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| 02:41.47 | xous | any linux sip client that I can rig to auto answer and play an audio source? |
| 02:49.32 | ectospasm | xous: why not use the Playback dialplan application? |
| 02:49.50 | ectospasm | probably much cleaner than such a rig you suggest |
| 02:49.52 | xous | don't actually need it to work with asterisk |
| 02:50.16 | xous | I will if I need |
| 02:51.36 | xous | If I did use asterisk I'd still end up sip trunking into another provider. |
| 02:53.15 | ectospasm | well, this is #asterisk |
| 02:53.20 | xous | it's basicially a hack to implment MOH on a shitty pbx |
| 02:53.44 | xous | yeah. couldn't find a generic voip or sip channel :P |
| 02:53.45 | ectospasm | let Asterisk do that. It's wonderful at being the glue between a legacy PBX and whatever |
| 02:54.39 | xous | http://www.voip-info.org/wiki/view/Asterisk+cmd+Ices |
| 02:54.46 | xous | what I'm looking at doesn't look very clean |
| 02:57.13 | *** join/#asterisk Yourname` (~Administr@unaffiliated/yourname/x-837320) |
| 02:57.55 | ectospasm | yeah, that looks like a bit of a hack. |
| 02:58.13 | xous | Freeswitch's implmentation looks better |
| 02:58.31 | xous | but the more complicated I make this the bigger pain in the ass it is to support. |
| 02:59.23 | xous | Our current "solution" is a Viking Coupler (shitty answering machine) + SIP ATA + client audio source |
| 02:59.32 | xous | which is unreliable and expensive. |
| 02:59.39 | xous | $600 setup cost for client. |
| 03:00.17 | ectospasm | I know less of what you're trying to do now. |
| 03:00.47 | xous | basically letting the client provide their own MOH. |
| 03:02.58 | ectospasm | xous: so where does the shitty PBX fit into that equation? |
| 03:03.40 | xous | shitty PBX initiates a SIP call to SIP ATA |
| 03:04.11 | xous | SIP ATA rings an analog line that's connected to the "viking coupler" |
| 03:04.28 | xous | the viking coupler answers and bridges the audio source with the line. |
| 03:05.56 | xous | ... troubleshooting this remotely is like trying to drive blind. |
| 03:06.04 | ectospasm | why not do something like PBX >SIP> Asterisk (which then plays MOH from audio source [like ICES2 you linked to]) |
| 03:06.20 | xous | That's what I was thinking |
| 03:06.32 | xous | but the asterisk implementation looked a bit sketchy |
| 03:08.44 | ectospasm | should be straightforward. Hardest part would be setting the audio source in musiconhold.conf |
| 03:08.57 | xous | how well would this scale? |
| 03:09.04 | ectospasm | what do you mean? |
| 03:09.12 | xous | if this works I suspect I could have over 100 sources. |
| 03:09.49 | xous | The default system hold music makes clients want to rip their earss out. |
| 03:11.01 | ectospasm | ...on Asterisk, or the legacy PBX? |
| 03:11.52 | xous | asterisk |
| 03:12.04 | xous | anecdote: 4-3Ghz/1Gb RAM: Native format_mp3 for MoH running on FreeBSD 5.4 with Asterisk CVS HEAD (July 2005). mp3 decode quality is great... up to 70 simultaneous MoH (8khz 16bit mono mp3) calls with about 80% idle. |
| 03:12.11 | xous | that's not too bad. |
| 03:12.37 | ectospasm | I find the latest round of default MOH music in Asterisk (opsound, I think it's called) to be quite pleasant. I was put on hold and heard pop hits. That made me want to commit suicide. |
| 03:13.22 | xous | hehe |
| 03:13.32 | xous | the stuff we got is from like 2000 |
| 03:13.40 | xous | it's like 80's bad porno music |
| 03:13.45 | xous | or maybe elevator music |
| 03:14.36 | ectospasm | yeah, check out the new stuff from say the 1.8.12.0 source. It's actually pretty good. |
| 03:14.46 | ectospasm | ...for free, at least. |
| 03:15.12 | xous | oh well |
| 03:15.20 | xous | I guess I gotta lab it out and see |
| 03:16.17 | xous | I bet the bastards will still charge $600 setup even if I bring the per user cost down to $30 |
| 03:16.34 | ectospasm | mo' profit! |
| 03:16.40 | xous | heh |
| 03:16.45 | xous | problem is if I build this |
| 03:16.57 | xous | I HAVE to maintain and troubleshoot everything. |
| 03:17.06 | xous | no other linux guys |
| 03:17.45 | xous | and I'd rather shoot myself in the face than let these guys on a linux box. |
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| 03:22.18 | xous | oh well. I'll figure out if it will work. |
| 03:22.44 | xous | then I'll decide if I'm going to see how much over time they'll give me for it. |
| 03:22.55 | xous | thanks. Helps to bounce ideas off someone. :P |
| 03:23.01 | ectospasm | no problem. |
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| 04:45.46 | wswartz | Is there any way I can get an incoming Google Voice call to ring a bunch of different users? |
| 04:45.56 | ectospasm | ring group? |
| 04:46.14 | ectospasm | Dial(SIP/1234&SIP/2234&SIP/3234...) |
| 04:46.25 | wswartz | Thanks, I'll start there. |
| 04:46.27 | [TK]D-Fender | your calls do whatever you tell them to. It;s your dialplan... |
| 04:48.14 | wswartz | Now with ring groups, anyone answering with their device makes it stop ringing for the others? |
| 04:51.48 | p3nguin | Yes, and with SIP phones, you will see a missed call on the phones that did not answer. |
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| 05:45.45 | din3sh | mrning all |
| 05:47.50 | ectospasm | if you could call it that |
| 05:52.59 | din3sh | call it what? |
| 05:53.00 | din3sh | :o |
| 05:54.17 | ectospasm | morning (-; |
| 05:56.07 | din3sh | its 10am here |
| 05:56.08 | din3sh | ;) |
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| 06:12.05 | BeeBuu | hi,all |
| 06:14.37 | BeeBuu | anyone help me please.i found my asterisk make big calls disconnected my program ( connected to the AMI), how can i fix this? |
| 06:23.41 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:26.44 | ruben23 | hi any successfully installed dahdi on a VPS..? any idea how to do it please |
| 06:35.16 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 06:39.31 | Yourname` | Hi. If Jitter is less than 2ms, wha could be causing calls being choppy? |
| 06:47.22 | bn-7bc | Yourname`: is there any transcoding going on? |
| 06:52.31 | Yourname` | bn-7bc: None. 711u.. |
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| 06:53.11 | Yourname` | bn-7bc: sorry was afkj |
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| 07:51.09 | bn-7bc | Yourname`: , np I cot detracted by youtube ?. anyway, how is the cpu load on the asterisk server |
| 07:51.25 | Yourname` | bn-7bc: Nothing zilch |
| 07:52.01 | Yourname` | bn-7bc: Wait, let me imagebin it to you |
| 07:53.59 | Yourname` | bn-7bc: http://imagebin.org/211926 |
| 07:54.08 | bn-7bc | Yourname`: hmm jitter is low ut what about rtt (is client an asterisk on the same lag)? |
| 07:55.17 | Yourname` | bn-7bc: Yeah jitter isnt big at all |
| 07:55.26 | Yourname` | bn-7bc: Client is a remote client. |
| 07:55.50 | Yourname` | bn-7bc: Remote -> Firewall ------internet------- Router -> Asterisk |
| 07:56.28 | bn-7bc | aha can you ping the remote firewall to get an rtt for me |
| 07:58.19 | Yourname` | bn-7bc: I'm at the remote site now.. so pinging the router |
| 07:59.25 | Yourname` | bn-7bc: http://pastebin.ca/2147223 |
| 08:00.02 | bn-7bc | hmm seems to be very good |
| 08:00.26 | bn-7bc | hold on, I will check in another hannel |
| 08:00.58 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 08:01.31 | qakhan | can i setup an ext when i dial it. it logoff agent |
| 08:03.59 | bn-7bc | Yourname`: seems the packets ar going different ways with jitter being <2ms but ping giving you south a spread in rtt |
| 08:04.30 | teloniusz | hi. My GSM operator sends music during ring signal. After passing it thorugh GSM gate and Asterisk I cannot hear this. What configuration/dial option controls this? |
| 08:05.02 | bn-7bc | hmm this is a bit over my head, how does asterisk measure jitter |
| 08:05.04 | teloniusz | (I'm dialing from a softphone through Asterisk and GSM gate to the GSM network) |
| 08:05.14 | Yourname` | bn-7bc: one sec |
| 08:09.13 | bn-7bc | Yourname`: is the client a soft phone an at a to an ip phone, and what make model |
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| 08:10.56 | bn-7bc | yha was ata nott "at a" |
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| 08:16.16 | Yourname` | bn-7bc: My apologies |
| 08:16.47 | Yourname` | bn-7bc: It's an xlite |
| 08:16.51 | Yourname` | softphone |
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| 08:22.34 | bn-7bc | hmm does xlite have any settings for jitter buffer or somesouch |
| 08:23.40 | bn-7bc | Yourname`: hold on I'm goggling a bit |
| 08:24.07 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 08:25.52 | Yourname` | bn-7bc: It does.. but i dont know what to put |
| 08:27.52 | bn-7bc | Yourname`: is it an iput field or a slider |
| 08:28.03 | Yourname` | bn-7bc: both :S |
| 08:30.02 | bn-7bc | Yourname`: try a call or tow with the slider at opposite ends of the scale and see way works better just remember where it is now |
| 08:32.35 | Yourname` | bn-7bc: gotcha |
| 08:33.58 | bn-7bc | Yourname`: : but as I have never had this problem myself I suggest you ask someone else (I'm still ore then willing to try to help, but I'm not shore I can be of any use) |
| 08:36.00 | Yourname` | bn-7bc: haha thanks for atleast trying!! |
| 08:37.07 | Yourname` | Really appreciate it |
| 08:38.06 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 08:40.24 | bn-7bc | Yourname`: Well glad to help, I hope you get it sorted, I know how annoying problems can be |
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| 08:41.58 | Yourname` | bn-7bc: yeah this one is a bitch, been bugging me for a couple days now |
| 08:45.08 | bn-7bc | Wel I have been screeching my head for a nounth wit a strange audio problem, an it seems the my provider has done something, so I'll get a new one soon |
| 08:45.47 | bn-7bc | annoying doing debug when the problem is at the providers end |
| 08:45.50 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:45.59 | jacc0 | good morning all!! |
| 08:46.01 | jacc0 | :) |
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| 09:03.28 | Yourname` | bn-7bc: Exactly! |
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| 09:38.44 | mirelab | hello, is there a way to logoff static Agent when there is no more callers in that queue, added to queue with AgentLogin? application |
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| 09:44.04 | din3sh | DEBUG[2248] netsock2.c: ...host '172.17.0.10' and port ''. |
| 09:44.12 | din3sh | what would this message mean? |
| 09:53.58 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 09:55.39 | mirelab | hello, is there a way to logoff static Agent when there is no more callers in that queue, added to queue with AgentLogin? application |
| 09:55.42 | mirelab | :) |
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| 10:04.22 | *** join/#asterisk timahvo1 (~rogue@41.80.245.34) |
| 10:04.41 | alexscott | hello there. Why asterisk (10.3) hangupcause is 34 when it receive a 403 ? |
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| 10:33.08 | *** join/#asterisk jamicque (~jamicque@80.50.125.74) |
| 10:36.01 | jamicque | Hi I have a PBX connected to my asterisk via PRI, for the incoming calls to asterisk from PBX digits are dialed in overlaped mode. I've changed the chan_dahdi.conf to overlap=yes and everything works as it should. However sometimes when the digits are slowly diled by the pbx user the asterisk can send not full number. As i've notced asterisk is waiting 3seconds and then makes the call, is |
| 10:36.01 | jamicque | there any way to mae this timeout longer? |
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| 10:49.17 | *** join/#asterisk [SySteM] (~antoine@ip-131.net-80-236-99.joinville.rev.numericable.fr) |
| 10:49.19 | [SySteM] | Hello |
| 10:49.31 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 10:49.35 | [SySteM] | I migrate to asterisk 1.6 from 1.4 |
| 10:49.42 | [SySteM] | asterisk dont load extensions.conf |
| 10:49.46 | [SySteM] | without any raisons |
| 10:49.57 | [SySteM] | if i start asterisk in verbose mode |
| 10:50.02 | [SySteM] | i make a reload |
| 10:50.14 | [SySteM] | not line with parsing extensions.conf |
| 10:50.33 | [SySteM] | it working only if i use extensions.conf from default package debian (with demo exts) |
| 10:51.03 | beebeeep | what happens when you type 'dialplan reload'? |
| 10:51.13 | beebeeep | in asterisk's CLI |
| 10:51.25 | [SySteM] | its parsing all file but not line with extensions.conf |
| 10:52.12 | [SySteM] | sorry, when i type "dialplan reload" : No such command 'dialplan reload' (type 'core show help dialplan reload' for other possible commands) |
| 10:52.34 | [SySteM] | but if i type "helo dialplan" i got help |
| 10:53.40 | beebeeep | looks like pbx_config.so module cannot load |
| 10:53.59 | WIMPy | jamicque: You're using Asterisk to forward calls from PRI to an ITSP? |
| 10:54.12 | beebeeep | try to reload: 'module reload pbx_config.so' |
| 10:54.39 | [SySteM] | [May 11 10:49:39] NOTICE[3611]: loader.c:686 ast_module_reload: The module 'pbx_config.so' was not properly initialized. Before reloading the module, you must run "module load pbx_config.so" and fix whatever is preventing the module from being initialized |
| 10:54.49 | [SySteM] | for information im on a new fresh install debian squeeze |
| 10:55.33 | beebeeep | mb try unload/load sequence? |
| 10:55.54 | [SySteM] | can you develop please |
| 10:56.43 | beebeeep | module unload pbx_config.so ; module load pbx_config.so |
| 10:57.09 | *** join/#asterisk wonderworld (~ww@dsdf-4db51205.pool.mediaWays.net) |
| 10:57.52 | [SySteM] | ok it give me one error in syntax extensions |
| 10:57.57 | [SySteM] | thank a lot beebeeep |
| 10:58.00 | [SySteM] | thanks :) |
| 10:58.58 | beebeeep | 1.4 to 1.6 is pretty painful upgrade ^) |
| 10:59.15 | beebeeep | but 1.6 to 1.8 is really seamless |
| 11:04.42 | [SySteM] | thanks again beebeeep |
| 11:04.54 | *** join/#asterisk darrenlooby (~darrenloo@cpc1-shep3-0-0-cust1012.8-3.cable.virginmedia.com) |
| 11:05.59 | darrenlooby | Hi Guys, I'm looking for a way to programatically check if a list of modules are loaded. Anyone know how? |
| 11:09.00 | beebeeep | asterisk -rx 'module show' ? |
| 11:09.42 | alexscott | I want to use hangupcause but i don't understand the mapping. HangupCause is 34 when asterisk receive a 404 or a 403, Why ? |
| 11:10.05 | WIMPy | alexscott: Q.850 / Q.931 |
| 11:10.44 | *** join/#asterisk wonderworld (~ww@dsdf-4db51205.pool.mediaWays.net) |
| 11:12.08 | alexscott | WIMPy: As i can see in documentation, 404 should be cause 1 unallocated and not 34 congestion |
| 11:12.40 | WIMPy | Yes, the translations aren't always optimal. |
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| 11:13.54 | alexscott | if i want to keep real status, i need to use 'storesipcause' ? |
| 11:14.07 | alexscott | do i* |
| 11:14.56 | Ice_Strike | hi |
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| 11:22.26 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 11:23.03 | darrenlooby | beebeeep, just ran "asterisk -rx 'module show'"... what I'm really looking for is a way of checking a list of module that need to load, and returning a list of those that didn't... so, failed to load for whatever reason. |
| 11:24.09 | darrenlooby | For instance, when we started asterisk, on of our fax modules didn't load. But, we only found out because we tried the fax server. Having a list of failed modules, would help to spot the problem earlier |
| 11:27.00 | beebeeep | mb u can get all modules from /usr/lib/asterisk/*.so, exclude noload modules from /etc/asterisk and check which modules are not loaded? |
| 11:27.17 | beebeeep | from /etc/asterisk/modules.conf of course* |
| 11:27.36 | Blowfish-nc | hey guys |
| 11:28.04 | Blowfish-nc | when initializing a new manager connection factory what username and password should i use? |
| 11:29.35 | Blowfish-nc | i tried the one from manager.conf, amportal.conf, web interface login details, host machine login details |
| 11:30.00 | Blowfish-nc | i can connect to the asterisk server but not authenticate |
| 11:30.42 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 11:30.48 | darrenlooby | beebeeep, so, say I has a list of module that should have loaded - all modules minus noload modules... how do I check this list against asterisk to see if they are loaded? Will I have to itterate through them? |
| 11:36.08 | Ice_Strike | When I issued the originate action (AMI) then EVENT will be respone.. how would i know which of those EVENT are related to the ACTION I have commit? |
| 11:36.47 | *** join/#asterisk Lo-lan-do (~roland@polymir.placard.fr.eu.org) |
| 11:40.14 | beebeeep | darrenlooby: save 'module show' output into temp file and do smth like: for module `cat modules_list.txt`; do grep -q $module actual_modules_list.txt || echo $module ; done |
| 11:40.48 | beebeeep | and you'll get list of modules that are in modules_list.txt and not in actual_modules_list.txt |
| 11:46.13 | Ice_Strike | beebeeep Do you know the answer to my question? |
| 11:47.51 | beebeeep | Ice_Strike: i'm not sure, but mb you should check ActionID of event? |
| 11:48.20 | beebeeep | i'm using UserEvent() in dialplan to track my originates :) |
| 11:49.00 | Ice_Strike | beebeeep I do get Event Response for ActionID |
| 11:49.12 | Ice_Strike | I mean Action Response |
| 11:49.17 | Ice_Strike | not for Event Reponse. |
| 11:50.43 | beebeeep | Ice_Strike: what kind of event? |
| 11:52.04 | Ice_Strike | see PM |
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| 12:24.15 | zkn | i have "setinterfacevar", "setqueueentryvar" and "setqueuevar" all set to yes. do you reckon it should be possible to get MEMBERNAME variable output to MONITOR_FILENAME? |
| 12:24.26 | zkn | so far I have not succeeded at that |
| 12:26.18 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 12:27.12 | zkn | also when I use Queue app with gosub option, the variables I set prior to the caller is sent to queue app, these variables are no longer available in the gosub context ... |
| 12:29.38 | *** join/#asterisk joshauh (~joshuah@harradence.net) |
| 12:29.53 | zkn | is that normal ? |
| 12:30.07 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 12:30.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 12:30.58 | bulkorok | I think yes... let me check |
| 12:31.03 | bulkorok | (global vars!) |
| 12:32.19 | bulkorok | here: https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance+Basics |
| 12:32.24 | bulkorok | I think this should help |
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| 12:49.29 | alexscott | why there is case 404: /* Not found */ ... ast_queue_hangup_with_cause(p->owner, AST_CAUSE_CONGESTION); and not AST_CAUSE_UNALLOCATED in chan_sip.c:20893 ? |
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| 12:55.32 | *** join/#asterisk ollii (~risker@port-87-193-161-154.static.qsc.de) |
| 12:55.36 | ollii | hey |
| 12:56.05 | ollii | could someone recommend a good tool to measure mos of asterisk? |
| 13:02.54 | beebeeep | human's ear :) |
| 13:04.37 | ollii | .. |
| 13:05.00 | ollii | i want to know how much good calls are possible on my hyper-v * 1.8 |
| 13:07.31 | bulkorok | wireshark!? |
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| 13:15.34 | mirelab | Is there a way to logoff static Agent when there is no more callers in that queue, added to queue with AgentLogin? application:) |
| 13:16.02 | *** join/#asterisk serafie1 (~erin@nat/digium/x-poarozkjckydybyv) |
| 13:16.52 | [TK]D-Fender | mirelab, there are CLI commands for this which yuo can also call from AMI |
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| 13:42.52 | rampage73 | ok have an asterisk Question, i have set up 10 digit did's as extensions and our calls are not being turned around or kept local if an extension on the asterisk box calls another extension on the asterisk box we used to use trixbox but have replaced it with plain asterisk with no GUI and I have no idea how trixbox did it but it is irrelevant anyway since we are not using it now. any help would be greatly appreciated. |
| 13:44.03 | *** join/#asterisk danfromuk (~IceChat77@2.30.230.217) |
| 13:45.28 | rampage73 | I should mention that everything else is working just fine outbound/inbound no issues other than we are doing a loopback according to our provider when one of our extensions calls another on the same box |
| 13:46.37 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:52.48 | WIMPy | alexscott: I have been told that it was due to some RFC or something. |
| 13:53.42 | [TK]D-Fender | rampage73, Your question isn't entirely clear... can you boil it down to the short version? |
| 13:55.03 | WIMPy | rampage73: Include your incoming context in to your phones/outgoing context. |
| 13:56.23 | *** join/#asterisk wonderworld (~ww@dsdf-4db51205.pool.mediaWays.net) |
| 13:56.43 | [TK]D-Fender | rampage73, show us an actual failed call |
| 13:56.45 | [TK]D-Fender | ~pb |
| 13:56.45 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:56.46 | [TK]D-Fender | ^^^ |
| 13:57.42 | rampage73 | the calls are not failing our provider is turning them around |
| 13:58.02 | ollii | that looks like a failed call for me |
| 13:58.39 | ollii | you wanted to keep it internal but it went out ... that is a failed call because it did not what you wanted to do |
| 13:58.51 | rampage73 | well ok i see your point |
| 13:59.08 | WIMPy | That is a missing configuration. |
| 14:00.31 | ollii | so please post a failed call and your dialplan |
| 14:00.32 | ollii | ~pb |
| 14:00.33 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:00.36 | rampage73 | WIMPy, is it that simple? man I make things complicated sometimes |
| 14:01.21 | rampage73 | WIMPy, correct me if i am wrong but that means I would list the extensions twice correct? once in incoming once in outgoing |
| 14:01.27 | [TK]D-Fender | rampage73, 11 steps to go! |
| 14:01.48 | rampage73 | [TK]D-Fender, ? |
| 14:01.59 | [TK]D-Fender | rampage73, show us what you've got. our description is vague and we on't like running blind |
| 14:02.03 | [TK]D-Fender | your* |
| 14:02.13 | [TK]D-Fender | Dangint, can't type for beans today... |
| 14:02.18 | ollii | *won't |
| 14:02.21 | [TK]D-Fender | @^#%&^ |
| 14:02.21 | ollii | ;) |
| 14:02.26 | rampage73 | [TK]D-Fender, thats ok me iether |
| 14:02.31 | rampage73 | *either |
| 14:02.38 | rampage73 | gees its catching |
| 14:03.44 | *** join/#asterisk atekinalp (~alper@85.105.28.77) |
| 14:03.51 | rampage73 | ok guys will go find or reproduce one be back in a bit thanks |
| 14:03.58 | *** part/#asterisk rampage73 (~rampage73@vpn.dctechonline.com) |
| 14:04.52 | atekinalp | hi i have a tdm400P card is it possible to configure it both as fxo and fxs |
| 14:04.54 | atekinalp | ? |
| 14:05.06 | beebeeep | of course |
| 14:05.23 | [TK]D-Fender | atekinalp, the cad has MODULES on it. That defines how many ports you have, and what type they are |
| 14:05.26 | [TK]D-Fender | card* |
| 14:07.09 | atekinalp | there are four module and they are red. |
| 14:08.19 | beebeeep | red is FXO |
| 14:08.20 | atekinalp | [TK]D-Fender: is that mean i can only configure modules as FXO or can i mix FSO/FXS |
| 14:08.34 | beebeeep | FXS modules are green |
| 14:08.34 | bulkorok | you can mix the modules... |
| 14:08.59 | atekinalp | bulkorok: how can i do that? |
| 14:09.30 | [TK]D-Fender | atekinalp, buy green modules |
| 14:09.31 | bulkorok | buy fxs modules and change |
| 14:09.34 | beebeeep | remove FXO modules and replace them with FXS :) |
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| 14:10.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:10.26 | atekinalp | so i can't use/configure red modules as FXS, write? |
| 14:10.34 | bulkorok | correct |
| 14:10.45 | bulkorok | this is what the green ones are for |
| 14:11.59 | beebeeep | and do not forget to connect additional power molex when using FXS cards |
| 14:13.44 | atekinalp | so modules are not hybrid so i can use both fxo and fxs. understand thanks. |
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| 14:30.03 | mirelab | [TK]D-Fender 03:16:46 PM |
| 14:30.05 | mirelab | "mirelab, there are CLI commands for this which yuo can also call from AMI" |
| 14:30.07 | mirelab | Yes I saw agent logoff <channel> i think, but was just wondering if there is an option for AgentLogin(), or queues.conf or agents.conf |
| 14:32.04 | [TK]D-Fender | mirelab, ORIGINATE <- |
| 14:32.34 | mirelab | [TK]D-Fender, kk i'll take a look |
| 14:32.36 | mirelab | thx :) |
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| 14:37.12 | mirelab | yes Originate might just do the trick , thanks a lot :) |
| 14:38.31 | *** join/#asterisk alexscott (~alexscott@94-84-15-217.reverse.alphalink.fr) |
| 14:38.34 | mirelab | cause i don't want agent to be stuck in 1 queue for a long time if there are no calls in that queue |
| 14:39.34 | mirelab | agent can leave pressing * but was just wondering if that can be done automaticly if no more callers in that queue |
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| 14:49.06 | cusco | hei |
| 14:51.21 | [TK]D-Fender | mirelab, I have almost never seen anyone even USING AgentLogin..... |
| 14:51.50 | [TK]D-Fender | mirelab, only in psycho-run outsourcing call-centers |
| 14:52.32 | _Corey_ | psycho-run? lol |
| 14:53.47 | mirelab | hehehe |
| 14:54.41 | mirelab | it's useful when you have 1 agent and a lot of callers in a lot of queus :P |
| 14:56.24 | [TK]D-Fender | mirelab, why not just add them dynamically? |
| 15:02.11 | mirelab | |
| 15:02.13 | mirelab | [TK]D-Fender, well that will have to be done eventually i guess, just need some planing and testing |
| 15:03.10 | [TK]D-Fender | If you automatically log them in when there are callers... then effectively they are ALWAYS logged in. |
| 15:03.34 | [TK]D-Fender | I really fail to see the point except for the use of DTMF to end a call and letting the next beep in... |
| 15:03.49 | [TK]D-Fender | Which still sounds wasteful |
| 15:04.24 | mirelab | but I need agent to chose the queue or event caller they want to answer and not automaticly :P |
| 15:06.04 | *** join/#asterisk Tim_Toady (~fuzzy@130.43.17.43.dsl.dyn.forthnet.gr) |
| 15:06.15 | mirelab | for example looking at some realtime app seeing all the queues stats and than agent picks the queue that he wants, by wait time or urgency or ... |
| 15:06.54 | mirelab | and having more than one server just complicate things lol |
| 15:07.22 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 15:07.33 | ThinkGNU- | good day to everyone |
| 15:07.41 | mirelab | but with originate might just solve few problems at once lol |
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| 15:11.11 | jaytee | anyone using Polycom phones with sip version 3.3.1RevF? seems like the configs have changed drastically from sip 3.2.4b and the configuration file management whitepaper they reference isn't up to date. |
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| 15:14.08 | danfromuk | jaytee, yes, it changed drastically |
| 15:14.17 | danfromuk | i made that mistake |
| 15:14.50 | Skaag | is there a command I can send after I answer a call on AGI that will instruct Asterisk to start recording the call immediately? |
| 15:17.32 | cusco | Monitor, MixMonitor |
| 15:18.03 | cusco | manager show commands |
| 15:18.07 | kaldemar | ll |
| 15:18.56 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:18.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:19.21 | cusco | I'm introuble with dahdi again... |
| 15:19.45 | cusco | basically... dahdi_scan shows span alarm=OK |
| 15:19.58 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 15:20.20 | cusco | pri show span: http://paste.debian.net/168333/ |
| 15:20.40 | cusco | http://paste.debian.net/168333/ |
| 15:20.43 | cusco | oops |
| 15:20.45 | cusco | Status: Down, Active |
| 15:20.51 | cusco | I dunno why |
| 15:23.28 | jaytee | danfromuk, how did you resolve it? Did you find a more up to date provisioning guide? |
| 15:24.20 | [TK]D-Fender | jaytee, The admin guid says it all |
| 15:25.38 | Skaag | cusco: thanks |
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| 15:27.25 | Skaag | I hate how voip-info.org is looking like a spam site |
| 15:27.42 | cusco | adblock |
| 15:27.47 | Skaag | yah... |
| 15:27.55 | Qwell | Skaag: It's always been a spam site. |
| 15:27.58 | Qwell | ~asteriskwiki |
| 15:27.58 | infobot | from memory, asteriskwiki is http://wiki.asterisk.org |
| 15:27.59 | jaytee | [TK]D-Fender, I'll check it out but I never found the Admin guide for previous sip versions to be particularly useful for the provisioning of the phones except to determine what settings did what. |
| 15:28.30 | [TK]D-Fender | Umm... th'as kinda the point... |
| 15:28.43 | [TK]D-Fender | it also describes how the <mac> & directory files work, etc.. |
| 15:28.46 | [TK]D-Fender | it IS complete... |
| 15:29.10 | jaytee | [TK]D-Fender, ok ok! I'll check it out! |
| 15:29.13 | *** join/#asterisk sickgrinder (~MattB@office2.vicidial.com) |
| 15:30.27 | Skaag | Qwell: that site uses the extremely confusing and ill conceived confluence wiki crap system… :-\ |
| 15:30.41 | Qwell | It's links. You click them. |
| 15:30.52 | Qwell | I don't understand what's confusing about that. O.o |
| 15:31.22 | Skaag | i didn't say it didn't contain useful information, or that given enough pain in the ass, you can't get to it |
| 15:31.31 | Skaag | it's just, too many clicks, confusing terms (spaces???), etc. |
| 15:31.37 | Skaag | but it's ok, I found what I wanted there |
| 15:31.40 | Skaag | :-) |
| 15:31.43 | Skaag | thanks for the link! |
| 15:32.05 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:33.52 | jeffspeff | how can i do pattern matching within an if statement? i'm trying to get the following to work -> exten=911,n,GotoIF($["${CALLERID(num)}" = "_16XX"]?FL) |
| 15:34.54 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v027-196.mobile.uci.edu) |
| 15:38.35 | beebeeep | jeffspeff: you can use DIALPLAN_EXISTS with special test context with all patterns that you need |
| 15:38.54 | Kobaz | so this was fun |
| 15:40.05 | beebeeep | i.e. GotoIf(${DIALPLAN_EXISTS(test-patterns,${CALLEID(num)})}?do-smth) |
| 15:42.46 | jeffspeff | beebeeep, I don't think that'll work for me. I'm needing to do some filtering when somebody dials 911. Example: the majority of the 16XX extensions reside in a florida office, however a few do not. So if somebody from extensions 1601 through 1620 dials 911, the CID(num) needs to be 1234 but if extension 1615 dials 911 then the CID(num) needs to be set to 5678. |
| 15:43.03 | beebeeep | or you can use REGEX() or even simple substring manipulations |
| 15:45.33 | [TK]D-Fender | jeffspeff, this is a simple OR operation in an expression.... |
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| 15:48.06 | beebeeep | jeffspeff: if some extensions need different outbound CID, i'd prefer to put them into separate context to make outbound rules clear without any complicated logic |
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| 15:50.24 | jeffspeff | beebeeep, the only time the CID needs to change is if calling 911. Some of our users work from home, but report to a branch office. There regular calls need to set CID to that of the main DID for their branch, but they call 911 then it needs to set CID to a different DID that's just theirs and has their address mapped to it |
| 15:51.27 | jeffspeff | I'd hate to have to make a bunch of extra contexts to accomodate this. we have 9 branch offices and about 40 or more people working from home. |
| 15:52.12 | beebeeep | jeffspeff: or just you cat specify setvar=CID_911=123456 in sip.conf for this peers and set custom CID while calling 911 if ${CID_911} is set :) |
| 15:52.54 | gusto | hm? |
| 15:53.02 | gusto | 911 is emergency number, right? |
| 15:53.20 | jeffspeff | yes |
| 15:53.27 | gusto | that would be a problem to 'test' emergency numbers on VoIP because that could be a 'real' emergency |
| 15:53.35 | *** part/#asterisk ollii (~risker@port-87-193-161-154.static.qsc.de) |
| 15:54.19 | carrar | so they say |
| 15:54.39 | carrar | Does your 911 provider,provide a 911 test did? |
| 15:54.42 | carrar | like 711 |
| 15:54.44 | jeffspeff | gusto, we have something different setup for the test calls. For testing the dialplan functionality we have it actually dial 933 instead of 911. |
| 15:54.46 | carrar | or something |
| 15:54.49 | jeffspeff | carrar, see above |
| 15:54.58 | carrar | SEE ABOVE!!! |
| 15:55.07 | p3nguin | ABOVE |
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| 15:55.13 | jeffspeff | beebeeep, i think i'm really liking the setvar idea |
| 15:55.26 | beebeeep | in russia many voip providers claims that you should not use them for emergency calls :) |
| 15:55.36 | gusto | of course not |
| 15:55.44 | carrar | haha |
| 15:56.17 | carrar | jeffspeff, whois your 911 provider? |
| 15:56.28 | gusto | but you can use any cell phone, also w/o any sim card in |
| 15:56.30 | WIMPy | Like most phone lines today aren;t suitible for emergency calls. |
| 15:56.54 | jeffspeff | carrar, all of our SIP termination is done through IntelePeer |
| 15:56.54 | gusto | maybe they should get rid of emergency calls completely |
| 15:57.03 | WIMPy | gusto: Depends on your location. In some countries that doesn;t work any more. |
| 15:57.16 | carrar | Can you craete customer DID's with addresses just for 911? |
| 15:57.22 | carrar | like BTN+EXTEN |
| 15:57.22 | gusto | WIMPy: where? |
| 15:57.31 | beebeeep | WIMPy: isn't it a part of GSM standart? |
| 15:57.37 | carrar | for a 14 digit number |
| 15:57.39 | WIMPy | gusto: Germany, Denmark. |
| 15:57.40 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:57.44 | WIMPy | beebeeep: Yes, it is. |
| 15:58.02 | gusto | WIMPy: i have a provider who gives me the option to enable emergency calls, but i disabledi t |
| 15:58.05 | gusto | it |
| 15:58.06 | WIMPy | So some networks are no longer GSM compliant, but rather GSM compatible. |
| 15:58.16 | beebeeep | huh |
| 15:58.28 | jeffspeff | carrar, no they won't do 14 digits... I'm having to get secondary DID's for all the seperate locations (including homes) to set the different addresses on. |
| 15:58.42 | WIMPy | They mada a law for that. |
| 15:58.47 | WIMPy | made |
| 15:58.49 | carrar | what a pain |
| 15:58.56 | gusto | what are we taling about now? VoIP emergency calls, or cell phones w/o inserted simcard? |
| 15:59.05 | jeffspeff | lol |
| 15:59.08 | carrar | Could use a E911 provider |
| 15:59.12 | carrar | and not your provider |
| 15:59.15 | Skaag | I have a simple AGI which answers a call, gets the channel to dial a SIP trunk, then hangs up. I have a logging command after the hangup line, but it seems to never be reached. Does that make sense? |
| 15:59.25 | WIMPy | call phones without a SIM. |
| 15:59.38 | gusto | ah, so that does not work now? |
| 15:59.53 | gusto | why? |
| 16:00.03 | WIMPy | By law you need a valid SIM, but so far any SIM will do. |
| 16:00.18 | gusto | what is a valid sim? |
| 16:00.28 | carrar | one that works? |
| 16:00.29 | Qwell | WIMPy: I don't think that's true |
| 16:00.30 | WIMPy | One that can register. |
| 16:00.34 | beebeeep | Skaag: deadagi? |
| 16:00.44 | WIMPy | Qwell: What? |
| 16:00.44 | Qwell | AFAIK, you don't even need a SIM at all |
| 16:00.58 | gusto | Qwell: that was what i was talking about |
| 16:01.13 | WIMPy | Qwell: Not by standard, but by law. See above. |
| 16:01.13 | Qwell | I've been wrong occasionally though. |
| 16:01.27 | WIMPy | Meeting... |
| 16:02.30 | Skaag | beebeeep: I have no idea |
| 16:02.42 | Skaag | beebeeep: I'm using phpagi |
| 16:04.28 | beebeeep | Skaag: how do you call AGI script from dialplan? |
| 16:04.49 | Skaag | AGI(router.php) |
| 16:05.11 | beebeeep | Skaag: A locally executed AGI script will receive SIGHUP |
| 16:05.15 | beebeeep | on hangup from the channel except when using DeadAGI |
| 16:05.29 | Skaag | I see! |
| 16:06.09 | Skaag | So all I need to do is replace AGI() with DeadAGI() and the script will be allowed to operate until it exits naturally. Right? |
| 16:06.15 | beebeeep | maybe :) |
| 16:06.20 | Skaag | haha :) |
| 16:06.27 | Skaag | ok thanks, I will experiment with it |
| 16:06.34 | beebeeep | good luck :) |
| 16:07.03 | Skaag | by the way is there a way to AGI events? such as the registration of a new extension? |
| 16:08.17 | beebeeep | you should use AMI to catch events |
| 16:09.14 | alexscott | I'm trying a fax from a T30 fax (g711) through an asterisk with t38gateway to an T38 ATA. After the re-invite, i don't have anymore RTP from asterisk to my T30 fax. I have the CLI in debug and i don't see any failures. There is no more RTP after the re-invite on the ATA side. |
| 16:09.19 | jeffspeff | beebeeep, your idea works great, you've saved me a lot of dialplaning! thanks |
| 16:09.58 | alexscott | Fax from T38 to T30 through asterisk works well with same equipments and configuration. |
| 16:12.14 | alexscott | a |
| 16:12.17 | alexscott | ops |
| 16:12.21 | *** part/#asterisk alexscott (~alexscott@94-84-15-217.reverse.alphalink.fr) |
| 16:12.59 | *** join/#asterisk alexscott (~alexscott@94-84-15-217.reverse.alphalink.fr) |
| 16:14.19 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 16:21.55 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:21.55 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:22.30 | *** join/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 16:25.52 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:26.38 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 16:27.13 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:31.05 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:32.01 | *** join/#asterisk nobodyshome (~dijib@bas10-kitchener06-1279682411.dsl.bell.ca) |
| 16:40.17 | dijib | hey all |
| 16:40.21 | dijib | whats soup in here? |
| 16:44.20 | cusco | im trying to figure whats wrong with PRI |
| 16:46.26 | *** join/#asterisk mKn0wt (~mtv@200.30.170.218) |
| 16:54.36 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 16:59.10 | alexscott | pffff ... i don't find anything ... I don't understand why there is no more RTP to my T30 equipment after t38 re-invite when i'm trying fax from t30 to t38 ... All it's okay on the other side .... There is no error or warning message during the call ... i don't know where to look to understand that ... What could be the reason to have a non working t38 to g711 bridge without any errors ? |
| 16:59.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:01.36 | alexscott | ~t38 |
| 17:01.36 | infobot | extra, extra, read all about it, t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 17:02.14 | alexscott | deadlink |
| 17:02.55 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:03.48 | alexscott | ~spandsp |
| 17:03.48 | infobot | well, spandsp is cool : http://www.soft-switch.org/installing-spandsp.html, or a softmodem that comes with app_rxfax and app_txfax for asterisk, or the shizzznit |
| 17:07.06 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:28.57 | FLeiXiuS | Whats the easiest way debug dtmf on a channel? |
| 17:35.10 | alexscott | dtmf outofband or inband ? |
| 17:36.40 | jrose_atDigium | via rubber bands |
| 17:37.35 | Qwell | FLeiXiuS: There's a dtmf log option that can be enabled in logger.conf, if I'm not mistaken. |
| 17:37.43 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 17:38.25 | jrose_atDigium | Yep, it's a log level. |
| 17:38.51 | jrose_atDigium | If enabled for console it should display I think. |
| 17:39.15 | jrose_atDigium | alternatively you could log it to a file of course. |
| 17:41.52 | Katty | ooooh i haz a chubby invite |
| 17:41.57 | leifmadsen | o.O |
| 17:42.14 | leifmadsen | at first my brain couldn't get to the word "invite" |
| 17:42.14 | Katty | yay beta testing |
| 17:42.28 | Katty | oh. i typoed. |
| 17:42.30 | Katty | it's cubby |
| 17:42.58 | Katty | http://lifehacker.com/5907654/cubby-is-like-dropbox-if-dropbox-also-had-free-unlimited-syncing-and-weve-got-invites <- relevant |
| 17:44.07 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 17:44.11 | timeshell | Greetings. |
| 17:44.15 | FLeiXiuS | Qwell, thanksQ! |
| 17:44.19 | alexscott | [May 11 19:43:27] DEBUG[28861]: res_fax.c:2699 fax_gateway_detect_t38: starting T.38 gateway for T.38 channel SIP/pcscf-1-cbv1-00000003 and G.711 channel SIP/routag1-00000002 |
| 17:44.59 | FLeiXiuS | logger set level DTMF on worked for a console fix |
| 17:45.32 | alexscott | i have this log (and some other) when re-invite is processing and no more rtp to the g711 side .... |
| 17:45.52 | timeshell | So, here's a situation I have brought before. When our internet connection drops, it causes our SIP trunks to outside providers to basically hang up. This in turn appears to hang up the entire SIP channel driver so that all connected phones lose their ability to further communicate with the server. |
| 17:48.11 | timeshell | Seems to me that if a SIP trunk is cut off it shouldn't affect the rest of the SIP communication. |
| 17:48.44 | timeshell | Our PSTN lines were still available, but were useless because all the phones couldn't communicate with the SIP chan until I deleted the SIP trunks. |
| 17:48.59 | Katty | do the sip devices register with the server locally? |
| 17:49.07 | timeshell | Yes. |
| 17:49.10 | talntid | Katty, thanks! I signed up =) |
| 17:49.14 | Katty | phone 192.168.0.x -> server 192.168.0.x |
| 17:49.19 | Katty | or similiar. subnetty, goodness. |
| 17:49.20 | timeshell | Correct |
| 17:49.21 | alexscott | Is there a specific version of spandsp to user with asterisk 10.x ? |
| 17:49.25 | Qwell | timeshell: Do they try to call out to the provider first? |
| 17:49.26 | Katty | can you ping the phones from the server? |
| 17:49.37 | Qwell | it may be DNS not resolving, causing this |
| 17:49.42 | Katty | talntid: you are welcomes. |
| 17:49.46 | timeshell | Qwell: It's probable that they did. |
| 17:49.52 | Katty | talntid: i added google drive and dropbox as cubbies |
| 17:49.56 | Katty | talntid: now i will have ALL THE THINGS |
| 17:50.09 | jrose_atDigium | That reminds me of something we were talking about a while back. |
| 17:50.20 | timeshell | Qwell: I've heard that before. I'd still submit that regardless of the reason, the server shouldn't hang up because one channel gets stuck. |
| 17:50.26 | jrose_atDigium | Like since it tries to do a dns lookup in the monitor thread without a timeout, the thread just blocks. |
| 17:50.33 | alexscott | Katty: do you have a working t38gateway setup with asterisk 10 :D ? |
| 17:50.34 | Qwell | yes, taht |
| 17:50.40 | timeshell | hang up meaning SIP-chan become unresponsive. |
| 17:51.06 | Katty | GASP, faxing?! |
| 17:51.17 | Katty | no, but i have a cassette tape out in my car |
| 17:51.18 | Katty | does that count? |
| 17:51.35 | alexscott | yup (its a joke because you wrote : "now i will have ALL THE THINGS" |
| 17:51.39 | timeshell | Qwell: It also stands to reason that if the internet itself is down that DNS wouldn't resolve. |
| 17:52.01 | jrose_atDigium | I'm pretty sure there is a workaround though. |
| 17:52.01 | Katty | alexscott: in relation to cubby being synced with dropbox and google drive. |
| 17:52.02 | timeshell | That really shouldn't interfere with the rest of the system. |
| 17:52.03 | Qwell | timeshell: You could always install a local dns cache |
| 17:52.15 | timeshell | Qwell: I suppose, but that's not really the point. |
| 17:52.18 | Qwell | or whatever magic jrose_atDigium has |
| 17:52.29 | Katty | he has a sandwich. |
| 17:52.29 | jrose_atDigium | Nah, it was just installing a local dns cache. |
| 17:52.38 | Qwell | there you go then! |
| 17:52.40 | timeshell | * shouldn't become unresponsive to everything else because of a problem with a single connection. |
| 17:52.45 | Katty | mmm dns cache sandwich. |
| 17:52.58 | Qwell | timeshell: it shouldn't, but it does. You're more than welcome to wait for the fix. |
| 17:53.09 | Qwell | (or, write the fix) |
| 17:53.16 | timeshell | Qwell: lol. Yep. Been waiting for 3 years. |
| 17:53.23 | jrose_atDigium | timeshell: The monitor thread for SIP multithreaded, so barring tearing it up completely, I don't think there is much we can safely do about that. |
| 17:53.25 | Qwell | You probably could've installed a dns cache in that time |
| 17:53.42 | jrose_atDigium | s/for SIP multithreaded/for SIP isn't multithreaded |
| 17:54.11 | jrose_atDigium | Basically, if something stalls the monitoring thread, it's stuck is what I'm saying. |
| 17:54.37 | timeshell | I understand what you're saying. I'm saying, that's a really bad thing. |
| 17:55.02 | timeshell | Is the business edition of asterisk subject to the same weakness? |
| 17:55.04 | Katty | there's that old thing, time vs money |
| 17:55.40 | Qwell | timeshell: Having your Internet connection go down isn't exactly a happy time. |
| 17:55.54 | Katty | YES IT IS |
| 17:56.02 | Katty | internet goes down, naps on desk |
| 17:56.12 | Qwell | Katty: but, no sleep reddit. Just think. |
| 17:56.14 | kenshin | hi, stupid question about nat and firewall, My Ata is weird (has 2 phone ports which can be assigned to different extensions, but with the same provider), long story short, had to set up asterisk to act like a *proxy*, it is running on the same computer that connect sot the internet, I'm not expecting any external clients, the only ones are the ata and the computer it's self (hylafax, via iaxmodem). |
| 17:56.15 | jrose_atDigium | It's a happy time to reflect peacefully and purge your mind of things like the telephone. |
| 17:56.17 | timeshell | No it isn't, but having phones go down when there are still PSTN lines available (or even intra office communication needed) is worse. |
| 17:56.37 | jrose_atDigium | timeshell: Yeah, I agree, it's ungood. |
| 17:56.39 | Qwell | timeshell: There are workarounds. |
| 17:56.43 | Katty | but i have cubby and google drive and dropbox! it shouldn't matter if athe internets go down! |
| 17:56.52 | Qwell | You've been offered a very good one. Complaining about it here is not useful. |
| 17:56.55 | timeshell | Especially when you're the only IT guy in the company and it happens on your day off when you're out of town. |
| 17:57.02 | Katty | oh hush your face |
| 17:57.03 | timeshell | (which happened to me last time)_ |
| 17:57.10 | Katty | i was driving to gatlinburg tn with clients calling me in a panic over raid issues |
| 17:57.15 | Katty | i don't wanna hear it |
| 17:57.23 | Katty | and i am the only IT GIRL |
| 17:57.26 | Qwell | Katty: and you didn't visit Huntsville. TSK, TSK. |
| 17:57.31 | jrose_atDigium | No need to get... Catty :3 |
| 17:57.34 | Katty | in my defense it was 2am |
| 17:57.45 | Katty | and i was asleep. |
| 17:57.45 | Qwell | Katty: as if we weren't all up :p |
| 17:57.52 | Katty | shhhhhhhhhhh |
| 17:57.57 | Katty | that's a secret. |
| 17:57.59 | timeshell | So, do the asterisk based appliances have the same vulnerability? |
| 17:58.20 | Qwell | timeshell: You tell us. |
| 17:58.28 | timeshell | I don't know. I don't have one. |
| 17:58.35 | timeshell | I have a sourcecode built server. |
| 17:58.40 | timeshell | I presume the answer is yet. |
| 17:58.42 | Qwell | This is a trivial problem to workaround. |
| 17:58.43 | timeshell | yes* |
| 17:58.54 | Katty | i want a server made of bacon. |
| 17:58.58 | Katty | and waffles. |
| 17:59.00 | jrose_atDigium | I don't know either. I imagine it would depend on whether or not they have a dns cache set up by default. |
| 17:59.04 | Katty | maybe a little syrup? |
| 17:59.05 | timeshell | Qwell: Matter of perspective. In concept yes. |
| 17:59.11 | kenshin | the fist time I set it up phone rings and no audio in either directions, So I opened OPEN_UDP="5004:5082 4569 10000:20000" |
| 17:59.12 | Qwell | Katty: if you did it right, it could cook your bacon for you |
| 17:59.21 | Katty | wait...what? |
| 17:59.22 | Qwell | kenshin: why 5004:5082? |
| 17:59.24 | jrose_atDigium | I don't think there is a significant difference in the architecture of chan_sip that would make a difference anyway. |
| 17:59.39 | Katty | Qwell: WHAT ARE YOU TRYING TO SUGGEST SIR |
| 17:59.45 | kenshin | I don;t know (found that on the internet) the funny thing is the moment I did that it worked |
| 17:59.50 | Qwell | Katty: Just what I said. |
| 18:00.00 | Katty | Qwell: trisky. |
| 18:00.05 | Katty | Qwell: tricksy. |
| 18:00.09 | kenshin | (may be coincidence) |
| 18:00.09 | Qwell | kenshin: you just need 5060 and whatever RTP ports in rtp.conf |
| 18:00.57 | kenshin | Qwell, why since asterisk should;t be listening on those port on the external interfaces Â… it is serving only internal clients |
| 18:01.12 | Qwell | kenshin: You don't have an external provider or anything? |
| 18:01.47 | kenshin | Qwell, well yes, but the firewall will permit any incoming connection if it is associated with an already open one |
| 18:01.52 | Qwell | your initial comment used the word Internet and provider several times |
| 18:02.04 | Qwell | They aren't associated, as far as the firewall goes. |
| 18:02.11 | Qwell | It has no idea that RTP is needed for SIP. |
| 18:02.21 | Qwell | and new incoming calls won't be associated with anything |
| 18:02.22 | kenshin | asterisk is registering with external providers (3-4) |
| 18:02.47 | Katty | i'm going to change all my rtp ports just to confuse Qwell |
| 18:02.52 | Katty | and then report an issue. |
| 18:03.02 | Qwell | Katty: Oh, I'll know. I'll know. |
| 18:03.15 | Katty | okay well just pretend so we can both have a good laugh. |
| 18:03.45 | kenshin | if I draw a parallel, I used to use my phone (cell) as a sip client) behind that same firewall (it connects to the same provider) and it worked at least for outgoing calls |
| 18:04.02 | Katty | cell client still is outside your network |
| 18:04.05 | Katty | via sprint or whatever |
| 18:04.08 | Katty | unless it's on a wan |
| 18:04.12 | Qwell | wifi probably |
| 18:04.17 | *** join/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 18:04.19 | Katty | but that's only in a building |
| 18:04.21 | Katty | which seems silly |
| 18:04.23 | Qwell | but, it's still not the same. incoming != outgoing |
| 18:04.35 | Katty | ya usually outgoing traffic isn't blocked |
| 18:04.36 | Qwell | and your phones client probably shoots first |
| 18:04.46 | Qwell | (which lets incoming traffic through) |
| 18:04.50 | Katty | it's when it's routed back into your network at goes OHAI FIREWALL PLZ PASS THIS AUDIO ON RTP PORT FOOOO |
| 18:04.59 | Katty | meanwhile firwall going herpderp, what's this. block. |
| 18:06.07 | kenshin | just to be sure not to nuke my working setup, I can block anything coming from these ports that is not coming from my providers ? |
| 18:06.17 | Katty | http://www.securitygeneration.com/wp-content/uploads/2010/12/firewall-lolcat.jpg <- firewall kitty demands RTP ports opened. |
| 18:06.31 | Qwell | kenshin: it depends. it's possible for your provider to reinvite to some other location |
| 18:06.40 | Qwell | that's probably unlikely for a provider though |
| 18:06.42 | Katty | kenshin: i know bandwidth.com hs a range |
| 18:06.54 | Katty | i think there was 2 or 3 ips the calls could come from |
| 18:07.03 | Katty | but just the same, rtp is pretty safe |
| 18:07.05 | Katty | it's just audio |
| 18:07.13 | Katty | and it's udp so no connections are actually made |
| 18:07.20 | Katty | just massive shoving of data up the ....well, you get the idea |
| 18:07.48 | Katty | regardless, i have incoming rtp connections locked down to just those 3 ips |
| 18:08.00 | kenshin | or simply add bindaddr => 192.168.x.x ? |
| 18:08.08 | Katty | i've never done it that way. |
| 18:08.15 | Katty | oh wait, you mean in sip.conf? |
| 18:08.21 | Katty | was it sip? |
| 18:08.27 | Katty | dagnabit, my memory |
| 18:08.38 | kenshin | yes |
| 18:09.05 | Katty | bindport |
| 18:09.22 | kenshin | I mean, I don;t want to be serving phone calls to the whole internet, |
| 18:09.24 | Katty | annnddd |
| 18:09.40 | Qwell | isn't that exactly what you're doing? |
| 18:09.40 | Katty | well they can't register unless you have port 5060 |
| 18:10.01 | Katty | if 5060 is open, they could try registering as a phoen to dial out through your system |
| 18:10.04 | Katty | which is bad juju |
| 18:10.11 | Katty | but rtp is just audio |
| 18:10.21 | p3nguin | They can dial out through the system without registering. |
| 18:11.13 | kenshin | so basically I can binaddr => 192.168.x.x in sip.conf (no need to block udp 5060 on firewall b/c art won;t be even listening) and forward RTP to 192.168.x.x |
| 18:11.19 | jrose_atDigium | registering makes you reachable rather than able to dial. |
| 18:11.43 | Qwell | How are you going to get incoming calls, if 5060 isn't forwarded? |
| 18:12.06 | Katty | i think he means udp 5060 |
| 18:12.08 | p3nguin | Cooperative NAT? |
| 18:12.08 | kenshin | Mmm, I keep thinking TCP :) |
| 18:12.09 | Katty | not tcp 5060 |
| 18:12.27 | Katty | I FOUND IT |
| 18:12.27 | Qwell | even if it were tcp, registering does not open a connection |
| 18:12.33 | Katty | externip, localnet, externhost, and nat settings |
| 18:12.34 | Katty | in sip.conf |
| 18:12.51 | p3nguin | I've got one client who does not forward 5060 in to the asterisk system, and receives calls inbound all day/night long. |
| 18:13.28 | Katty | kenshin: but you probably already have those setup |
| 18:13.28 | kenshin | registering (over tcp) doesn;t open a persistent connection to provider ?? |
| 18:13.34 | Qwell | kenshin: no it does not |
| 18:13.38 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 18:13.45 | kenshin | I was assuming that. |
| 18:14.12 | kenshin | so What I did is what is supposed to be. |
| 18:14.28 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.195) |
| 18:14.34 | Qwell | other than the excessive amount of ports, yes |
| 18:14.55 | Katty | seems like i did a writeup on this |
| 18:14.57 | p3nguin | Now I'm curious how it works. |
| 18:15.10 | Katty | oh hey i did |
| 18:15.12 | Qwell | p3nguin: keepalives and such |
| 18:15.20 | p3nguin | I don't forward ports for phones, ever, but calls to phones always make it in. |
| 18:15.39 | p3nguin | The same seems to be true for calls from the provider to asterisk -- it is the same server/client model. |
| 18:15.40 | Qwell | if your qualify is low enough, and your firewall isn't too anal, it can work |
| 18:15.49 | Katty | kenshin: http://dl.dropbox.com/u/25951554/42ND%20GEEK%20STREET_%20Setting%20Up%20A%20Basic%20Remote%20Asterisk%20User.pdf <- maybe that will help. |
| 18:16.05 | kenshin | thank Katty |
| 18:16.14 | Qwell | p3nguin: it really just depends on how long your firewall will keep the hole open |
| 18:16.22 | Qwell | some close it really really quickly |
| 18:16.47 | kenshin | (what I don;t want is ppl attempting to register and succeeding in doing so) |
| 18:16.51 | p3nguin | That system in behind a Comcast business gateway appliance. I have no idea who makes it. |
| 18:17.06 | *** join/#asterisk Neptu (~Neptu@c-6c96e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 18:17.16 | Katty | kenshin: it's not /exactly/ what you're doing, but outlines the routing info and port stuffs |
| 18:18.31 | Qwell | kenshin: You can limit the forwarding to just your provider, but understand that with some providers, that may present problems. |
| 18:18.43 | p3nguin | That can be summarized in a single line. Forward 5060 UDP and whatever port range is listed in rtp.conf (also UDP) to the asterisk system. Done. |
| 18:18.56 | Katty | p3nguin: is it your documentation? |
| 18:18.58 | Katty | p3nguin: no? |
| 18:19.01 | Katty | p3nguin: then shush your face. |
| 18:19.13 | Katty | better yet, you write one! |
| 18:19.17 | *** join/#asterisk nix8n82 (~none@24.143.10.144) |
| 18:19.18 | p3nguin | I just did. |
| 18:19.22 | Katty | and when you share it to help others... |
| 18:19.24 | Katty | then we'll talk! |
| 18:19.30 | p3nguin | I just wrote it and shared it. |
| 18:19.34 | p3nguin | Published. |
| 18:19.35 | Katty | excellent. |
| 18:20.02 | kenshin | Qwell, yeah I'll need to go through logs to see if they use any other ip's |
| 18:20.19 | Katty | kenshin: you can often call them and just ask for the ips or the range |
| 18:20.26 | Katty | kenshin: that way you don't miss one |
| 18:20.30 | kenshin | oh btw fax ? Send|Rcv] FAX work on sip ? |
| 18:20.39 | Qwell | use T.38 |
| 18:20.54 | Qwell | (if your provider supports it..) |
| 18:20.56 | p3nguin | Yes. But ulaw isn't nearly as good as T.38. |
| 18:21.02 | kenshin | one I can call the other I can't (bno support for free phones ) |
| 18:21.52 | p3nguin | I don't have any provider support for t.38, so I use ulaw with SIP. I have pretty reliable faxing this way. |
| 18:22.05 | kenshin | Qwell, I mean if those 2 work what the need for Hylafax for ex ? |
| 18:22.33 | Qwell | anymore? very little |
| 18:22.54 | kenshin | (basically can I send receive faxes just with asterisk , no add stuff -- fax machine, hylafax or the like) |
| 18:23.06 | Qwell | sure |
| 18:23.19 | Qwell | ~faxforasterisk |
| 18:23.19 | infobot | well, faxforasterisk is Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 18:23.27 | Qwell | Or there's res_fax_spandsp too. |
| 18:23.33 | kenshin | (I was browsing yesterday, & I read a lot about NVBackground detect and some other app) with I can't seem to find for 10.x.x |
| 18:23.39 | kenshin | are they still needed ? |
| 18:23.57 | beebeeep | recieveFAX/sendFAX are doing well last versions |
| 18:24.32 | p3nguin | Just use SIP's faxdetect setting. |
| 18:24.58 | kenshin | fax detect thing is need to run fax extn wight ? |
| 18:25.06 | kenshin | otherwise it does;t bother ? |
| 18:25.20 | Qwell | fax with Asterisk has come a long way in the last few years. |
| 18:25.36 | p3nguin | Yes. If you enable faxdetect for a peer, if fax tone is detected, the call will jump to extension 'fax' in the current context. |
| 18:26.10 | Qwell | even though it should've been deprecated 15 years ago... |
| 18:26.10 | Qwell | stupid lawyers, and banks, and doctors. |
| 18:27.04 | talntid | yeah |
| 18:27.07 | talntid | and state agencies. |
| 18:27.22 | kenshin | it does the same as email but email is better |
| 18:27.48 | *** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net) |
| 18:28.38 | *** join/#asterisk timahvo1 (~rogue@197.178.136.80) |
| 18:28.55 | kenshin | so asterisk is capable of handling fax the only prob is to spool them (for sending) or get them (hylafax is one of many to do just that ?) |
| 18:29.38 | p3nguin | What do you want to use hylafax for? |
| 18:31.00 | beebeeep | kenshin: what's the problem to recieve it? i'm usually recieving TIFF and sending it to email through local MTA using System() app |
| 18:31.21 | dijib | p3nguin: ever messed around with zoneminder? |
| 18:31.25 | p3nguin | Yes. |
| 18:31.32 | p3nguin | I use it. |
| 18:31.39 | dijib | ever had issue in getting a camera going? |
| 18:31.43 | p3nguin | No. |
| 18:32.02 | dijib | damn ive got an issue getting an old logitech usb camera going |
| 18:32.06 | kenshin | beebeeep, one way of doing it, I just exploring options here |
| 18:32.10 | p3nguin | Plug it in. Configure the source to the correct input. Monitor the feed. |
| 18:32.12 | dijib | kernel detects and loads driver |
| 18:32.21 | dijib | cant get zm to use it |
| 18:32.26 | p3nguin | What kind of card do you have? |
| 18:32.27 | dijib | gives signal errors |
| 18:32.37 | kenshin | p3nguin, queuing faxes (the only thing I used to know/used in the past ) |
| 18:32.38 | dijib | how re you by the way? |
| 18:32.44 | kenshin | (very distant past) |
| 18:32.48 | beebeeep | dijib: does it supports v4l ? |
| 18:32.51 | p3nguin | typical |
| 18:33.18 | *** join/#asterisk goddva (~glarsen@77.40.154.242) |
| 18:36.10 | dijib | yes beebeeep |
| 18:36.22 | dijib | your typical p3nguin ? |
| 18:36.27 | dijib | hows the truck doing? |
| 18:38.00 | p3nguin | My typical what? |
| 18:38.36 | kenshin | btw, if I'm only using sip (do I need the res of config files ?? ) or just sip.conf extension.conf and voicemail.conf ?) |
| 18:38.40 | p3nguin | Like... "That's your typical p3nguin. He's just a regular guy." |
| 18:39.31 | p3nguin | If you are only using chan_sip and only using voice mail and dial plan, then you might be able to get away with not using any res_*.conf files. |
| 18:39.46 | p3nguin | But I recommend that you configure whatever you need to configure for whatever features you want to use. |
| 18:40.06 | p3nguin | If you don't want certain things, remove the configs and noload the modules. |
| 18:41.41 | timeshell | Qwell: Do you deem that dnsmasq is sufficient to be used locally on an Asterisk server? |
| 18:41.52 | Qwell | yes |
| 18:41.59 | timeshell | And what would happen when TTL expires? |
| 18:42.01 | [TK]D-Fender | kenshin, There are a lot of other important config files that, if missing, will kill your server |
| 18:42.17 | [TK]D-Fender | timeshell, my guess : death. |
| 18:42.25 | kenshin | like? |
| 18:42.28 | timeshell | lol |
| 18:42.31 | timeshell | Qwell? |
| 18:42.37 | Qwell | timeshell: try it, let us know |
| 18:43.11 | kenshin | do I really need all of this : http://pastebin.com/Qks4GawK |
| 18:43.13 | dijib | you said typical after i asked how you were..... i assumed you were answering my query |
| 18:43.15 | timeshell | Will try it. Problem right now is office people have bad taste for our asterisk PBX due to probs like this. |
| 18:43.45 | timeshell | I've been trying to look at Asterisk based appliances from Digium, but if we end up with the same problem it'll be hard to convince them to go with it. |
| 18:43.45 | leifmadsen | get a better administrator :) |
| 18:43.55 | kenshin | sip yes (iax may be internally I haven;t decided yet, but the rest ?) |
| 18:44.19 | timeshell | Or a more solid product? |
| 18:45.09 | timeshell | I don't mean to knock *, I personally like using it. |
| 18:46.16 | p3nguin | leifmadsen: +1 |
| 18:47.06 | *** join/#asterisk gusto (~gusto@88.128.214.123) |
| 18:47.23 | leifmadsen | I mean to knock people, but it's funny that people will type out full sentences and thoughts then randomly through in a * instead of just typing out a word |
| 18:47.33 | leifmadsen | s/through/throw |
| 18:47.34 | gusto | hey |
| 18:47.55 | gusto | when i was a child, i had an ISA modem 33,6 K and that did Fax and Voice, so i could make calls through that one |
| 18:48.19 | gusto | could a modem be a DAHDI device for analogue phone line? |
| 18:49.01 | leifmadsen | no |
| 18:49.33 | gusto | no? |
| 18:49.50 | timeshell | asterisk |
| 18:49.52 | timeshell | asterisk asterisk asterisk |
| 18:50.00 | p3nguin | Asterisk, you say? |
| 18:50.03 | timeshell | Maybe |
| 18:50.04 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 18:50.13 | [TK]D-Fender | kenshin, You want IAX2 trunk mode? How about meetme? More configs. Want actual VOICE with calls arriving over SIP? Better have rtp.conf. Want * to load at all? Better have asterisk.conf and modules.conf. That's for STARTERS. |
| 18:50.45 | [TK]D-Fender | kenshin, You should probably take a REAL look at what's there and what uses it.... |
| 18:51.49 | leifmadsen | gusto: no. |
| 18:52.20 | leifmadsen | gusto: yes[1] [1] if you program your own driver that will interact with the particular modem you have -- non-trivial |
| 18:52.34 | gusto | ok ok, i have none |
| 18:52.40 | gusto | was only a question |
| 18:52.45 | leifmadsen | and I answered it |
| 18:52.55 | leifmadsen | you seemed to deny my first answer of "non" |
| 18:52.56 | gusto | that modem i had back then 33,6 that worked on win98 and one could do calls through it |
| 18:53.08 | p3nguin | You answered it too thoroughly. |
| 18:53.09 | gusto | well, i was not sure if you actually mean it |
| 18:53.19 | gusto | yes, or that too |
| 18:53.20 | leifmadsen | perhaps I wasn't clear... |
| 18:55.05 | gusto | so ... what do we do now? |
| 18:55.24 | leifmadsen | we dance |
| 18:55.27 | kenshin | [TK]D-Fender, for now I only need my 2 phones working like old POTS phones (for old ppl that do no know anything about anything), nothing really fancy |
| 18:55.50 | [TK]D-Fender | Want your calls to be able to RING to your caller? There's indications.conf |
| 18:56.02 | [TK]D-Fender | kenshin, Go really look. |
| 18:56.09 | [TK]D-Fender | I'm suspecting you haven't so far |
| 18:59.46 | *** join/#asterisk itbroke (hater@lolunix.org) |
| 19:03.15 | itbroke | Hello, I'm having some problems getting the cisco 7960 to work with asterisk. The phone registers fine and outbound calls work fine. Inbound calls however do not reach the phone. Has anyone experienced this problem before? Thanks for any help in advance. |
| 19:03.26 | p3nguin | Using SCCP or SIP? |
| 19:03.29 | itbroke | SIP |
| 19:03.40 | p3nguin | Check your dial plan. |
| 19:04.28 | itbroke | I set the sip debug on.... the error message was the congested busy... which yes is what happens when you don't have the extension defined. |
| 19:04.41 | p3nguin | Check your dial plan. |
| 19:04.45 | itbroke | but I thought I should see an invite to the phone... which doesn't happen |
| 19:04.52 | itbroke | ok I will re-review the dial plan thanks p3n |
| 19:05.17 | kenshin | [TK]D-Fender, I played around so far only with sip.conf extention |
| 19:05.21 | p3nguin | Feel free to pastebin extensions.conf |
| 19:05.22 | kenshin | and voicemail |
| 19:05.23 | p3nguin | ~pb |
| 19:05.24 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:15.37 | kenshin | (I'm rebuilding asterisk with span support) http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Fax_id263810.html speaks about res_fax_digium.so is the same as res_fax.so ? |
| 19:15.51 | Qwell | no |
| 19:16.09 | Qwell | res_fax_digium is a FAX technology provider. res_fax is the API it uses |
| 19:16.10 | p3nguin | Both modules should be used. |
| 19:16.43 | p3nguin | And spandsp isn't needed if you use those two modules. |
| 19:17.14 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 19:17.52 | kenshin | I don;t have fax_digium |
| 19:18.00 | Qwell | ~faxforasterisk |
| 19:18.01 | infobot | rumour has it, faxforasterisk is Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 19:18.55 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 19:21.22 | *** join/#asterisk albertoandrade (~albertoan@186.206.7.82) |
| 19:21.52 | FLeiXiuS | I'm having an issue with DTMF with one of my xlite4 clients. |
| 19:21.55 | jrose_atDigium | infobot uses the British spellings of things apparently. |
| 19:22.03 | FLeiXiuS | Some of my clients can pass DTMF and others cant. Any ideas? |
| 19:22.28 | *** join/#asterisk cbdev (~cbdev@hieristdas.internetzuen.de) |
| 19:23.32 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 19:24.47 | *** join/#asterisk din3sh (din3sh@41.136.247.125) |
| 19:27.05 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:30.15 | jrose_atDigium | FLeiXiuS: Are there any patterns you can identify for users that can and can't? |
| 19:30.27 | jrose_atDigium | certain phones or channel drivers used for instance |
| 19:31.17 | jrose_atDigium | Are they all xlite clients? |
| 19:32.55 | jrose_atDigium | First thing you should do if they are all xlite using SIP is probably tell which method for sending DTMF you are using. |
| 19:34.08 | FLeiXiuS | jrose_atDigium, All xlite, RTP is working. I can send/receive audio. Just no DTMF |
| 19:34.30 | FLeiXiuS | Im wondering if its a group policy setting / firewall issue. But that hardly seems possible due to audio coming and going |
| 19:35.23 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 19:37.49 | jaytee | [TK]D-Fender, do you have a second? |
| 19:38.32 | kenshin | FLeiXiuS, (codecs may be and dtmf mode ???) |
| 19:39.16 | kenshin | last time I run into this it was the ATA doing something funny (default config was use in band even though it was using g729 ) |
| 19:44.47 | kenshin | Thanks Qwell (I'll try it eventually) for now I'll go with spandsp.so (less of a hassle to install), but is there a fundamental diff ??? |
| 19:45.10 | kenshin | (by less of a hassle I mean USE=span; emerge asterisk :)) |
| 19:46.06 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 19:46.49 | jaytee | anyone using Polycom phones with UC SIP 3.3.0 or higher? In the provisioning Polycom says to enter the IP of the phone in the reg-basic.cfg file in the reg.1.address. In previous versions I have not had to enter this info as all the phones were setup using DHCP from Windows server passing the Option 66 boot address of the Asterisk server running vsftpd and they were all set to dynamic in |
| 19:46.49 | jaytee | SIP.CONF in Asterisk. Is that field mandatory or can I just fill in the user and password and then fill in the voIP.server.1.address in sip-basic.cfg |
| 19:47.47 | Deeewayne | anyone have much luck w/ the latest Asterisk 1.8 and SIP TLS SRTP ? |
| 19:51.20 | jeffspeff | jaytee, why are you using the UC firmware instead of the SIP firmware? |
| 19:52.55 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:53.10 | [TK]D-Fender | jaytee, 66 is the boot server for provisioning, not the SIP SERVER |
| 20:00.28 | jaytee | [TK]D-Fender, I know. I'm saying that I use DHCP from a Windows server and that the Option 66 is set to the provisioning server in the DHCP scope options there. |
| 20:00.31 | *** join/#asterisk albertoandrade (~albertoan@186.206.7.82) |
| 20:01.37 | jaytee | jeffspeff, The only available downloads for SIP 3.3.1RevF firmware are the UC files. I don't see any of the older SIP firmware files for that version available. The phones the client ordered are 335's and came with 3.3.1RevF. |
| 20:02.06 | jeffspeff | jaytee, what phone model? |
| 20:02.19 | jaytee | Polycom Soundpoint IP 335 |
| 20:03.28 | jeffspeff | jaytee, have you checked this out? http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 20:05.16 | jaytee | on the Polycom model 321/331 phones I got in a month ago that came with 3.2.4RevB the SIP firmware files came with the old phone1.cfg file, etc and when I edited the config files for each phone I didn't put an IP address in the reg.1.address field like the updated guide to provisioning says to put in the reg-basic.cfg file |
| 20:05.22 | [TK]D-Fender | jaytee, The UCS breakout takes a little getting used to. I've done it maybe once or twice .... |
| 20:05.53 | jaytee | I just don't want to have to setup the phones with static IP's. That would suck on a large installation. |
| 20:06.31 | jeffspeff | jaytee, i'm using IP335's and firmware 3.2.6 right now and i don't have to define any phone IP's in any of my files |
| 20:06.52 | jeffspeff | but i'm using SIP firmware not the UCS |
| 20:07.30 | jaytee | jeffspeff, yes but it all changed in 3.3.0 apparently. |
| 20:07.54 | jeffspeff | jaytee, so why not just use the 3.2.6 then? |
| 20:08.11 | jaytee | according to the matrix I could downgrade to 3.2.4 or 3.2.6 but my understanding is that voids the warranty. |
| 20:08.55 | jeffspeff | the IP 3335's ship with 3.2.4 on them, i don't see how reverting to original firmware could void the warranty |
| 20:08.56 | leifmadsen | I think you can still override it per cfg file |
| 20:09.36 | jaytee | my client purchased 3 of the 335's and they all came with 3.3.1RevF |
| 20:10.30 | jaytee | leif, I know. Actually I got the original provisioning setup I've used in the past from a really great book on Asterisk written by three crazy guys. :-) |
| 20:11.26 | jaytee | I'll call my voip vendor and ask them to verify if I downlevel whether it voids the warranty. |
| 20:11.52 | jaytee | but first I guess I'll just read more of the Admin 4.0 guide |
| 20:12.12 | jaytee | s/4.0/3.0 |
| 20:15.43 | leifmadsen | jaytee: :) |
| 20:15.57 | leifmadsen | I'm going through the 3.2 -> 3.3/4.0 transition now and it's a bit of a pain |
| 20:16.07 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 20:16.13 | leifmadsen | configs have change a fairly significant amount |
| 20:18.24 | jaytee | yeah, they sure have. I'm going to have to rewrite script file I wrote to automate creating the provisioning files. |
| 20:18.41 | leifmadsen | been doing that myself |
| 20:18.56 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:23.24 | [TK]D-Fender | checkout time, BBL |
| 20:23.37 | jaytee | leif, do you use static ip addresses when you setup phones? |
| 20:25.45 | *** join/#asterisk enzo (~enzo@extranet1.source-rh.com) |
| 20:25.51 | enzo | hello |
| 20:27.30 | enzo | I've very old pci cards (zaptel) for FXO and FXS, I'd like to buy a new one (all in one if possible) |
| 20:27.42 | Katty | my mind has just been blown by quantum gravity |
| 20:27.43 | *** join/#asterisk iheffner (~codon@iheffner.com) |
| 20:27.56 | Katty | i have not now re-think my entire existance |
| 20:28.47 | jeffspeff | Katty, what are you talking about? |
| 20:28.50 | jaytee | wow, that kinda sucks |
| 20:29.02 | enzo | I've seen the TDM410P and adding some modules, what do you think for that ? |
| 20:29.09 | Katty | specifically? |
| 20:29.13 | Katty | 4dimension beings watching me. |
| 20:29.32 | Katty | i now wonder if other objects are 4 dimensional |
| 20:29.39 | jaytee | wishes he had some of what Katty's been smoking. |
| 20:29.55 | Katty | i've been smoking SCIENCE! |
| 20:30.05 | enzo | I'm in France, I'd like to connect to 2 anagogic lines and to have 1 or 2 phones connected, so TDM422 or something like that, |
| 20:30.11 | enzo | => any feedback from those cards ? |
| 20:30.22 | jaytee | SCIENCE? Is it green and does it smell "skunky"? |
| 20:30.34 | iheffner | does anyone know of a tool that allows me to write tests against an AGI script and a check responses in a manner similar to how SIPp can test/debug SIP signalling issues? |
| 20:31.29 | jrose_atDigium | iheffner: You might be able to apply some of the agi stuff in the Asterisk test suite. |
| 20:31.52 | jrose_atDigium | I'm not really sure since I don't work a great deal with AGI. |
| 20:32.42 | Katty | 2d, a sheet of paper. |
| 20:32.56 | Katty | 3d...passes through 2d paper slice...the 2d people see....a slice of your midsection |
| 20:33.07 | Katty | 3d, picture you in a room. |
| 20:33.19 | Katty | 4d, picture a TON of rooms, stacked side by side...different realities |
| 20:33.41 | Katty | like you can see everything going on the 2d paper...the 2d peeps only see a slice of you |
| 20:33.56 | Katty | so a 4d object wandering across the 'rooms', you can only see a slice of their midsection as it passes through |
| 20:34.04 | jaytee | depressing, the orange *geek tshirt I got from my Advanced Asterisk class back in 2008 is getting really ratty and they don't have a replacement in the online Digium Store. |
| 20:34.07 | Katty | and the 4d object can see each of these little room realities |
| 20:35.03 | jrose_atDigium | jaytee: The Asterisk shirts are really just meant to be a generic swag thing. I don't think we do any actual merchandising :P |
| 20:35.53 | iheffner | Katty: consider this: time is a 4th dimension. a 4th dimension being would able to see you as an object from before your conception through all of your offspring. |
| 20:35.56 | jrose_atDigium | I doubt anyone would get upset if you had threadless or something screen print a shirt with the Asterisk logos and such. |
| 20:36.06 | jrose_atDigium | I could be wrong though, that isn't my legal opinion. |
| 20:36.09 | Katty | shape is the 4th dimension, dear |
| 20:36.16 | Katty | but time is up there, too |
| 20:36.33 | Katty | shape being a whale swimming through our reality boxes |
| 20:36.40 | _Corey_ | jaytee: Come to Astricon, get a t-shirt |
| 20:36.42 | Katty | we see a ..tailfin...or a tube...maybe a nose tip |
| 20:36.48 | Katty | but we don't really get the big picture |
| 20:37.19 | jaytee | jrose_atDigium, I got a lot of cool swag when I took the class. I had Jared Smith as my instructor and I got to meet Mark Spencer, Danny Windham, Russell Bryant and Qwell. |
| 20:38.13 | jrose_atDigium | That's pretty nifty. I didn't know jsmith ever taught the training courses. |
| 20:38.44 | jaytee | he did then but I think his consulting business swallowed up any open time for teaching. |
| 20:38.58 | talntid | at least it swallows. |
| 20:39.05 | jeffspeff | oh dang! |
| 20:40.52 | jaytee | well, I'm done for the day. Gonna go find some really hoppy ale and get "unsober" for a bit. |
| 20:41.02 | talntid | hey |
| 20:41.03 | talntid | wait |
| 20:41.06 | jaytee | what? |
| 20:41.10 | talntid | aren't you from dalnet #helpdesk? |
| 20:41.23 | jaytee | nope, never hungout on dalnet |
| 20:41.28 | talntid | ah, ok |
| 20:41.29 | talntid | :) |
| 20:41.34 | talntid | same nick, different person |
| 20:41.36 | jaytee | just freenode and blitzed |
| 20:41.42 | talntid | have a good one, John. |
| 20:42.00 | jaytee | ? how'd you know my first name? |
| 20:42.23 | jaytee | aha, did a whois |
| 20:42.23 | talntid | you just looked like a John |
| 20:42.27 | talntid | :) |
| 20:42.48 | jaytee | didn't realize my name showed up on a whois. Doesn't on blitzed. |
| 20:42.52 | talntid | ah |
| 20:43.11 | jaytee | well, have a great weekend everyone! |
| 20:43.58 | *** join/#asterisk enzo_ (~enzo12@extranet1.source-rh.com) |
| 20:49.06 | woleium | lo peeps :-) |
| 20:49.43 | woleium | Has anyone migrated from extension mode to device and user mode with a live system? |
| 20:50.53 | _Corey_ | woleium: Sounds like a #FreePBX question |
| 20:51.16 | woleium | you are probably correct :-) |
| 20:53.08 | *** join/#asterisk Holos (~cosmond@dyn-dial-mb-216-168-111-227.nexicom.net) |
| 20:53.14 | *** part/#asterisk Holos (~cosmond@dyn-dial-mb-216-168-111-227.nexicom.net) |
| 20:57.18 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 20:59.00 | FLeiXiuS | kenshin, Is there a way to force a DTMF codec for SIP users? |
| 20:59.21 | kenshin | (in my case I set it to auto, ) |
| 20:59.28 | kenshin | takes whatever is there |
| 21:01.33 | kenshin | (I mean on the client -- ata -- ) |
| 21:01.44 | kenshin | and in sip.conf |
| 21:02.03 | p3nguin | fleixius: In your own words, what is a DTMF codec? |
| 21:02.17 | *** join/#asterisk eicto (~eicto@144-71.dsl.aichyna.com) |
| 21:02.27 | FLeiXiuS | Meh I didnt mean code c ;-) |
| 21:02.42 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-eppcmwpcoyhpobdv) |
| 21:08.02 | Katty | GOOD NIGHT MY LOVLIES |
| 21:08.07 | Katty | i am off to go camping, and get drunk |
| 21:08.16 | Katty | and wander up and down the campground sing merry little songs |
| 21:08.28 | Katty | with marshmallows and amaraetto |
| 21:08.37 | _Corey_ | Katty: have fun :) |
| 21:08.43 | Katty | hugs _Corey_ |
| 21:08.45 | Katty | you too! :> |
| 21:08.47 | *** join/#asterisk darkcomposer (a6f80e26@gateway/web/freenode/ip.166.248.14.38) |
| 21:08.56 | leifmadsen | Katty: get drank for me! |
| 21:15.07 | carrar | and me too! |
| 21:15.16 | carrar | Thats some serious drinking you need to do! |
| 21:15.20 | talntid | and ME! |
| 21:25.33 | *** join/#asterisk mazpe (~lesterm@173.221.153.90.nw.nuvox.net) |
| 21:25.59 | timeshell | FLeiXiuS Are you referring to INBAND? |
| 21:32.27 | FLeiXiuS | timeshell, Im issue is I'm having soft clients not passing DTMF |
| 21:33.00 | FLeiXiuS | Using xlite. Not sure why, some clients work and others don't. All can send/receive audio - so RTP is working... |
| 21:33.14 | FLeiXiuS | Thats why I figured dtmf was getting compressed in an audio codec. |
| 21:33.35 | FLeiXiuS | So I forced rfc2833 on both ends. |
| 21:33.38 | FLeiXiuS | Still nothing. |
| 21:33.56 | bluregard | am I missing something or is Switch => not really documented anywhere? |
| 21:37.43 | bluregard | I'm only able to find bits and pieces here and there |
| 21:46.33 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 21:49.30 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ameyiinwoyoacdln) |
| 21:49.33 | *** join/#asterisk mazpe_ (~lesterm@173-9-155-49-BusName-Miami.hfc.comcastbusiness.net) |
| 21:54.42 | blizzow | I have a redfone, and am using a compiled copy of asterisk 1.8.7. 1. I start getting this error message flooding my logs that says "res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer clicks" It floods my logs at probably more than 40k lines per second and makes the logs grow to 50-250GB per day. Anyone know how I might fix this? |
| 22:06.03 | *** part/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 22:09.51 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 22:10.47 | kenshin | FLeiXiuS, what do these clients have in common ? |
| 22:12.22 | timeshell | FLeiXiuS Force INBAND on the XLITE? |
| 22:12.28 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 22:12.35 | timeshell | And in user settings in asterisk |
| 22:19.27 | *** join/#asterisk jlcaicedo (~jlcaicedo@186.146.216.79) |
| 22:21.52 | mazpe | An eternal fight trying to get Polycom phones to authenticate with my ntp server.. i'm getting the following from tcpdump: http://pastebin.com/vRq5weXN |
| 22:29.36 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:5cac:4f42:46db:8203) |
| 22:31.07 | mazpe | driving me insane! |
| 22:34.23 | mazpe | does it matter that the phones are running on a separate vlan? do i have to tell ntpd about the vlan? |
| 22:38.38 | jpsharp | ntpd has to be listening on an interface that is attached to that vlan. |
| 22:39.35 | jpsharp | But you don't have to specifically tell NTP about a vlan. |
| 22:41.21 | mazpe | jpsharp: well the the server has two interfaces eth0 is the public one and eth1.200 is what i use as a vlan for the phones |
| 22:41.35 | mazpe | same box is the pbx and where the ntp server is running. |
| 22:44.38 | jpsharp | The phones don't have any problem talking to services other than NTP? |
| 22:45.39 | mazpe | nope.. |
| 22:45.49 | mazpe | dial in and out just fine. |
| 22:46.38 | jpsharp | So most likely an NTP config file. Can you pastebin ntp.conf? |
| 22:46.39 | carrar | restart ntp and drop your filters |
| 22:47.45 | carrar | ntpq -c peers |
| 22:48.02 | jpsharp | And is your NTP synced to something? I have a recollection of my polycoms not NTP syncing until NTP was synced and became something besides stratum 16. |
| 22:48.07 | carrar | hopefully your * peer is st 1-4 |
| 22:48.42 | mazpe | jpsharp: http://pastebin.centos.org/38828 |
| 22:48.54 | mazpe | iptables is empty |
| 22:49.13 | carrar | the cli command? |
| 22:49.14 | mazpe | jpsharp: yeah it is synced |
| 22:50.13 | mazpe | carrar: http://pastebin.centos.org/38830 |
| 22:51.39 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 22:51.39 | *** mode/#asterisk [+o file] by ChanServ |
| 22:52.12 | carrar | remove all but the server lines, key and drifefile lines from your config and restart |
| 22:53.12 | carrar | also change your phone configs |
| 22:53.56 | carrar | actually jsut the driftfile and server lines is good enough |
| 22:55.21 | mazpe | the phone phone configs? |
| 22:55.33 | carrar | also from another host try ntpq -c peers IPHEREOFYOURPBX |
| 22:55.56 | mazpe | this is what i have on the sip.cfg |
| 22:55.57 | mazpe | http://pastebin.centos.org/38829 |
| 22:56.45 | mazpe | brb |
| 22:57.16 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 22:58.35 | carrar | so |
| 22:58.38 | carrar | also from another host try ntpq -c peers 192.168.1.254 |
| 22:59.41 | carrar | ntp over unix box using 802.1Q tagging works fine FYI |
| 22:59.51 | carrar | I also do that |
| 23:05.39 | *** part/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 23:12.51 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 23:14.16 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 23:17.12 | dijib | im back, i had to drive when i was here last |
| 23:17.15 | dijib | on the road |
| 23:40.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 23:47.29 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:51.06 | *** join/#asterisk twanny796 (~twanny@46.11.51.90) |
