquidave_ | jacc0: thanks for the reply, i really appreciate....they are not on the same subnet, let me send you the debug output |
| 08:34.22 | oquidave_ | kaldemar: here is the debug output |
| 08:34.30 | oquidave_ | kaldemar: http://pastebin.com/Fqt5pyVH |
| 08:34.46 | oquidave_ | jacc0: here is it is....http://pastebin.com/Fqt5pyVH |
| 08:39.19 | jacc0 | I susspect subnet/gateway configuration or firewall configuration : is there any firewall in between? |
| 08:39.39 | oquidave_ | jacc0: do you know of any web tool that allows sharing images, so that i send you the network configuration? Thanks |
| 08:40.21 | jacc0 | http://picturepush.com/upa |
| 08:40.36 | oquidave_ | the remote sip server(10.1.16.72) is behind a router |
| 08:40.49 | oquidave_ | jacc0: great |
| 08:43.48 | oquidave_ | jacc0: the site is not loading on my end, do you have an alternative site...thanks |
| 08:44.26 | jacc0 | !google: upload image without registration |
| 08:45.08 | jacc0 | and find gazilion |
| 08:45.14 | jacc0 | :p |
| 08:48.03 | jacc0 | I need to go offline: technical problems @ a customer |
| 08:48.05 | jacc0 | sorry |
| 08:50.27 | oquidave_ | jacc0: yes also need to do something briefly, we shall discuss when you are back online, but thanks alot |
| 08:51.22 | jacc0 | it's public holiday over here |
| 08:51.23 | oquidave_ | kaldemar: can we pause a bit for a moment. just something has interrupted, however, thanks alot for the help, i greatly appreciate |
| 08:51.32 | jacc0 | I'll be going to watch the wailers later on :) |
| 08:52.10 | oquidave_ | jacc0: can u send me your email address for future reference...thanks |
| 08:52.19 | jacc0 | jaccovantuijl@gmail.com |
| 09:10.55 | ectospasm | notes jacc0's e-mail as spam target... |
| 09:17.36 | *** join/#asterisk the_5th_wheel (~edd@webster.cybertek.co.za) |
| 09:18.24 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fc.bcn.adamo.es) |
| 09:30.20 | the_5th_wheel | Howdy. Im stumped. I have peers setup between two asterisk boxes. Box A is 1.4 box B is 1.6. Both are set as peers. Both have ost set to the IP of the other box. There is no natting happening. It seems to ignore the context when I dial from Box A to B. Box B is is connected to ourbound providers. |
| 09:34.30 | ectospasm | the_5th_wheel: do you have tcpdump packet captures of the attempts? SIP debug? |
| 09:38.19 | the_5th_wheel | ectospasm: I can quickly make some. Which side do you want the dump of? |
| 09:39.04 | ectospasm | either side should be OK |
| 09:39.26 | the_5th_wheel | ectospasm: one moment. |
| 09:42.08 | the_5th_wheel | ectospasm: Do you want it in Pcap? Or what format |
| 09:42.34 | ectospasm | pcap is OK. Put it somewhere so I can get it with wget |
| 09:42.55 | ectospasm | also, a pastebin of SIP debug from that side would be helpful. |
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| 10:36.55 | sawgood | anyone home? |
| 10:37.02 | \DSAFEW\ | nope |
| 10:38.17 | sawgood | Is there a 'way' to see from the * CLI output (variables) without having to first edit them into the dialplan (/etc/asterisk/extensions.conf) ... by this I mean for example: |
| 10:38.36 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 10:38.38 | sawgood | Say, I want to see the current Caller ID number and name inside the CLI for all calls .... |
| 10:39.35 | sawgood | so I can see information on the fly |
| 10:40.23 | ectospasm | the_5th_wheel: do you have the SIP debug? The INVITE from 209...10 fails because the destination 031277..@41...149 does not exist |
| 10:41.00 | oquidave_ | kaldemar: are u there? |
| 10:41.50 | kaldemar | oquidave_: not for long |
| 10:43.55 | the_5th_wheel | ectospasm: the issue is its ignorniong the context |
| 10:44.27 | orn | sawgood: If the call is active, you could use core show channel [channel you want to look at] |
| 10:44.50 | sawgood | orn: thank you |
| 10:44.56 | ectospasm | the_5th_wheel: I've seen this before, can't remember how to fix it though. |
| 10:45.56 | sawgood | Asterisk by default writes to the system log a complete call record, for example: /var/log/asterisk/cdr-csv/ |
| 10:46.05 | sawgood | What if there are no files in this directory? |
| 10:47.04 | the_5th_wheel | ectospasm: it seems to be sending the username from the sip handset instead of the sip peer its coming from, and then confusing it |
| 10:47.41 | pierre` | does someone know good DECT bases compatible with asterisk ? |
| 10:48.00 | sawgood | Snom M3 is solid |
| 10:48.48 | orn | pierre`: Doro IP880 is good as well (basically the same unit as the Snom M3) |
| 10:49.05 | ectospasm | pierre`: we have a panasonic KX-TPA50 DECT handset that works |
| 10:49.07 | pierre` | (i need to support =~ 20 phones with 3 or 4 central bases) |
| 10:49.15 | ectospasm | pierre`: I did not set it up, however |
| 10:49.28 | orn | pierre`: And they need to be able to move between base stations? |
| 10:49.30 | ectospasm | not sure the model of the base stations. |
| 10:49.32 | pierre` | orn: yep |
| 10:49.43 | orn | pierre`: Then you'll need to look at something a bit more expensive, like perhaps Aastra |
| 10:49.44 | sawgood | Panasonic makes a cool DECT 6.0 unit with a base and 6 handsets |
| 10:49.51 | pierre` | i was reading doc on that : http://www.opcom.fr/product_info.php?cPath=13_19&products_id=86 |
| 10:49.56 | ectospasm | the Panasonic can move between stations OK, but I haven't tested it |
| 10:50.06 | sawgood | The base has one button access to each of the 6 DECT receivers |
| 10:50.21 | orn | ectospasm: With call handover? |
| 10:50.21 | pierre` | sawgood: do you have a reference ? |
| 10:50.45 | sawgood | I've seen the unit at IT EXPO West, and receently in a LAB ... It worked well |
| 10:50.50 | ectospasm | orn: all I know is I can walk throughout the building and get a signal almost anywhere |
| 10:50.53 | schmidts | pierre` take a look at the panasonic they are really nice, but i think you maybe need more like a ascom system but thats really expensive (>400$ for each handset) |
| 10:51.06 | pierre` | (we tried TOIP over wifi but it sucked, so now, we try DECT) |
| 10:51.06 | ectospasm | I haven't yet needed to answer a call while using the DECT |
| 10:51.11 | ectospasm | But it never seems to lose its signal. |
| 10:52.10 | oquidave_ | kaldemar: i needed to upload the network diag of my sip peering servers |
| 10:52.27 | pierre` | thank for all the informations, i will RTFM now :) |
| 10:53.29 | sawgood | would it be possible to tell Asterisk to write the Asterisk Hangupcause code to the Master.csv file in /var/log/asterisk/cdr-csv |
| 10:55.43 | oquidave_ | kaldemar: here is the link to the network diagram....https://docs.google.com/document/d/19ouic-GafuHQ3_j6sf14ulJlPpDOjQMvh2RnVmlEdAo/edit?hl=en&authkey=COjnh6QM |
| 10:55.56 | pierre` | I will read documentations on snom m3 and panasonic KX-TGP500 and KX-TGP550 systems |
| 10:59.03 | orn | pierre`: From what I've read, it doesn't seem like either one of those supports handovers between base stations. I believe you'll have to look at much more expensive solutions. |
| 10:59.13 | orn | pierre`: You might have to look at something like Aastra's SIP-DECT 142. |
| 10:59.53 | kaldemar | oquidave_: does 172.18.0.34 have a route for 10.1.16.72 via 172.18.0.33? |
| 11:00.00 | pierre` | orn: i will propose 2-3 solutions with compromise of prices and fonctionnalities |
| 11:02.11 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 11:02.31 | oquidave_ | kaldemar: yes, it does, i can ping 10.1.16.72 from my box(172.18.0.34) |
| 11:03.20 | oquidave_ | kaldemar: i added it(10.1.16.72) on my routing table using 172.18.0.33 as the gateway |
| 11:05.13 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:06.21 | kaldemar | oquidave_: actually, you said earlier that you're getting "SIP/2.0 504 Server Time-out, Via: SIP/2.0/UDP 172.18.0.34:5060", that comes from the SIP GW. |
| 11:07.38 | kaldemar | oquidave_: the 504 leads to the conclusion that the problem is behind SIP GW. |
| 11:08.05 | kaldemar | oquidave_: you had an issue with connectivity from the asterisk box to SIP GW, but that was already fixed earlier. |
| 11:13.44 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 11:14.30 | oquidave_ | kaldemar: is that affirmed by this line....Reliably Transmitting (no NAT) to 10.1.16.72:5060: |
| 11:14.58 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 11:15.26 | oquidave_ | kaldemar: so you are saying that i can connect to the remote sip server(10.0.16.72), but that remote server can not get back to me? |
| 11:17.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:17.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:21.22 | kaldemar | oquidave_: "Reliably Transmitting (no NAT) to 10.1.16.72:5060" only says that asterisk sends a packet to 10.1.16.72, nothing more. |
| 11:22.40 | kaldemar | oquidave_: "Via: SIP/2.0/UDP 172.18.0.34:5060" tells that 172.18.0.34 has most likely been involved in the dialog. "<--- SIP read from UDP:10.1.16.72:5060 --->" tells that 10.1.16.72 has responded you. |
| 11:23.49 | kaldemar | eh, .34 was you. anyway, the remove server seems to get back to you so that is not the problem. the problem is something behind 10.1.16.72 that causes the 504 Server Time-out. |
| 11:25.57 | oquidave_ | kaldemar: so i can confidently say that it's not a firewall on my side since i have opened up port 5060 though when i run netstat, there is not connection established with a foreign host |
| 11:30.29 | oquidave_ | kaldemar: thanks alot, let me contact my remote sip provider to look to the details |
| 11:33.18 | oquidave_ | kaldemar: thanks for the help, i would appreciate it if you availed me with you email. thanks |
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| 12:17.04 | bintut | waves |
| 12:17.31 | EmleyMoor | Caller ID from my BT (DAHDI) line is being wiped out by an "extra" polarity reversal. On 1.6.2.9, is there anything I can do to fix it? |
| 12:19.19 | bintut | anyone here running asterisk 1.8.3.3 from the upstream with sip/tls working using comodo's possitivessl? i'm always getting 401 in my asterisk server. |
| 12:21.21 | bintut | i used bria in my iphone4 and snom 300 as my sip client using tcp/tls port 5061 but failed. |
| 12:21.44 | bintut | any idea? |
| 12:24.24 | bintut | is this a problem with my setup that i use a commercial ssl certificate instead of generating my own or is this a bug in the current asterisk 1.8.3.3 from the upstream .deb package? please advice. thank you. ;-) |
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| 12:32.46 | EmleyMoor | is reinstating a patch in chan_dahdi.c |
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| 12:36.37 | EmleyMoor | will be surprised if it works, though |
| 12:40.20 | EmleyMoor | It doesn't <!> |
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| 12:47.49 | b0ot | Anyone here ever use the spa8000 |
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| 12:56.45 | beek | mornin' jaytee |
| 12:56.59 | beek | waves to katty |
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| 12:58.51 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:06.12 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:09.55 | pierre` | orn: do you know how much phones can be connected to a single aastra rfp l32ip base ? I only find informations about simultaneous voice channels. |
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| 13:14.45 | *** join/#asterisk OldMonk (~raju@59.178.187.65) |
| 13:15.05 | OldMonk | hi. any way to prevent multiple machines logging into a single SIP account? |
| 13:15.50 | *** join/#asterisk datalay (~datalay@unaffiliated/datalay) |
| 13:18.04 | *** join/#asterisk bchia (~chatzilla@nat/digium/x-dnrgkxaznljzgona) |
| 13:20.18 | leifmadsen | OldMonk: not really, unless those remote machines are logging into your server, in which case you need to use ACL to block one of them (or a firewall) |
| 13:21.04 | OldMonk | leifmadsen: i could use a firewall if the clients had a fixed IP, but is there a way with variable IPs? |
| 13:21.35 | OldMonk | [firewall here being native iptables on the SIP server] |
| 13:22.42 | leifmadsen | OldMonk: you could probably use something like fail2ban and write some custom rules to block one of the offenders somehow |
| 13:23.05 | pigpen | Does anyone know if Panasonic's new DECT sip base & phone support hints/presence/buddywatch/whatever ? |
| 13:23.05 | leifmadsen | it's hard, because someone registering again from another location is not invalid, it just sucks when you don't want it |
| 13:23.07 | OldMonk | hmm, that would require deep packet inspection to get the auth ID |
| 13:23.22 | leifmadsen | you could write a rule that checks logs to see if someone is already registered, and if from a different IP, to reject / ban them |
| 13:24.05 | leifmadsen | I'm not entirely sure how else to approach it at this point |
| 13:24.11 | leifmadsen | never had to solve that problem |
| 13:25.11 | OldMonk | looks like there's no non-trivial way to do it |
| 13:25.15 | pigpen | leifmadsen, yeah, you would almost have to run a script that would have to dump packets, then determine which one is the "one" and block the others in iptables. |
| 13:25.18 | leifmadsen | OldMonk: right |
| 13:25.32 | leifmadsen | I don't see how you could develop a trivial way to do it |
| 13:26.13 | pigpen | leifmadsen, anyway to narrow it down by mac addres? |
| 13:26.21 | pigpen | just thinking out loud. |
| 13:26.24 | leifmadsen | only way I can think, is to NOT have the phones register, and use something like dyndyn.org to update the IP address and then set the hostname statically |
| 13:26.37 | pigpen | IP adddress is no good. for that matter over nat mac is lost. |
| 13:26.41 | leifmadsen | pigpen: certainly could if they register with their mac addresses :) |
| 13:26.42 | orn | pierre`: I believe you can have 12 handsets, and 8 simultaneous calls |
| 13:26.45 | OldMonk | ok, just thinking out aloud here -- if you set call-limit (or equivalent) for a sip account, would it apply to the aggregate of all peers logged into that account or each peer logged into that account? |
| 13:27.08 | leifmadsen | OldMonk: call-limit just limits the number of simultaneous calls for the peer, not the registration at all |
| 13:27.11 | orn | pierre`: Per access point that is, but if you add access points the theoretical limit is 512 handsets |
| 13:27.33 | OldMonk | leifmadsen: i know that, trying to make it difficult for the person who's logged into a duplicate account |
| 13:27.52 | leifmadsen | how are you determining it to be a duplicate account? |
| 13:28.28 | leifmadsen | if someone registers with one account, and someone else registers, the registration of the first account is blown away from memory and delivery of calls to the registration end point is to the last registration |
| 13:28.46 | leifmadsen | the authentication of end points placing calls does not relate to the registration |
| 13:29.05 | OldMonk | there's only outbound on these accounts, no incoming |
| 13:29.39 | leifmadsen | if you're not actually worried about the registration, but rather the placing of calls from the end point (which is an entirely separate matter from the registration part), then just use GROUP() and GROUP_COUNT() in the dialplan |
| 13:29.59 | leifmadsen | when you say "incoming" I presume you mean incoming form the viewpoint of the phone/device |
| 13:30.05 | leifmadsen | and not incoming from the viewpoint of asterisk |
| 13:30.16 | leifmadsen | if that is the case, you don't need registration stuff at all |
| 13:30.17 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:30.17 | pierre` | orn: thanks, i can't find informations about the maximum phone numbers per base. |
| 13:30.25 | OldMonk | leifmadsen: not really, but it's one way to (a) detect the problem and (b) have a control over it |
| 13:30.26 | pierre` | orn: thanks, i can't find informations about the maximum handsets numbers per base. |
| 13:30.32 | leifmadsen | the phone will know where you asterisk server is and will attempt to authenticate when placing the call |
| 13:30.51 | leifmadsen | OldMonk: I don't understand what you mean by (a) and (b) |
| 13:30.56 | leifmadsen | I'm not even sure I understand what the real problem is |
| 13:30.57 | orn | pierre`: I think it is 12 |
| 13:31.14 | OldMonk | i mean, i want to limit simultaneous logins, but if there's no easy way to do i want to be able to at least detect it and, hopefully, disincentivise my callers from trying to use another person's account |
| 13:31.16 | leifmadsen | are you worried about multiple devices placing calls via the same account? |
| 13:31.24 | leifmadsen | please define "login" |
| 13:32.00 | *** join/#asterisk CheeYang (~CheeYang@175.144.126.191) |
| 13:32.01 | leifmadsen | it sounds like you're just worried about multiple end points placing calls via the same account |
| 13:32.09 | leifmadsen | which is not at all related to registrations |
| 13:32.14 | leifmadsen | in fact a registration is not even necessary |
| 13:32.22 | OldMonk | leifmadsen: i have 1000 telecallers, and i want to ensure that they use their own account and not someone else's. there's no centralised auth in place, so passwords are trivial, telecaller A can easily guess telecaller B's password |
| 13:32.31 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:32.39 | leifmadsen | that sounds like an implementation issue |
| 13:32.53 | leifmadsen | but if you just want to limit the number of simultaneous calls on a single account, use GROUP() and GROUP_COUNT() |
| 13:33.12 | CheeYang | Hi, how may I retrieve the PSTN trunks status from Asterisk AMI? |
| 13:33.13 | OldMonk | leifmadsen: ok, will try that -- at least they won't share passwords then |
| 13:33.30 | leifmadsen | writing you a simple dialplan to show what I mean |
| 13:36.00 | leifmadsen | OldMonk: http://pastebin.com/YvUXhTYZ |
| 13:36.02 | MrTelephone | 1.8.4rc3 segfaults because of odbc at runtime. Is this happening to everyone? |
| 13:36.09 | leifmadsen | MrTelephone: nope |
| 13:36.21 | leifmadsen | not everyone is using ODBC I'm sure |
| 13:36.48 | pierre` | orn : I found a obscur AAstra presentation in France with 12 sigalisation sessions, do 12 seems to be the correct number of supported handsets per base. |
| 13:37.00 | pierre` | s/sigalisation/signalisation/ |
| 13:37.14 | OldMonk | leifmadsen: got it, thanks |
| 13:37.52 | MrTelephone | Just wondering if it might be a problem with my voicemail table |
| 13:38.50 | leifmadsen | MrTelephone: more likely is a bug you may be hitting in the unixODBC drivers -- make sure you're using the latest from the unixodbc.org site and not from your distributions packages |
| 13:38.59 | leifmadsen | that's the first place to usually start with ODBC crashes |
| 13:44.44 | MrTelephone | I wonder how sony got hacked |
| 13:45.07 | MrTelephone | Probably someone who wasn't happy with their pay on the inside |
| 13:45.12 | OldMonk | MrTelephone: they used their distribution's ODBC drivers |
| 13:45.27 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 13:45.45 | MrTelephone | I just seen oracle got hacked now |
| 13:46.16 | *** part/#asterisk serafie (~erin@nat/digium/x-vrxlwcioldmkkrus) |
| 13:46.49 | m4xx | is it possible to do math in a dialplan? like set(current=${EPOCH}) then further in the dialplan gotoif(${current}-${EPOCH}=${x}?con1,exten1,prio1:con2,exten2,prio2); |
| 13:47.57 | kaldemar | m4xx: sure. either with $[] or with func MATH. |
| 13:48.00 | MrTelephone | My unixodbc drivers are 2.2 and the current is 2.5. I'm guessing that is pretty old. |
| 13:48.46 | m4xx | awesome |
| 13:49.48 | MrTelephone | *thumbs up* |
| 13:50.55 | kaldemar | m4xx: actually your example was only missing $[] around ${current}-${EPOCH}=${x} |
| 13:51.07 | m4xx | =] |
| 13:51.14 | m4xx | what about multiple conditions |
| 13:51.23 | leifmadsen | you only get true:false |
| 13:51.53 | leifmadsen | you may have to elaborate |
| 13:52.00 | m4xx | yes, one second =] |
| 13:52.59 | m4xx | so i couldnt have set(diff=$[${EPOCH}-${current}) ... GOTOIF(${diff} >= ${x} && ${diff} <= $y?...:...); |
| 13:53.16 | leifmadsen | you can certainly do that |
| 13:53.19 | leifmadsen | single ampersand |
| 13:53.23 | m4xx | awesome |
| 13:53.30 | leifmadsen | this is all covered in the dialplan chapters of.... |
| 13:53.31 | leifmadsen | ~thebook |
| 13:53.32 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 13:53.33 | m4xx | sorry, i'm trying to read and learn by doing at the same time |
| 13:53.48 | *** join/#asterisk sereal (~jjrh@2001:4830:16ca:1:21a:6bff:fe6a:3cd2) |
| 13:53.52 | m4xx | almost finished with chapter 5 |
| 13:54.01 | m4xx | 6 is next |
| 13:54.04 | m4xx | which probably covers that? |
| 13:54.25 | leifmadsen | it'll either in be dialplan basics, or the other dialplan chapter |
| 13:55.06 | m4xx | leifmadsen, have you written any other books? this one is informative and entertaining in the same token =P |
| 13:56.47 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 13:58.43 | leifmadsen | Asterisk Cookbook |
| 13:59.02 | leifmadsen | that's about it for now |
| 13:59.23 | fullstop | leifmadsen: I'm curious, what do you author the asterisk book in? latex? |
| 13:59.48 | leifmadsen | xmlmind (docbook) |
| 14:00.12 | leifmadsen | source is kept in subversion |
| 14:01.02 | OldMonk | leifmadsen: oh, i've read _Asterisk - The Future of Telephony_ -- thanks! |
| 14:02.03 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 14:02.05 | leifmadsen | :) |
| 14:02.16 | OldMonk | is _The Definitive Guide_ available for download as PDF? |
| 14:02.23 | leifmadsen | not for free |
| 14:02.31 | OldMonk | how about the source? :) |
| 14:02.32 | leifmadsen | you can purchase it from O'Reilly though |
| 14:03.06 | OldMonk | to tell the truth i prefer tech books on my computer and other reading in paper |
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| 14:09.00 | *** part/#asterisk Miriup (~dirk@pdpc/supporter/active/miriup) |
| 14:12.58 | fullstop | leifmadsen: I understand.. it looks like O'Reilly runs the docbook through adobe framemaker. |
| 14:13.04 | leifmadsen | exactly |
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| 14:13.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:13.14 | leifmadsen | they take the source, then do their stuff outside of docbook for page setting and stuff |
| 14:13.24 | leifmadsen | it takes a long time to get through the process. That book was in production for 2-3 months |
| 14:13.39 | fullstop | That's interesting.. that they use a closed-source typesetting package. |
| 14:14.11 | fullstop | I'm sure that they have their reasons, though. |
| 14:15.13 | leifmadsen | probably been using it for years, and just because they are pro-open source doesn't make the open source apps the best :) plus I'm sure its easier to find people who are trained on that software |
| 14:17.17 | OldMonk | plus they're likely to have a well-defined production process that they apply to all books, regardless of whether the book is to be published as CC or proprietary |
| 14:17.38 | MrTelephone | Adobe makes good stuff |
| 14:18.48 | fullstop | except for flash |
| 14:20.19 | MrTelephone | Flash is ok? |
| 14:20.26 | MrTelephone | Better than java |
| 14:20.58 | beek | MrTelephone: Really? |
| 14:21.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:21.08 | fullstop | Java applets were awful, but I don't like flash. |
| 14:21.08 | MrTelephone | yeah, how many websites programmed in java screw up on you? |
| 14:21.27 | beek | Javascript != Java |
| 14:21.45 | fullstop | client side java isn't that great in a browser. Server side java is wonderful. |
| 14:22.20 | MrTelephone | I find it really slow. Is facebook chat script javascript or java? |
| 14:22.34 | m4xx | javascript |
| 14:22.35 | fullstop | facebook is javascript and php |
| 14:22.40 | MrTelephone | Farmville is javascript? |
| 14:22.45 | m4xx | <3 hpp ;x |
| 14:22.55 | fullstop | I've never played farmville, but I suspect that it is flash. |
| 14:24.37 | MrTelephone | We bought this netenforcer years ago to manage traffic. It's interface was java or javascript. I had to load the client side java runtime version 2.* for it to work probably. Then install version 6 for current sites to work. |
| 14:24.54 | MrTelephone | I've never had to go back and edit perl scripts for them to work with new versions of stuff |
| 14:25.35 | fullstop | You are confusing client side and server side things. |
| 14:28.04 | MrTelephone | Maybe it's something to look into. Server side java. I'm guessing people like it because there is a lot of support files for it? |
| 14:28.26 | MrTelephone | I tried to make a blackberry application a while back but the java development kit didn't work with the version of eclipse they had for download. |
| 14:29.02 | MrTelephone | Do you have an android tablet? |
| 14:31.30 | MrTelephone | So asterisk 1.8.4rc3 works with unixodbc 2.3 and libmyodbc 3.51.15r409-4 without a segfault. Leifmadson, thanks for the hint. |
| 14:32.01 | leifmadsen | MrTelephone: awesome, no worries. That's always where I start. |
| 14:32.05 | leifmadsen | OldMonk: and you doubted me :) |
| 14:32.25 | MrTelephone | Are you not developers and know this already? I thought you were just jerking my chain |
| 14:32.44 | OldMonk | leifmadsen: nah, i was just being facetious about the OT discussion :) |
| 14:32.57 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:33.05 | leifmadsen | heh, no I've done a lot of res_odbc installs, and ODBC related crashes almost almost stem from the unixODBC packages being used locally |
| 14:33.09 | MrTelephone | OldMonk hit it home with the 'sony got hacked because they use distro releases of stuff' |
| 14:33.21 | leifmadsen | upgrading unixOBDC almost always fixes crash issues |
| 14:33.26 | leifmadsen | :) |
| 14:33.30 | leifmadsen | OldMonk: :D |
| 14:33.50 | *** part/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 14:33.56 | Kobaz | i've moved away from odbc because of various problems. i never had a crash but i have db handle leaks |
| 14:34.22 | MrTelephone | kobaz, what did the leaks do to you and how did you notice them? |
| 14:34.57 | leifmadsen | I have a system using unixodbc HEAVILY on an ABE C.2.4.1 release and it is rock solid |
| 14:35.01 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 14:35.05 | leifmadsen | been in production for over 2 years |
| 14:35.07 | leifmadsen | moy: o/ |
| 14:35.28 | *** join/#asterisk gruvfunk (~chatzilla@user-160uac8.cable.mindspring.com) |
| 14:35.28 | MrTelephone | Yeah I have to use ODBC for clustering. |
| 14:35.52 | Kobaz | MrTelephone: it would lose connection to postgres every so often |
| 14:35.56 | Kobaz | MrTelephone: but it wouldn' |
| 14:36.17 | Kobaz | but it wouldn't drop the connection to postgres completely... so after a few days of running it hit the max postgres connections that i had (200) |
| 14:36.24 | leifmadsen | I use MySQL in the backend and it is master-master replicated |
| 14:36.53 | Kobaz | so i wrote a script as a stopgap, to kill idle postgres connections older than 24 hours |
| 14:36.56 | moy | leifmadsen: \o |
| 14:37.24 | MrTelephone | did you try using mysql kobaz? |
| 14:37.29 | Kobaz | of course not |
| 14:37.33 | Kobaz | why would i use mysql |
| 14:37.44 | leifmadsen | troll |
| 14:37.46 | Kobaz | hehe |
| 14:37.47 | MrTelephone | I never used postgres before |
| 14:37.56 | leifmadsen | I used pgsql for years, then switched to mysql |
| 14:38.00 | MrTelephone | postgres? What kind of name is that |
| 14:38.01 | leifmadsen | they both work fine |
| 14:38.05 | Kobaz | well we have thousands of lines of code in postgres stored procedures and triggers |
| 14:38.22 | Kobaz | we can't exactly 'try using mysql' |
| 14:38.26 | leifmadsen | right |
| 14:38.39 | leifmadsen | your comment implied that mysql is shit and you're stupid for using it |
| 14:38.55 | Kobaz | and i wouldn't want to anyway |
| 14:38.55 | Kobaz | hehe |
| 14:38.55 | Kobaz | well |
| 14:38.55 | MrTelephone | lol |
| 14:38.55 | leifmadsen | that's how I read it anyways ;) |
| 14:38.55 | Kobaz | bet tool for the job |
| 14:38.55 | leifmadsen | right |
| 14:39.03 | Kobaz | but i personally don't like mysql after i've gotten bit several times |
| 14:39.10 | Kobaz | s/bet/best |
| 14:39.25 | Kobaz | but hehe, but yeah mysql is a little lacking.. but that's another story :P |
| 14:39.31 | MrTelephone | Well kobax it's another one of those looking for a needle in a haystack situations. It sounds like maybe with all the extra code you have written may have been affecting the database engine somehow? |
| 14:39.40 | Kobaz | MrTelephone: nope |
| 14:39.54 | Kobaz | work flawlessly with native postgres connection libraries |
| 14:40.09 | Kobaz | odbc-postgres on the version i used, leaks connection handles |
| 14:40.40 | MrTelephone | So you switched to IMAP or local storage? |
| 14:40.49 | Kobaz | and if some perl code in a postgres stored procedure causes a connection to leak, that would be really really bad |
| 14:41.03 | Kobaz | i use local storage for voicemail |
| 14:41.35 | Kobaz | leifmadsen: so why'd you switch to mysql? |
| 14:41.51 | MrTelephone | leifmadsen, do you ever see the asterisk developers adding support in asterisk to compete with the location() feature in openser? |
| 14:42.23 | MrTelephone | I always thought mysql had the higher user base. They might goto shit now that oracle bought them out? |
| 14:42.47 | jaytee | I tried using mysql but the system already had hersql on it and she wouldn't let me install it. she bitches all the time whenever I run a simple query and nags constantly about me not applying updates. |
| 14:42.49 | Kobaz | it does have a higher user base |
| 14:42.58 | beek | jaytee: :D |
| 14:43.00 | Kobaz | but if you want enterprise type features, mysql just doesn't have them |
| 14:43.11 | beek | Kobaz: That's not as true as it once was. |
| 14:43.13 | MrTelephone | Give me an example |
| 14:43.13 | Kobaz | they have a higher user base because of advertising |
| 14:43.25 | MrTelephone | Simply Accounting uses MYSQL now |
| 14:43.26 | Kobaz | MrTelephone: do cascading deletes work yet? |
| 14:43.29 | OldMonk | MrTelephone: of enterprise features? |
| 14:43.39 | leifmadsen | Kobaz: I find it easier to work with |
| 14:43.40 | Kobaz | that's not even enterprise |
| 14:43.43 | OldMonk | how about a field with more than 65535 bytes? |
| 14:43.46 | leifmadsen | MrTelephone: I'm unfamiliar with openser |
| 14:43.47 | beek | Kobaz: But if you choose to use the enterprise features, MySQL doesn't run any faster than PostgreSQL |
| 14:44.12 | beek | FWIW, I still use PostgreSQL except in those rare occasions (read: wordpress) where MySQL is the preferred database. |
| 14:44.23 | leifmadsen | this isn't #database-debate |
| 14:44.26 | Kobaz | hehe |
| 14:44.29 | beek | PG has never failed me. |
| 14:44.35 | Kobaz | one of my biggest peves is silent-trucation |
| 14:44.43 | OldMonk | basically mysql kept convincing users that feature <foo> was not needed, until the next version when they introduced <foo> and suddenly it became the hottest feature for any RDBMS to have ;) |
| 14:44.46 | Kobaz | which you can turn on some warnings now, in the new versions so it's not silent |
| 14:44.49 | Kobaz | but it still lets your data in |
| 14:45.03 | OldMonk | ok, no more mysql-bashing from me |
| 14:45.34 | fullstop | mysql uses the apple technique. Who want's CUT AND PASTE? |
| 14:45.36 | OldMonk | it's fun though... mysql, php, vi... the list goes on ;) |
| 14:45.41 | fullstop | ugh.. wants |
| 14:45.49 | fullstop | I don't know where that apostrophe came from. |
| 14:46.37 | Kobaz | oh |
| 14:46.43 | Kobaz | okay, one more thing and then i'm done |
| 14:46.54 | Kobaz | in mysql, you can't do DDL in a transaction |
| 14:47.07 | fullstop | The same is true for oracle. |
| 14:47.10 | fullstop | DDL commits |
| 14:47.12 | Kobaz | that's a must-have for us |
| 14:47.38 | MrTelephone | leifmadsen: If dns serv records spread users out on 2 asterisk boxes how do I get one asterisk box to check the other to see if the user is connected to it? |
| 14:47.40 | fullstop | you alter tables in real-time? |
| 14:47.53 | leifmadsen | MrTelephone: dundi |
| 14:47.56 | MrTelephone | DDL, quit talking chinese |
| 14:48.05 | Kobaz | MrTelephone: schema changes |
| 14:48.24 | Kobaz | fullstop: sometimes, if we need to emergency fix a performance issue... but mainly it's for doing atomic upgrades |
| 14:48.31 | OldMonk | Kobaz: but... but... why? |
| 14:48.50 | Kobaz | if something breaks half way through an upgrade, you don't want your system in a semi-undefined start |
| 14:49.04 | Kobaz | *state... damn fingers. |
| 14:49.13 | OldMonk | i quote (myself): "a schema that changes ever so often on production isn't a schema, it's a schoolbook scribbling" |
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| 14:50.21 | MrTelephone | leifmadsen: thanks, I'll have to check it out. All the clustering manuals don't mention dundi so I'm a bit skeptical about it working but I'll give it a try. |
| 14:50.33 | leifmadsen | I've used it lots |
| 14:50.35 | leifmadsen | works great |
| 14:50.46 | leifmadsen | I've even written documentation on it ;) |
| 14:50.51 | OldMonk | MrTelephone: there's a couple of excellent howtos on the 'net -- vivek kapoor and another one |
| 14:50.58 | Kobaz | OldMonk: if you don't have a system in place for schema changes, you're going to screw youself in the long run... and transaction ddl is very awesome |
| 14:50.59 | MrTelephone | ok |
| 14:51.05 | Kobaz | anyways... back to axeterisk |
| 14:51.19 | MrTelephone | And does it still work if you have dialplans where it dials 3 phones but 2 of them are on the other server? |
| 14:51.26 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:51.37 | MrTelephone | Mysql Scores 3, Postgresql 0 |
| 14:51.39 | MrTelephone | j/k |
| 14:51.40 | leifmadsen | MrTelephone: when you know how dundi works and what it does, you'll be able to answer that question |
| 14:51.52 | MrTelephone | yes or no leif |
| 14:51.53 | MrTelephone | :( |
| 14:52.09 | leifmadsen | yes |
| 14:52.34 | MrTelephone | You could have saved yourself typing in Keystrokes = len(MrTelephone: when you know how dundi works and what it does, you'll be able to answer that question) - len(yes) |
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| 14:52.58 | MrTelephone | and visual basic is better than java too :P |
| 14:52.58 | OldMonk | Kobaz: still, you'd do schema changes on staging then apply to production when you know it works |
| 14:53.29 | beek | vi rules! |
| 14:53.50 | OldMonk | rolls up his sleeves and invites beek into the ring |
| 14:54.21 | beek | Don't tell me -- emacs? |
| 14:54.23 | aberrios | cheers beek on |
| 14:54.35 | OldMonk | beek: need you ask? |
| 14:54.54 | beek | So OldMonk -- shall we say "edit" at 20 paces? |
| 14:54.56 | Kobaz | OldMonk: and you simulate all the real-world conditions that your app will see in the lab? |
| 14:55.15 | OldMonk | beek: meet you at high noon |
| 14:55.33 | MrTelephone | If you lose connection to odbc and your using realtime for sip peers will asterisk use fall back to memory? |
| 14:55.35 | beek | aberrios: Want to be my second? |
| 14:55.38 | OldMonk | Kobaz: in terms of schema, sure |
| 14:55.48 | Kobaz | OldMonk: are you 100% certain your production release will work in all situations, given any usage |
| 14:56.03 | aberrios | beek: Hit it. |
| 14:56.05 | MrTelephone | Nothing is 100% |
| 14:56.21 | OldMonk | Kobaz: nah, you could be sony ;) |
| 14:56.30 | Kobaz | OldMonk: schema is one thing but then there's views and stored procedures |
| 14:56.55 | Kobaz | no, i'm not going to rewrite a 5gb table during peak hours |
| 14:57.11 | Kobaz | but I will replace a poorly performing view if the server is on it's knees |
| 14:57.31 | MrTelephone | Why not use your database just for storage and write externally the scripts you need to manipulate the data? |
| 14:57.44 | Kobaz | because that's the "old way" of doing things |
| 14:57.50 | MrTelephone | Then you don't have a stroke when something goes wrong |
| 14:58.06 | Kobaz | you still have a stoke when something goes wrong, no matter how you develop your system |
| 14:58.22 | MrTelephone | 3 packs of ciggarettes after the fact :( |
| 14:58.30 | Kobaz | by putting all of our technology in the db layer, it's become an amazing platform for rapid development |
| 14:58.38 | Kobaz | well not all, but most of the heavy lifting |
| 14:59.30 | MrTelephone | I totally understand where your coming from. Database engines are more like operating systems now. |
| 14:59.36 | *** join/#asterisk peep_ (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 15:00.08 | peep_ | yo yo yo |
| 15:00.21 | aberrios | wonders wether the fastest dCAP practical was done with emacs, vi or nano (or summat else). |
| 15:00.24 | MrTelephone | What I don't see is how mysql or any database knows that there has been data corruption. I don't know much about the engine itself but if a record is missing is there a warning of some sort? |
| 15:00.33 | peep_ | any chance that the world famous russellb is around? |
| 15:00.59 | beek | famous, or infamous? |
| 15:01.00 | leifmadsen | peep_: he's in a meeting |
| 15:01.36 | peep_ | beek: Infamous perhaps |
| 15:03.25 | OldMonk | "world-infamous"? |
| 15:03.30 | *** join/#asterisk glam (~glam@113.88.231.72) |
| 15:05.06 | Kobaz | leifmadsen: i'm curious, why did you wind up switching to mysql? |
| 15:05.51 | leifmadsen | Kobaz: like I said, I find it significantly easier to work with |
| 15:06.17 | Kobaz | sh |
| 15:06.17 | Kobaz | ah |
| 15:06.22 | Kobaz | i missed that |
| 15:06.38 | peep_ | leifmadsen: Im late to the party, easier than what? |
| 15:06.44 | leifmadsen | pgsql |
| 15:06.44 | Kobaz | postgres |
| 15:07.10 | leifmadsen | just how my brain works |
| 15:07.25 | Kobaz | some stuff is a little nicer admin wise |
| 15:07.28 | Kobaz | show create table |
| 15:07.33 | leifmadsen | yep |
| 15:07.42 | Kobaz | but i wound up writing my own export system for pg |
| 15:07.45 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:07.57 | Kobaz | i can get the ddl for anything now, it's quite nice |
| 15:08.20 | OldMonk | Kobaz: you can always get that from a dump anyway |
| 15:08.34 | Kobaz | ah yes but dump will get you the data in inconsistent orders |
| 15:08.56 | Kobaz | so doing db schema diffs and such, is a royal pain |
| 15:10.24 | OldMonk | true |
| 15:10.26 | MrTelephone | How do I switch my keyboard layout to chinese |
| 15:11.03 | oquidave_ | hello people |
| 15:11.11 | Kobaz | MrTelephone: that's an asterisk question? |
| 15:11.43 | oquidave_ | does anyone know why an Outbound trunk is unreachable |
| 15:11.54 | Kobaz | the pigs stole it |
| 15:12.29 | oquidave_ | when i sip show peers, one of my peers(the trunk) is unreachable, any idea why? thanks |
| 15:12.42 | Kobaz | your box can't reach it |
| 15:12.47 | peep_ | oquidave_: Because you cant reach it |
| 15:12.51 | peep_ | DAMN beat me to it |
| 15:13.00 | Kobaz | either your own machine is blocking it, something in the middle is blocking it, or the remote server is blocking it |
| 15:13.16 | OldMonk | oquidave_: could be wrong auth or network issues or misconfiguration |
| 15:13.17 | Kobaz | maybe the goats and the pigs are fighting over it |
| 15:13.56 | oquidave_ | okay, but i ping the remote sip server ip, it's can see it |
| 15:14.00 | *** join/#asterisk cesar_CR (~cesar@201.193.82.8) |
| 15:14.05 | Kobaz | or you have the wrong hostname, or wrong ip address |
| 15:14.10 | Kobaz | could be anything really |
| 15:14.18 | OldMonk | oquidave_: try sip set debug on or something |
| 15:16.34 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 15:27.47 | MrTelephone | leifmadsen, you have your mysql master replicated but if one fails how do you switch them out. Are you using heartbeat with failover scripts? |
| 15:29.42 | oquidave_ | okay i think i figured it out, it was something in my sip.conf file |
| 15:30.37 | oquidave_ | but now am getting SIP read from UDP:10.1.16.72:5060 --->SIP/2.0 504 Server Time-out Via: SIP/2.0/UDP 172.18.0.34:5060;...what does that mean? |
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| 15:39.22 | gruvfunk | oquidave_ do you have network restrictions? can 10.1.16.72 access port 5060 UDP on 172.18.0.34? |
| 15:39.41 | gruvfunk | (check firewall, router, proxy etc) |
| 15:47.44 | oquidave_ | Unmonitored gruvfunk: yes i have just confirmed |
| 15:48.49 | gruvfunk | then pastebin your configs so we can see... do you have an ACL? allow/deny somewhere? |
| 15:49.06 | MrTelephone | means the server didn't respond to the sip request |
| 15:49.19 | MrTelephone | oops that was already answered |
| 15:49.29 | oquidave_ | gruvfunk: let me paste my configs thanks |
| 15:49.33 | OldMonk | leifmadsen: in the Guide, Architecture chapter, chan_alsa in the table has an extra | prefix |
| 15:49.55 | MrTelephone | is 172.18.0.34 public? |
| 15:50.10 | gruvfunk | is 172.18.0.24 the asterisk server/pbx ? |
| 15:50.28 | MrTelephone | You may have a NAT issue there if 172 is public |
| 15:50.44 | *** join/#asterisk linuxgeek (~uppal@121.100.50.75) |
| 15:50.49 | gruvfunk | s/24/34 |
| 15:52.19 | MrTelephone | 172.16.0.0->172.31.255.255 |
| 15:54.02 | OldMonk | later |
| 15:54.33 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 15:54.39 | oquidave_ | gruvfunk: here it is....thanks.....http://pastebin.com/yx1c7GZZ |
| 15:56.34 | gruvfunk | oquidave_ I think you want NAT, line 11: nat=yes |
| 15:57.03 | gruvfunk | tell us more about your setup, the environment |
| 15:57.26 | gruvfunk | how are these 2 IP addresses separated ? (physically/logically) |
| 15:58.39 | oquidave_ | gruvfunk: am using opensuse 11.3, asterisk 1.6 and the two ip address are physically seperated...let me send you the network diagram |
| 15:58.48 | *** join/#asterisk manji (~manjiki@ppp-2-84-9-127.home.otenet.gr) |
| 15:59.00 | jaytee | gawd! Dialogic Powermedia HMP 3.0 is an unruly beast |
| 15:59.38 | sxpert | jaytee: you don't say ;) |
| 15:59.42 | oquidave_ | gruvfunk: here is the the link ....https://docs.google.com/document/d/19ouic-GafuHQ3_j6sf14ulJlPpDOjQMvh2RnVmlEdAo/edit?hl=en&authkey=COjnh6QM |
| 16:01.45 | gruvfunk | and what kind of network is this 172 link? |
| 16:02.02 | gruvfunk | private vpn? |
| 16:02.22 | oquidave_ | gruvfunk: it's an E1 connection |
| 16:02.50 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 16:02.58 | Aut0ExeC | sup guys |
| 16:03.07 | gruvfunk | oquidave_ private though, right? |
| 16:03.10 | gruvfunk | not a public network? |
| 16:03.29 | oquidave_ | gruvfunk: no its not |
| 16:03.33 | oquidave_ | gruvfunk: 10.1.16.72 is the remote sip server and my server is 172.18.0.34 |
| 16:03.53 | gruvfunk | if this is a private network, you have to look at the router/firewall configurations of said network |
| 16:03.59 | gruvfunk | something is blocking |
| 16:08.42 | oquidave_ | gruvfunk: what is the quickest way to find out if a port is open or not on a box? |
| 16:09.56 | Aut0ExeC | oquidave_: netstat |
| 16:10.28 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 16:10.29 | oquidave_ | Aut0ExeC: what do i expect to see when i netstat |
| 16:10.39 | Aut0ExeC | which ports are open lol |
| 16:10.46 | Aut0ExeC | just kidding |
| 16:10.50 | Aut0ExeC | the status of active ports |
| 16:10.53 | Aut0ExeC | which are listening |
| 16:11.01 | Aut0ExeC | currently active |
| 16:11.01 | Aut0ExeC | etc etc |
| 16:11.07 | gruvfunk | or use netcat |
| 16:11.10 | Kobaz | netstat -lntup |
| 16:13.07 | Aut0ExeC | oquidave_: if its a remote box then use netcat to do a scan.. or nmap |
| 16:13.21 | gruvfunk | nods |
| 16:13.25 | Aut0ExeC | :) |
| 16:15.31 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 16:19.03 | oquidave_ | Aut0ExeC: i have to scan a remote box or router frm my box |
| 16:19.19 | Aut0ExeC | then nmap or netcat |
| 16:19.57 | Aut0ExeC | more than likely u have netcat already in ur arsenal |
| 16:20.16 | oquidave_ | Aut0ExeC: when i netcat from my box, i get a prompt Cmd line: |
| 16:20.41 | Aut0ExeC | no |
| 16:20.42 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:21.05 | Aut0ExeC | use options with it... nc -v -w 2 -z target [portrange] |
| 16:22.50 | Kobaz | pets the netcat |
| 16:22.55 | Aut0ExeC | lol |
| 16:23.07 | oquidave_ | Aut0ExeC: i only need to find if 5060 is open on a remote box 10.1.16.72...can u provide a sample syntax for that...thanks |
| 16:23.29 | Aut0ExeC | <PROTECTED> |
| 16:23.42 | Aut0ExeC | put ur IP as target |
| 16:24.05 | Aut0ExeC | remote IP |
| 16:25.46 | oquidave_ | Aut0ExeC: somethg like.....netcat -v -w 2 -z 172.18.0.33 5060 |
| 16:26.03 | Aut0ExeC | yes |
| 16:26.11 | Aut0ExeC | that should work |
| 16:27.26 | oquidave_ | Aut0ExeC: okay i get ......172.18.0.33: inverse host lookup failed: ......(UNKNOWN) [172.18.0.33] 5060 (sip) : Connection refused |
| 16:27.36 | Aut0ExeC | then its closed |
| 16:27.40 | Aut0ExeC | :) |
| 16:29.50 | oquidave_ | Aut0ExeC: Thanks, the irony is that is call up some support guy at the remote site and they tell me that the port is open!!!!! |
| 16:30.23 | Aut0ExeC | yeah you should call again |
| 16:31.42 | Aut0ExeC | 172 is internal address tho right? |
| 16:31.47 | Aut0ExeC | u sure u have the right IP? |
| 16:33.19 | oquidave_ | Aut0ExeC: 172.18.0.33 is the remote box am scanning i thought from my box 172.18.0.34 |
| 16:33.29 | Aut0ExeC | oh ok |
| 16:33.47 | oquidave_ | Aut0ExeC: we are great, right? |
| 16:33.55 | Aut0ExeC | yes |
| 16:39.53 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 16:40.17 | oquidave_ | Aut0ExeC: have a good night |
| 16:40.36 | oquidave_ | gruvfunk: have a good night too |
| 16:41.37 | oquidave_ | kaldemar: thanks too, have a good night, am out |
| 16:43.34 | oquidave_ | and to asterisk irc channel members God bless, been very helpful |
| 16:45.17 | Eitan | im scared to ask this question. so i have multiple sip trunks coming in from say differnt providers. Need to get them to my server with 4 NIC's what happens when i run out of nics, would i use a sip gateway to combine them or somethign? or do i need more nics? |
| 16:45.48 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 16:45.57 | leifmadsen | why can't each provider come in on the same nic? |
| 16:46.07 | leifmadsen | they are not T1s... you can have many provides on 1 single interface |
| 16:46.16 | leifmadsen | you don't need more than 1 nic |
| 16:47.26 | leifmadsen | you don't need a physical interface per provider |
| 16:47.28 | Eitan | each provider is comming in on a differnt line |
| 16:47.42 | leifmadsen | define "different line" |
| 16:47.43 | Eitan | on a different interface, because we have to go with t1's due to our location |
| 16:48.04 | Eitan | we cant get a circuit that can handle more than 1.5mb |
| 16:48.10 | leifmadsen | so? |
| 16:48.13 | leifmadsen | it's just a data connection |
| 16:48.15 | sxpert | Eitan: sure you can... pull a fiber ;) |
| 16:48.17 | leifmadsen | put them all into a router |
| 16:48.24 | Eitan | that was my question |
| 16:48.40 | Eitan | so just all come into a router |
| 16:48.47 | Eitan | then form there into 1 nic |
| 16:48.49 | Eitan | thank you |
| 16:49.04 | Eitan | router has to support SIP correct? |
| 16:57.15 | Qwell | no |
| 16:57.21 | Qwell | router has to support Internets. |
| 17:00.26 | Aut0ExeC | isnt sip just a protocol |
| 17:00.30 | Aut0ExeC | router doesnt care right? |
| 17:01.02 | Eitan | ok, just making sure |
| 17:01.36 | Eitan | just trying to figure out how that works, cause looking online its called mutilink ppp |
| 17:01.41 | Eitan | doesnt work with different carriers. |
| 17:02.48 | Aut0ExeC | sounds like ur worried about bandwith |
| 17:03.02 | Aut0ExeC | having all calls going over 1 nic |
| 17:03.10 | Aut0ExeC | vs 4 or more |
| 17:03.42 | Aut0ExeC | i would think even if u have more than 1 trunk... still boils down to how many calls are connected |
| 17:04.07 | Aut0ExeC | at the same time that is |
| 17:04.26 | *** join/#asterisk jkroon (~jkroon@41.0.41.26) |
| 17:06.34 | *** join/#asterisk JerJer (~jj@asterisk/original-h323-guy/JerJer) |
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| 17:15.26 | *** part/#asterisk clintc (~clintc@n128-227-87-124.xlate.ufl.edu) |
| 17:16.09 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 17:20.48 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 17:22.12 | jkroon | what are the possible reasons, that after receiving a SIP 401 response from a server, with a secret and authuser set that asterisk won't attempt to authenticate? |
| 17:23.33 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:25.23 | jkroon | would the receipt of a 100 Trying before the 401 Unauthorized cause the problem? asterisk 1.6.2.16.3 ?!? |
| 17:26.24 | *** join/#asterisk dan__t (~dant@72.233.89.95) |
| 17:26.28 | dan__t | Good morning. |
| 17:27.52 | dan__t | Working with a few things lately such as Monitor and MixMonitor applications. They work fine. I'd like to eventually store the products of these applications in specific directories, the path of which I need to eventually pull from a database. Anyone have any good examples on how to interactively query a (in this case, a MySQL) database for external information for an implementation such as this? |
| 17:28.12 | jkroon | core show function MYSQL |
| 17:28.17 | dan__t | My initial research tells me that I'm best off going the OBDC route. I'm fine with that... just not done this before. |
| 17:29.02 | dan__t | I'll take a look. |
| 17:29.35 | dan__t | Kind of related I guess, is "realtime" the storage of configuration in a SQL database? |
| 17:30.04 | dan__t | I understand static and dynamic configuration - do those ever get mixed, or is it strictly realtime and non-realtime |
| 17:34.38 | lpmusic | dan__t: you can mix them, I have both realtime and static dialplans and realtime and static sip devices, haven't seen any problems |
| 17:39.30 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 17:40.34 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 17:41.53 | *** join/#asterisk imcdona (imcdona@2001:470:e8f1:1:4462:7036:c469:abee) |
| 17:42.18 | dan__t | Yea that's kind of what I was getting at. |
| 17:42.50 | dan__t | With regards to macros.... i'm thinking macros would be statuc that use some sort of dynamic data? This is all very new to me so I'm sure I'll eventually figure it out, but I would like to start off on the right path. |
| 17:43.13 | dan__t | Like, nobody stores a macro in a database right - why bother? just make the macro retrieve whatever needed data from the database? |
| 17:43.42 | leifmadsen | right |
| 17:43.45 | leifmadsen | that's what func_odbc is for |
| 17:43.56 | leifmadsen | a combination of GoSub() and func_odbc is great |
| 17:43.56 | dan__t | got it |
| 17:44.06 | dan__t | Excellent, that's exactly what I was hoping to hear. |
| 17:44.18 | leifmadsen | check the Database Integration chapter in Asterisk: The Definitive Guide for a good example of how to do that. The example given is hot-desking |
| 17:44.31 | leifmadsen | Keep the dialplan static, make the data dynamic |
| 17:44.33 | dan__t | I do remember that. I'll get refreshed later this evening. |
| 17:44.34 | leifmadsen | that's my approach |
| 17:44.38 | dan__t | Understood. |
| 17:47.52 | *** join/#asterisk scubes13 (~scubes13@66.64.147.100.nw.nuvox.net) |
| 17:49.41 | jkroon | hi guys, on a B410P when trying to set a hwgain I get an innapropriate ioctl for device, eg, dahdi set hwgain tx 94 -4 results in Unable to set the hardware gain for channel 94: Inappropriate ioctl for device |
| 17:49.47 | scubes13 | I have a pbx behind a pfsense firewall... i have multiple users outside of the firewall that are connecting back in... my connections seem a bit on and off... anyone have a similar setup or know of somewhere I can get some details on settings I need in place? as far as I know, I have port forwarded all the ports I need to.... |
| 17:50.01 | jkroon | and doing a swgain just "fails" ... any ideas what I can do about this? |
| 17:50.22 | jkroon | I really, really need to drop the tx gains ... |
| 17:50.38 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:50.38 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:51.20 | jkroon | also, it sounds like I've got a ground-loop (audio mix terminology) on my rx - replaced cables, and also tested with a different PABX system, problem when linking with ast, not with PABX. |
| 17:51.32 | imcdona | I've been working with the Polycom engineers on an issue related to the way polycom's handle directed call pickup on a BLF line appearance. Polycom is saying that Asterisk is not sending properly formatted SIP packets. See http://pastebin.com/Ga81qNEA Is this an Asterisk bug or a bug in the latest polycom firmware? |
| 17:53.12 | imcdona | The SIP packet they show in that pastebin is supposedly from Asterisk 1.4 |
| 17:54.49 | *** part/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:54.53 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:54.53 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:56.01 | malcolmd | hard to say; i'm not intimately familiar with the formatting - most anyone's going to go have to look to reference documentation to see if there's actually an error, of if polycom just wants the message to come a certain way to make them happy. |
| 17:56.17 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 17:57.09 | imcdona | RFC 4235 section 4.1 I beleive |
| 17:59.50 | *** join/#asterisk andyoutside (6161c2a5@gateway/web/freenode/ip.97.97.194.165) |
| 18:00.08 | andyoutside | ok does this tell anyone usefull info http://pastebin.com/G9nQHU4k |
| 18:00.20 | andyoutside | asterisk is not acking as it should |
| 18:00.37 | andyoutside | http://pastebin.com/G9nQHU4k |
| 18:00.39 | andyoutside | er |
| 18:00.47 | andyoutside | <PROTECTED> |
| 18:03.50 | gruvfunk | andyoutside what is not working, what are you experiencing? |
| 18:03.55 | jkroon | what is the effect of the LBO values in dahdi/system.conf? |
| 18:06.41 | *** join/#asterisk joshaidan (~brianj@host-216-26-197-55.tbaytel.net) |
| 18:06.53 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 18:08.27 | andyoutside | phones stop working |
| 18:08.35 | andyoutside | it acts if there is no activity |
| 18:08.45 | andyoutside | and data does not update wen I look in asterisk -r |
| 18:09.05 | andyoutside | calls coming in get a busy signal |
| 18:09.16 | gruvfunk | sounds like you have system trouble |
| 18:09.22 | andyoutside | bingo |
| 18:09.40 | andyoutside | but I cannot seem to find the cause |
| 18:09.46 | gruvfunk | too many variables to troubleshoot, did it ever work, when/why did it stop working? |
| 18:09.55 | andyoutside | I updated astrisk to 1.8.3.3 |
| 18:10.04 | andyoutside | from 1.8.0 |
| 18:10.35 | andyoutside | I had upgraded to centos 5.6 from 5.5 but that was before the astrisk update |
| 18:10.48 | andyoutside | but the system had not been restarted. |
| 18:11.11 | leifmadsen | back out so many changes -- go back to the exact version you were just running |
| 18:11.17 | leifmadsen | test 1 change at a time |
| 18:11.28 | *** join/#asterisk lupestro (~chatzilla@c-75-68-76-84.hsd1.nh.comcast.net) |
| 18:11.29 | andyoutside | ok what is the best way to do that? |
| 18:11.51 | leifmadsen | I don't understand the question... |
| 18:11.54 | leifmadsen | reinstall 1.8.0 |
| 18:12.00 | leifmadsen | test, does that fix the problem? |
| 18:12.05 | leifmadsen | if no, then it was an OS change likely |
| 18:12.18 | leifmadsen | if yes, then find out what change causes your break |
| 18:12.21 | andyoutside | also It worked when I updated to 1.8.3.3 and restared asterisk it workked |
| 18:12.42 | leifmadsen | too many variables are now present to determine what happened |
| 18:12.54 | andyoutside | but when the day came and we had havey traffic it crashed |
| 18:13.26 | andyoutside | So I rebooted it and then had to fix a dahdi issue |
| 18:13.36 | andyoutside | sense then it has been randomly crashing |
| 18:15.36 | carrar | perhaps hire a consultant |
| 18:16.42 | andyoutside | We may in the end |
| 18:18.23 | gruvfunk | sage advice: don't upgrade unless you need to, and then if you do, don't upgrade a production system - migrate to an upgraded platform |
| 18:19.58 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 18:20.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:21.53 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 18:23.01 | *** join/#asterisk cusco (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 18:23.03 | cusco | hail |
| 18:23.21 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 18:23.31 | cusco | there is a variable that I may have before using Queue() that gives me the estimate hold time for a queue, right? |
| 18:24.16 | cusco | foundit QUEUEHOLDTIME |
| 18:24.45 | cusco | ow but that is after Queue() |
| 18:25.27 | gruvfunk | cusco were you working on QUEUEPRIO yesterday? |
| 18:26.26 | cusco | yes |
| 18:26.32 | cusco | turns out I don't need it anymore |
| 18:26.57 | cusco | can I know the estimated hold time prior to joining a queue ?? |
| 18:27.20 | anonymouz666 | I don't think you can, that's calculated by app_queue |
| 18:27.32 | gruvfunk | unless you used queueprio again |
| 18:27.47 | cusco | didn't think about that |
| 18:28.00 | gruvfunk | maybe, i'm thinking (join the queue, pull out with prio, present queueholdtime, rejoin queue with prio) |
| 18:28.05 | cusco | so call goes in queue with a timeout of 1 sec, with lower prio |
| 18:28.30 | cusco | thats akward but perhaps it works! |
| 18:29.24 | gruvfunk | perhaps not the right way, but you did say you wanted to present it before joining the queue |
| 18:29.57 | cusco | yes |
| 18:30.07 | *** join/#asterisk jenna (~jjones@unaffiliated/jenna) |
| 18:30.11 | cusco | I must have a blank moh file too |
| 18:30.12 | cusco | lol |
| 18:30.14 | cusco | ok |
| 18:30.17 | cusco | I'll try that |
| 18:30.31 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 18:30.40 | cusco | but... |
| 18:30.47 | cusco | how do I read that var in the dialplan? |
| 18:30.52 | McBoingbo | How do I tell what version of SIP Firmware my Asterisk supports? |
| 18:31.06 | cusco | it works after queue () |
| 18:31.08 | cusco | probably |
| 18:31.51 | anonymouz666 | McBoingbo: Asterisk supports RFC3261 that defines the SIP Protocol. SIP version 2.0. |
| 18:32.46 | andyoutside | gruvfunk: I updated to for a bug that was fixed |
| 18:33.44 | McBoingbo | Im confused...so SIP firmware, how do I determine what Asterisk 1.2 lets say, will support up to for Polycom phones? |
| 18:37.56 | cusco | gruvfunk: my problem is that a queue() will always have a event in queue_log (real time) with ENTERQUEUE |
| 18:38.07 | cusco | can I tell somehow asterisk to not inster that? |
| 18:41.19 | *** join/#asterisk joshaidan (~brianj@host-216-26-197-55.tbaytel.net) |
| 18:43.54 | jenna | A person asked me " He has good a fiber cable running from telco to his data center, serving the 2 E1s on that. How can he hook up that fiber thingy (with 2 E1s on it) to setup a asterisk based PABX." I have no answer to. I mean I think this setup requires media converters. but not sure. Any ideas/suggestions ? |
| 18:48.09 | *** join/#asterisk the_nation (~bla@adsl-75-19-113-197.dsl.yntwoh.sbcglobal.net) |
| 18:49.18 | TuNoD_68 | Jenna: You only need the server cards for the E1's. They might be brought up from CO on fiber, but ultimatly they are terminated into dmark which would should have a standard copper E1 port |
| 18:50.39 | the_nation | I have a question. I have google voice forwareded to a number of voip lines and of course gtalk. I would like to have them all registered in asterisk for redundancy, but a little confused as to how the calls will be handled. All trunks will be ringing at pretty much the same time. How should I devise a proper handling of this situation? |
| 18:51.07 | jenna | TuNoD_68, u mean the media converters e.g http://www.omnitron-systems.com/products/8700_t1e1_iconverter.php |
| 18:52.08 | jenna | TuNoD_68, wouldn't it be cheaper to ask the telco to send in the E1 copper interfaces with RJ-45 crimped rather than them bringing in the fiber interfaces ? |
| 18:53.08 | TuNoD_68 | Jenna: My t1's are brought in on fiber.... then they are terminated at the dmark. The dmark had standard 8-pin connector for my equipment |
| 18:53.23 | jenna | TuNoD_68, but another issues what if one asks the telco to give them more E1 (i.e. upgrade the channels) . |
| 18:53.48 | TuNoD_68 | interface is the same.... channels are logical |
| 18:55.10 | jenna | TuNoD_68, hmm. So what you are saying is they would just logically allocate more E1 (from the telco NOC) & I would have to use dmark to expand them & hook them into my copper E1 digium cards ? |
| 18:57.48 | TuNoD_68 | Jenna: no. T1 = 24 total channels E1 = 30 channels... the circuit is t1 or e1. The number of channels in use are the (for lack of a more laman term) the lines. You would config your equipment (i.e. PBX) for the lines on the circuit. |
| 19:00.12 | *** join/#asterisk artista_frustrad (~fernando@200.247.43.2) |
| 19:00.42 | artista_frustrad | I've had an asterisk 1.6 runing for months without complications |
| 19:00.44 | *** join/#asterisk ferdna (~yup@67.10.220.35) |
| 19:01.21 | artista_frustrad | out of the blue it is spitting timed out errors when registering to my sip provider |
| 19:02.14 | jenna | TuNoD_68, I understand the T1/E1 difference. The setup Im studying is this. 2 E1s have been allocated to a site & been given(relayed to datacenter) thru a single fiber. From what I seen on digium/sangoma site there are only copper cards with RJ-45 interfaces available. So I guess ones needs to converter/translate that single fiber interface into 2 copper E1s to hook them into the digium (dual)span card to make use of it. right ? |
| 19:02.43 | Qwell | artista_frustrad: Is your SIP provider up? |
| 19:03.01 | artista_frustrad | vono - a brasilian cheap provider |
| 19:03.07 | Qwell | so, probably not. |
| 19:04.13 | artista_frustrad | Qwell, the provider is up |
| 19:04.40 | anonymouz666 | wait |
| 19:04.43 | artista_frustrad | using vono's softphone it connects |
| 19:04.45 | anonymouz666 | I know this provider |
| 19:05.11 | *** join/#asterisk ixyd_ (~denzs@ip-78-94-178-44.unitymediagroup.de) |
| 19:06.17 | TuNoD_68 | Jenna: the t1/e1 would have to be terminated at a dmark..... if you extend from the dmark over fiber, then yes you would need media converters on both ends of the fiber run. if the fiber is coming straight from the CO then they should provide a dmark at the end of the fiber. |
| 19:06.32 | anonymouz666 | ~newbook |
| 19:06.32 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 19:06.38 | anonymouz666 | ~thebook |
| 19:06.38 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 19:06.53 | anonymouz666 | ~cookbook |
| 19:06.53 | infobot | well, cookbook is "Over 1,500 time-saving recipes and hints for busy modern computer users" at http://dsl.org/cookbook/ |
| 19:07.23 | anonymouz666 | where's the cookbook link? |
| 19:07.35 | Aut0ExeC | ~cookbook |
| 19:07.35 | infobot | cookbook is probably "Over 1,500 time-saving recipes and hints for busy modern computer users" at http://dsl.org/cookbook/ |
| 19:07.45 | leifmadsen | ~asteriskcookbook |
| 19:07.45 | infobot | The Asterisk Cookbook is available for purchase as an eBook from O'Reilly at http://oreilly.com/catalog/0636920018551 or via Amazon. The book is available under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and can be read online at http://ofps.oreilly.com/titles/9781449303822/ |
| 19:08.06 | Aut0ExeC | why is linux cookbook in here? |
| 19:08.12 | Aut0ExeC | wow |
| 19:08.42 | tzanger | because much like linux itself it's only good for starting fires |
| 19:08.53 | Aut0ExeC | ahh ok |
| 19:08.59 | tzanger | lack of proper power/thermal management is why linux fails it |
| 19:09.19 | jenna | TuNoD_68, thanx. yup. the CO should provide the dmark. the my understanding is that they have just provided a fiber switch (weird) I guess I need to read more on the scenario. |
| 19:09.28 | Qwell | tzanger: that reminds me of my favorite marketing term |
| 19:09.28 | coppice | linux power management is very good these days |
| 19:09.40 | tzanger | coppice: shhh, you're ruining my troll |
| 19:09.40 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 19:09.43 | Qwell | tzanger: "variable thermal control" read: it has a fan. |
| 19:09.51 | *** join/#asterisk dichis (~dichis@mail.topfactoring.ro) |
| 19:10.04 | coppice | well, its about as good as windows, however good or bad that may be |
| 19:10.33 | Aut0ExeC | please dont say windows here right after linux |
| 19:10.58 | dichis | Hello guys, anybody cares to help somebody with some ISDN questions? |
| 19:11.01 | jaytee | I'm running Windurs Seben |
| 19:11.06 | Qwell | ~ask |
| 19:11.07 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:11.16 | Aut0ExeC | jaytee: nice.. asterisk running on it? |
| 19:11.26 | cusco | thermal info available in /proc/acpi/thermal_zone/* |
| 19:12.42 | dichis | I have a ISDN-PRI and I receive a ALERTING with progress inband, and Asterisk generates RINGING and does not play the inband message from the provider |
| 19:13.19 | jaytee | Aut0ExeC, nope. I only run Asterisk on CentOS 5.x or RHEL 5.x cuz I'm a linux bigot |
| 19:14.45 | *** join/#asterisk movozo (~Mo@p57AA04CE.dip0.t-ipconnect.de) |
| 19:16.48 | dichis | so... the question: what can I do to get the inband progress after an ALERTING ISDN message? |
| 19:23.27 | jenna | TuNoD_68, thanx for the hints/ideas. |
| 19:24.12 | *** join/#asterisk Gary_B (u610@gateway/web/irccloud.com/x-itzsihynmgmqaetl) |
| 19:24.31 | cusco | so... how can I know estimated hold time for a queue prior to joining it? |
| 19:24.34 | ixyd_ | hi guys, can anyone show me an example for using Queue()'s gosub parameter? i tried macroname(arg1,arg2) and ¯oname(arg1, arg2) for gosub. macroname is defined with "macro macroname(arg1, arg2)..." but iam getting "No such label 'macroname' in extension 's' in context 'app_queue_gosub_virtual_context'" when the queue call gets connected :( |
| 19:25.20 | leifmadsen | GoSub != Macro |
| 19:26.14 | leifmadsen | it is likely trying to run a subroutine via GoSub() at the context name you specify, extension 's', priority '1' |
| 19:26.50 | leifmadsen | so if you name your context [subFoobar], then you should set the option in the Queue() application to be 'subFoobar' and [subFoobar] should contain an extension named 's' |
| 19:27.21 | tzanger | leifmadsen: are you still crazy for subroutines? I haven't ever used them yet |
| 19:27.25 | tzanger | (subs over macros) |
| 19:27.59 | leifmadsen | tzanger: yes i use subroutines most of thet ime |
| 19:28.02 | leifmadsen | the time* |
| 19:28.32 | leifmadsen | no chance of crashing asterisk with nested subroutined because of lack of stack space |
| 19:28.47 | ixyd_ | i will try that thx! |
| 19:29.06 | tzanger | heh you know how long I've been using asterisk... I've never once had a dialplan so complex that it was in any kind of danger of doing that |
| 19:29.14 | leifmadsen | :) |
| 19:29.16 | leifmadsen | welcome to my world |
| 19:29.47 | ixyd_ | hm and what about the arguments? :) is there no need to declare them like macros? :) |
| 19:30.32 | ixyd_ | should i mention that iam using ael? ;) |
| 19:31.22 | *** part/#asterisk Gary_B (u610@gateway/web/irccloud.com/x-itzsihynmgmqaetl) |
| 19:33.53 | leifmadsen | I don't think you get to pass arguments with that functionality |
| 19:34.19 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 19:34.19 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 19:34.32 | ixyd_ | i hoped you would not say that ;) |
| 19:34.49 | leifmadsen | because it executes on the CALLING parties channel, you can just use channel variables you set prior to calling Queue() |
| 19:36.22 | jenna | leifmadsen, great job on that book. its a smooth read. |
| 19:36.30 | artista_frustrad | alguem do vono ai? |
| 19:38.24 | leifmadsen | jenna: thanks! It wasn't just me :) |
| 19:38.32 | leifmadsen | russellb helped a bunch too :) |
| 19:38.49 | russellb | i did nothing |
| 19:39.00 | jenna | leifmadsen, yeah well he is not around. |
| 19:39.05 | jenna | ooops |
| 19:39.08 | leifmadsen | burn |
| 19:39.09 | russellb | :-) |
| 19:39.10 | leifmadsen | he does that |
| 19:39.26 | russellb | leifmadsen: i'm watching you! |
| 19:39.49 | jenna | russellb, you too russ. excellent book. swell job |
| 19:39.50 | leifmadsen | weird... |
| 19:40.39 | gruvfunk | artista_frustrad quem em a vono? provider? |
| 19:42.34 | *** join/#asterisk b0ot (~Jinxed---@147.177.56.244) |
| 19:42.50 | b0ot | Anyway have any trouble using the polycom 335 with dtmf-relay? |
| 19:43.03 | mickecarlsson | thanks seanbright with all his heart. Issue 19090 fixed. |
| 19:43.28 | seanbright | nice |
| 19:43.47 | seanbright | it may not be fixed though |
| 19:43.48 | seanbright | :-) |
| 19:43.49 | mickecarlsson | It also fixed an issue with appleraisin trunk |
| 19:43.58 | ixyd_ | thank you so much leifmadsen! :) |
| 19:44.11 | mickecarlsson | It was actually when I tested appleraisin trunk that I found it |
| 19:44.23 | russellb | jenna: thanks :-) |
| 19:44.31 | seanbright | not familiar with appleraisin |
| 19:44.35 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 19:44.40 | mickecarlsson | Well, oej is :-) |
| 19:44.42 | russellb | wonders what happened to the days when people named branches after the actual code they contain |
| 19:44.45 | leifmadsen | it's an oej branch |
| 19:44.53 | seanbright | oh |
| 19:44.56 | seanbright | well in the case |
| 19:44.57 | seanbright | revers |
| 19:44.58 | leifmadsen | pretty sure it's full of fruit |
| 19:45.00 | seanbright | reverts* |
| 19:45.02 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 19:45.07 | seanbright | damn... funnier if i pulled it off the first time |
| 19:45.09 | mickecarlsson | So, now I can go on with FreePBX and realtimne voicemail for that release |
| 19:45.18 | seanbright | well, hold on |
| 19:45.32 | seanbright | i want to get it reviewed |
| 19:45.46 | seanbright | i want to make sure that modifying MYSQL_ROW in place is OK |
| 19:45.46 | mickecarlsson | I am on the FreePBX Development Team |
| 19:46.39 | mickecarlsson | OK |
| 19:46.42 | seanbright | i am not |
| 19:47.00 | mickecarlsson | Oh, OK was a reply to yor comment :-) |
| 19:47.06 | seanbright | ah |
| 19:47.22 | seanbright | my whole point is: there is no telling what that patch might do |
| 19:47.33 | russellb | eat babies? |
| 19:47.33 | seanbright | so before we celebrate let me get it reviewed |
| 19:47.40 | seanbright | that'd be sweet |
| 19:50.18 | seanbright | meh... i might just commit it |
| 19:50.35 | mickecarlsson | I have done some more tesing |
| 19:50.39 | mickecarlsson | testing that is |
| 19:51.16 | mickecarlsson | On the realtime astdb that I test, I added more than 1000 users, that added over 6000 entries in the mysql astdb database |
| 19:51.34 | mickecarlsson | I then updated a lot of entries, no issues detected |
| 19:53.32 | artista_frustrad | gruvfunk, yes.. I'm desperate |
| 19:54.05 | gruvfunk | artista_frustrad I dont' know vono, but what's your issue perhaps we can help |
| 19:54.08 | artista_frustrad | have no sip and my cel just died |
| 19:55.00 | artista_frustrad | gruvfunk, [2011-05-05 16:54:19] NOTICE[1695]: chan_sip.c:12357 sip_reg_timeout: -- Registration for 'myaccount@vono.net.br' timed out, trying again (Attempt #136) |
| 19:55.30 | artista_frustrad | second time this week both my accounts just stop registering |
| 19:55.45 | artista_frustrad | but softphones are able to register |
| 19:57.45 | gruvfunk | are softphones on a different network? |
| 19:58.23 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:59.58 | artista_frustrad | gruvfunk, no.. but they are not the ones I use to connect to asterisk.. they are a custom software from the provider |
| 20:00.16 | gruvfunk | :) |
| 20:00.45 | gruvfunk | so, does vono support Asterisk? or do you know others using it that way? maybe this provider "locks it down"? |
| 20:01.22 | *** join/#asterisk \DSAFEW\ (~DSAFEW_@ip72-208-176-219.ph.ph.cox.net) |
| 20:01.27 | artista_frustrad | gruvfunk, they support it, put vary vague instructions on their site on how to configure vono + asterisk |
| 20:01.48 | artista_frustrad | and it has been working for about a year |
| 20:02.32 | seanbright | mickecarlsson: patch is being changed |
| 20:02.35 | seanbright | mickecarlsson: fyi |
| 20:02.57 | mickecarlsson | ok on the patch change |
| 20:03.55 | gruvfunk | artista_frustrad have you changed something recently that broke it? or perhaps vono has changed something on their side |
| 20:06.14 | *** join/#asterisk jetlag (~jetlag@pool-173-61-207-173.cmdnnj.east.verizon.net) |
| 20:07.21 | artista_frustrad | gruvfunk, I've made a few changes but It did not break back then.. I'm suspecting it is a network issue but I'm not able to pinpoint it |
| 20:08.06 | artista_frustrad | and to make matters worse I'm out of town .. I'm connecting via ssh |
| 20:10.13 | *** join/#asterisk devil_evoxxx (~d3v1l@host137-92-dynamic.0-79-r.retail.telecomitalia.it) |
| 20:10.17 | devil_evoxxx | hi all guy |
| 20:11.13 | andyoutside | run wire shark and seeif you can see anything odd |
| 20:11.35 | devil_evoxxx | i have a problem with Music on Hold..it start playng but i can't listen the song |
| 20:11.39 | devil_evoxxx | it say -- Started music on hold, class 'nocturne', on SIP/portato3-00000006 -- SIP/12-00000007 is ringing |
| 20:11.43 | devil_evoxxx | any tips? |
| 20:14.25 | cusco | hi... |
| 20:14.29 | devil_evoxxx | P.s. if i answer the phone i can hear |
| 20:14.33 | cusco | I can run a macro when using queue() right ? |
| 20:14.34 | devil_evoxxx | hi cusco |
| 20:15.00 | cusco | devil_evoxxx: did you answer() the call first? |
| 20:15.18 | cusco | gruvfunk: remember my queue-hold-time question? |
| 20:15.48 | cusco | I could run a macro while joining a queue and maybe I could get hold of the queueholdtime variable in there, right? |
| 20:16.12 | cusco | can I then in the macro, after evaluating that var, remove the member from queue ? |
| 20:18.45 | mickecarlsson | seanbright: I am heading for the sack (local time in Sweden is 10:18 pm) just ping me here or over at #freepbx-dev if you need me. Please update ticket with new pathc for testing. |
| 20:20.06 | gruvfunk | cusco: hoping others can help answer you |
| 20:20.12 | seanbright | mickecarlsson: good times |
| 20:22.22 | cusco | ok I gotta go now |
| 20:22.32 | cusco | I will have a look at this macro possibilittie later |
| 20:22.34 | cusco | thanks |
| 20:24.45 | *** join/#asterisk kuj (~kuj@c-24-9-82-245.hsd1.co.comcast.net) |
| 20:37.28 | *** join/#asterisk mcargile (~mikec@office2.vicidial.com) |
| 20:39.30 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 20:47.49 | *** join/#asterisk fantasma (~root@204.152.202.19) |
| 20:47.53 | fantasma | ! book |
| 20:47.57 | fantasma | !book |
| 20:48.02 | fantasma | i hear there's a book here? |
| 20:48.24 | Deeewayne | ~thebook |
| 20:48.24 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 20:48.46 | mcargile | damn beat me to it |
| 20:48.59 | fantasma | oh that's a pay book |
| 20:49.05 | fantasma | :( |
| 20:49.12 | Deeewayne | ~book |
| 20:49.12 | infobot | For more information about the Asterisk book, see ~thebook |
| 20:49.51 | fantasma | anybody down for spoonfeeding information? |
| 20:49.57 | mcargile | http://ofps.oreilly.com/titles/9780596517342/ |
| 20:50.04 | mcargile | that is the whole book |
| 20:50.08 | mcargile | its creative commons |
| 20:50.10 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 20:50.14 | Deeewayne | ~roulette |
| 20:50.14 | infobot | ACTION watches deeewayne pull the trigger: Click! |
| 20:50.28 | fantasma | infobot? |
| 20:50.33 | fantasma | infobot, x is y |
| 20:50.35 | fantasma | infobot x |
| 20:50.40 | fantasma | that's not a real infobot |
| 20:50.49 | mcargile | it's special |
| 20:51.15 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 20:51.46 | fantasma | be zippy |
| 20:51.50 | fantasma | infobot be zippy |
| 20:51.50 | infobot | yow are you this evening? |
| 20:52.06 | fantasma | that thing is hacked up the ass |
| 20:52.11 | fantasma | infobot weather KJFK |
| 20:52.23 | fantasma | not bad |
| 20:52.38 | ChannelZ | get a room |
| 20:52.47 | fantasma | ok |
| 20:52.54 | fantasma | but first can someone spoonfeed me asterisk info? |
| 20:53.16 | ChannelZ | Oh, is that what channel we're in? :P |
| 20:53.28 | fantasma | i need to set up a # to receive calls and play a wav file |
| 20:54.23 | ChannelZ | Is that a question or a job offer or.. |
| 20:55.20 | fantasma | look i have no idea how to get started with this, the FAQ's available aren't very helpful |
| 20:55.38 | fantasma | if you wanna give me RTFM fine but at least point me to a good one |
| 20:55.45 | ChannelZ | well, it's a large question and you've given no info |
| 20:55.51 | fantasma | what info you want |
| 20:55.57 | fantasma | i have nothing except a linux box in some data cneter |
| 20:55.59 | ChannelZ | Do you have hardware? Are you going to do this VoIP through an ITSP? |
| 20:56.00 | fantasma | and asterisk compiled on it |
| 20:56.27 | fantasma | my hardware is a linux box |
| 20:56.46 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
| 20:57.06 | ChannelZ | I meant telephony hardware vs VoIP |
| 20:57.23 | mcargile | fantasma: I did point you to a good manual. http://ofps.oreilly.com/titles/9780596517342/ It is the complete Asterisk: The Definitive Guide, 3rd Edition book. The book itself is released under the creative commons |
| 20:57.30 | fantasma | yes but it costs $50 |
| 20:57.42 | fantasma | oh |
| 20:57.43 | fantasma | no it doesn |
| 20:57.44 | fantasma | 't |
| 20:57.45 | fantasma | lol |
| 21:00.29 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 21:01.46 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 21:03.24 | Jcook_5xData | I have a stupid ? we are running asterisk 1.6, a user reported that the system call them back because they did not leave a voicemail. does this seem plausible? |
| 21:05.56 | ChannelZ | Well you can make * do just about anything, but no that's not any sort of default behavior or something you can even do with a simple setting or anything |
| 21:06.43 | ChannelZ | Are they sure they didn't accidentally hit a button and parked the call, and it timed out and rang them back? |
| 21:08.51 | Jcook_5xData | that what I am thinking |
| 21:08.59 | ChannelZ | or are they analog phones? |
| 21:11.06 | Jcook_5xData | the one that rang back was the on the asterisk side are all polycom |
| 21:11.28 | leifmadsen | did it really ring, or did the voicemail indicator just come on and make a noise? |
| 21:12.10 | *** join/#asterisk csd-199 (~asdf@189.248.49.243) |
| 21:15.20 | csd-199 | Hi. I have an Asterisk box up and running... using CentOS. Now I want to teach my coworkers to make simple changes, for that, I want to use a web administration way, what is a good web administration option for asterisk? |
| 21:15.27 | Jcook_5xData | he said the it call back then the voice mail message was there on answer |
| 21:16.03 | mcargile | csd-199: FreePBX |
| 21:16.52 | csd-199 | ok, but can I use it with an already installed asterisk? |
| 21:18.19 | seraphie | csd-199: not if you have configured anything. FreePBX will overwrite all your config changes. |
| 21:18.38 | leifmadsen | FreePBX builds everything |
| 21:19.26 | leifmadsen | I'm working on a gui in my free time *cough* whose purpose is to provide a function of saving information for asterisk in a database, which can then be accessed from asterisk but does not write any configurations to asterisk directly |
| 21:19.30 | leifmadsen | there may be others |
| 21:19.48 | leifmadsen | the Asterisk GUI is much less invasive in that regard than FreePBX |
| 21:19.48 | Jcook_5xData | ChannelZ, I have had this thing happen to me in the passed I think it more the analog phone does not properly destroy the channel. But that just a guess |
| 21:20.40 | seraphie | Yeah, Asterisk GUI at least won't blow away the configuration you've already put in place, but it won't necessarily reflect all the configuration you've done, either, and let you modify it. |
| 21:23.26 | Jcook_5xData | +1 for asterisk-gui Nice user interface for managers but, still allow you to hand code and not worry about being over written like freepbx or the like |
| 21:23.53 | seraphie | and it actually supports 1.8 now. |
| 21:24.08 | mcargile | So I am trying to come up with a way to mess with the people scanning my asterisk boxes for sip accounts. I do not want to ban them. Instead I want to give them dummy accounts that are in a context that just screws with them. Things like route the call right back to them. |
| 21:26.20 | csd-199 | ok, so asterisk gui keeps the configuration clean, right? |
| 21:26.55 | seraphie | it will not overwrite anything you have done |
| 21:26.57 | seraphie | it will add things. |
| 21:27.16 | seraphie | So it depends on your definition of "clean" :) |
| 21:27.36 | mcargile | it wont be pretty, but it wont be a mess either |
| 21:27.40 | Jcook_5xData | csd-199, It uses all std asterisk configs if that what you mean |
| 21:28.03 | *** part/#asterisk bchia (~chatzilla@nat/digium/x-dnrgkxaznljzgona) |
| 21:28.44 | Jcook_5xData | csd-199, I believe it create just one to keep track for asterisk-gui stuff |
| 21:31.14 | Jcook_5xData | One thing I like about is the file explore it lets you edit the config in the web browser so you do not have to ssh in and vim/nano ect.. |
| 21:38.09 | *** join/#asterisk engrxyz (~puitpyitr@host81-150-217-173.in-addr.btopenworld.com) |
| 21:41.05 | *** part/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 21:41.52 | *** join/#asterisk jetlag (~jetlag@pool-173-61-207-173.cmdnnj.east.verizon.net) |
| 21:42.52 | *** join/#asterisk imcdona (imcdona@2001:470:e8f1:1:4462:7036:c469:abee) |
| 21:45.08 | *** join/#asterisk the_nation (~bla@adsl-75-19-113-197.dsl.yntwoh.sbcglobal.net) |
| 21:50.11 | *** join/#asterisk micols (~ident@rlogin.dk) |
| 22:03.29 | *** part/#asterisk artista_frustrad (~fernando@200.247.43.2) |
| 22:13.26 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 22:15.54 | *** join/#asterisk silvanov (~none@24-176-5-183.dhcp.klmz.mi.charter.com) |
| 22:17.57 | silvanov | having a wierd issue with a pbx system. We've got Aastra phones, a pbxinaflash system, and a linsys ata. The IVR was working properly for some time, but recently (last few days), the answering machine on the ATA is breaking the IVR and asking for a remote access code. It is off, and part of a base station that we want to migrate away from shortly, but not sure how to temporarily fix or patch |
| 22:17.57 | silvanov | this issue. |
| 22:19.45 | silvanov | anyone willing to lend me a hand? |
| 22:21.03 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 22:23.51 | *** join/#asterisk cHarNe2 (~cHarNe2@gateway/shell/bshellz.net/x-ylxvkhuyyxzbpkkn) |
| 22:24.42 | Freeaqingme | silvanov, sounds like you have an issue iwth your ata or answering machine? |
| 22:25.01 | silvanov | yes. |
| 22:26.56 | silvanov | If this isn't the right place for help, maybe someone could point me in the right direction. Been working at this all day :S |
| 22:27.06 | Freeaqingme | silvanov, I'd say it's kinda impossible to help you from here, esp since its no asterisk issue |
| 22:27.13 | Freeaqingme | okay, what kind of access code is it asking for? |
| 22:29.21 | silvanov | the answering machine remote access code (to listen to messages). This wasn't the functionality we had before, and it happens after about 1.5-2 mins in que. |
| 22:29.58 | silvanov | Nothing has changed to my knowledge, and the answering machine is off. Not sure how to keep it from "picking up the line" as it seems. |
| 22:29.59 | Freeaqingme | can it be that your caller id is used to check if you have access to the machine, and that the caller id changed? |
| 22:31.17 | silvanov | Caller ID hasn't changed, and it happens from an inside line or any outside line. |
| 22:32.59 | Freeaqingme | well |
| 22:33.03 | Freeaqingme | something changed I reckon |
| 22:33.57 | silvanov | seems that way, but no setting or config change on either the pbx or ATA has occured. :S |
| 22:34.20 | Freeaqingme | fair enough |
| 22:34.36 | Freeaqingme | perhaps someone else can help you, but I'd say that with the info we've got it's pretty impossible |
| 22:34.47 | Freeaqingme | also, you're aware asterisk has built in voicemail functionality? |
| 22:35.26 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 22:35.41 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:36.15 | silvanov | Yep. We use the voicemail, not the answering machine normally, which is why this is a problem lol. It's bypassing the que and instead of after a period, sending them to VM, the answering machine takes over. |
| 22:48.50 | silvanov | In any case, thanks for trying @freeeaqingme |
| 22:50.33 | *** part/#asterisk csd-199 (~asdf@189.248.49.243) |
| 22:51.27 | Freeaqingme | silvanov, yw |
| 22:51.42 | Freeaqingme | silvanov, also, type 'Fre' (without quotes), and then press tab |
| 22:51.49 | Freeaqingme | welcome to irc ;) |
| 22:58.21 | silvanov | Freeaqingme thanks! |
| 22:59.34 | cHarNe2 | hi guys,i found this guide and it seems nice, but wonder if it is current. http://jeremy-mcnamara.com/asterisk/how-to-configure-asterisk/ |
| 22:59.42 | cHarNe2 | cant find a postdate |
| 23:00.33 | *** join/#asterisk anygivenname (~anygivenn@41.32.28.166) |
| 23:01.10 | anygivenname | need help....Asterisk is running but sip app is unable to connect to it |
| 23:02.50 | cHarNe2 | anygivenname: sip app? |
| 23:03.35 | *** part/#asterisk lupestro (~chatzilla@c-75-68-76-84.hsd1.nh.comcast.net) |
| 23:03.55 | anygivenname | cHarNe2: xlite on PC & Sipdroid on Android |
| 23:03.55 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 23:05.14 | cHarNe2 | anygivenname: ohh, u sure all ports and stuff like that is ok? |
| 23:05.16 | seraphie | cHarNe2: I wouldn't think it's current. It's talking about Asterisk 1.4, which is security-only now, and on the same page is an article about Digium acquiring Switchvox, which happend in 2007. |
| 23:05.32 | cHarNe2 | :S |
| 23:05.42 | seraphie | use Asterisk 1.8. |
| 23:05.56 | cHarNe2 | will the guide work do you think? |
| 23:06.14 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 23:06.20 | anygivenname | cHarNe2: I opened port 5060 to the server's IP & also ports 10000 to 20000 |
| 23:06.49 | seraphie | cHarNe2: http://ofps.oreilly.com/titles/9781449303822/ and http://wiki.asterisk.org |
| 23:07.04 | anygivenname | cHarNe2: is there any .conf files that I need to change ? |
| 23:07.21 | cHarNe2 | anygivenname: i dont know any about asterisk |
| 23:07.42 | cHarNe2 | anygivenname: but im rly godd at networking and it soung like you have a problem there. |
| 23:07.57 | cHarNe2 | seraphie: ok, will check em out |
| 23:08.06 | anygivenname | cHarNe2: pls guide me |
| 23:08.48 | cHarNe2 | anygivenname: you how to scan ports? |
| 23:09.08 | anygivenname | cHarNe2: no |
| 23:09.24 | cHarNe2 | anygivenname: $nmap -sS <IPADRESS> -PN -p5060 |
| 23:09.55 | anygivenname | cHarNe2: ipadress of the server ? |
| 23:09.58 | cHarNe2 | ye |
| 23:10.21 | cHarNe2 | anygivenname: so you need to run that command from another location |
| 23:11.38 | anygivenname | not from on the server ? |
| 23:11.43 | cHarNe2 | no |
| 23:11.56 | cHarNe2 | that will work |
| 23:11.59 | anygivenname | cHarNe2: I only have linux on the server |
| 23:12.18 | cHarNe2 | :S ok |
| 23:12.59 | anygivenname | cHarNe2: I mean I dont have any other linux machine except the server |
| 23:13.20 | cHarNe2 | ye, that gonna be tricky |
| 23:13.38 | cHarNe2 | cus i dont have admin on the host im on right now |
| 23:13.57 | cHarNe2 | gimme a sec and ill boot a virtual |
| 23:14.23 | cHarNe2 | if i can remember the password that is.. |
| 23:16.18 | cHarNe2 | and update it.. 250mb |
| 23:16.52 | cHarNe2 | anygivenname: send me ur public ip of the server in pm |
| 23:18.51 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 23:20.07 | *** join/#asterisk ttpears (~ttpearso@gw.teamgleim.com) |
| 23:21.03 | ttpears | anyone have expierence compiling for i386 on x86_64? |
| 23:21.32 | ttpears | Getting an error everytime I 'make', I'm sure it's something simple |
| 23:22.34 | ttpears | Well, relatively simple anyway, I've specified the --host and --build |
| 23:26.59 | *** join/#asterisk l1nuxman (~l1nuxman@CPE0021296828b2-CM00111ae6f860.cpe.net.cable.rogers.com) |
| 23:27.15 | *** join/#asterisk anygivenname (~anygivenn@41.32.28.166) |
| 23:27.47 | cHarNe2 | well, he's port is open anyway, |
| 23:28.01 | cHarNe2 | anygivenname: ur port is open |
| 23:28.10 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:28.33 | l1nuxman | can someone help me I'm trying to set up voicemail to send an email with the attachment but its a no go yet. I installed msmtp and it works when I do 'msmtp -a default my@email.com < myfile.wav |
| 23:28.54 | anygivenname | cHarNe2: 5060 is opened & routed to server's ip |
| 23:29.11 | cHarNe2 | kk, then u need to check the logs |
| 23:29.19 | l1nuxman | I have my mailbox works except for mail. my extension is '1 => 4321,Household Mailbox,my@email.com' . What should the mailcmd be? |
| 23:29.22 | cHarNe2 | or trace using tcpdumpo |
| 23:29.25 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 23:29.31 | cHarNe2 | and view in wiredhark |
| 23:30.42 | anygivenname | cHarNe2: I dont understand |
| 23:31.05 | cHarNe2 | first you need to check your logfiles |
| 23:31.15 | *** part/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:31.30 | anygivenname | cHarNe2: which ? how ? |
| 23:32.01 | cHarNe2 | i dont know where asterisk keeps it's logs |
| 23:32.19 | cHarNe2 | probobly somewhere in /var/logs/. |
| 23:32.33 | cHarNe2 | and then u gonna wan't to tail it |
| 23:32.45 | cHarNe2 | as u try to connect |
| 23:33.33 | anygivenname | what do u mean tail it ? |
| 23:33.57 | cHarNe2 | do you know what a commandline is? |
| 23:34.35 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 23:35.09 | anygivenname | yes |
| 23:36.12 | cHarNe2 | good, do you know how to navigate your filesystem? |
| 23:37.19 | cHarNe2 | well, i need to sleep now |
| 23:37.28 | cHarNe2 | u wanna check the logs |
| 23:37.39 | anygivenname | I see the log now |
| 23:38.01 | anygivenname | seems hundereds of people were trying to hack |
| 23:38.17 | anygivenname | but failed cause asterisk is not working |
| 23:38.41 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:42.02 | anygivenname | sudo /usr/sbin/astman 127.0.0.1 |
| 23:42.15 | anygivenname | connection refused |
| 23:44.50 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 23:46.42 | anygivenname | is there anyway Asterisk would have a web interface instead of ur silly command lines |
| 23:47.03 | anygivenname | is there anyway Asterisk would have a web interface instead of using silly command lines |
| 23:47.38 | dan__t | braces |
| 23:49.37 | paulc | Use the force Luke |
| 23:57.10 | gruvfunk | says silly silly commands, we're all lost in a sea of silly commands |
| 23:58.57 | p3nguin | anygivenname: There are several web-based interfaces available for Asterisk, but none of them are supported in this channel. We apparently only know how to use silly command lines. |