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| 00:36.34 | trollboy | howdy I see where "asterisk -rx reload" will connect to the existing asterisk instance and open the cli and do a reload.. any way to do it with out getting the cli? I just want to load a new conf get a "Conf file is bad" or "all is good" |
| 00:36.59 | trollboy | as I'm writing an application to switch/modify conf's via a UI |
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| 00:53.36 | cxreg | die res_timing_pthread die |
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| 01:40.10 | *** join/#asterisk joncolby (~Joncolby@74.7.230.201) |
| 01:40.28 | joncolby | Hello |
| 01:40.38 | joncolby | I need help with a digit map |
| 01:40.44 | joncolby | can someone help me ? |
| 01:41.37 | joncolby | is anyone in here ? |
| 01:42.24 | joncolby | anyone ? |
| 01:42.45 | WIMPy | ~ask |
| 01:42.45 | 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. |
| 01:44.28 | joncolby | ok, I have two sip servers configured on my asterisk box. I have 4 digit extentions. extentions are limited to 1xxx - 7xxx |
| 01:44.40 | joncolby | there are no extentions that start with 8 or 9 |
| 01:45.30 | joncolby | i created two rules, one rule says that if the number starts with a 9 to use sip server 1, and rule 2 says if the number starts with 8 to use sip server 2 |
| 01:45.53 | joncolby | now, the rule for the number 9 works, but the one for 8 does not. |
| 01:46.29 | joncolby | as soon as you dial 8 and then 3 more digits, it tries to send it as a 4 digit extention and makes a busy sound. |
| 01:46.54 | joncolby | i can give screen shots of the configuration |
| 01:46.58 | joncolby | or copy and paste |
| 01:48.07 | WIMPy | screen shot? If you're using a GUI you'd better ask in the appropriate channel. |
| 01:50.07 | *** join/#asterisk mindCrime_ (~chatzilla@12.155.80.90) |
| 02:11.25 | joncolby | no gui, i was just going to take a screenshot of my screens text. |
| 02:11.51 | joncolby | basicaly i just need to get the default settings that apply to the number 9 as a prefix to apply to number 8 as well. |
| 02:12.27 | joncolby | i need it to stop trying to send 8xxx as an extention, and start sending the phone number out on the seccond sip server. |
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| 03:56.11 | thehar | ~book |
| 03:56.11 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 04:01.45 | joeyjones | dman, i still get weirdness from sipdroid |
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| 04:22.23 | joeyjones | weird, sipdroid seems to not send audio when receiving a call |
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| 04:36.35 | trollboy | anyone see my question? |
| 04:37.01 | trollboy | howdy I see where "asterisk -rx reload" will connect to the existing asterisk instance and open the cli and do a reload.. any way to do it with out getting the cli? I just want to load a new conf get a "Conf file is bad" or "all is good" as I'm writing an application to switch/modify conf's via a UI |
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| 05:35.56 | sizzers | hey guys... question about voicemail to email |
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| 05:37.49 | sizzers | my voicemails are ending up in gmail spam, which is related to the fact that the server is acting as a mail server via sendmail, and the messages are coming from "asterisk pbx" <asterisk@localhost.localdomain> ... |
| 05:37.56 | sizzers | is this correct? |
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| 05:46.20 | ChannelZ | probably |
| 05:54.03 | trollboy | I'm with ChannelZ on that one.. |
| 05:54.12 | trollboy | have you tried whitelisting that email address? |
| 05:55.13 | ChannelZ | or changing it |
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| 06:05.11 | sizzers | yah trollboy, i think i can whitelist it, but there should be some easy method to tell asterisk to simply act as an SMTP client and send the email through my normal mailserver right? |
| 06:05.15 | sizzers | it has to be that easy right? |
| 06:05.18 | joeyjones | ChannelZ: any idea why i ould be able get a 2-way echo test in sipdroid but when zoiper calls the extension for sipdroid i get no audio out of sipdroid? |
| 06:05.50 | trollboy | sizzers, lol |
| 06:06.10 | sizzers | lol trollboy... i thought not |
| 06:06.10 | ChannelZ | sizzers: with sendmail? nothing is easy with sendmail |
| 06:06.11 | p3nguin | sizzers: Configure your local sendmail to relay through gmail. |
| 06:06.32 | ChannelZ | p3nguin: sounds like it already is |
| 06:06.56 | sizzers | i think channelz's statement is a good prelude to this conversation thats about to happen |
| 06:06.57 | trollboy | ChannelZ, not properly if its coming from asterisk@localhost.localdomain |
| 06:07.11 | p3nguin | I thought he said it was sending email to a gmail address. I'm talking about using your gmail credentials and relaying through gmail. |
| 06:07.15 | ChannelZ | joeyjones: yeah, if it's doing a re-invite trying to get both peers to talk directly to each other, and they're not able to talk to each other correctly |
| 06:07.42 | sizzers | currently, it's not relaying... there appear to be some tutorials to make it do that,... but i don't think thats the route we want to go |
| 06:07.56 | p3nguin | It's easy to do with postfix, but i haven't even attempted with sendmail. |
| 06:08.19 | ChannelZ | trollboy: that has more to do with his host config being broken |
| 06:08.21 | sizzers | we've looked at doing it with postfix and ssmtp, but both appear to share the same over-complication |
| 06:08.39 | sizzers | sendmail certainly retaining the crown |
| 06:08.40 | ChannelZ | just uses msmtp |
| 06:08.58 | p3nguin | It should take you about five minutes to configure postfix to use your gmail credentials and set up a gmail relay. |
| 06:09.10 | ChannelZ | mailcmd=/usr/bin/msmtp -t -f asteriskvm@mydomain.com |
| 06:09.12 | sizzers | if i had ever configured postfix, or if it was installed on my servers, perhaps |
| 06:09.43 | sizzers | but i know absolutely nothing on postfix |
| 06:10.53 | sizzers | my frustration is this.... these are mail server agents that can be reconfigured to act as clients..... is there no simple native smtp client app... with a single config file... where i can put in a username, password, smtp-server, SSL yes/no, and launch with a simple 1-liner? |
| 06:11.01 | p3nguin | Maybe the Arch Linux wiki would help you, even though I'm sure you aren't using Arch. |
| 06:11.04 | sizzers | it should be that stupidly easy i think |
| 06:12.37 | ChannelZ | http://pastebin.com/cGYY1YSr |
| 06:12.49 | ChannelZ | Easy-peasy. Or you can complain the rest of the night |
| 06:13.33 | sizzers | thanks channelz |
| 06:13.41 | sizzers | where do i put the smtp credentials? |
| 06:13.50 | ChannelZ | 'man msmtp' for all the lovely bits you can put in the config |
| 06:14.19 | sizzers | you provided a config sample... i didn't know if htat was customized for me or what |
| 06:14.36 | ChannelZ | not unless you're really "mydomain.com" :) |
| 06:14.41 | ChannelZ | etc |
| 06:15.19 | ChannelZ | but that is a minimal config. In my case I send to an smtpd running on a different local server so it doesn't need auth |
| 06:15.40 | ChannelZ | but it's easy enough to add whatever you need into the config |
| 06:17.03 | ChannelZ | you might make a separate config file just for your asterisk to use if other people can login to that system; use the -C option and put it somewhere/change its perms such that only asterisk/root can read it |
| 06:19.24 | ChannelZ | (and actually you could probably pass everything you needed as args on the commandline, but you can decide whether you want to/whether its wise to for your case |
| 06:20.08 | sizzers | well... as soon as i find the yum repo hosting it i'll try it |
| 06:20.09 | sizzers | thanks man |
| 06:21.36 | ChannelZ | http://rpm.pbone.net/index.php3?stat=3&search=msmtp&srodzaj=3 |
| 06:22.18 | sizzers | i tried that... there's dependencies missing |
| 06:22.41 | p3nguin | Guess you should fulfill them. |
| 06:22.58 | ChannelZ | hmm. Well you could always build it, it can't be that complex |
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| 06:23.24 | sizzers | lol... u guys so crazy |
| 06:23.56 | shtrb | is there a way to call to sippura 2100 that did not do a register or proxy (p2p call) ? |
| 06:26.19 | ChannelZ | If you knew its IP or whatnot, probably |
| 06:27.06 | shtrb | I know the ip , but I can't find any listening open port (nmap) |
| 06:27.32 | ChannelZ | is it a SIP phone? It aught to be listening on 5060 unless it's been changed in the config |
| 06:27.35 | shtrb | so if it is possible it might have some setting that I can't find (the reason I caome here) |
| 06:27.56 | shtrb | its an ATA I know It should but for now it not |
| 06:29.03 | ChannelZ | well I dunno about that device specifically but I guess root around the config and see if it's turned off or something |
| 06:29.29 | ChannelZ | If it has multiple ports (FXO and FXS for instance) they are generally configurable separately, can be enabled/disabled, etc |
| 06:29.58 | ChannelZ | (you were looking for UDP with nmap, yes?) |
| 06:30.04 | shtrb | yep |
| 06:30.11 | shtrb | and tcp |
| 06:31.58 | shtrb | found it !!! |
| 06:32.33 | ChannelZ | splendid |
| 06:32.45 | shtrb | now the question is how to change the filtering .. |
| 06:32.49 | sizzers | jeeeebus... file /usr/share/info/dir from install of msmtp-1.4.6-1.i386 conflicts with file from package info-4.8-14.el5.i386 |
| 06:33.07 | ChannelZ | Aren't package managers great? |
| 06:33.18 | sizzers | eff my life |
| 06:35.06 | sizzers | wel... this appears to be the best place to start over |
| 06:35.07 | sizzers | http://wiki.mutt.org/?LightSMTPagents |
| 06:35.58 | ChannelZ | ./configure;make;make install too hard? |
| 06:36.08 | sizzers | yess |
| 06:36.15 | ChannelZ | but yes there are certainly others to choose from |
| 06:37.55 | sizzers | looks like esmtp is the winner here |
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| 08:12.07 | DelphiWorld | hi |
| 08:12.14 | DelphiWorld | anyone using a android phone with asterisk? |
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| 08:25.46 | DelphiWorld | any droidster? |
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| 08:28.44 | joeyjones | DelphiWorld: i've eben having shitty luck with getting sipdroid to be fully dunctional |
| 08:28.49 | joeyjones | *functional |
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| 09:43.34 | skrusty | morning all |
| 09:46.10 | adelas | hey |
| 09:46.16 | adelas | can someone enlighten me about T.38 faxing?I have Asterisk 1.4.36 connected to a T1 PRI line, a Fax Machine A connected ATA SPA2102 device that is T.38 faxing aware. So when MachineA sends fax -->SPA2102 encapsulates the T.30 fax into a VoIP T.38 fax -->sends to astererisk server --> Asterisk does something???, sends out to T1 Line -->...-> Fax MachineB Receieves fax. Where or how does the T.38 play? I see it being created but |
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| 09:58.37 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 09:59.09 | coppice | adelas: you need a T.38 gateway. asterisk doesn't do that by default, but if you look in the bugtracker there are widely used patches for it |
| 09:59.29 | adelas | well, its just a passthrough? |
| 09:59.49 | adelas | what i was trying to see is why do i even bother with the ata device that supports t.38 when my provider is a T1 PRI line |
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| 10:10.06 | coppice | if you want to go fax->ATA->asterisk->T1 you can do that just as a passthrough operation, provided you use a-law or u-law between the ATA and asterisk. however, the results can be quirky, because the timing of the ATA is not synchronised to the timing of the PSTN. T.38 between the ATA and asterisk decouples the timing, amongst other things, but the asterisk will need to support T.38 gateway... |
| 10:10.08 | coppice | ...operation |
| 10:10.34 | krion | hey guys |
| 10:10.50 | adelas | thanks, that sums up the answer i'm looking for |
| 10:10.50 | krion | i wonder if anyone face a trouble when holding a call |
| 10:11.39 | krion | scenario is, you receive a call, speak to the caller, hold the call, regain the call (sorry for my english) and the caller doesn't hear you |
| 10:12.16 | krion | witch pcap decoded of the trouble you can here both caller and calle |
| 10:12.32 | krion | so i guess a trouble when bridging channel ? am i wrong ? |
| 10:15.26 | krion | felle alone |
| 10:15.34 | krion | feel |
| 10:16.18 | adelas | umm |
| 10:16.24 | adelas | the call hang up? |
| 10:17.18 | adelas | i had issues where the calls would hang up, and that was b/c of a config setting in my chan_dahdi.conf |
| 10:17.33 | adelas | sometimes you would know, sometimes not |
| 10:17.38 | adelas | i changed busycount=3 to busycount=8 |
| 10:17.43 | adelas | and problem solved |
| 10:18.27 | adelas | not sure if you mean your still hearing in and out with the callee? |
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| 10:44.10 | krion | adelas: yeah, on the pcap i can hear both |
| 10:44.26 | krion | but they can hear each other using the phone :) |
| 10:44.32 | krion | they can't |
| 10:45.17 | krion | i should have done a show channel when it happem |
| 10:47.53 | krion | the things is there are too many potential factor |
| 10:48.13 | krion | the sip trunk provider, the phone, the asterisk, the session border controller... |
| 10:52.34 | krion | in fact since the rtp flux pass the trunk sip, the sbc, the phone it should be a trouble with my asterisk somehow |
| 10:54.45 | krion | there is transcoding between asterisk (g711) and sbc (g729) too |
| 10:56.26 | krion | [2010-11-29 11:52:50] DEBUG[29511] rtp.c: Ooh, format changed from unknown to alaw |
| 10:56.44 | krion | OoOoh :) Could it be that ? |
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| 11:14.13 | krion | i don't know why but i'm sure it's a trouble on how the bridge is handled |
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| 11:37.38 | krion | hum, in fact it can be on the sbc |
| 11:37.55 | krion | it's too hard to troubleshoot voip, smtp is much easier :) |
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| 13:10.11 | Aqituado | Hi, is there some what of setting a "cooldown" period for registrationattemps on a Extention ??? |
| 13:10.16 | Aqituado | i have random chinese IP's trying to broteforce their way into my box. |
| 13:10.50 | Aqituado | so if i could tell asterisk that it is only allowed to make one Registration attempt a second ;) |
| 13:10.59 | petern_ | try fail2ban |
| 13:11.01 | Aqituado | instead of the 60 attemps they are making. |
| 13:11.22 | Aqituado | petern_ ive read about it... but how long is a IP blocked then its blocked ? |
| 13:11.37 | petern_ | that's configurable |
| 13:11.41 | Aqituado | k |
| 13:12.09 | Aqituado | is it hard to set up ? cause im not exactly l33t ;) |
| 13:12.20 | petern_ | not particularly |
| 13:12.29 | Aqituado | k |
| 13:12.41 | petern_ | i followed http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk |
| 13:13.02 | petern_ | installed fail2ban using my package manager, then started from "Configure Fail2Ban" |
| 13:13.18 | Aqituado | would have been so easy if it was just an option on the extention :) like: "timeBetweenAttemps=(given in Seconds)" |
| 13:14.07 | petern_ | but you want to ban based on source host, not extension |
| 13:14.19 | *** join/#asterisk ukine_work (~ukine@152-144.97-97.tampabay.res.rr.com) |
| 13:14.24 | Aqituado | well. they tend to focus on same extention numbers |
| 13:14.32 | Aqituado | and i only have 6 extentions =) |
| 13:14.55 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 13:14.56 | Aqituado | so if i set max 1 registration attempt per sec (per extention) they should be hold back |
| 13:14.58 | petern_ | they generally try lots of a extensions with common passwords |
| 13:15.09 | petern_ | and then try some random extensions with lots of different passwords |
| 13:15.31 | petern_ | you really don't want to bother generating traffic for each request, so it's better to just filter them all out completely |
| 13:15.36 | Aqituado | i can se in my logs that they start by trying to identify wich extentions are used :) |
| 13:15.52 | Aqituado | then try with passswords more intensly |
| 13:16.02 | petern_ | blocked 3GB of incoming traffic the other day... |
| 13:16.02 | Aqituado | ill try to give it a go with fail2ban |
| 13:16.07 | Aqituado | damn ;) |
| 13:16.16 | petern_ | 6 million packets, heh |
| 13:16.34 | Aqituado | i only got 1.2GB of full log in 3 month . |
| 13:16.53 | Aqituado | most is registrationattempts. |
| 13:16.57 | petern_ | this is network traffic, not log size, heh |
| 13:17.01 | Aqituado | hehe =) |
| 13:17.09 | Aqituado | seems like they realy want to play with you. |
| 13:17.23 | Aqituado | so i should be quite happy with my low traffic ;) |
| 13:17.45 | Aqituado | but i gues i should be doing this filtering sooner rather than later. Cause im switching to a Fiber connection where i dont have Flat Rate. |
| 13:17.55 | Aqituado | so i gues its better in the long run |
| 13:18.11 | russellb | I just set up fail2ban a few days ago, I found it really easy actually |
| 13:18.16 | petern_ | well, the traffic still comes in to you, unfortunately |
| 13:18.28 | petern_ | but you still save a shed load not having to send out failure replies |
| 13:18.40 | Aqituado | but dont they stop when they realice that you block them ? |
| 13:18.54 | petern_ | sometimes, sometimes not |
| 13:18.58 | Aqituado | <PROTECTED> |
| 13:19.20 | Aqituado | well then i gues i should be gratefull that i have ~100 IP's i can move around on :D |
| 13:19.46 | Aqituado | so if on IP is fucked with incomming requests, then i can just move to a different ip |
| 13:22.26 | Aqituado | well with an alphanumeric password with special carracters with length 10, i should be quite safe (other than load on my box, and traffic) |
| 13:27.17 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 13:27.57 | *** join/#asterisk joachim_- (~joachim@post.comvie.no) |
| 13:29.06 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:29.22 | joachim_- | <PROTECTED> |
| 13:32.03 | joachim_- | I get that always after ReceiveFAX, it stores my faxes in TIFF files, but wont send to my mail.. Seems like everything stops after ReceiveFAX |
| 13:33.37 | viraptor | could someone tell me about running digium g729 codecs with multiple asterisk copies on one host - can I somehow split the licenses between the instances? will it work at all? |
| 13:34.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:36.29 | WIMPy | viraptor: I think you'd best ask Digiums support on that one. |
| 13:37.21 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:38.07 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:38.07 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:38.16 | viraptor | WIMPy: yeah, I know - just hoped to get a response from here today instead of waiting for official support to wake up ;) |
| 13:39.42 | *** join/#asterisk oDesk (~f@77.31.16.105) |
| 13:39.53 | oDesk | hello All |
| 13:40.28 | oDesk | my asterisk is behind NAT + dynamic DNS and everything works fine |
| 13:40.42 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:40.58 | oDesk | except that i can't register any device to the Asterisk locally if the internet connection is down |
| 13:41.07 | oDesk | what would be the problem ? |
| 13:41.53 | oDesk | anyone ? |
| 13:44.20 | joachim_- | <PROTECTED> |
| 13:44.25 | joachim_- | I get that always after ReceiveFAX, it stores my faxes in TIFF files, but wont send to my mail.. Seems like everything stops after ReceiveFAX |
| 13:45.17 | leifmadsen | oDesk: should be fine as long as the switch or router doesn't come into play. You may need to also make sure you have localnet enabled so that Asterisk doesn't try to connect to it as a though it were remote (on the internet) |
| 13:45.37 | leifmadsen | joachim_-: just looks like the dialplan fell through or the channel hung up |
| 13:46.12 | leifmadsen | joachim_-: you'd have to show the full call flow and dialplan logic in a pastebin to determine whether that is expected or not |
| 13:46.40 | leifmadsen | after the ReceiveFAX though if the incoming channel hangs up, then you'll have to process the emailing aspects somewhere other than after the ReceiveFax() application |
| 13:50.24 | oDesk | leifmadsen: i don't understand what you mean by the localnet |
| 13:50.31 | leifmadsen | joachim_-: please use a real pastebin and give the link to the channel -- I'm working on writing documentation right now and can't focus on your full problem right now |
| 13:50.39 | leifmadsen | oDesk: you're using SIP? |
| 13:50.47 | oDesk | leifmadsen: yes |
| 13:50.58 | joachim_- | leifmadsen: ok. will do |
| 13:51.03 | leifmadsen | oDesk: localnet is an option in sip.conf you should make sure is configured |
| 13:51.26 | oDesk | leifmadsen: i'll check, isn't it into sip_nat.conf ? |
| 13:51.41 | leifmadsen | oDesk: oh you're using a GUI based system -- sorry I'm not sure where it would be stored in that case |
| 13:52.11 | leifmadsen | oDesk: you might be able to get more help from #freepbx or #trixbox depending on what system you're using |
| 13:53.58 | oDesk | leifmadsen: i'm on freepbx yes, thank you for the hint |
| 14:04.41 | Katty | yawns |
| 14:04.46 | leifmadsen | Katty: STOP! |
| 14:05.13 | Katty | yawwwwwwwwwwwwwwwnnnnnnnnnnnnnnnnsssssssssssssssssssssssssss |
| 14:06.39 | chuckf | is suddenly tired |
| 14:09.25 | *** join/#asterisk Jasnejac (kvirc@81.91.107.235) |
| 14:11.21 | joeyjones | lol |
| 14:11.51 | leifmadsen | has 5 cups of coffee available to him |
| 14:12.01 | russellb | leifmadsen: still? |
| 14:12.26 | russellb | also, http://www.digium.com/en/company/2010-holidays.php |
| 14:12.31 | russellb | happy holidays, #asterisk ! :-) |
| 14:12.33 | leifmadsen | russellb: no, I drank 3 of them already |
| 14:12.34 | WIMPy | leifmadsen: You need them to go through the issues again today? |
| 14:12.56 | leifmadsen | WIMPy: NO! I finished all 88 new issues over 2 days yesterday! |
| 14:13.13 | leifmadsen | I still have 430+ emails to go through though for feedback :( |
| 14:13.21 | leifmadsen | so to answer your question, yes. |
| 14:13.36 | leifmadsen | poor russellb and malcolmd :) |
| 14:13.41 | WIMPy | I had the impression that it was even more the day before. |
| 14:14.37 | leifmadsen | I did 55 issues yesterday (new issues for triage) and about 35 new issues yesterday |
| 14:14.53 | leifmadsen | s/new issues yesterday/new issues the day before/ |
| 14:15.22 | WIMPy | And updated quite some others as well. |
| 14:15.30 | leifmadsen | nah, just focused on new issues |
| 14:15.39 | leifmadsen | I haven't reviewed items in feedback yet :( |
| 14:16.00 | leifmadsen | there will probably be 4-8 new issues for me today too |
| 14:16.17 | leifmadsen | WIMPy: we appreciate your patches and your feedback on issues though! |
| 14:17.13 | WIMPy | Not counting the two wring ones. |
| 14:17.27 | leifmadsen | meh, I've filed many issues that ended up not being issues :) |
| 14:17.44 | WIMPy | I guess so. |
| 14:17.47 | leifmadsen | I'd rather people file an erroneous issue that we have to close later, than to not file a real issue that we can't resolve |
| 14:18.02 | leifmadsen | or something lik ethat |
| 14:19.10 | WIMPy | Yes, I felt like having to open one for cahn_iax, but I wasn't able to find any usefull information on what was going on. |
| 14:19.51 | WIMPy | It _looks_ to me as if it was in the latest 1.6.2 versons but not in 1.8. |
| 14:21.31 | leifmadsen | WIMPy: well that's good then. 1.6.2 will go away soon enough :D |
| 14:21.44 | leifmadsen | if it was in 1.8, then I'd definitely want it looked at |
| 14:21.54 | leifmadsen | unfortunately chan_iax2 doesn't get as much love as chan_sip these days |
| 14:22.02 | malcolmd | yeah, but 1.6.2 has bug fix support until april, iirc |
| 14:22.21 | leifmadsen | malcolmd: right, but it depends how much information can be supplied and how important it is to resovle |
| 14:22.21 | *** join/#asterisk joachim_- (~joachim@post.comvie.no) |
| 14:22.48 | leifmadsen | malcolmd: and whether it can get resolved before april :) |
| 14:23.03 | malcolmd | word |
| 14:23.13 | WIMPy | Well, the effect was pretty bad. I had really bad or no audio at all. |
| 14:23.48 | leifmadsen | ah, then yes, I would consider that pretty bad, so if you had information on being able to reproduce it, that'd be ideal |
| 14:23.51 | WIMPy | Wat I observed was a single call eating almos 300kbit of bandwidth and two calls exceeding the available 1 mbit. |
| 14:24.03 | leifmadsen | if we can reproduce it, then it should be easy enough to resolve |
| 14:24.09 | leifmadsen | that's really the whole trick :) |
| 14:25.22 | WIMPy | I had it between (1.6.2.10-RCsomething or 1.6.2.14) and 1.8.0. Connections between 1.8.0 and (1.4.21 or Zoiper) worked however. |
| 14:25.58 | leifmadsen | malcolmd: ya, April 27 I think is the date |
| 14:26.05 | WIMPy | So I upgraded the 1.6.2 box to 1.8.0. |
| 14:26.09 | leifmadsen | :) |
| 14:26.34 | WIMPy | Well, no SVN actually, as 1.8.0 also had an iax bug that was resolved short after release. |
| 14:26.42 | leifmadsen | I think upgrading 1.6.2 to 1.8 is about the easiest major version transition we've had so far |
| 14:26.45 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 14:26.57 | leifmadsen | WIMPy: 1.8.1 will be out today that hopefully has that change |
| 14:27.20 | leifmadsen | and 1.8.2-rc1 later this week I believe |
| 14:29.02 | FlashDeluxe | Hi @ all! Should dahdi 2.4.0 work with asterisk 1.8.0 and junghanns quadbri? |
| 14:29.36 | WIMPy | FlashDeluxe: Yes, but TE mode only. For NT mode you need a patch. |
| 14:30.57 | FlashDeluxe | WIMPy: For gods sake i do not need NT mode for this installation :) |
| 14:31.27 | WIMPy | In that case going DAHDI should be the easiest choice. |
| 14:32.12 | FlashDeluxe | Thats the reason why i`ll take it :) |
| 14:32.27 | Katty | hmm. |
| 14:33.20 | Katty | i think i need a new boyfriend |
| 14:36.15 | Chainsaw | Katty: Are the terms & conditions on the current model not favourable? |
| 14:36.26 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-mymlppyfitbjqgfb) |
| 14:37.33 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 14:37.41 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:37.50 | Katty | Chainsaw: i think i got a defective model |
| 14:38.31 | Chainsaw | Katty: Ah yes. Return it while you still can. |
| 14:38.38 | chuckf | Katty: wait for the christmas gift before returning.... |
| 14:42.00 | Katty | heh |
| 14:42.06 | Katty | i'd feel bad. |
| 14:43.25 | joachim_- | http://forums.digium.com/viewtopic.php?f=1&t=76401&sid=0e5bcae18bb72836a3dc3b5438a3487c |
| 14:44.46 | Chainsaw | joachim_-: You need to use cat instead of nano. That $ cutoff looks a bit odd. |
| 14:45.30 | Chainsaw | joachim_-: It looks quite similar to what I have. I wait 2 seconds before ReceiveFAX and don't have a HangUp call though. |
| 14:46.13 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:46.13 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:46.56 | joachim_- | Chainsaw: My apologies for the nano cut, its not the issue anyhow.. So you think a wait is needed before receivefax? i tried to put a wait(3) after Receivefax. didnt work ofcourse |
| 14:47.08 | Chainsaw | joachim_-: Between Answer & ReceiveFAX, yes. |
| 14:47.18 | Chainsaw | joachim_-: I have Wait(2). Also, I don't have a HangUp call after ReceiveFAX. |
| 14:47.34 | joachim_- | Chainsaw: Thanks.Will try that right now! :) |
| 14:48.14 | Amnesia | could anyone here help me |
| 14:48.22 | Chainsaw | Amnesia: If you'd explain to us what's wrong, perhaps? |
| 14:48.27 | Amnesia | when I dial my line which is connected to the asterisk pbx |
| 14:48.48 | Chainsaw | Amnesia: This line. Is it analog? Does a telecoms company provide it? Which one? |
| 14:48.51 | Amnesia | I get a woman saying: congratiulation, you've succesfully configured asterisk blablabla |
| 14:49.05 | Amnesia | it;'s a digium analog card yes, dahdi |
| 14:49.50 | Chainsaw | Amnesia: Right. So this line works, and you enter the dial plan. |
| 14:49.54 | Chainsaw | Amnesia: What is the problem? |
| 14:50.13 | Amnesia | as I said |
| 14:50.32 | Amnesia | I keep getting the message that it's successfully configure, instead the phones connected to the pbx should ring |
| 14:50.55 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 14:51.14 | Amnesia | http://filebin.ca/bqvam/asterisk.rar |
| 14:51.27 | joachim_- | Chainsaw: Didnt help. It still wont send mail.. It continues to say: == Spawn extension (macro-mailfax, s, 5) exited non-zero on 'Local/4733501771@from-trunk-6fcc;2' in macro 'mailfax' |
| 14:51.30 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.151.106) |
| 14:51.55 | joachim_- | Chainsaw: s,5 is the ReceiveFAX line |
| 14:52.09 | Chainsaw | joachim_-: And your HangUp call is gone for the moment? |
| 14:52.17 | joachim_- | Chainsaw: yup |
| 14:52.34 | Chainsaw | joachim_-: exten => h,1,System(/etc/asterisk/scripts/mailfax.sh "${REMOTESTATIONID}" "${FAXNO}" "${FAXMAIL}" "${FAXFILE}" "${FAXBITRATE}" "${FAXRESOLUTION}") |
| 14:52.39 | Chainsaw | joachim_-: That's what I have. |
| 14:52.51 | Amnesia | Chainsaw: any idea? |
| 14:53.19 | Chainsaw | Amnesia: I believe your DAHDI line is in a different context. |
| 14:53.39 | Amnesia | I used dahdi_genconf.. |
| 14:53.57 | Chainsaw | Amnesia: Sure. The DAHDI inbound calls will end up in a different context in the dial plan then inbound calls from your internal handsets. |
| 14:53.57 | joachim_- | Chainsaw: i tried with exten => s,n .... going to try h,1 now |
| 14:54.17 | Chainsaw | Amnesia: For security, this is a *very* good thing. As you don't want to offer some Israeli guy with a cellphone free calling to South Korea. |
| 14:54.43 | Chainsaw | Amnesia: It does mean that you need to write your dial plan with that in mind. |
| 14:54.48 | Amnesia | hmkay |
| 14:54.57 | Amnesia | hah |
| 14:55.07 | Amnesia | but I selected context=incoming |
| 14:55.18 | Amnesia | ugh selected is configured* |
| 14:56.11 | Amnesia | http://pastebin.com/D6tBXNrd |
| 14:56.17 | Amnesia | that's dahdi-channels.conf |
| 14:56.19 | *** join/#asterisk bullium (~wbradshaw@216.68.250.30) |
| 14:58.25 | Amnesia | :/ |
| 14:58.33 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 14:58.52 | Chainsaw | Amnesia: Okay, so what extension are you calling? |
| 14:59.03 | Amnesia | incoming |
| 14:59.10 | Chainsaw | Amnesia: That's a context, not an extension. Please try again. |
| 14:59.24 | Amnesia | ah sorry |
| 14:59.44 | Amnesia | ehm I dunno tbh:/ |
| 14:59.56 | Amnesia | http://pastebin.com/NK8Ypaeu |
| 15:00.01 | Amnesia | that's my extensions.conf |
| 15:01.15 | Amnesia | any major screwups? |
| 15:01.59 | Chainsaw | Amnesia: You've issued dialplan reload since you changed that file? |
| 15:02.12 | Amnesia | yep |
| 15:03.02 | Amnesia | <PROTECTED> |
| 15:03.16 | Amnesia | that's all of asterisks config folder |
| 15:03.40 | Chainsaw | Amnesia: core set debug 10 |
| 15:03.43 | Chainsaw | Amnesia: core set verbose 10 |
| 15:03.55 | Chainsaw | Amnesia: Place your call again and have the console open. See what it does. It might even admit why. |
| 15:05.15 | Amnesia | k sec |
| 15:06.31 | Amnesia | http://pastebin.com/G382vtg3 |
| 15:07.04 | Amnesia | exten 's' o0? |
| 15:11.37 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:12.39 | kaldemar | Amnesia: exten s is fine, the contexts are not. your call is landing in a context dahdikaart, which your dialplan doesn't have. |
| 15:13.40 | Amnesia | hm I changed the context in chan_dahdi.conf |
| 15:13.44 | Amnesia | and now it's working |
| 15:14.20 | Amnesia | [TK]D-Fender pointed me on it |
| 15:14.44 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:15.34 | *** join/#asterisk Dovid (Dovid@213.8.118.62) |
| 15:15.48 | Dovid | how do i do show commmand FOO in 1.8.X ? |
| 15:16.03 | Dovid | nm |
| 15:17.25 | *** join/#asterisk Tim_Toady (~moi@188.4.87.53.dsl.dyn.forthnet.gr) |
| 15:20.30 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 15:21.13 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:23.51 | leifmadsen | Amnesia: going out DAHDI or SIP? |
| 15:23.56 | Amnesia | dahdi |
| 15:24.03 | leifmadsen | Amnesia: did you set the callerID? |
| 15:24.07 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:24.14 | Amnesia | hm nope |
| 15:24.14 | leifmadsen | Set(CALLERID(name)=Jimmy Pop) |
| 15:24.26 | leifmadsen | so asterisk is just using the default then, that's why you see 'asterisk' |
| 15:24.33 | Amnesia | but what if the number is someone I don't know |
| 15:24.34 | Amnesia | ? |
| 15:24.39 | leifmadsen | eh? |
| 15:24.43 | leifmadsen | you said when you place a call |
| 15:25.07 | Amnesia | yep |
| 15:25.12 | leifmadsen | then I don't understand your question |
| 15:25.23 | Amnesia | when I call the asterisk pbx with my mobile phone |
| 15:25.39 | leifmadsen | ok, ask your question again |
| 15:25.45 | Amnesia | the voip telephone connected to the asterisk pbx displays a called id of "asterisk" |
| 15:25.57 | Amnesia | caller* |
| 15:25.57 | leifmadsen | is the VOIP phone SIP? |
| 15:26.05 | Amnesia | ehm |
| 15:26.13 | leifmadsen | it likely is |
| 15:26.14 | Amnesia | Linksys Ip phone SPA941 |
| 15:26.16 | Amnesia | think so yes |
| 15:26.25 | leifmadsen | enable sendrpid=yes and trustrpid=yes in sip.conf |
| 15:26.38 | Amnesia | in general? |
| 15:26.40 | leifmadsen | yes |
| 15:26.53 | leifmadsen | you can do it per peer if you want too, but i Just enable it in general |
| 15:27.18 | *** join/#asterisk ctaloi (~ctaloi@nat-66-218-1-160.usadatanet.com) |
| 15:27.55 | Amnesia | keeps stating from: asterisk |
| 15:28.11 | Amnesia | and added it to the general section |
| 15:28.17 | leifmadsen | and you did a 'sip reload' ? |
| 15:28.26 | leifmadsen | what is the incoming callerID on the DAHDI line? |
| 15:28.31 | leifmadsen | Is asterisk getting callerID info? |
| 15:28.48 | Amnesia | I restarted the entire asterisk daemon |
| 15:28.58 | leifmadsen | does chan_dahdi have callerID functionality enabled? |
| 15:29.11 | Amnesia | [Dec 8 16:26:36] WARNING[5847]: chan_dahdi.c:8850 ss_thread: CallerID returned with error on channel 'DAHDI/1-1' |
| 15:29.16 | leifmadsen | check the console and add debug information using ${CALLERID(all)} function |
| 15:29.34 | leifmadsen | Amnesia: you're not getting callerID then, or it's coming in in a format asterisk doesn't understand |
| 15:29.37 | Amnesia | think so yes |
| 15:29.43 | Amnesia | http://pastebin.com/6aZbpb3A |
| 15:29.48 | Amnesia | thats chan_dahdi |
| 15:31.51 | Amnesia | http://pastebin.com/A1LZQcTQ |
| 15:32.18 | leifmadsen | Amnesia: you're still not showing what callerID is coming in on the channel |
| 15:32.23 | leifmadsen | I suggest you do something like: |
| 15:32.24 | leifmadsen | Answer() |
| 15:32.27 | leifmadsen | Wait(2) |
| 15:32.35 | leifmadsen | Verbose(2,Incoming callerID: ${CALLERID(all)} |
| 15:32.42 | Amnesia | thx |
| 15:32.53 | leifmadsen | with the correct closing parenthesis on Verbose() |
| 15:34.15 | Amnesia | hm, trying that now |
| 15:34.45 | Amnesia | hm |
| 15:35.22 | Amnesia | I tried using this as my extensions.conf: http://pastebin.com/v1vjuBWs |
| 15:35.29 | Amnesia | afaik the same as you said |
| 15:35.36 | Amnesia | but still getting the same in my console |
| 15:35.44 | Amnesia | (with debug and verbose set to 10) |
| 15:36.00 | Amnesia | my apologese if I'm dumb as fuck atm |
| 15:37.12 | leifmadsen | Amnesia: I want to see the result of you adding that, not the dialplan |
| 15:37.14 | Amnesia | also tried an extra parenthese after the Verbose line you stated, since I wasnt sure it hadnt had to be closed |
| 15:37.15 | leifmadsen | from the console |
| 15:37.45 | leifmadsen | I gave you the dialplan, I know what it looks like :) |
| 15:37.52 | Amnesia | http://pastebin.com/MKNnct98 |
| 15:38.01 | Amnesia | (A) |
| 15:38.10 | leifmadsen | Amnesia: you didn't reload the dialplan |
| 15:38.16 | Amnesia | I did |
| 15:38.17 | Amnesia | :o |
| 15:38.20 | leifmadsen | Amnesia: you gave me the exact same thing as before |
| 15:38.26 | leifmadsen | the dialplan I gave you is not executing |
| 15:38.43 | leifmadsen | the only reason would be because you didn't add it, or you didn't reload the dialplan |
| 15:39.05 | leifmadsen | Amnesia: or you did it wrong |
| 15:39.10 | leifmadsen | exten => s,n,Wait(2) |
| 15:39.11 | leifmadsen | etc... |
| 15:39.15 | leifmadsen | you forgot the 's' extension |
| 15:39.19 | leifmadsen | (and priority) |
| 15:39.36 | leifmadsen | and the priority numbers should be 'n' (other than '1') so you don't have to renumber |
| 15:39.41 | leifmadsen | what version of asterisk are you using? |
| 15:39.44 | *** join/#asterisk stope (~nobody@sud-cable-cmts3-69-60-242-213.vianet.ca) |
| 15:39.46 | Amnesia | I ah |
| 15:39.48 | Amnesia | sorry |
| 15:40.49 | leifmadsen | don't be sorry |
| 15:41.11 | leifmadsen | Amnesia: http://pastebin.ca/2013872 |
| 15:41.34 | Amnesia | wtf |
| 15:41.39 | Amnesia | now I'm getting the demo again |
| 15:41.46 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:41.56 | leifmadsen | Amnesia: http://ofps.oreilly.com/titles/9780596517342/ch05.html |
| 15:42.24 | leifmadsen | Amnesia: and http://ofps.oreilly.com/titles/9780596517342/ch09.html |
| 15:43.33 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 15:45.28 | Amnesia | finally-.-:P |
| 15:45.41 | Amnesia | <PROTECTED> |
| 15:45.44 | Amnesia | <PROTECTED> |
| 15:46.16 | leifmadsen | Amnesia: you're not getting callerID then -- perhaps your phone company hasn't enabled it |
| 15:46.29 | leifmadsen | Amnesia: perhaps you're not paying for that service on that line :) |
| 15:46.43 | Amnesia | so hm, could be it's the fax line |
| 15:47.04 | leifmadsen | I'd check with the phone company to see if CID is enabled. It likely isn't based on the info provided. |
| 15:47.13 | Amnesia | will do |
| 15:47.40 | Amnesia | thx for helping me out leifmadsen |
| 15:47.44 | leifmadsen | npo |
| 15:47.59 | leifmadsen | bill is in the mail |
| 15:48.15 | Amnesia | I'm a trainee @ this company and thought it'd be able to program |
| 15:48.25 | Amnesia | now I'm configuring asterisk:p:( |
| 15:48.30 | leifmadsen | asterisk is fun though :) |
| 15:48.35 | leifmadsen | you just need to read some more documentation |
| 15:48.39 | leifmadsen | what version of asterisk are they running? |
| 15:48.47 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 15:48.47 | Amnesia | hm well not really that motivated tbh |
| 15:49.16 | leifmadsen | asterisk takes motivation -- it's a huge toolkit :) |
| 15:49.30 | *** join/#asterisk jhirley (~chatzilla@c-75-74-13-194.hsd1.fl.comcast.net) |
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| 15:49.47 | Amnesia | 1.6.2.7 |
| 15:49.56 | Amnesia | yep |
| 15:50.07 | Amnesia | planning to set it up later at my own place though:) |
| 15:50.33 | Amnesia | but perhaps I'll try freeswitch then:) |
| 15:54.29 | brainiac | I have a Grandstream that won't transfer blind or in supervise mode. Any ideas? |
| 15:57.53 | asteriskATmarmuD | gxw? |
| 15:58.31 | asteriskATmarmuD | no idea, got gxw4008 but seems to work fine. I am not really using transfers, i think |
| 15:58.35 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:58.54 | brainiac | I don't know the model num I'm using |
| 15:59.07 | asteriskATmarmuD | what is it used for |
| 15:59.13 | brainiac | I have a customer who's machine it is |
| 15:59.16 | asteriskATmarmuD | are you connecting analog phones? |
| 15:59.25 | brainiac | yes |
| 16:00.10 | asteriskATmarmuD | problem may be, that the gwx itself is a pbx. you can call every connected phone from another phone on the gxw |
| 16:00.20 | brainiac | A Linksys on the same PBX works fine |
| 16:00.43 | asteriskATmarmuD | sorry, no clue |
| 16:01.09 | brainiac | ok, well thx anyway |
| 16:01.41 | asteriskATmarmuD | CLI output may help someone who is more experienced with asterisk + grandstream gxw |
| 16:01.50 | asteriskATmarmuD | good luck |
| 16:02.02 | brainiac | I've got that logged |
| 16:02.04 | brainiac | thax |
| 16:03.35 | WIMPy | Here isa howto for any issue: |
| 16:03.40 | WIMPy | http://xkcd.com/627/ |
| 16:10.12 | ectospasm | WIMPy: heh |
| 16:11.13 | Chainsaw | WIMPy: The IT secret! Stop divulging that information immediately! |
| 16:12.33 | WIMPy | I prefer less work. |
| 16:13.01 | JonnyD_work | brainiac: did you default the phone back to factory setttings? |
| 16:16.55 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
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| 16:20.06 | justdave | anyone know if there's a way on a Polycom 501 to temporarily change the default line button that's picked up if you just pick up the phone without hitting a button? |
| 16:20.26 | justdave | (hopefully something that doesn't involve a reboot to actually swap the buttons around or something) |
| 16:20.59 | justdave | this is just an end-user question at this point... got two lines, need to make several phone calls on the second one, and it's way too easy to forgot to switch lines before each call |
| 16:22.23 | leifmadsen | justdave: not that I'm aware of, think it always picks up line 1 |
| 16:22.56 | leifmadsen | best off just dialing the number with the handset down and hit line 2 before picking up the handset |
| 16:23.04 | justdave | yeah, that's what I'm suspecting... been poking around in the config panel on the phone itself for a bit and not finding anything that sounds like that |
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| 16:24.31 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 16:24.45 | justdave | I work from home, and have a Polycom on my desk in my office... line 1 is my work extension, line 2 is my home phone line. :) My wife is a girl scout leader and needs to make several scout-related calls... was going to be nice and let her use the office phone with the headset. |
| 16:24.55 | justdave | not sure if she could figure out to have to hit line 2 every time though :) |
| 16:25.19 | leifmadsen | justdave: I'd just reboot the phone then |
| 16:25.26 | leifmadsen | change the line order around for now |
| 16:25.36 | justdave | yeah, that's what I'm thinking. |
| 16:25.42 | justdave | it's not that hard |
| 16:25.43 | leifmadsen | you'd be done by now :) |
| 16:25.48 | justdave | swap the regs in the config and reboot :) |
| 16:26.11 | aimka | hi |
| 16:26.17 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 16:26.36 | aimka | is there a way to define a template for a category with a static relatime configuration ? |
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| 16:28.37 | pabelanger | aimka: no |
| 16:28.52 | aimka | snif, ok, thanks pabelanger |
| 16:29.17 | pabelanger | aimka: Ya, there are some limitation with realtime ATM |
| 16:30.59 | aimka | yes, there is always a limitation when it comes to realtime :-/ |
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| 16:47.53 | Spirits-Sight | what is the best database system to use in a voice-interactive / keypad type directory system? |
| 16:49.10 | *** part/#asterisk SaS1 (~Adium@ip-87-86-154-70.easynet.co.uk) |
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| 17:04.41 | Spirits-Sight | [TK]D-Fender: thanks for your assistance, its greatly appearated! |
| 17:07.09 | alphaz | hello people~, i was wondering if someone could help me out abit to figure out how to set callerid. i have 2 asterisk boxes. conntected by sip say A and B, B has the outbound lines, and A is internal. on A i set an extention for outbound caller id to say 000 0000, when that extention calls out, which goes through B, in B it always sets the callerid to the serverA.. i tried fiddling with the |
| 17:07.09 | alphaz | macro-user-callerid but it did nothing :\ |
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| 17:33.44 | yonahw | are there any guides on patching asterisk? I am experiencing a bug that was fixed upstream. Looking for a guide on creating a local patch especially gotchas and stuff. Do people usually patch or just download from head etc |
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| 17:39.58 | gnarf | so, I have a cisco 7975 here, and I'm getting a "Error Verifying Config Info" on the stats messages, is there any way for me to figure out exactly what that error is complaining about? |
| 17:40.10 | gnarf | the xml file validates in an xml validator |
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| 17:51.39 | trumee | anybody using asterisk on Asus RT N16 router. Can G729 codec be installed on it? |
| 17:51.42 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
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| 17:57.59 | brainiac | how can I drop calls from the CLI? |
| 17:58.17 | WIMPy | channel request hangup |
| 17:59.27 | sol326 | anyone have any suggestions on equipment that works well for a 4 line system either components or prebuilt box - ata to start using asterick?? |
| 18:00.22 | sol326 | interested in getting rid needing a phone company other than a service provider.... which I guess I can't get around :( |
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| 18:57.26 | sol326 | D-Fender: Thanks |
| 18:59.14 | cxreg | can you use res_timing_dahdi with no hardware support? |
| 18:59.45 | WIMPy | yes |
| 19:08.26 | alphaz | <alphaz> hello people~, i was wondering if someone could help me out abit to figure out how to set callerid. i have 2 asterisk boxes. conntected by sip say A and B, B has the outbound lines, and A is internal. on A i set an extention for outbound caller id to say 000 0000, when that extention calls out, which goes through B, in B it always sets the callerid to the serverA.. i tried fiddling |
| 19:08.26 | alphaz | with the macro-user-callerid but it did nothing :\ |
| 19:11.22 | *** join/#asterisk zigzapgabe (~gabe@c-71-57-35-169.hsd1.il.comcast.net) |
| 19:13.28 | *** join/#asterisk PoTe (~PoTe@rev-200-40-119-222.netgate.com.uy) |
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| 19:15.25 | PoTe | I need to call a web service implemented in a python library, this needs to happen as soon as a call is sent from a queue to the agent. what I need to know is how to get in the AGI the info about the agent that picked up the phone. |
| 19:15.51 | PoTe | What I am currently doing is inserting the AGI as an argument in the queue command |
| 19:16.32 | PoTe | But if I call the CHANNEL variable all I get is the zap channel that the caller is using, I cannot get the SIP channel being used by the agent. |
| 19:16.37 | PoTe | Any suggestions? |
| 19:17.25 | p3nguin | alphaz: Set(CALLERID(num)=1234567890) |
| 19:17.44 | alphaz | i want server A to set it. |
| 19:18.01 | alphaz | when someone from server A calls. i'm watching the log and it does do the SET(callerid(num)=1234567890 |
| 19:18.10 | alphaz | but on server b |
| 19:18.12 | alphaz | it never sees this |
| 19:18.42 | alphaz | because to server B, server A is just an extension. so it uses from-internal context |
| 19:18.44 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 19:18.46 | alphaz | which uses macro-user-callerid |
| 19:19.38 | WIMPy | alphaz: Then put it in to an apropriate context. And don't set a callerid in the sip peer. |
| 19:19.52 | alphaz | so i tried fiddling with macro-user-callerid |
| 19:20.11 | alphaz | and replaced the amp part where it tries to set it based on db to |
| 19:20.19 | alphaz | Set(CALLERID(all)="${CALLERID(name)}" <${CALLERID(number) |
| 19:20.23 | WIMPy | But when I read about that macro I get a feeling, you should ask in #freepbx or #trixbox. |
| 19:20.27 | alphaz | and that doesnt seem to do anything |
| 19:21.40 | p3nguin | I stopped caring when he tried to convince me that a second asterisk server was an extension. |
| 19:21.59 | alphaz | why not..? |
| 19:22.18 | WIMPy | Possible, but not sensible. |
| 19:22.34 | alphaz | well on server A its set as trunk. but on ser |
| 19:22.45 | WIMPy | And otherwise I'd go for IAX. That's what it was made for. |
| 19:28.02 | alphaz | well iax aside.. cuz i dont really wanna get into that right now.. trying to sip to work, but i mean clearly asterisk will send a callerid. when calling using a trunk. eg. i call from server B to provider, the provider takes in the callerid i set from server B.. so the callerID i send from server A to server B, clearly is sent.. its just that server B is ignoring it because .. well i guess |
| 19:28.02 | alphaz | i dont know waht the variable is that is sent to server B? |
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| 19:35.23 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 19:35.36 | eduzimrs | where is chan_sip.c in * 1.6 ?? |
| 19:36.25 | WIMPy | eduzimrs: Have you tried things like find, locate or maybe tree? |
| 19:36.28 | p3nguin | Did you consider using `find' to find it? |
| 19:37.59 | WIMPy | Actually it shouldn't even be neccessary when looking at what directories exist. |
| 19:38.11 | leifmadsen | it's in the channels/ subdi |
| 19:38.12 | leifmadsen | r |
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| 19:39.53 | alphaz | hmmm getting closer... i'm guessing its in something like.. ${SIPHeadersomething...... |
| 19:40.21 | eduzimrs | yeap´ |
| 19:40.28 | eduzimrs | i tried |
| 19:47.28 | alphaz | alright! 1 step away :| stupid cut function... |
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| 19:48.50 | justdave | hate to sound like a broken record (and don't want to be pushy about it, I'm just curious) but how's the 1.8.1 release coming along? :) |
| 19:48.51 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 19:49.36 | p3nguin | There's an RC, so it must be coming right along. |
| 19:50.35 | WIMPy | heard a rumour that it is to be expected later on today. |
| 19:50.45 | eduzimrs | wich dir is chan_sip.c in * 1.6 ??? |
| 19:50.55 | p3nguin | (1338.12) <@leifmadsen> it's in the channels/ subdi |
| 19:50.55 | p3nguin | (1338.12) <@leifmadsen> r |
| 19:51.44 | sol326 | I got one suggestion from d-fender but was hoping others might chime into who they use for ITSp? |
| 19:52.17 | p3nguin | ~itsplist-us |
| 19:52.17 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 19:53.08 | alphaz | Woo Hoo. finally works. |
| 19:53.48 | alphaz | hehe cut cut SIP_HEADER(FROM) sigh... |
| 19:54.42 | WIMPy | Do we give out prizes for doing easy things in the most complicated way? |
| 19:54.54 | alphaz | i hope so :D |
| 19:55.02 | alphaz | how is it "easy"??? |
| 19:55.08 | alphaz | is there an easier way to do this? |
| 19:55.10 | alphaz | non iax? |
| 19:55.25 | WIMPy | Configure the peer correctely. |
| 19:55.30 | WIMPy | But I already told you earlier. |
| 19:55.37 | alphaz | what do you mean by correctly? |
| 19:55.38 | p3nguin | no way! |
| 19:55.40 | sol326 | I'm wondering how .01c/m doesn't end up being more than the monthly rates...... anyone know?? |
| 19:56.04 | WIMPy | alphaz: Then put it in to an apropriate context. And don't set a callerid in the sip peer. |
| 19:56.13 | eduzimrs | i tried |
| 19:56.17 | eduzimrs | wich dir is chan_sip.c in * 1.6 ??? |
| 19:56.25 | p3nguin | Rates are typically .01 dollars per minute, not cents per minute. |
| 19:56.32 | p3nguin | (1350.46) <eduzimrs> wich dir is chan_sip.c in * 1.6 ??? |
| 19:56.32 | p3nguin | (1350.54) <p3nguin> (1338.12) <@leifmadsen> it's in the channels/ subdi |
| 19:56.32 | p3nguin | (1350.54) <p3nguin> (1338.12) <@leifmadsen> r |
| 19:56.36 | alphaz | i didnt set a callerid in the sip peer. |
| 19:56.43 | sol326 | right... |
| 19:56.52 | sol326 | what I meant |
| 19:56.57 | WIMPy | p3nguin: You don't give up that easily, do you? |
| 19:57.18 | WIMPy | Somehow today feels like monday in here. |
| 19:57.20 | p3nguin | It doesn't take a genius to look in the directory where he was told it was. |
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| 19:57.33 | p3nguin | It also doesn't take a genius to use find. |
| 19:58.04 | WIMPy | And it doesn't take a genious to read the answers one gets. |
| 19:58.30 | alphaz | doesnt take genious to read answers is true. but sometimes takes genious to understand the answers -_- |
| 19:58.49 | WIMPy | Not that one. |
| 19:58.50 | p3nguin | It shouldn't take a genius to spell genius, either. |
| 19:58.53 | sol326 | or at least experience with the terminology... |
| 19:59.19 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.80) |
| 19:59.55 | alphaz | well because i need it to be easily manageable in freepbx. thats why i need one side to be an extension.. |
| 20:00.42 | WIMPy | alphaz: And why not set them both as "trunk"? |
| 20:02.02 | alphaz | thats a good question. |
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| 20:02.40 | alphaz | probably because i dont want to make my own complicated context? cuz its probably not a 2-3 liner context.. |
| 20:02.53 | alphaz | again. with my limited knowledge.. thats why i did it like that |
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| 20:07.29 | bmoraca_work | i need to find a new wholesale sip provider (uhg). anyone have a suggestion, preferably someone you've worked with? |
| 20:10.49 | JerJer | heh |
| 20:16.04 | WIMPy | Is someone here using a hint on a confbridge? It doesn't seem to work any more. |
| 20:26.43 | sol326 | looks like bandwidth.com might be one of the better price structures... at least most open 5 numbers unlimited $30/m |
| 20:28.05 | sol326 | whenever companies start charging you for every feature it gives them an open door to manipulate the bill and you need to track their shit more... plus I'm seasonal so the few busy months are crazy phones going 12 hours a day constantly |
| 20:31.40 | bullium | I just downloaded the AsteriskNOW-1.7.1-x86_64 off of asterisk.org and the install basically just installs centos 5.5 from what I can see...I thought it was supposed to install centos asterisk and freepbx during the install? |
| 20:33.01 | *** part/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:34.47 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 20:34.47 | p3nguin | bullium: That's the idea. |
| 20:35.18 | p3nguin | bullium: You're looking at about 15 minutes from booting the CD to booting the installed system. |
| 20:35.32 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 20:38.01 | bullium | p3nguin: ok but asterisk and freepbx were not installed |
| 20:38.18 | p3nguin | Sounds broken. Did you follow the guide? |
| 20:38.51 | p3nguin | I've only had to install it once, and it went flawlessly. |
| 20:39.19 | bullium | LOL the screen on the guide never shows up...when the vm booted from the disk it simply booted the centos installer and then did the entire setup without any intervention on my part |
| 20:39.43 | p3nguin | I'm talking about the guide found on asterisk.org. |
| 20:40.03 | bullium | p3nguin: after it finishing installing it reboots and I'm sitting at a text based login prompt |
| 20:40.37 | p3nguin | Yeah. That seems okay to me. |
| 20:40.46 | p3nguin | Nothing wrong with that. |
| 20:40.53 | bullium | p3nguin: so am I, the quick start guid shows a nice boot splash screen with options, I'm never given a screen that looks like the one on asterisk.org's site |
| 20:41.08 | p3nguin | hmm |
| 20:41.16 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 20:43.51 | p3nguin | So you never saw this? http://www.asterisk.org/images/AsteriskNOW-1.7.0_Boot_Menu.png |
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| 20:45.00 | p3nguin | Did you verify the md5 sum of the file you downloaded? |
| 20:46.19 | bullium | p3nguin: nope, never saw it...I think it may be something with the VMWare workstation Easy Install feature |
| 20:49.06 | bullium | p3nguin: yep it was the easy install feature...disabled it and got the menu you linked too, thanks |
| 20:49.27 | justdave | sees 1.8.1 on the website |
| 20:49.41 | p3nguin | yep |
| 20:49.48 | JerJer | * Multiple fixes related to Local channels. <--- that scares me :) |
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| 21:03.00 | sol326 | itsplist-us |
| 21:03.06 | sol326 | ~itsplist-us |
| 21:03.06 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 21:08.54 | justdave | where do I find version number requirements for prereqs? |
| 21:09.12 | justdave | I have a couple things menuselect is claiming I don't have, which I actually do. |
| 21:09.21 | justdave | and I'm wondering if it's because the version I have is too old |
| 21:09.41 | justdave | and how I would tell what version it's looking for |
| 21:10.16 | WIMPy | Did you ./configure again? |
| 21:10.59 | justdave | they were already there before I ran configure |
| 21:11.12 | justdave | they weren't installed afterwards |
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| 21:14.26 | lordvadr | I'm having a tough time with some far-end hybrid echo. Raw tx/rx on the PRI are synced up perfectly so you don't notice it, but by the time the audio makes it to the end user, they do. Seems to me that the hardware (mg2) echocan on the card should be able to remove this but it doesn't. Anybody have any suggestions? |
| 21:16.40 | frigidzephyr | mg2 echocan on a card? what card is it? |
| 21:17.27 | lordvadr | Openvox, but I have a digium one as well. |
| 21:17.38 | lordvadr | or that's at least what proc says |
| 21:18.19 | lordvadr | is mg2 a software echo canceller? |
| 21:19.14 | WIMPy | yes |
| 21:19.43 | WIMPy | But as far as I understand it whatever is displayed has no meaning. |
| 21:20.02 | WIMPy | The only place that tells you if HWEC is working is dmesg after loading the driver. |
| 21:20.14 | WIMPy | And the lightshow inside the box. |
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| 21:23.03 | lordvadr | WIMPy: I'm not seeing anything. Any idea what it would say? dmesg|grep -i echo gives me nothing. I see where the module was loaded but it doesn't say anything about echo |
| 21:23.42 | frigidzephyr | WIMPy: with the digium boards you can tell if HWEC is working on an active channel by looking in /proc/dahdi/<span> |
| 21:24.44 | lordvadr | dahdi board: 1 WCT1/0/1 Clear (In use) (SWEC: MG2) |
| 21:24.50 | WIMPy | lordvadr: grep for VPM |
| 21:24.59 | lordvadr | openvox board: 1 D115/0/1/1 Clear (In use) (SWEC: MG2) (EC: VPM450M) |
| 21:25.12 | WIMPy | frigidzephyr: Need to remember that one. |
| 21:25.23 | frigidzephyr | the "(EC: VPM450M)" means hardware echocan is active on that channel |
| 21:25.35 | frigidzephyr | the SWEC shows what is configured in dahdi/system.conf |
| 21:25.55 | frigidzephyr | you'll also see EC: VPM150M for the VPMADT032 module on digium single span boards and analog boards |
| 21:27.12 | frigidzephyr | lordvadr: you can have software ec configured, but hardware will always override it if the board is not damaged or disabled via a kernel module param |
| 21:27.19 | WIMPy | And I think it makes sense to configure no EC. As far as I can see that has no effect on HWEC but otherwise a instance of te SWEC is initialised per channel. |
| 21:28.20 | lordvadr | ok, so I have hardware echo can on both boards. First question is what's the difference between the vpm150 (digium board) and vpm450 (openvox board)? |
| 21:29.05 | frigidzephyr | vpm150 on the digium board is probably a VPMADT032. I don't know anything about the openvox boards |
| 21:29.20 | lordvadr | Ok, next question is why doesn't it work? |
| 21:29.46 | lordvadr | It's definately hybrid echo, and I can hear it on the rx side of the PRI. |
| 21:30.09 | frigidzephyr | you get support with your digium boards, you can call them up and they will look into it |
| 21:30.42 | lordvadr | I thought it'd be a better learning experience to ask you guys. |
| 21:31.37 | lordvadr | or is it just not working well enough? How do I disable the hardware ec? |
| 21:32.23 | lordvadr | or could the software one be part of the problem? |
| 21:32.29 | frigidzephyr | for troubleshooting sake, you can enable and disable with vpmsupport=1|0 , passing that to the kernel module |
| 21:32.51 | frigidzephyr | the echo could just be crazy bad echo |
| 21:33.47 | frigidzephyr | make sure you are using the very latest dahdi drivers |
| 21:33.55 | frigidzephyr | as they have newer firmware for the digium echo can boards |
| 21:34.04 | frigidzephyr | 2.4.0 at least |
| 21:38.33 | lordvadr | well, these are in production so I have to do this all at night |
| 21:38.37 | lordvadr | thanks for the info |
| 21:39.01 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:39.01 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 21:42.34 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.1 (2010/12/08), 1.6.2.15 (2010/12/08), 1.4.38 (2010/12/08), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 21:42.52 | leifmadsen | Asterisk 1.8.1, 1.6.2.15, and 1.4.38 are now available. Please see http://www.asterisk.org for more information. |
| 21:43.03 | atan | ...what's new in 1.8.1? ^_^ |
| 21:43.39 | leifmadsen | atan: bug fixes |
| 21:43.52 | atan | There a list of them somewhere? =) |
| 21:44.01 | leifmadsen | atan: yes, look at the ChangeLog |
| 21:44.04 | atan | I ran into troubles with a few things, not sure if they are fixed or what =D |
| 21:44.11 | leifmadsen | that's what the ChangeLog is for |
| 21:44.20 | leifmadsen | check the download page for a link to the file |
| 21:44.32 | atan | ty http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1 |
| 21:44.55 | leifmadsen | yes I know the link :) |
| 21:45.14 | atan | ;-) good good |
| 21:45.22 | leifmadsen | I released it :) |
| 21:46.03 | atan | If I installed Asterisk on Debian with apt-get, would apt-get update it to 1.8.1? |
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| 21:47.59 | atan | Oh the hell. Scratch that totally. I have 1.4.x.... pfft! |
| 21:49.57 | leifmadsen | haha |
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| 21:56.42 | citywok | Hmmm. has anybody uses 1.8.0 w/ an Aastra handset succesfully? my aastra's won't receive calls. doing a sip debug it looks like * doesn't even try. |
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| 22:15.33 | Godfather_ | i'm having troubles with my tdm410p, i got in the cli this message ( ss_thread: Got event 17 (Polarity Reversal).. ) in incoming calls, and get a "piiiii tone" in the dialer handset... ? |
| 22:15.38 | citywok | hmm nevermind, freepbx is being stupid. |
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| 23:06.13 | peted20 | Anyone have a good recommendations for an FXO SIP Gateway that has a great hw echo canceller? Looking for 1-2FXO and 8FXO models. |
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