| 00:04.25 | telnettech_lap | ok so i want to check the database to see if a number exists in the blacklist family.....hpw exactly to check this in the dialplan? |
| 00:04.37 | telnettech_lap | do I want to use DB_EXIST? |
| 00:07.05 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:08.02 | pabelanger | telnettech_lap: func_odbc.so |
| 00:08.40 | telnettech_lap | this is the astDB |
| 00:09.05 | telnettech_lap | i dont have a seperate DB....just a very small system for the house |
| 00:11.08 | leifmadsen | telnettech_lap: I think there is a BLACKLIST() function isn't there? |
| 00:11.15 | leifmadsen | or PRIVACY() or something like that? |
| 00:11.40 | telnettech_lap | i got it....i finally found the note that i made to myself to get this working |
| 00:12.00 | telnettech_lap | facebook is a great notpad when you used multiple PC's |
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| 00:18.08 | E-bola | What has to be present for me to be able to enable res_timing_timerfd in make menuconfig? I thought i just needed a newer kernel (im on 2.6.36 and still cant enable it) |
| 00:20.15 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 00:20.52 | *** join/#asterisk angav (~angav@189.190.0.229) |
| 00:22.38 | angav | Hi eveyone; Centos 5.5/Asterisk 1.6.2/FreePBX 2.0.7.5: I want to know what audio codec is using a SIP phone since disallow=all and allow=gsm but phone does not support gsm and still works. |
| 00:23.18 | raden | who messaged me a long long time ago icon was flashing but history wont go back that far |
| 00:26.10 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 00:26.13 | E-bola | hmm i guess maybe this kernel was compiled without timerfd :/ |
| 00:26.25 | JunK-Y | yo! |
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| 00:29.48 | FuriousGeorge | are there any tricks in the voicemail.conf file to make email messages sent from my server (behind NAT) to look less like spam? |
| 00:30.00 | FuriousGeorge | i've already tried changing the FROM: line |
| 00:30.19 | FuriousGeorge | gmail detects it anyway, and even if I mark it as notspam, the next one goes straight to the spam box |
| 00:30.50 | E-bola | its more about ur mta on the server |
| 00:31.02 | E-bola | if u send a mail as root from the console, doesnt that look like spam as well |
| 00:33.53 | FuriousGeorge | E-bola: doesn't seem to go through at all |
| 00:33.57 | FuriousGeorge | let me try again |
| 00:35.06 | FuriousGeorge | E-bola: you're right, it does look like spam |
| 00:35.39 | FuriousGeorge | E-bola: so you think there are some settings i can change in sendmail perhaps that would make my mail look less spammish? |
| 00:35.50 | E-bola | no, i know there are |
| 00:35.57 | E-bola | and its not related to asterisk much |
| 00:36.09 | E-bola | so go read ur sendmail manual or ask sone sendmail ppl :) I use exim |
| 00:36.47 | FuriousGeorge | E-bola: thanks |
| 00:37.25 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:38.13 | chuckf | FuriousGeorge: one thing that you can do is have sendmail/postfix/whatevermta relay the mail from your box to gmail's smtp server and use that as the sending mta |
| 00:38.51 | FuriousGeorge | chuckf: in other words, configure sendmail to be an smtp client, and use another smtp server to send the message? |
| 00:39.05 | FuriousGeorge | perhaps i would use ssmtp instead of sendmail |
| 00:39.07 | chuckf | FuriousGeorge: yep |
| 00:39.31 | FuriousGeorge | chuckf: thanks |
| 00:39.37 | FuriousGeorge | chuckf: i think that |
| 00:39.41 | FuriousGeorge | 's my best bet |
| 00:41.33 | E-bola | for me all i had to do was make my mta use a real hostname as the sending hostname instaid of the hosts own non Full qualified name |
| 00:41.39 | E-bola | then it passed my filter |
| 00:42.02 | E-bola | instaid of sending from asterisk@asterisk1 |
| 00:42.10 | E-bola | it sends from asterisk@asterisk1.domain.com |
| 00:42.15 | E-bola | was all it took for me |
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| 00:43.43 | chuckf | E-bola: sometimes it is that easy. But for my needs I just relay to an outside smtp server |
| 00:45.11 | FuriousGeorge | E-bola: i think the problem with that might be my ISP and the fact that I don't have a real FQDN with return address here |
| 00:45.17 | FuriousGeorge | and maybe NAT |
| 00:45.36 | E-bola | if ur going to use it as an smtp server |
| 00:45.43 | E-bola | yoiu should have both a FQDN and a reverse dns |
| 00:46.01 | E-bola | unless ur only sending to ur own e-mail servers, in that case you can ofcourse just whitelist the asterisk sending ip |
| 00:46.22 | E-bola | but ur prolly better off setting up some sort of client to handle it, like chuckf mentioned |
| 00:46.28 | FuriousGeorge | E-bola: I can use a c-name to get a FQDN, but what about return address? my messages will still look spammy, no? |
| 00:46.50 | FuriousGeorge | E-bola: actually, i have dyndns setup already so that should count as a FQDN, no? |
| 00:46.54 | E-bola | FuriousGeorge: not as much, but yes all sending smtp servers should come from an IP with a reverse dns pointer |
| 00:47.13 | E-bola | FuriousGeorge: correct as long as u make ut smtp server use that domain name |
| 00:47.31 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:47.32 | FuriousGeorge | E-bola: and that is a sendmail setting, not in voicemail.conf, right? |
| 00:48.02 | E-bola | its an exim setting, so im guessing yes |
| 00:48.54 | E-bola | is sick and tired of crappy timing sources and virtualization |
| 00:49.08 | E-bola | Im going back to an oldfashioned normal server |
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| 00:56.29 | boodu | I have some problem to configure mISDN for my openvox b400e, I can use it but the moh don't work and i don't why but i can passe just one call (but receive 6) |
| 00:56.29 | *** join/#asterisk digitalirony (www-data@my.ass.looks.just.like.your-face.info) |
| 00:57.02 | boodu | why the moh don't work with my config :O ? |
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| 01:03.10 | telnettech_lap | guys.....any idea how to turn off the sip messages that are showing on the CLI? I tried sip set debug off command but i am still seeing SIP messages |
| 01:03.19 | telnettech_lap | i have the verbosity turned to 3 |
| 01:04.39 | ManxPower | telnettech_lap, remove debug from the console => line in /etc/asterisk/logger.conf, then do a logger reload |
| 01:05.04 | telnettech_lap | what about the sipdebug parameter in sip.conf |
| 01:05.12 | ManxPower | that must be new in 1.8 |
| 01:05.23 | telnettech_lap | yes i am using 1.8 |
| 01:05.37 | telnettech_lap | that looks to do it |
| 01:07.50 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:08.20 | telnettech_lap | having issues with the BLACKLIST function though....dont seem to have it setup correctly yet |
| 01:08.50 | boodu | I use that for my calls : exten => _0NXXXXX,1,Dial("mISDN/g:intern/${EXTEN:1}") |
| 01:09.03 | boodu | why this is limited at one call ? :0 |
| 01:09.57 | ManxPower | telnettech_lap, have you read the UPGRADE*.txt files? |
| 01:10.14 | telnettech_lap | i have yet to find it |
| 01:10.29 | ManxPower | try the doc/ directory of the Asterisk source |
| 01:10.42 | ManxPower | maybe the main asterisk source directory. |
| 01:10.51 | ManxPower | you won't get very far without reading it. |
| 01:11.52 | ManxPower | All the information you will find online will be 1.6.x or 1.4 specific and you will have no way to translate the information to apply to 1.8 unless you read those files. |
| 01:14.21 | *** join/#asterisk digitalirony (www-data@my.ass.looks.just.like.your-face.info) |
| 01:14.58 | *** join/#asterisk Marquel (~Marquel@84.16.240.149) |
| 01:22.14 | WIMPy | boodu: "intern"? Are you using NT mode tere? |
| 01:23.09 | boodu | WIMPy, no |
| 01:23.45 | boodu | i have <port mode="te" link="ptmp">1</port> type line in my mISDN.conf |
| 01:23.48 | WIMPy | Good. It did have a reputation for being rather unstable. |
| 01:23.58 | boodu | ok |
| 01:24.38 | WIMPy | And even if you only use TE mode, I'd suggest using either dahdi or lcr. |
| 01:25.10 | WIMPy | But if you want to continue with cahn_misdn, tell us, what happens. |
| 01:25.15 | boodu | I don't success use dahdi with my card :( |
| 01:26.15 | WIMPy | Isn't openvox one of the standard HFC-4S (or 8S) ones? |
| 01:26.23 | boodu | It seems to be recognize correctly but don't work |
| 01:26.41 | mmlj4 | hey ManxPower |
| 01:26.50 | ManxPower | hello mmlj4 |
| 01:26.57 | boodu | WIMPy, i think this is an hfc card |
| 01:27.23 | mmlj4 | I just interviewed to be the lead PBX tech in a shop that does oilfiled work, and is a CLEC to boot |
| 01:27.41 | ManxPower | cool |
| 01:28.06 | mmlj4 | they like me, I'm supposed to do a second interview with the owner |
| 01:28.12 | mmlj4 | so at least I have a shot |
| 01:30.55 | WIMPy | boodu: jep. That one sure looks like all others. |
| 01:30.57 | ManxPower | Stirling finished ripping out all the Cisco stuff I installed and replacing it with Cisco stuff? |
| 01:32.15 | JoeT | Anyone know how to get Asterisk to accept an extension number greater than one digit? |
| 01:33.01 | boodu | WIMPy, you think the openvox B400E can works perfectly with Dahdi ? |
| 01:33.17 | WIMPy | JoeT: By not specifying only one. But if you're looking for a more helpful answer, describe what you're trying to do. |
| 01:33.48 | boodu | I will try again with Dahdi |
| 01:33.53 | WIMPy | boodu: Yes, it should work with recent versions of dahdi, but only in TE mode so far. |
| 01:34.14 | JoeT | Trying to build an IVR menu option to dial an extension (3 digits) and transfer the call to that extension (and to vm if not answered). I'm using WaitExten, but it only accepts one digit. |
| 01:34.19 | boodu | ok, but with misdn why my moh doesn't work |
| 01:34.34 | WIMPy | I personally prefer to use LCR, but that's not fully 1.8 compatibel, yet. |
| 01:34.40 | boodu | and why can put just one call and receive multicall |
| 01:34.48 | boodu | LCR ? |
| 01:35.29 | WIMPy | boodu: I have no idea, but I already asked, you to describe what's happening. |
| 01:36.48 | WIMPy | misdn was replaced with misdn2 over two years ago. But there is no channel driver for Asterisk and misdn2. But you can use misdn2 with LCR and connect it to Asterisk via chan_lcr. |
| 01:37.12 | WIMPy | http://www.linux-call-router.de/ |
| 01:37.56 | ManxPower | JoeT, then you are doing something wrong. |
| 01:38.00 | WIMPy | Looks a littel complicated, but it's quite easy if you only use it as an interface. |
| 01:38.22 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:38.23 | ManxPower | JoeT, WaitExten will match as soon as it is sure there cannot be any matches by accepting more digits. |
| 01:38.31 | WIMPy | And the nice thing is that misdn2/lcr can be completely reconfigured at runtime. |
| 01:39.07 | boodu | moh on call works fine in intern, but when i receive a call from misdn it's ringing |
| 01:39.14 | WIMPy | In fact you cann even add new physical interfaces without disturbing anything running. |
| 01:39.25 | ManxPower | JoeT, You should either 1) not have IVR options that overlap with extensions or 2) make them select an option to dial an extension, then accept the extension outside of the IVR |
| 01:39.36 | a1fa | how different is Asterisk 1.8 vs 1.6? |
| 01:39.48 | WIMPy | boodu: I'm not sure, waht you mean. |
| 01:39.49 | ManxPower | WIMPy, other than the silly problem of sparks when you insert/remove the PCI card? |
| 01:39.52 | JoeT | OK, I figured out what I was doing wrong... Thanks. :) |
| 01:40.38 | WIMPy | ManxPower: There are boards that can handle that. And for the hobbyist you can use USB. |
| 01:43.37 | a1fa | shit.. how hard is it to go from 1.4 version to .16? |
| 01:43.41 | a1fa | err 1.8 |
| 01:44.46 | ManxPower | a1fa, it is all explained in the UPGRADE*.txt files included in every Asterisk source tarball |
| 01:44.56 | a1fa | no way |
| 01:44.57 | a1fa | :( |
| 01:45.10 | boodu | thx WIMPy to answer |
| 01:45.18 | boodu | I'm going to re |
| 01:45.31 | a1fa | i'll probably loose the voice changer features |
| 01:45.33 | a1fa | they come in so handy |
| 01:45.44 | ManxPower | why don't you read and find out |
| 01:45.51 | a1fa | i am dl now |
| 01:46.33 | a1fa | note to self |
| 01:46.39 | a1fa | backup config files |
| 01:46.43 | a1fa | kthnkxbye |
| 01:47.50 | *** join/#asterisk digitalirony (www-data@my.ass.looks.just.like.your-face.info) |
| 01:47.56 | ManxPower | ~toolkit |
| 01:47.56 | infobot | Remember, Asterisk isn't really a PBX: Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 01:49.16 | a1fa | how long for *Now to catchup |
| 01:51.24 | ManxPower | your statement fails to parse |
| 01:53.06 | ManxPower | If you have an AsteriskNOW question, then ask it on #asterisknow |
| 01:53.54 | telnettech_lap | ahhhhh!!!! finally success in finding what i did with the docs that are loaded with the tarball |
| 01:53.59 | telnettech_lap | now to read |
| 01:54.18 | a1fa | this is going to be a learning curve |
| 01:54.21 | boodu | Here my log : http://pastebin.com/aFC5EZcu when i try to put a second call |
| 01:55.49 | WIMPy | boodu: The other end tells you that tere is no channel available. |
| 01:57.26 | WIMPy | boodu: You don't try to call yourself there, do you? |
| 01:57.52 | boodu | no I don't call me |
| 01:58.15 | boodu | there are 3 ports isdn |
| 01:59.03 | boodu | I don't understand why it doesn't work |
| 01:59.40 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 01:59.59 | boodu | WIMPy, do you have some suggestion ? |
| 02:00.17 | WIMPy | As far as I remember "I IND" means received from the network. So that's where the cause comes from. |
| 02:01.22 | boodu | "I IND" ??? |
| 02:01.30 | WIMPy | Are those 3 seperate lines or one interface? |
| 02:01.45 | WIMPy | In the log you pasted. |
| 02:01.55 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 02:02.04 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:02.13 | boodu | in [intern] there are port 1,2,3 msns=* |
| 02:02.23 | boodu | 3 line |
| 02:02.33 | WIMPy | "I SEND" = physically transmitted, "I IND" = physically received. |
| 02:03.03 | boodu | 3 lines are connected at my card B400 |
| 02:03.45 | WIMPy | Unfortunaletly the debug output is not that specific. Have you tried to dial out on a specific channel instead of the group? |
| 02:04.06 | boodu | no |
| 02:04.11 | boodu | good idea |
| 02:04.27 | boodu | now i can't try but in few hours i can |
| 02:05.50 | *** join/#asterisk digitalirony (www-data@my.ass.looks.just.like.your-face.info) |
| 02:05.53 | *** join/#asterisk Marquel (~Marquel@84.16.240.149) |
| 02:06.09 | WIMPy | But I'd still suggest, you try some recent software. |
| 02:07.14 | boodu | i have dahdi 2.3.0.1 |
| 02:07.39 | boodu | and misdn 1.1.9.1 |
| 02:08.43 | WIMPy | dahdi sounds ok, but misdn was replaced for a reason. That's no just a name change like from zaptel to dahdi. |
| 02:08.51 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 02:10.19 | a1fa | that was a pain-free upgrade |
| 02:10.45 | coppice | misdn sounds a little like misery for a good reason |
| 02:11.14 | WIMPy | Yes, but that applies to the old version. |
| 02:12.00 | WIMPy | And I think it's rather senseless that it made it's way into Asterisk 1.8.. |
| 02:12.07 | coppice | they've been saying that since it was isdn4linux :-) |
| 02:12.31 | WIMPy | i4l worked |
| 02:12.32 | *** join/#asterisk ectospasm (ectospasm@188.72.223.139) |
| 02:12.33 | a1fa | hm |
| 02:12.42 | a1fa | how do i noload dundi? |
| 02:13.00 | WIMPy | Unless you tried to do voice with an HFC card. And I think that was never cared for. |
| 02:13.09 | coppice | i4l was a disaster |
| 02:13.24 | WIMPy | But I used i4l with HSCX cards for many years. |
| 02:13.48 | coppice | some some simple data applications it was OK. for anything else it was a disaster |
| 02:13.56 | a1fa | nvm |
| 02:13.57 | WIMPy | No. Except for the fact that HFC drivers were never completed it worked perfectely. |
| 02:14.18 | coppice | and how many voice calls did you make with i4l? |
| 02:14.43 | WIMPy | I used it for many years. |
| 02:15.14 | WIMPy | Until I had to ditch the HSCX cards for the lack of ISA slots. |
| 02:16.26 | WIMPy | It was mainly just for voicemail, but it worked for call back and call through as well. |
| 02:17.44 | a1fa | lol |
| 02:17.50 | a1fa | my * is segfaulting |
| 02:18.31 | coppice | when anyone says |
| 02:18.32 | coppice | I have DTMF issues using i4l/misdn/lcr, can you help |
| 02:18.34 | coppice | I have echo issues using i4l/misdn/lcr, can you help, or |
| 02:18.36 | coppice | I have FAX issues using i4l/misdn/lcr, can you help |
| 02:18.37 | coppice | people mostly ignore the question |
| 02:19.44 | WIMPy | I haven't senn any of these questions here. |
| 02:20.08 | coppice | we get them a *lot* |
| 02:20.14 | WIMPy | But DTMF detection is definitely over sensitive in the default configuration. |
| 02:20.28 | WIMPy | Who is we? |
| 02:21.26 | coppice | the DTMF detector within misdn is completely useless, and they have been really obstructive about putting a proper detector in it. even passing the audio through to a decent detector is very troublesome, as misdn can't provide a clean audio stream |
| 02:21.35 | WIMPy | I do rather have the impression that LCR is quite unknown. |
| 02:22.24 | WIMPy | Just to make that clear: Are you talking about misdn1 or misdn2? |
| 02:23.28 | a1fa | its amazing what stale modules do to the stability :) |
| 02:23.32 | coppice | i4l, misdn1, misdn2, nothing improves. isdn4bsd seems so much more reliable, its a pity nobody ever tried porting that to linux |
| 02:23.33 | a1fa | if you have autoload=yes |
| 02:25.26 | WIMPy | coppice: I'm not sure, you're really up to date there. If misdn2 still has troubles, they're hidden away VERY well. |
| 02:28.15 | coppice | they aren't well hidden from people using echo canceller, or FAX or DTMF |
| 02:29.46 | WIMPy | Well, I wouldn't know how to use fax with linux at all. |
| 02:30.19 | WIMPy | All the Asterisk soulutions also only support G3, which is usually not exactely what you want. |
| 02:30.38 | coppice | huh? what would you want other than G3? |
| 02:30.45 | WIMPy | G4 |
| 02:31.04 | coppice | hardly anything supports G4 |
| 02:31.19 | coppice | that's why I never bothered implementing it |
| 02:31.27 | WIMPy | All the fax servers and copiers do. |
| 02:31.58 | coppice | no they don't. G4 is ISDN FAX. the vast majority of FAX over ISDN is G3 or SuperG3 |
| 02:32.05 | WIMPy | Who do you exchange faxes with? Usually bigger companies and they most certainly have G4. |
| 02:32.24 | coppice | I've never actually seen a G4 machine |
| 02:32.31 | WIMPy | Not over here. |
| 02:33.26 | WIMPy | Just take any el cheapo Windows fax software. |
| 02:33.45 | coppice | nobody has ever asked my to add G4 support |
| 02:34.09 | coppice | can you name a Windows FAX package that supports G4? |
| 02:34.17 | WIMPy | Which fax solution are you working on? |
| 02:34.50 | WIMPy | The RVS com package should have one. Or Fritz fax. |
| 02:35.02 | WIMPy | But I'm not the windows guy. |
| 02:35.34 | coppice | Fritz fax does FAX in hardware, and transparently handles G3 and G4. You will rarely see one exchanging in G4 mode |
| 02:35.38 | WIMPy | But I guess I might some of those CDs somewhere that came with some hardware or the other. |
| 02:36.19 | coppice | adding G4 support is pretty easy, but nobody asks for it |
| 02:36.24 | WIMPy | Depeands. Some of the FritzCards are also just te standard HFC-S ones. |
| 02:37.41 | WIMPy | Maybe that's because they aren't even considerig Asterisk? |
| 02:38.39 | coppice | what the Linux FAX packages lack is SuperG3 (ie V.34 FAX), which is heavily used. Patent issues mean only a commercial solution is possible, but even FAX for Asterisk lacks V.34 support |
| 02:38.51 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 02:44.24 | WIMPy | Oh, and another thing that is missing in all variations is G.722. |
| 02:44.50 | WIMPy | That's a question I hear quite a lot. |
| 02:48.24 | *** join/#asterisk coppice (~chatzilla@m121-202-14-5.smartone-vodafone.com) |
| 02:59.00 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
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| 03:58.12 | Bloudermilk | Hi everyone. I was hoping somebody could clear up something for me regarding AMI |
| 03:59.04 | Bloudermilk | I'm reading the voip-info wiki page on AMI, and at one point it mentions "Only one action may be outstanding at a time". Then, about a paragraph down, it says " That way the client can easily match Action and Response packets while sending Actions at any desired rate without having to wait for outstanding Response packets before sending the next action" |
| 03:59.25 | Bloudermilk | It was my initial understanding that I can send out a stream of actions without waiting for their response |
| 03:59.29 | Bloudermilk | Is this true? |
| 04:00.36 | WIMPy | I have shoved requests into AMI without waiting. But I can't say if I was just lucky that it worked, as I only did that for testing. |
| 04:00.42 | titter | Is there a way to test/see why my CDR isn't writing to the DB? I see no errors in the console, do I need to disable the CSV file as well? |
| 04:00.55 | titter | DB* = mysql |
| 04:01.04 | WIMPy | But there might be a difference between acknowledgements and responses. |
| 04:17.53 | *** join/#asterisk delroy (~delroy@S0106001310246ce7.ss.shawcable.net) |
| 04:23.40 | boodu | re |
| 04:39.12 | *** join/#asterisk [hC] (~voxter@76.77.73.130) |
| 04:40.25 | [hC] | am i only just noticing that even if i am using g729 on both legs, if i have a SIP call that is connecting to an IAX call, a g729 transcode will still happen? |
| 04:41.26 | delroy | Hey, I am building an Asterisk system for my office and want to know what phones people like best... opinions? |
| 04:41.38 | [hC] | delroy: look into Aastra or Polycom |
| 04:42.50 | delroy | I like aastra as its a canadian company as am I, and i hear they keep more up to date on firmware. whats the sound quality like for each? same? |
| 04:42.54 | WIMPy | delroy: We don't know your taste nor your requirements. But I like Snom 3xx. |
| 04:43.43 | delroy | the only experience i have had so far is with linksys 942s and I liked them except the company needs more buttons |
| 04:43.53 | WIMPy | Well, the real cheap ones can have bad sound. |
| 04:43.58 | delroy | and an obvious transfer button |
| 04:44.12 | WIMPy | What about side cars? |
| 04:44.20 | [hC] | aastra 6731i/55i/57i are my favourites |
| 04:44.29 | [hC] | I am also canadian, and have been running an ITSP for 5 years |
| 04:44.39 | [hC] | Ive got boxes and boxes of old phones that didnt pass the test. |
| 04:44.47 | WIMPy | How many buttoons? With LEDs? |
| 04:45.09 | delroy | programmable - leds no necessary |
| 04:45.34 | delroy | which ones DIDNT pass the test and should be avoided |
| 04:45.56 | delroy | cobtw, from sask |
| 04:46.14 | WIMPy | Allnet is shit and people rant a lot about grandstream as well. |
| 04:46.45 | delroy | thought the aastra lcd sidecar looksed nice for the reception |
| 04:46.54 | WIMPy | Snom has 12 free buttons and most of the function keys can be reprogrammed as well. |
| 04:47.25 | ChannelZ | SNOG |
| 04:47.40 | *** join/#asterisk CRCinAU (~CRCinAU@zeus.crc.id.au) |
| 04:47.51 | delroy | [hC]: any cdn pricing similar to telephonydepot.com in canada? |
| 04:47.53 | *** part/#asterisk CRCinAU (~CRCinAU@zeus.crc.id.au) |
| 04:47.54 | riscphree | I've used the Aastra 6731i and Polycom 321 phones, both are very simple to use |
| 04:48.00 | *** join/#asterisk CRCinAU_ (~CRCinAU@zeus.crc.id.au) |
| 04:48.08 | CRCinAU_ | afternoon all |
| 04:48.27 | CRCinAU_ | just a quick question.... |
| 04:48.33 | delroy | our target price for office sets is 150 ish |
| 04:48.43 | [hC] | delroy: for buying phones? probably best to use someone else. we are set up regionally in vancouver, and dont carry a ton of stock, its usually order when it comes up |
| 04:48.56 | [hC] | delroy: the aastra 6731i is about 120-130 |
| 04:48.58 | CRCinAU_ | in sip.conf, I'm having to set fromuser=blah the same as username=blah or defaultuser=blah |
| 04:49.28 | CRCinAU_ | should this be the way it works? or is it not quite right here? |
| 04:49.46 | CRCinAU_ | using 1.6.2.12-rc1 atm |
| 04:50.14 | delroy | any other cdn suppliers of aastra? |
| 04:50.26 | delroy | who is a good place to buy from in canada |
| 04:50.28 | delroy | ? |
| 04:50.37 | ChannelZ | username= is no more |
| 04:50.59 | CRCinAU_ | well, it is, and it works, but it says its depreciated. |
| 04:51.46 | CRCinAU_ | my question is, why if I set defaultuser=blah, do I also need to set fromuser=blah - shouldn't having the defaultuser=blah imply to use that username as the from? |
| 04:52.21 | ChannelZ | Read sip.conf - they're two different options |
| 04:52.57 | CRCinAU_ | see, I would have also thought that the default username without ANY options would be the username in the REGISTER string |
| 04:53.01 | ChannelZ | why it works we can only guess, knowing nothing of how the whole peer is configured and your ITSP |
| 04:53.09 | delroy | does the aastra have rj11 headset jack on the back? |
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| 04:54.54 | boodu | re |
| 04:55.00 | ChannelZ | boot |
| 04:55.11 | CRCinAU_ | this is a quick rundown of my config: http://fpaste.org/DNUp/ |
| 04:56.18 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 04:56.54 | CRCinAU_ | see, I would have thought that the auth username is whatever is between => and : in the register string |
| 04:57.21 | CRCinAU_ | but without both defaultuser and fromuser set in the peer config, I get the phone number used as the username |
| 04:57.53 | CRCinAU_ | probably obtained frmo the /phonenumber part at the end of the register string? |
| 04:58.11 | ChannelZ | That's what you register with, but that's only part of the story. Registration does little more than tell the remote side who you are and what your IP is |
| 04:58.29 | ChannelZ | How you send each other calls is a separate matter |
| 04:59.01 | CRCinAU_ | so if defaultuser and fromuser aren't set, where does it get the username from? |
| 05:01.57 | CRCinAU_ | or does that come from the device making the call? |
| 05:01.59 | ChannelZ | Depends on them... your caller ID name if it's set |
| 05:02.16 | ChannelZ | or they maybe don't care and match you by your IP |
| 05:03.08 | ChannelZ | SIP is kind of a mess |
| 05:03.56 | CRCinAU_ | :\ |
| 05:04.31 | CRCinAU_ | oh well, it seems redundent - but I'll just leave it there as it works lol |
| 05:04.54 | ChannelZ | My ITSP needs to see my DID in the From: header but otherwise knows I have a static IP so I don't need to auth |
| 05:05.57 | CRCinAU_ | nods |
| 05:06.33 | CRCinAU_ | it took me 2 days to figure out wtf was going on - as I was under the impression that defaultuser=blah would also change the fromuser - as it kinda makes sense that it would |
| 05:06.52 | CRCinAU_ | but without both, I get rejected by a couple of providers. |
| 05:07.19 | CRCinAU_ | but to confuse things, I'd just changed from building asterisk by hand to a prebuilt package |
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| 05:09.42 | ManxPower | have you read the UPGRADE*.txt files included in the Asterisk source? |
| 05:10.04 | ManxPower | historically the register statement had NOTHING WHATSOEVER to do with the rest of sip.conf |
| 05:10.40 | CRCinAU_ | yeah - I know that - but I figured it had to get values from *somewhere* and I couldn't think of any other real sane place to get them |
| 05:10.42 | ManxPower | That has, apparently changed, but the UPGRADE*.txt files would have all the information about it. |
| 05:12.05 | ChannelZ | You really should just ask what your ITSP(s) require of you. You've made it work, but by throwing darts, and we have no idea what is necessary or why either. |
| 05:12.44 | [TK]D-Fender | [01:04]<ChannelZ>My ITSP needs to see my DID in the From: header but otherwise knows I have a static IP so I don't need to auth <- fromuser= |
| 05:12.56 | CRCinAU_ | yeah - I'll talk to him later... but for now, I gotta run... thanks for th ehelp... |
| 05:12.59 | *** part/#asterisk CRCinAU_ (~CRCinAU@zeus.crc.id.au) |
| 05:14.05 | ChannelZ | Actually I just tested it and they don't seem to care, probably because of my static. I put "poopypants" in and it doesn't seem to mind |
| 05:22.13 | Bloudermilk | Does anyone know of an ITSP that can do high outbound volume (thousands of call setups per second) that doesn't require white labeled IPs? |
| 05:23.40 | ChannelZ | no but I can sell you some nice white Avery labels you can stick over the top of your yellow IPs |
| 05:24.31 | Bloudermilk | ChannelZ: Let me talk to the heads upstairs about that. If I throw some buzzwords into your pitch I just might be able to sell them on it |
| 05:24.47 | [TK]D-Fender | Level3 |
| 05:24.57 | ChannelZ | Synergistic. Leverage. Comingle. |
| 05:25.15 | Bloudermilk | *whitelisted |
| 05:25.48 | Bloudermilk | Thanks guys—will look into those |
| 05:26.19 | Bloudermilk | High-volume and non-whitelisted outbound traffic don't seem to mix |
| 05:26.38 | ChannelZ | mine weren't ITSPs. Those were my buzzwords. |
| 05:27.17 | ChannelZ | but they sound like they probably could be :) |
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| 05:30.34 | [TK]D-Fender | checkout time, later all |
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| 06:01.17 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 06:16.09 | Bloudermilk | ChannelZ: Haha. I didn't even catch onto that. They definitely sound like they could be |
| 06:17.52 | boodu | WIMPy, re when i try to use a specific it's work for a second line but when i use the context I can use just one channel ! why ? |
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| 06:22.25 | boodu | Dial("mISDN/g:isdn/${EXTEN:1}") is not a good syntax ? |
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| 06:39.07 | boodu | :(( |
| 06:39.48 | boodu | may be a problem with the version of misdn |
| 06:39.50 | boodu | ? |
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| 07:06.33 | boodu | is it possible to try to call with each port one by one ? |
| 07:06.39 | ChannelZ | dunno never seen mISDN. It's really a tech type of it's own? |
| 07:06.50 | boodu | if one is not ok use 2 then 3 then 4 ? |
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| 07:07.23 | ChannelZ | well the channel driver has to support grouping to do it in an automatic fashion |
| 07:07.38 | ChannelZ | otherwise you have to build your dialplan to look at each channel and act accordingly |
| 07:11.13 | boodu | I want to configure a dialplan because the automatic fashion doesn't ork |
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| 07:29.14 | ChannelZ | you can look at the application ChanIsAvail |
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| 07:43.34 | boodu | thx ChannelZ i will look |
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| 07:50.12 | schmidts | good morning |
| 07:56.08 | SeTTleR | morning |
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| 08:03.06 | ChannelZ | throws raisin bran around the channel |
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| 08:13.38 | schmidts | channelz flower power trip? |
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| 08:32.30 | ChannelZ | OT, but any recommendations on a decent Wireless Access Point? I have a Linksys WAP54G and at random all of my clients stupid Mac laptops cannot access it. Full signal, just seems to refuse to work. |
| 08:33.07 | ChannelZ | At my wits end when other devices (Windows laptops, my phone, etc) are functioning just fine during these Mac outages so all I have left to try is a different WAP |
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| 08:37.47 | shamelessn00b | how can I use the pattern _X! to cater multiple digit extensions read from the user by background() command? |
| 08:38.14 | shamelessn00b | _. works fine |
| 08:38.46 | shamelessn00b | * jumps to _X! as soon as the user presses a single digit |
| 08:39.08 | joako | Yes because e.g 3 is a match for _X! |
| 08:39.20 | shamelessn00b | it is for _. as well |
| 08:39.35 | shamelessn00b | 333 is a match for _X! as well |
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| 08:41.51 | kaldemar | shamelessn00b: _X. |
| 08:42.30 | kaldemar | shamelessn00b: ! matches immediately |
| 08:42.49 | shamelessn00b | ok |
| 08:42.52 | shamelessn00b | thanks |
| 08:46.25 | Godfather_ | ChannelZ, you can use a wrt54gl for it |
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| 08:48.55 | shamelessn00b | and why wouldn't this work _[X.] |
| 08:49.42 | ChannelZ | what's the point of that? |
| 08:50.06 | ChannelZ | X. isn't a range |
| 08:51.39 | shamelessn00b | I made a tool, which reads from a csv and generates extensions.conf |
| 08:52.04 | shamelessn00b | treating the dialplan as tree, nodes/edges |
| 08:53.15 | shamelessn00b | edges are conditions when one node has to go to another node, a node is a combination of lines in the dialplan |
| 08:53.32 | ChannelZ | Lovely. Doesn't explain why you'd put a wildcard in a character class |
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| 09:11.58 | kjs | hiiiya |
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| 09:40.32 | shamelessn00b | ChannelZ: for instance if a node has to go to extension 1,2,3 and 4 and do the same thing, I'd simply write _[1234] |
| 09:41.20 | shamelessn00b | _[1] is same as exten 1,1,whatever() |
| 09:42.27 | shamelessn00b | I added an exception for the extension 'X.', the code no longer surrounds it with brackets, its working now |
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| 10:45.04 | metiu | hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected |
| 10:45.23 | metiu | I checked the SIP dialogue and at some point the server sends a BYE message to one party |
| 10:45.32 | metiu | I have no timeout set |
| 10:45.53 | metiu | the two linphones register with a name which is defined as dynamic in sip.conf |
| 10:49.49 | metiu | the call terminates on the caller's side |
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| 10:53.31 | metiu | I'm using asterisk 1.8.0 and linphone 3.99 |
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| 11:40.18 | joel_oliveira | hi all |
| 11:40.22 | joel_oliveira | here I am again |
| 11:40.36 | joel_oliveira | is it possible to check if a user is registered to recieve calls in asterisk? |
| 11:46.44 | hrhrhr | sip show peers |
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| 11:52.22 | jmkgreen | we are occasionally seeing an AGI result=-1 coming from Asterisk. This may be resulting from prompts being cut off as they are streamed to the handset. Can anyone advise how best to debug this? |
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| 11:58.48 | joel_oliveira | hrhrhr: thanks, but I was thinking more on checking that on the dialplan, i.e. when some user places a call, the asterisk tryes to check if he's already registered |
| 12:00.50 | Chainsaw | joel_oliveira: ChanIsAvail might serve you well here. |
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| 12:01.43 | Chainsaw | joel_oliveira: That's a "would I be able to send a call to SIP/54321?" question. Unless you set up explicit call limits, it will not consider whether the destination is already on a call, just whether it is registered. |
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| 12:26.38 | schangmi | Hi |
| 12:27.30 | schangmi | I have a basic asterisk setup with two phones. I ahve noticed that all RTP trafic goes through asterisk although both phones and asterisk are on the same subnet |
| 12:27.47 | schangmi | asterisk appears to be acting as a proxy as far as RTP trafic is concerned... |
| 12:28.34 | schangmi | is there a way of "telling" asterisk that RTP trafic should be exchanged directly from phone 1 to phone 2 ? |
| 12:28.40 | pabelanger | schangmi: directmedia=yes |
| 12:29.00 | russellb | unless you have also told asterisk to do something that requires access to the media :-) |
| 12:29.23 | russellb | like watching for DTMF to activate features, recording, other things |
| 12:32.29 | schangmi | pabelanger I just googled a bit (which I should have done before coming here), and the right parameter seems to be canreinvite which needs to be set to yes |
| 12:33.11 | schangmi | pabelanger so, which one should I go with ? canreinvite or directmedia ? |
| 12:33.45 | russellb | they are the same thing |
| 12:33.48 | russellb | it got renamed |
| 12:33.53 | russellb | but the code will support both |
| 12:34.18 | pabelanger | what russellb said |
| 12:35.50 | [TK]D-Fender | Unless you're using a branch older than they think you are which was never mentioned. |
| 12:36.06 | russellb | that would be unheard of |
| 12:36.15 | schangmi | ok, thanks a lot |
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| 12:38.41 | [TK]D-Fender | russellb: It was unheard. Quite astute! |
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| 12:49.08 | metiu | hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected... |
| 12:49.18 | metiu | the call terminates on the caller's side |
| 12:49.23 | metiu | I'm using asterisk 1.8.0 and linphone 3.99 |
| 12:54.23 | fauxalliance | metiu, there may indeed be a problem... standard procedure dictates that you send a pastebin, detailed.... we actually need to _SEE_ whats on the go.... |
| 12:54.39 | fauxalliance | metiu, are you using any * type distro or framework? |
| 12:58.36 | metiu | fauxalliance: a pastebin of asterisk -rvvvvvvvvddddddd and sip set debug on? is it ok? distro is debian (testing) but I compiled asterisk 1.8.0 myself |
| 12:59.00 | fauxalliance | metiu, thats sounds good |
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| 13:02.22 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 13:02.58 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 13:04.07 | Katty | shivers |
| 13:04.18 | fauxalliance | Katty, snow? |
| 13:04.47 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 13:04.49 | Katty | 8C with drizzle |
| 13:05.02 | Katty | or 41F if you prefer that measurement |
| 13:05.16 | verywiseman | how can i know ip for the sip devices that connect to asterisk? |
| 13:05.37 | Katty | verywiseman: sip show peers |
| 13:06.20 | Katty | checks crittercam |
| 13:06.27 | metiu | fauxalliance: http://asterisk.pastebin.com/jN4R32KG |
| 13:06.43 | Katty | damp and gloomy out there too )= |
| 13:06.47 | metiu | that's the output of asterisk with debug |
| 13:07.51 | *** join/#asterisk sbszulu (~dundubala@41.14.30.244) |
| 13:13.50 | metiu | fauxalliance: and this http://asterisk.pastebin.com/x9fSEU7X is the output of linphone with debug on a similar call |
| 13:14.08 | Katty | no birdies this morning :< |
| 13:14.28 | metiu | two linphones calling each other directly are happily up since one hour ago |
| 13:14.54 | Chainsaw | Katty: Just a loud aeroplane here. But indeed, no birds :/ |
| 13:16.39 | Katty | :<< |
| 13:16.54 | Katty | Chainsaw: i think they are already working on a new time change |
| 13:16.55 | pabelanger | metiu: So, what is your issue? |
| 13:17.28 | metiu | if I place a call through asterisk, the call gets dropped on the caller side after around 20s |
| 13:17.45 | Chainsaw | Katty: I propose hibernation until the weather improves. It's grey and just miserable :/ |
| 13:18.11 | metiu | no clue on what causes it, asterisk just says that Dial exited |
| 13:18.13 | [TK]D-Fender | metiu: I see a LOT of OK's to taht invite like it isn't ACKing it |
| 13:18.27 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 13:18.41 | metiu | is that an option on asterisk? |
| 13:19.06 | Katty | Chainsaw: i approve of this proposition. wake me up in march. |
| 13:19.14 | Chainsaw | Katty: Will do :) |
| 13:19.17 | metiu | right now I set canreinvite=yes |
| 13:19.32 | Katty | Chainsaw: sadly we'd miss christmas tho. and thanksgiving. |
| 13:19.39 | Katty | Chainsaw: birdies are showing up on crittercam now |
| 13:20.15 | Chainsaw | Katty: I think both are overrated though. |
| 13:20.25 | Katty | Chainsaw: oh, i don't know about that. |
| 13:20.30 | Katty | Chainsaw: commercially, certainly so |
| 13:20.52 | Chainsaw | Katty: I can't even walk through the mall without being attacked with christmas songs & decorations. It's barely November! |
| 13:20.55 | Katty | Chainsaw: but when you leave all that BS aside, they are lovely holidays. |
| 13:21.23 | [TK]D-Fender | metiu: Perhaps its failing the hand-off. check your firewalls, etc |
| 13:21.41 | Katty | we should do a christmas card list |
| 13:21.56 | Katty | anyone interested in christmas cards this year? |
| 13:22.02 | Chainsaw | Absolutely :) |
| 13:22.17 | Chainsaw | International christmas cards would make it more fun. I have a franking machine here. |
| 13:22.23 | Katty | of course. |
| 13:22.28 | Katty | international card are the best |
| 13:22.37 | metiu | [TK]D-Fender, well no firewalls, I have the same behavior even if I place a call from a linphone on the same host as asterisk |
| 13:22.42 | Katty | i wonder if we could get a sign up sheet somewhere |
| 13:22.46 | fauxalliance | wraps up a giant fruitcake... who wants to be the 'lucky' one? |
| 13:23.04 | n3hxs | likes fruitcake |
| 13:23.16 | Katty | Qwell: ping |
| 13:23.17 | fauxalliance | Katty, that'd be cool... like secret santa, but only exchange cards |
| 13:23.29 | Katty | yeah cards only else it gets too expensive on shipping. |
| 13:23.43 | Katty | you can buy a big pack of cards for very little, and send them to lots of people |
| 13:24.00 | fauxalliance | Katty, post cards are neat too.... and my office has a postage machine ;) |
| 13:24.15 | Katty | yeah that's a neat idea. |
| 13:24.49 | Katty | leifmadsen: ping. |
| 13:25.00 | fauxalliance | Katty: +1 |
| 13:25.14 | Katty | Deeewayne: ping |
| 13:25.16 | Katty | surely someone's awake. |
| 13:25.31 | Katty | russellb: ping! |
| 13:25.38 | russellb | yes? |
| 13:25.46 | Katty | oh goody. someone who is awake. |
| 13:26.00 | Katty | russellb: can we start an asterisk christmas card sign up page, somewhere |
| 13:26.05 | Katty | russellb: publicly accessible |
| 13:26.05 | russellb | lol |
| 13:26.10 | Katty | russellb: for irc. |
| 13:26.28 | russellb | google spreadsheet with a form? |
| 13:26.37 | russellb | you have the power |
| 13:26.47 | fauxalliance | is all over Katty's idea, like leifmadsen's proverbial fat kid on a smarty |
| 13:27.09 | Katty | russellb: that makes the list. |
| 13:27.28 | Katty | i guess i could just spam the channel all the time |
| 13:27.29 | fauxalliance | Katty, people pick and choose whom to send to? or the computer delegates |
| 13:27.39 | Katty | fauxalliance: you send to the whole list |
| 13:27.58 | russellb | i am totally confused as to what we're talking about |
| 13:28.02 | Katty | fauxalliance: i guess you could pick and choose your favorites tho |
| 13:28.03 | russellb | "christmas cards" ... "list" ... |
| 13:28.04 | fauxalliance | Katty, wow... a whole lotta postage... but, nice... |
| 13:28.50 | leifmadsen | fauxalliance: :) |
| 13:28.56 | fauxalliance | as long as no one comes knocking on my door christmas eve looking for help with their trixbox ;P |
| 13:29.05 | Katty | i'll put something on my blog |
| 13:29.40 | leifmadsen | you have a blog?! :) |
| 13:30.08 | fauxalliance | thats the twelfth commandment, thou shalt not blog... |
| 13:30.39 | *** join/#asterisk iulhk (~iulhk@115.186.41.8) |
| 13:30.39 | pabelanger | fauxalliance: Ya, that popped into my head. A list of addresses for people in #asterisk, however before somebody shows up on your front step looking for support :) |
| 13:30.54 | pabelanger | s/however/how long/ |
| 13:31.08 | Chainsaw | This is why I use my work address. |
| 13:31.14 | fauxalliance | pabelanger, fortunately, i am on an isolated island... |
| 13:31.32 | fauxalliance | but i still get strange calls on my GV number... |
| 13:32.08 | Katty | can someone take the asterisk logo and put a santa claus hat on it |
| 13:32.18 | Katty | that'd be super. |
| 13:32.27 | fauxalliance | facepalms... 89.4% packet loss |
| 13:33.11 | Katty | infobot: Christmas |
| 13:33.11 | infobot | christmas is probably the feast for the birth of Jesus Christ our Savior, or commercialised, or on December 25th. unless you aren't christian, when christmas is an opportunity to screw dumb christians for lots of money, or nono, this is all wrong, christmas is on December 24th |
| 13:33.16 | Katty | infobot: Card |
| 13:33.24 | Katty | infobot: Christmas Card? |
| 13:34.03 | Katty | infobot: christmas card? |
| 13:34.04 | infobot | extra, extra, read all about it, christmas card is If you're interested in being on the Asterisk Christmas Card List, go here: http://42ndgeekstreet.blogspot.com/2010/11/asterisk-christmas-card-list.html |
| 13:35.59 | Chainsaw | has posted |
| 13:36.19 | fauxalliance | "Hello LEC... the kids are rooting around in that box on the pole by the Torbay road bus stop again...." http://tinyurl.com/29zjojq (on the pole) |
| 13:37.09 | fauxalliance | why would they leave my leased copper wire out in the public like that.... |
| 13:38.19 | Katty | i could send it to private |
| 13:38.24 | Katty | erm set it to private |
| 13:38.33 | Katty | but then people would have to give me their email address in order to post |
| 13:38.36 | Katty | would you prefer that? |
| 13:38.42 | krion | for the one who read me yesterday, i've to specify context and ext in my outcall file |
| 13:38.51 | Chainsaw | Katty: It's not a secret where I work :) |
| 13:39.42 | *** join/#asterisk sbszulu (~dundubala@41.16.8.105) |
| 13:39.47 | tzanger | I got me a gt blueberry muffin today :-) |
| 13:40.06 | fullstop | gt? |
| 13:40.12 | Katty | well it's a bad idea to have peoples addresses sitting around... |
| 13:40.24 | fullstop | grand touring? |
| 13:40.34 | tzanger | grand traverse pie company |
| 13:40.57 | tzanger | fauxalliance: I don't get the link |
| 13:41.04 | Katty | ok it's set to private |
| 13:41.07 | tzanger | it's a google street view in NL with a bunch of ads |
| 13:41.54 | Katty | infobot: christmas card |
| 13:43.47 | *** join/#asterisk coppice (~chatzilla@116.92.195.24) |
| 13:43.50 | krion | can i simulate "pick up/blf" with outcall file ? |
| 13:44.11 | Katty | infobot: christmas card |
| 13:44.12 | infobot | extra, extra, read all about it, christmas card is If you're interested in being on the Asterisk Christmas Card List please see Katty for an invite to http://42ndgeekstreet.blogspot.com/2010/11/asterisk-christmas-card-list.html |
| 13:47.25 | metiu | fauxalliance: so no clue on why asterisk drops the call? |
| 13:50.30 | *** join/#asterisk Gast1984 (~Gast1984@212.80.243.19) |
| 13:50.42 | Gast1984 | good morning |
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| 13:51.23 | ManxPower | metiu, did you do what [TK]D-Fender told you to do? |
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| 13:53.42 | Gast1984 | can someone tell me if it possible to make a playback like "Please hold the Line" and there is one forwarding of 2 phone an these 2 phones shall ring at the same time as playback played |
| 13:54.40 | Gast1984 | i´ve the prob. that first the playback is running and when ist stop the phones ring |
| 13:54.49 | [TK]D-Fender | Gast1984: yes, make a MoH class that only has a single recording in the folder with that along with a "ring sound" and use m() |
| 13:54.51 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 13:56.05 | Gast1984 | moh=musiconhold?exit |
| 13:56.31 | Gast1984 | sorry the exit should go to the cli ^^ |
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| 13:57.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:58.29 | Gast1984 | can you please discribe how to make a moh class? is it [macro-musikply] and than i call the macro in my dialplan? |
| 13:59.32 | ManxPower | Gast1984, are any of the ports FXO ports? |
| 13:59.36 | [TK]D-Fender | Gast1984: Who said anything about a MACRO? |
| 13:59.52 | [TK]D-Fender | Gast1984: Use m() in your DIAL to set the MoH class to use INSTEAD of normal ringing |
| 14:00.11 | ManxPower | Gast1984, Trixbox/FreePBX/AsteriskNOW support is a different channel |
| 14:00.11 | Gast1984 | i´m only work at intranet |
| 14:00.23 | Gast1984 | asterisk 1.4.21 at debian |
| 14:01.14 | ManxPower | <ManxPower> Gast1984, are any of the ports FXO ports? |
| 14:01.14 | Gast1984 | the problem is we have some 1.2 asterisk server, but this person who have make him is not longer avialable, an i have to learn howto asterisk (: |
| 14:01.18 | [TK]D-Fender | ManxPower: Stop jumping the gun on those... give them enough rope, THEN yank once you see real evidence. |
| 14:01.41 | ManxPower | [TK]D-Fender, I'd rather just shoot them and avoid the rest of it. |
| 14:01.55 | [TK]D-Fender | ManxPower: You mean the "find out if they're really guilty"? :p |
| 14:02.16 | Gast1984 | what is a fxo port? |
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| 14:37.01 | titter | cesar_CR: do not disturb feature on the phone should work as well |
| 14:37.02 | p3nguin | DND could work, but it could also do something else with the call instead of just skipping it. |
| 14:37.02 | p3nguin | Depends on several factors. |
| 14:37.02 | titter | Correct |
| 14:37.03 | p3nguin | For example, setting DND to Busy could send the call to the voicemail of the operator. |
| 14:37.10 | p3nguin | If that happens, reconfiguration might be necessary. |
| 14:37.11 | cesar_CR | p3nguin, titter ok I will try to see which one if for me, thank you ! |
| 14:37.11 | p3nguin | The DND on the phone will be easy to test. |
| 14:37.12 | titter | Only reason I suggested that, I know my Polycom's will continue on in the dialplan during DND |
| 14:37.35 | titter | I forget if I changed a cfg or not for that |
| 14:37.37 | cesar_CR | p3nguin, sure, but that will depend on my config |
| 14:38.12 | p3nguin | titter: You probably don't have a dialplan that sends DIALSTATUS = BUSY calls to voicemail. |
| 14:38.14 | cesar_CR | titter, I will have to test, if I have the seme behavoir I'am using xlite |
| 14:39.08 | *** join/#asterisk zmitya (~mitya@gw.gammatelecom.hu) |
| 14:39.17 | zmitya | hi gents |
| 14:39.18 | kjs | what is that system called that looks up a number to see if it is a voip number before placing the call over the pstn ? |
| 14:39.39 | zmitya | can asterisk play an announcement in early media ?? |
| 14:40.01 | zmitya | I mean without answering the call first |
| 14:40.02 | p3nguin | zmitya: Try using Playback() and its noanswer option. |
| 14:40.18 | zmitya | p3nguin: ok thanks, let me check |
| 14:40.42 | p3nguin | And, of course, don't use Answer() before that. :) |
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| 14:44.31 | zmitya | p3nguin: it seems to be working well, thanks !! |
| 14:44.47 | p3nguin | Great. |
| 14:45.07 | zmitya | thanks, cheers |
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| 14:46.51 | metiu | p3nguin: I tried setting the sip clients as in the attached sip.conf http://asterisk.pastebin.com/zXT0m9d1 |
| 14:47.10 | Gast1984 | can someone tell me why Playback() make some error when i want to play my waitingmusic? |
| 14:47.20 | metiu | p3nguin: when I sip show channel ##########, the client is still expected on 5060 |
| 14:47.21 | titter | p3nguin: Exactly, I have nothing set for that. The normal dialplan handles voicemail once the timeout is reached. |
| 14:47.33 | metiu | and * sends it to 5060 instead of the received port |
| 14:48.21 | p3nguin | gast1984: Don't use Playback() for music... use MusicOnHold() or WaitMusicOnHold() to play music. |
| 14:48.31 | Gast1984 | hx |
| 14:48.34 | Gast1984 | thx* |
| 14:48.34 | jmkgreen | anyone know how to find the current linux kernel timer value? Somewhere in /proc ? |
| 14:48.59 | p3nguin | metiu: Ditch the templates during testing. |
| 14:49.37 | metiu | p3nguin: ok I'll rewrite it |
| 14:50.13 | titter | wouldn't nat need to be set to no since the hosts are on the same internal subnet? |
| 14:50.17 | Gast1984 | i have make some [macro-waiting] exten => s,1,startmoh(class1) exten => s,2,startmoh(class2) but it always play the musik from calls 2 at first |
| 14:52.20 | Gast1984 | class* |
| 14:55.35 | *** join/#asterisk aberrios_ (~aberrios@195.171.4.82) |
| 14:56.02 | metiu | p3nguin: http://asterisk.pastebin.com/f0BkFXfU new sip.conf, same problems |
| 14:57.29 | Katty | infobot: christmas card |
| 14:57.29 | infobot | extra, extra, read all about it, christmas card is If you're interested in being on the Asterisk Christmas Card List please see Katty for an invite. |
| 14:57.46 | p3nguin | sees katty |
| 15:00.04 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:01.34 | metiu | p3nguin: ok problem "solved": I moved asterisk to port 5062 and returned the linphones to 5060... then the call gets through and the invites are exchanged correctly |
| 15:02.22 | thehar | frolics about |
| 15:02.24 | thehar | morning friends |
| 15:02.24 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
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| 15:02.36 | Katty | hai thehar! |
| 15:02.46 | [TK]D-Fender | Gast1984: Did you restart * after creating this 2nd class? |
| 15:02.51 | Katty | thehar: we decided to do a christmas card exchange...if you wanna participate. if not that's cool too |
| 15:03.13 | thehar | hehe hiiiiiiiiii |
| 15:03.17 | thehar | cards? |
| 15:03.18 | thehar | yes! |
| 15:03.21 | Katty | yes. cards. |
| 15:03.21 | thehar | i will happily join |
| 15:03.56 | SaiSoma | ok, i have a rather serious issue that started after my upgrade to 1.8.0 |
| 15:04.34 | SaiSoma | i changed two lines on an extension in sip.conf (added sendrpid=yes and trustrpid=yes) and did a sip reload |
| 15:05.08 | SaiSoma | got this: http://pastebin.com/xkCFMT0d from the console |
| 15:05.17 | SaiSoma | and all calls stopped |
| 15:05.20 | SaiSoma | and still are |
| 15:05.25 | SaiSoma | core commands don't work |
| 15:05.30 | SaiSoma | sip commands don't work |
| 15:05.38 | Gast1984 | i hab make a reload |
| 15:05.42 | Gast1984 | had* |
| 15:05.47 | SaiSoma | i kill the asterisk process, even rebooted the server for the heck of it. same deal |
| 15:06.01 | SaiSoma | asterisk*CLI> sip show peers |
| 15:06.02 | SaiSoma | No such command 'sip show peers' (type 'core show help sip' for other possible commands) |
| 15:06.27 | SaiSoma | i had to restore from backup last time this happened:( |
| 15:06.44 | [TK]D-Fender | Gast1984: Then show us the problem |
| 15:07.02 | hrhrhr | SaiSoma: restart asterisk |
| 15:07.17 | SaiSoma | hrhrhr: i have. and restarted the box. as i stated aboce |
| 15:07.20 | SaiSoma | above* |
| 15:07.25 | hrhrhr | sorry, can't read |
| 15:07.48 | hrhrhr | i had an issue once where sip functionality appeared to have died |
| 15:08.04 | SaiSoma | not just sip . i don't have even the "core" commands |
| 15:08.45 | SaiSoma | only started with the 1.8.0 upgrade, related? i haven't made any significant config changes (minor sip.cfg changes and dialplan changes), and even restoring backup configs doesn't help |
| 15:08.48 | *** join/#asterisk shapr (~shapr@nat/digium/x-xujoqzckflxxuqpk) |
| 15:09.51 | Katty | fender are you going to do christmas card exhange this year |
| 15:10.06 | shapr | grins |
| 15:10.13 | Gast1984 | the porblem is that i have 2 mp3 but thy are playded not in the right sequence |
| 15:10.40 | Gast1984 | sorry for my bad englich |
| 15:10.55 | Gast1984 | exit |
| 15:11.11 | Katty | music on hold? |
| 15:11.21 | Gast1984 | sorry false keybord -.- it to latte for me to work simultan |
| 15:11.31 | Gast1984 | yes moh |
| 15:11.37 | Katty | Gast1984: random=no |
| 15:12.04 | Gast1984 | must i write random=no in the moh.conf? |
| 15:12.04 | Katty | Gast1984: and the names of your files should reflect the order you want them to play in |
| 15:12.09 | Katty | Gast1984: yes. |
| 15:12.20 | Gast1984 | i will try it |
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| 15:14.52 | SaiSoma | ok, had to delete the asterisk binary and reinstall |
| 15:15.55 | tzafrir | Gast1984, moh.conf? musiconhold.conf? |
| 15:16.01 | Gast1984 | it dont work, the first mp3 beginn with a "d" and the second with a "w" but ist plays not the one with "d" first :/ |
| 15:17.02 | Gast1984 | exit |
| 15:17.30 | *** part/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 15:19.21 | [TK]D-Fender | Gast1984: maybe * was already playing the "D" one before and it is CONTINUING on the "w"? |
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| 15:21.09 | kjs | hmmm, something strange going on here... when i login to my voicemail to set my name or any other messages, it gives me the standard "ay your msg after the tone" then it instantly plays "press 1 to accept this msg..." |
| 15:21.17 | kjs | it was working fine, hmm |
| 15:22.47 | Gast1984 | how to you mean this with *? |
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| 15:23.30 | kjs | ok fixed it, was a permission issue... |
| 15:23.56 | kjs | strange that I get so many issues like this after installing with the rpm perhaps src is still the way to go. |
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| 15:28.30 | Katty | kjs: i prefer source. |
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| 15:29.16 | p3nguin | As strange as it may seem, the RPM is made from the same source. |
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| 15:36.38 | kjs | p3nguin: i have had a few permissions issues from the rpm install. |
| 15:37.17 | kjs | other than that it's been fine |
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| 15:47.06 | SaiSoma | gah. it's doing it again, after i applied the patch here: https://issues.asterisk.org/view.php?id=18192 |
| 15:47.11 | SaiSoma | tried to resatrt |
| 15:47.19 | goddva | I have 15 phones, all of them are using a Attendent Console that subscribe to all of the 15 phones. After a restart of asterisk, I often do get a bunch of LAGGED messages, and it can takes serveral minutes before everything is working fine again... |
| 15:47.37 | goddva | Is this because of the big amount of subscriptions? |
| 15:48.49 | [TK]D-Fender | goddva: WHO is saying "lagged"? |
| 15:49.07 | goddva | most of the peers |
| 15:49.42 | [TK]D-Fender | goddva: and where are they locate relative to * and your "Attendant"? |
| 15:50.26 | goddva | same switch (if that was answereing your question) |
| 15:50.44 | goddva | expect for 3 of the peers... |
| 15:52.05 | [TK]D-Fender | goddva: What devices? |
| 15:52.50 | goddva | phones? |
| 15:52.55 | goddva | Cisco SPA525 |
| 15:53.11 | goddva | with an attenden console SPA500s |
| 15:54.20 | [TK]D-Fender | goddva: try increasing their qualify time |
| 15:54.45 | *** join/#asterisk sbszulu (~dundubala@41.14.228.85) |
| 15:54.52 | goddva | I have tried turning qialify off, does help a bit.. |
| 15:55.10 | goddva | but I still have the problem... |
| 15:56.15 | goddva | I think most of the phones are trying to both register and send the subscribe at the same time.. all are connected to the same switch, and they all use POE.. |
| 15:56.38 | goddva | For me it looks like its too much for asterisk to handle.. but that again dosent make sense.. |
| 15:57.27 | goddva | I have been trying to recreate the problem on another server, and I can only reproduce the problem by adding some random delay on the network |
| 16:00.06 | SaiSoma | ok, so i can't get asterisk to load completely in 1.8.0 and this is a production server:(. http://pastebin.com/Np4xiEPk for more details. i'm stumped |
| 16:00.58 | goddva | [TK]D-Fender: any tips/ideas? |
| 16:01.41 | [TK]D-Fender | goddva: Not offhand... |
| 16:02.06 | Katty | infobot: christmas card |
| 16:02.07 | infobot | christmas card is probably If you're interested in being on the Asterisk Christmas Card List please see Katty for an invite. |
| 16:02.26 | russellb | I still don't get how this works, heh |
| 16:02.38 | russellb | what does being on this list mean |
| 16:02.48 | SaiSoma | is there an order to how modules load or anything perhaps? |
| 16:02.57 | Katty | russellb: it means you get cards in the mail |
| 16:03.00 | Chainsaw | russellb: You will get a christmas card. With a hand-written seasonal greeting. |
| 16:03.04 | Katty | russellb: and you ship cards, to people on the list, in the mail |
| 16:03.04 | russellb | from who? |
| 16:03.05 | Chainsaw | russellb: Snail mail mind you. |
| 16:03.09 | Katty | russellb: people on the list |
| 16:03.10 | Chainsaw | russellb: Other people on the list! |
| 16:03.26 | russellb | so it's like ... signing up to have friends? |
| 16:03.30 | titter | lolol |
| 16:03.35 | Katty | not exactly. |
| 16:03.45 | Katty | you can only send it to your friends on the list, if you like |
| 16:03.59 | russellb | got it. |
| 16:04.12 | russellb | I guess I want to be on it? I'm willing to put my work address on there, at least. :-) |
| 16:04.12 | kjs | what list |
| 16:04.23 | kjs | I dont have any friends on "the list" ? |
| 16:04.26 | Chainsaw | russellb: I also used my work address, yes. Just as an anti-stalker measure. |
| 16:04.40 | Katty | kjs: they are mostly people from the irc chat room. |
| 16:04.43 | Chainsaw | kjs: If you're on the list, I'm happy to send you a card. Whether I know you or not. |
| 16:04.49 | Chainsaw | is friendly like that |
| 16:04.51 | kjs | feel free to stalk me. |
| 16:05.02 | russellb | I wonder if I can get some epic cards made with the asterisk logo |
| 16:05.07 | chuckf | wonders if he can pick his stalker |
| 16:05.27 | Chainsaw | chuckf: Karen Gillen can only stalk one person at once I'm afraid. |
| 16:05.42 | ruyo | Any idea what kind of processor do I need for an Asterisk with 2 PRI with ~100 concurrent calls? |
| 16:05.43 | kjs | picks the hot blonde stalker girl with thought asterisk was something else completely when she signed up |
| 16:06.02 | Chainsaw | ruyo: Are you planning on any conference bridge facilitieS? |
| 16:06.05 | ruyo | Is an i3 enough? |
| 16:06.14 | kjs | ruyo: an p3 800 x 2 |
| 16:06.19 | Chainsaw | ruyo: Will you be transcoding those calls? |
| 16:06.32 | ruyo | I'm not counting on transcoding, no. |
| 16:06.43 | ruyo | Maybe some conferencing, but not much. |
| 16:06.49 | Katty | russellb: it'd be pretty fancy if you could get a logo with a santa hat on it for the blog page |
| 16:06.49 | Chainsaw | ruyo: I believe an i3 should cope with that workload, yes. |
| 16:07.03 | Chainsaw | ruyo: If transcoding comes into play, 100 concurent calls might be more... interesting. |
| 16:07.17 | kjs | dont transcode ftw. |
| 16:07.21 | ruyo | Ok, taking a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning gave me the idea that an i3 would sufice. |
| 16:07.32 | ruyo | Thanks. |
| 16:07.38 | Chainsaw | ruyo: (But you could then always offload it to that hardware adapter that Digium sell. Then you'll definitely be on russellb's christmas card list!) |
| 16:07.43 | kjs | is there a way to bench an * box? |
| 16:08.12 | hrhrhr | like everything in the world of linux, it should require no more than a p1 200 mmx |
| 16:08.16 | Katty | kjs: did you want an invite to the exchange list? |
| 16:08.17 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 16:08.17 | Chainsaw | kjs: Sure, start a large teleconference, announce it here and get people to call in. |
| 16:08.25 | ruyo | What adapter? :o |
| 16:08.31 | kjs | Chainsaw: ok hang on. |
| 16:09.18 | *** join/#asterisk delroy (~delroy@tba.usask.ca) |
| 16:10.33 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:10.54 | *** join/#asterisk rneese (~rneese@c-76-125-160-55.hsd1.pa.comcast.net) |
| 16:11.14 | rneese | whi is in charge of dahdi now days |
| 16:11.41 | rneese | we need to replace libnewt that it uses to build |
| 16:11.53 | rneese | as libnewt is not being maintained |
| 16:12.24 | tzafrir | rneese, replace it with? |
| 16:12.42 | rneese | thats the question |
| 16:12.46 | russellb | libnewt is optional |
| 16:12.50 | russellb | so it doesn't really matter |
| 16:13.03 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 16:13.17 | rneese | well it seems dahdi wont build with out it |
| 16:13.39 | rneese | everytime i have built it it has a dep of newt |
| 16:13.49 | rneese | but oyour saying its not needed |
| 16:13.50 | russellb | it's for an optional utility. |
| 16:13.58 | tzafrir | rneese, who is "we"? What do you use DAHDI for? |
| 16:13.59 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 16:14.13 | rneese | openzap cards with freeswitch |
| 16:14.40 | rneese | freetdm on freeswitch uses it |
| 16:15.42 | rneese | and on bsd there is discussion since its no longer being maintined to pull libnewt in the near future |
| 16:16.18 | rneese | and the dahdi build has a dep . but if its not needed we can pull it out |
| 16:16.31 | *** join/#asterisk JunK-Y (~junky@64.15.77.94) |
| 16:16.50 | russellb | well, you won't have dahdi_tool anymore, which is unfortuante. |
| 16:17.00 | russellb | but whateva'. |
| 16:17.12 | rneese | unless we come up with a lib replacement |
| 16:17.21 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 16:17.25 | russellb | that means rewriting the tools that use it ... unlikely. |
| 16:17.30 | russellb | libnewt works fine. |
| 16:18.33 | p3nguin_ | kjs: Did you announce the conference yet? I had to disconnect for a minute. |
| 16:18.54 | rneese | but if its not being maintained then why not pull the src right into the tool and rm thee need for the lib |
| 16:19.19 | russellb | because that would be silly. :-) |
| 16:19.26 | russellb | we don't want to maintain it, either. |
| 16:19.32 | russellb | it works. everyone packages it. |
| 16:19.46 | russellb | If your distribution wants to remove it and make less programs accessible, have at it. |
| 16:19.49 | russellb | Seems stupid to me. |
| 16:20.21 | kjs | is there a way to get my sip address to be the same as my email address? it has to be ext@hostname right ? |
| 16:20.54 | *** part/#asterisk rneese (~rneese@c-76-125-160-55.hsd1.pa.comcast.net) |
| 16:21.53 | russellb | heh, bye? |
| 16:23.03 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 16:23.21 | p3nguin_ | If your email domain is the same as where your SIP device is, and if you configure an extension on it to be the same as your username portion of your email address, it's possible. |
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| 16:31.42 | kjs | how can i find the correct syntax for meetme on asterisk 1.8 |
| 16:31.55 | [TK]D-Fender | kjs: "coer show application meetme" |
| 16:32.01 | [TK]D-Fender | core* |
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| 16:37.24 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 16:40.24 | kjs | p3nguin_: yeah it's not, which sucks. |
| 16:40.44 | p3nguin_ | kjs: Are you using a public email service? |
| 16:41.08 | kjs | p3nguin_: no, i host my email. |
| 16:41.25 | p3nguin_ | kjs: Then you can create your SIP URI to be the same as your email address. |
| 16:41.46 | kjs | p3nguin_: would it involve putting my * box and my email server on the same IP address? |
| 16:41.53 | p3nguin_ | kjs: If you control the DNS for the domain and if you control the PBX, you can do it. |
| 16:42.06 | kjs | I control both. |
| 16:42.37 | kjs | currently my * box is on a subdomain. |
| 16:42.52 | p3nguin_ | kjs: Use an SRV record. |
| 16:42.58 | kjs | like, voip.mydomain.com |
| 16:43.53 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 16:44.10 | kjs | p3nguin_: cool looking it up on wikipedia now. |
| 16:44.21 | p3nguin_ | kjs: Create SRV _sip._udp.mydomain.com to appropriately reflect the actual hostname of the SIP device. |
| 16:46.28 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
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| 16:56.41 | *** part/#asterisk VonGodric (~albeva@109.231.205.3) |
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| 17:06.36 | delroy | Any handset recommendations? whats the best quality brand? |
| 17:06.42 | citywok | ~phones |
| 17:06.42 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 17:07.22 | *** part/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 17:07.23 | kjs | p3nguin_: that worked, hot |
| 17:08.00 | Chainsaw | delroy: I agree with the bot. Polycom. |
| 17:08.03 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 17:08.40 | kjs | p3nguin_: I spoke to soon it does not look like it's working |
| 17:09.36 | p3nguin_ | kjs: If you want help, you'll have to be a lot more specific than "it's not working." |
| 17:10.13 | delroy | thanks |
| 17:13.03 | kjs | p3nguin_: it's a bit weird, the srv record is there, I have checked it with dig and it looks sane yet when i try to dial it, it just fails and does not hit the box. |
| 17:13.10 | kjs | asterisk box* |
| 17:13.23 | kjs | TTL is set to 100 seconds as well. |
| 17:13.54 | citywok | Chainsaw: i like aastra phones, and at astricon last week a ton of booths had aastra phones in their displays. lol |
| 17:14.14 | Chainsaw | citywok: Sure, they score high in the handset survey. |
| 17:14.22 | Chainsaw | citywok: But I wouldn't say they're the best ones out there. |
| 17:14.40 | citywok | Probably not, but they are a great value :P |
| 17:14.41 | Chainsaw | (You see Cisco 7960 *all* the time on TV, but that is the power of sponsorship deals) |
| 17:14.47 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 17:17.05 | *** join/#asterisk [netman] (~netman@59.Red-79-146-230.dynamicIP.rima-tde.net) |
| 17:18.10 | kjs | could someone try and ring my sip address to see if it's working ? |
| 17:18.37 | p3nguin_ | If I knew the URI, I'd test it. |
| 17:18.47 | WIMPy | kjs: Post it ad see it you can handle the load :-) |
| 17:20.24 | kjs | I would but it's my email address and no doubt this channel is publicly archived ;) |
| 17:20.59 | SaiSoma | is there an order to how modules load or anything perhaps? apparently a module in 1.8.0 doesn't like me |
| 17:21.06 | WIMPy | Multimedia testing |
| 17:23.11 | kjs | p3nguin_: did you get that msg ? |
| 17:23.43 | Jasnejac | SaiSoma: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244725.html |
| 17:25.56 | p3nguin_ | kjs: Nope. |
| 17:28.35 | *** join/#asterisk abel408 (429863dc@gateway/web/freenode/ip.66.152.99.220) |
| 17:29.12 | *** join/#asterisk moltar_net (~Roman@67.69.160.130) |
| 17:29.42 | abel408 | Hey everyone! Is anyone familiar with progress in band and wants to help me debug sip packets? I've been stuck on a ring back problem forever! |
| 17:30.33 | Katty | if anyone is interested in joining the asterisk christmas card exchange, please let me know. |
| 17:30.45 | delroy | When it comes to trunking to analog lines, what are peoples preferences? I have had a lot of problems in the past with echo and noticed that with DHADI and Digium card I no longer have problems... Opinions? |
| 17:30.57 | ukine_work | what's the asteriskgui 1.7.1 default user/pass, and if possible where can i find a more comprehensive summary on asteriskgui? |
| 17:31.36 | ukine_work | got it..admin/password |
| 17:31.40 | [TK]D-Fender | ukine_work: AsteriskGUI is not being maintained, nor has it had a maintainer for over a year now |
| 17:32.11 | [TK]D-Fender | ukine_work: There are less than a handful of active users in their channel at all |
| 17:32.15 | [TK]D-Fender | ukine_work: #asteriskgui |
| 17:32.21 | [TK]D-Fender | ukine_work: #asterisk-gui <- rather |
| 17:34.03 | pabelanger | anybody used ekiga with asterisk recently? |
| 17:34.31 | ukine_work | [TK]D-Fender, ty. will reinstall w/ freepbx |
| 17:35.05 | p3nguin_ | pabelanger: I did a few days ago. |
| 17:35.12 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 17:35.44 | pabelanger | p3nguin_: mind if I ask which version of asterisk? |
| 17:37.12 | p3nguin_ | pabelanger: 1.4.36 |
| 17:37.38 | WIMPy | pabelanger: I also did with 1.6.2.9. |
| 17:38.20 | n3hxs | Katty, are you trying to make sure you still get "carded"? |
| 17:38.44 | Katty | n3hxs: statement does not parse. |
| 17:39.11 | n3hxs | d in joining the asterisk christmas card exchange |
| 17:39.12 | Katty | n3hxs: well more like it parses, but i don't get it |
| 17:39.28 | n3hxs | Play on words with Carded -- Christmas Card.. |
| 17:39.42 | n3hxs | Like you were trying to buy a drink |
| 17:39.55 | Katty | :P\ |
| 17:40.00 | n3hxs | LOL |
| 17:40.03 | Katty | who is d. |
| 17:40.14 | Katty | or what is d |
| 17:40.21 | Katty | DON"T CONFUSE ME WITH THE FACTS |
| 17:40.33 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 17:40.35 | n3hxs | Copied from your statement earlier. |
| 17:40.39 | n3hxs | Fact! |
| 17:41.20 | Katty | :P |
| 17:41.29 | delroy | Analog line termination - what are people using in commercia setups? |
| 17:41.33 | delroy | commercial |
| 17:41.39 | Chainsaw | delroy: Patton gateways. |
| 17:41.40 | pabelanger | p3nguin_: WIMPy: Ok, thanks. I'm running into an issue with ekiga and asterisk on the same box, configuration issue. Will roll back to 1.4 and see what happens |
| 17:41.56 | [TK]D-Fender | Chainsaw: They're plural now? ;) |
| 17:41.58 | Chainsaw | delroy: (4118 for my analog stuff; 4634 to talk to telcos over ISDN) |
| 17:42.09 | Chainsaw | [TK]D-Fender: Always have been, there's a pair at each office. |
| 17:42.14 | [TK]D-Fender | :p |
| 17:42.46 | delroy | any echo problems? |
| 17:43.04 | WIMPy | pabelanger: you should use gconfed to put ekiga on a different port. |
| 17:44.11 | Katty | distributes baby carrots and grapes |
| 17:45.00 | p3nguin_ | pabelanger: Did you change the port of ekiga using gconf-editor? |
| 17:45.07 | delroy | are the sip gateways better than pci card from sagnoma or digium? |
| 17:45.12 | pabelanger | WIMPy: Ya, I don't think the port is the issue, but it looks like my mutli NIC on my system is |
| 17:45.22 | [TK]D-Fender | delroy: Generally no. |
| 17:45.39 | [TK]D-Fender | delroy: What are your actual requirements? |
| 17:45.47 | WIMPy | pabelanger: By default it will be. |
| 17:46.24 | p3nguin_ | pabelanger: Since both Asterisk and ekiga will want to use 5060, it will be a problem until you change ekiga to use another port. |
| 17:46.34 | delroy | [TK]D-Fender: Have 3 locations that need max 4 lines and one that has 8 lines. |
| 17:46.37 | WIMPy | In fact, starting ekiga prevented the running Asterisk from communicationg with the anything. |
| 17:46.53 | delroy | Client picky about quality so echo will no be tolerated |
| 17:47.10 | Katty | i would use a sip providor for that |
| 17:47.15 | [TK]D-Fender | delroy: Sangoma B600d |
| 17:47.17 | Katty | personally |
| 17:47.27 | Chainsaw | delroy: I like the fallback options that a separate device offers me. |
| 17:47.40 | Chainsaw | delroy: Also, it means I get to talk SIP to Asterisk, which happens to be the most robust stack it has. |
| 17:47.49 | Chainsaw | delroy: (Echo is not a problem, no) |
| 17:48.16 | [TK]D-Fender | Chainsaw: Several thoughts : Fax, cost, analog Flash, etc |
| 17:48.45 | Chainsaw | [TK]D-Fender: T.38 works well, I grant you that one, that works fine, do go on. |
| 17:48.47 | delroy | I recently deployed an Audiocodes box and client reports echo every so often |
| 17:49.17 | [TK]D-Fender | Chainsaw: How would * signal the Patton to do a "flash"? |
| 17:49.46 | Chainsaw | [TK]D-Fender: There's config for it, no doubt there's support. |
| 17:49.59 | Chainsaw | [TK]D-Fender: I can even join phone conferences with my pulse-dial phone. |
| 17:50.04 | [TK]D-Fender | Chainsaw: default = doubt :) |
| 17:50.16 | Chainsaw | [TK]D-Fender: (Because it sends DTMF for the digits I dial with it) |
| 17:50.21 | [TK]D-Fender | ChaAnd that latte = FXS, not FXO |
| 17:50.31 | [TK]D-Fender | latter* |
| 17:52.59 | delroy | is the firmware complicated with the pattons? The audiocodes is a nightmare |
| 17:53.48 | [TK]D-Fender | delroy: I wouldn't say "nightmare", but not as clear as others, that's for use. give Mediatrix a whirl |
| 18:01.55 | *** join/#asterisk deonv (~adium@196.1.28.226) |
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| 18:07.09 | _omer | any suggestion ?? http://www.pastebin.ca/1980856 |
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| 18:33.53 | alex_voip | hello if i buy 10 g729 licenses, that is 10 x $10, how many calls can i have up at the same time? The description says 1 concurrent call, but the details says it's licensed on a per-channel basis. |
| 18:34.44 | russellb | 10 = 10 decoders and 10 encoders |
| 18:34.50 | russellb | so ... it depends |
| 18:34.51 | [TK]D-Fender | alex_voip: 10 licenses = 10 concurrent transcodes |
| 18:36.38 | alex_voip | heh ok so it's not the sum of encoders and decoders as in 6/4 = 10 and i'm out of licenses |
| 18:37.07 | [TK]D-Fender | alex_voip: Correct |
| 18:37.49 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 18:38.10 | [TK]D-Fender | alex_voip: You could be using a decoder alone if * is doing a Playback of a non G.729 file to a G.729 channel (not a "call"), or an encoder if recording to g.729 from a non G.729 channel, etc |
| 18:38.24 | [TK]D-Fender | alex_voip: Hence why its not always 1-1 |
| 18:38.40 | [TK]D-Fender | alex_voip: Your typical "call" will be a matched pair of course |
| 18:39.04 | coppice | or two licences if you want to log a call that is passing through |
| 18:41.12 | alex_voip | [TK]D-Fender, thanks that clears it up for me...i interpreted the store's description of "per-channel" to mean you needed to buy one for encoding and 1 for decoding |
| 18:41.47 | alex_voip | [TK]D-Fender, now i understand that each license comes with one of each |
| 18:42.09 | [TK]D-Fender | alex_voip: The assumption of sorts is that to bridge to a non G.729 channel you'd need to decode audio FROM it, and encode TOWARDS it. They could have been a bit clearer for you |
| 18:42.53 | alex_voip | [TK]D-Fender, i get that, what is not clear is that each license includes a decoding AND encoding license |
| 18:43.10 | coppice | that pairing of encode and decode is the basis of the patent licencing |
| 18:43.45 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:44.03 | Pio | [TK]D-Fender, you got any tips for tweaking the quality of calls on the PSTN line on that SPA 3102? |
| 18:44.24 | [TK]D-Fender | Pio: www.voxilla.com <- check out their forums, they have lots of guides there |
| 18:44.31 | Pio | yeah i found some of that stuff googling last night |
| 18:44.36 | [TK]D-Fender | Pio: I don't have that much personal experience having to tweak them |
| 18:45.13 | Pio | yeah. i have mine sounding passable now.. i spent an hour or two on it last night.. i made it waaay worse then i gradually was able to get it better |
| 18:45.15 | coppice | the PSTN port on an SPA3102 is very troublesome |
| 18:45.18 | Pio | still could be better though |
| 18:45.20 | Pio | yeah sure is |
| 18:47.02 | coppice | what is your problem? echo is the usual one with the SPA3102 |
| 18:47.07 | Pio | yeah echo |
| 18:47.46 | Pio | if i turn the echo cancellation off its insane |
| 18:47.49 | coppice | you won't solve it. the echo canceller in that device is badly broken. lots of ITSPs have ripped out large numbers and given up on them |
| 18:47.57 | bmoraca_work | is the TCE400P any good? |
| 18:48.03 | Pio | yeah .. well i've got it better than it was at least |
| 18:48.06 | Pio | it seems at best its barely passable |
| 18:49.03 | [TK]D-Fender | bmoraca_work: I'm sure it does its job. Does it fit your needs? |
| 18:49.10 | bmoraca_work | it might |
| 18:49.19 | bmoraca_work | i'm just curious about other peoples' experience with it |
| 18:49.50 | Pio | i gotta admit i dont exactly understand the nature of the echo.. like why is there echo at all |
| 18:49.51 | coppice | if you need G.723.1 your only choices are that, and the new sangoma card |
| 18:49.53 | [TK]D-Fender | bmoraca_work: Given the cost I'd say you'd want to need to just about max it out to get your money's worth. |
| 18:50.10 | [TK]D-Fender | coppice: Oh? What's the model #? |
| 18:50.32 | coppice | model number of what? |
| 18:50.39 | bmoraca_work | well, the draw for me is the ability to offload g729 encoding and decoding from the CPU and software-based codecs |
| 18:51.06 | coppice | bmoraca_work: why do you care about offloading? |
| 18:51.39 | bmoraca_work | because i don't trust the quality and i hope to get more capacity from my server |
| 18:51.46 | *** join/#asterisk josephnexus (~josephnex@71-209-40-81.bois.qwest.net) |
| 18:51.51 | josephnexus | anyone have any ideas on using flash transfer as part of an IVR? |
| 18:52.14 | josephnexus | or, even better, as part of a follow me if the call originated on a POTS line? |
| 18:52.32 | coppice | 120 G.729 channels is <1.5 cores of a quad core CPU. the digium card is not very high capacity. the sangoma card goes up to something like 500 channel |
| 18:53.22 | bmoraca_work | my system is two single core Xeons. upgrading the server is more expensive than adding this DSP card |
| 18:55.54 | coppice | I guess single core xeons means P4s. you could save the cost of a new server in reduced power bills :-) |
| 18:56.57 | [TK]D-Fender | coppice: I found the Sangoma cards you were mentioning... nice denisty options for it. |
| 18:57.00 | [TK]D-Fender | density* |
| 18:57.08 | [TK]D-Fender | D100-480E Up to 480 Transcoding Sessions |
| 18:57.24 | delroy | crap - i just ordered a spa3102 to play with !!! arg! |
| 18:57.27 | coppice | I thought you were a sangoma user. they've been out for a while now. |
| 18:58.11 | [TK]D-Fender | coppice: I am historically, but haven't had to buy anything for some time, and missed any news releases they may have made.. just not up to date is all. |
| 18:58.34 | coppice | it supports quite a few codecs. the digium cards only support G.729A and G.723.1 |
| 19:00.08 | bmoraca_work | coppice: power's a flat cost for me. i don't pay per kwh. |
| 19:00.44 | Katty | if anyone is interested in joining the asterisk christmas card exchange, please /query me. |
| 19:01.07 | riscphree | haha what |
| 19:01.56 | bmoraca_work | hmm...the sangoma one is roughly the same cost, and supports ilbc...interesting |
| 19:03.36 | josephnexus | katty... are we all exchanging our fxo/fxs cards? |
| 19:03.42 | josephnexus | what of the modules? |
| 19:03.45 | josephnexus | :-P |
| 19:04.00 | Katty | no just christmas cards. |
| 19:04.10 | josephnexus | twas a joke |
| 19:06.23 | WIMPy | postcareds are much more useful |
| 19:07.45 | drfreeze | Hi. Does anyone have experience with turning up the mic gain on polycom phones? |
| 19:07.48 | drfreeze | I am trying to get a 335 to pick up a voice 15ft away |
| 19:08.09 | Katty | WIMPy: you can do that too |
| 19:08.43 | _omer | any suggestion ?? http://www.pastebin.ca/1980856 |
| 19:09.46 | WIMPy | Katty: I didn't mean to say that I pfind postcards useful. |
| 19:10.04 | [TK]D-Fender | drfreeze: gains are easily findable in the sip.cfg |
| 19:19.40 | *** join/#asterisk devmod (~devmod__@c-76-100-208-204.hsd1.md.comcast.net) |
| 19:20.32 | devmod | Hello, I just installed ast 1.8 and even when I set debug and verb to 9 - I don't see any msgs on the console. any ideas? |
| 19:22.51 | putnopvut | devmod: check logger.conf and look for a console => line |
| 19:23.24 | putnopvut | If you don't have "debug" on that line, you won't see debug messages on the console. |
| 19:23.40 | putnopvut | Verbose, though...that should show up as long as you have verbosity set high. |
| 19:26.25 | devmod | right, that is what I expected but I see nothing even when calling in to an extension |
| 19:28.22 | Jasnejac | I have a D100-480E to test. Sounds like it will be fun |
| 19:28.35 | Nugget | http://www.geekosystem.com/britains-got-telnet/ |
| 19:29.27 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 19:30.29 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 19:33.49 | drfreeze | I am trying to setup BLF on polycom phones. |
| 19:33.57 | drfreeze | Following instructions at http://forums.contribs.org/index.php?topic=42805.0 |
| 19:34.13 | drfreeze | So far I have done the following: http://pastie.textmate.org/private/axdsddvi5fysvcfa6lpga |
| 19:34.41 | drfreeze | But, instead of the icon that is a profile of a person, I get the 10 dot icon |
| 19:35.43 | drfreeze | Anyone know what that icon means? |
| 19:36.00 | [TK]D-Fender | drfreeze: means you didn't enable buddy-watch on the contact |
| 19:38.44 | drfreeze | The menu shows Buddy Watch Enabled, but that is on the Watcher phone |
| 19:39.10 | drfreeze | Does presence need to be enabld on the watchee phone? |
| 19:41.15 | joel_oliveira | hi, is it possible to have the status, that is shown when the 'sip show peers' action is executed on the CLI, available on the dialplan as a function so I can save it to the CDR? I mean: to get the peer status upon the placing of a call? |
| 19:42.05 | [TK]D-Fender | drfreeze: no. pastebi your sip.conf and extensions.conf that are supposed to support this. |
| 19:42.46 | *** join/#asterisk moltar_net (~Roman@CPE940c6dac4ffb-CM001ade8cc08e.cpe.net.cable.rogers.com) |
| 19:43.53 | drfreeze | I have not made any changes to extensions.conf |
| 19:43.59 | drfreeze | Am posting sip.conf |
| 19:45.24 | [TK]D-Fender | drfreeze: EXTENSIONS.CONF PLEASE..... |
| 19:46.10 | drfreeze | http://pastie.textmate.org/private/xquqv8q6zc9w8bdi7jrqia |
| 19:47.42 | [TK]D-Fender | drfreeze: sip.conf entries need to be type=peer |
| 19:47.45 | [TK]D-Fender | drfreeze: EXTENSIONS.CONF PLEASE..... |
| 19:51.51 | drfreeze | [TK]D-Fender: http://pastie.textmate.org/private/det7r0uabwyrjaclrlpew |
| 19:54.00 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 19:54.08 | drfreeze | One thing that puzzles me is that the directory xml file does not have the <bw> or <bb> properties |
| 19:54.17 | drfreeze | but, the phone says that bw is enabled |
| 19:54.36 | p3nguin_ | Your "long distance" extension patterns are fail. |
| 19:54.39 | [TK]D-Fender | drfreeze: You don't even HAVE any hints to subscribe against <------------ |
| 19:55.14 | ManxPower | drfreeze, then your phone has not downloaded the updated config file. |
| 19:55.36 | [TK]D-Fender | Nope.. the phone has not UPLOADED its changes to the server <- |
| 19:55.44 | [TK]D-Fender | Silly backwards people... |
| 19:55.51 | ManxPower | [TK]D-Fender, that depends on who makes the changes, him or the phone |
| 19:56.13 | ManxPower | But if the phone says he has buddies then that is what the phone thinks the config is |
| 19:56.19 | [TK]D-Fender | ManxPower: "phone downlaods" don't change the contexts of a file on the provisioning server. |
| 19:56.30 | [TK]D-Fender | ManxPower: re-read teh direction of that a few more times... |
| 19:56.43 | ManxPower | <drfreeze> One thing that puzzles me is that the directory xml file does not have the <bw> or <bb> properties |
| 19:56.44 | ManxPower | <drfreeze> but, the phone says that bw is enabled |
| 19:56.46 | ManxPower | you mean that? |
| 19:57.15 | drfreeze | yes |
| 19:57.28 | ManxPower | I read that as the config file on the server does not have buddies defined, but the phone shows buddies defined. I am assuming he is trying to "undefine" buddies. |
| 19:57.35 | drfreeze | if I look at the properties for a directory listing, Buddy Watch is Enabled |
| 19:57.58 | ManxPower | drfreeze, do you want buddies enabled or disabled? |
| 19:58.13 | drfreeze | I reboot the phone and look at the directory xml file, and it does not have the <bw> property. Maybe because <bw>1</bw> is the default |
| 19:58.15 | [TK]D-Fender | drfreeze: that just means it hasn't written the directory BACK to the server. |
| 19:58.27 | [TK]D-Fender | drfreeze: which is irrelevant. YOU HAVE NOTHING YOU CAN WATCH |
| 19:58.35 | drfreeze | ManxPower: enabled - I want to see the status of a 'buddie's phone |
| 19:58.53 | ManxPower | drfreeze, then the phone is not uploading your contacts change, like [TK]D-Fender said. |
| 19:59.01 | ManxPower | have you checked the FTP logs? |
| 19:59.10 | [TK]D-Fender | IRRELEVANT MOVE ALONG |
| 19:59.30 | ManxPower | [TK]D-Fender, he is going to have to resolve the issue at some point |
| 19:59.35 | [TK]D-Fender | ManxPower: Nope. |
| 19:59.37 | drfreeze | So, what does the poster mean by 'heading' of sip.conf? [General] ?? |
| 19:59.43 | [TK]D-Fender | ManxPower: Comlpetely not required |
| 19:59.52 | ManxPower | so when he fixes the hints the phone will magically upload the contacts? |
| 20:00.30 | [TK]D-Fender | ManxPower: the phone reports it has contacts. Writing back to FTP is a backup bonus |
| 20:01.12 | [TK]D-Fender | [15:58]<[TK]D-Fender>drfreeze: which is irrelevant. YOU HAVE NOTHING YOU CAN WATCH <------------------- |
| 20:01.40 | ManxPower | Odd, there being a non-trixbox user here asking for help. 8-| |
| 20:01.58 | ManxPower | drfreeze, buddy watch is NOT a default. |
| 20:02.22 | *** join/#asterisk imox1234 (~imox1234@p4FC5C53E.dip0.t-ipconnect.de) |
| 20:02.24 | drfreeze | I have changed the contact file and reboot the phone to see if it uploads the file |
| 20:02.26 | *** join/#asterisk ukine_droid (~ukine@64.134.182.140) |
| 20:03.03 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:03.27 | devmod | Any idea why I can't see any messages (debug and verbose = 9) when connecting to an existing asterisk 1.8 instance |
| 20:04.20 | ManxPower | drfreeze, I'm starting to agree with [TK]D-Fender stop working on this part and work on the other part that is required to make this work |
| 20:04.31 | ManxPower | devmod, did you read the UPGRADE*.txt files? |
| 20:06.03 | drfreeze | ManxPower: fine. What needs to be done? |
| 20:06.17 | devmod | ManxPower: oh k thx |
| 20:06.41 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 20:06.43 | drfreeze | Both phones have sip.conf modified and the sip.conf has been updated |
| 20:07.39 | ManxPower | define the hints in extensions.cof |
| 20:07.39 | [TK]D-Fender | drfreeze: haev I not been clear enough? |
| 20:08.11 | drfreeze | nope |
| 20:08.22 | drfreeze | As far as I can tell, there is a 4 step process |
| 20:08.37 | [TK]D-Fender | drfreeze: You have not fucking HINTS in your dialplan. Even when you phone is READY to watch it has nothing to ASK TO WATCH |
| 20:08.48 | [TK]D-Fender | HINT <-------------------- |
| 20:08.52 | drfreeze | I have done all four on one phone and 3 on the other (I assume the watched phone does not need to have the watcher as a buddy) |
| 20:09.50 | [TK]D-Fender | drfreeze: that guide shows how to do the PHONE side, it means jack shit about the ASTERISK side |
| 20:11.14 | ManxPower | drfreeze, please pastebin the hint lines from your extensions.conf |
| 20:11.22 | [TK]D-Fender | ManxPower: HE HAS NONE |
| 20:11.29 | ManxPower | without hints in extensions.conf buddy watch will never work. |
| 20:11.34 | [TK]D-Fender | I've beaten this point to fucking DEATH now. |
| 20:11.38 | ManxPower | [TK]D-Fender, the he will have to either admit that or pastebin them |
| 20:11.43 | [TK]D-Fender | HE DID |
| 20:11.56 | ManxPower | [TK]D-Fender, put him on /ignore then |
| 20:11.59 | [TK]D-Fender | [15:47]<[TK]D-Fender>drfreeze: EXTENSIONS.CONF PLEASE..... |
| 20:12.01 | [TK]D-Fender | [15:51]<drfreeze>[TK]D-Fender: http://pastie.textmate.org/private/det7r0uabwyrjaclrlpew |
| 20:13.11 | drfreeze | take break man |
| 20:13.23 | ManxPower | yup, no hints are defined in extensions.conf so buddy watch / presence is guaranteed to not work. |
| 20:14.46 | drfreeze | right, found some docs on that |
| 20:15.00 | drfreeze | I guess once hints are turned on, then I'll get the buddy icon |
| 20:15.20 | WIMPy | Hmm. The phones could watch each other without using Asterisk, I guess. |
| 20:15.21 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:15.53 | devmod | ManxPower: read the updates*.txt and nothing there |
| 20:17.08 | pabelanger | I know MACRO_CONTEXT exists for Marco, but what about gosub? Aside from passing the CONTEXT as an argument |
| 20:20.22 | ManxPower | devmod, how about the sample config file? |
| 20:20.46 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 20:21.47 | *** part/#asterisk Gibby_away (~gibby@204.118.10.244) |
| 20:26.05 | *** join/#asterisk jhirley (~chatzilla@c-75-74-13-194.hsd1.fl.comcast.net) |
| 20:27.24 | bneff | why would I get a 401 Unauth when my client uses UDP, but TCP works fine? |
| 20:28.08 | [TK]D-Fender | checkout time, BBIAB |
| 20:28.16 | ManxPower | you will always get an unauth the first try, that is the way MD5 Digest works |
| 20:29.07 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 20:29.12 | *** join/#asterisk knot (~knotsucke@unaffiliated/devemo) |
| 20:29.57 | bneff | I see that in the traces, but it will never auth when the client is UDP |
| 20:30.09 | bneff | could it be that a firewall in the path is faking this? |
| 20:30.55 | drfreeze | if 523 dials 522, and 522 has 'hint' in the extensions.conf, then how does that help 523 know when 522 makes a call? |
| 20:32.16 | bneff | hmm..no firewall, its actually hitting asterisk via sip debugging |
| 20:33.48 | ManxPower | drfreeze, asterisk will not monitor the state of an extension without a hint. |
| 20:34.16 | paulc | Anyone know an XMPP client that supports rendering of "formatting".. like <span> and the like? (XHTML-IM is the standard, I think?) - Pandion gives me bold using *bold here* type formatting, which might be enough.. but I wouldn't mind a splash of colour here and there either.. |
| 20:34.29 | ManxPower | drfreeze, the phone asks asterisk to notify it when the buddy watched extension state changes. Asterisk will not do this without a hint configured for the monitored extension |
| 20:36.31 | drfreeze | is the 'hint' step done when extesions.conf is loaded? I'm trying to figure out how the hint extension is priorizted wrt 1,2,n,n,... |
| 20:38.32 | ManxPower | it isn't |
| 20:38.36 | ManxPower | it is a special priority |
| 20:38.47 | drfreeze | ok, found this: in extensions.conf. "hint" extensions and "real" extensions are |
| 20:38.54 | drfreeze | separate entities |
| 20:38.59 | ManxPower | there you ho |
| 20:39.00 | ManxPower | go |
| 20:39.29 | ManxPower | hint = monitor the state of this extension |
| 20:40.28 | shapr | sets up the D-Fender channel on his PRI |
| 20:41.04 | ManxPower | drfreeze, have you looked at the voip-info info about hints and buddies? |
| 20:42.07 | drfreeze | ManxPower: yes. got it working now. All was missing was hints |
| 20:42.19 | drfreeze | That wasn't in the first three articles I read |
| 20:46.09 | *** join/#asterisk Zairus (~charrit69@193.111.165.83.dynamic.mundo-r.com) |
| 20:46.34 | ManxPower | one of the few topics voip-info.org is the best place to get information about is hints. |
| 20:49.09 | citywok | voip-info is really good, but a lot of the content is pretty out dated. it will be nice to see the new wiki work if it keeps current |
| 20:49.59 | Zairus | hi |
| 20:51.37 | *** join/#asterisk [netman] (~netman@59.Red-79-146-230.dynamicIP.rima-tde.net) |
| 20:51.42 | Zairus | anybody knows if there is a way for making the user abble to accept the call and the bridge it with a established channel? |
| 20:51.49 | Zairus | I can explain it better |
| 20:53.01 | Zairus | I have an incoming call in moh, and I need to pass the call to a mobile, but the mobile could be switched off... |
| 20:54.26 | Zairus | then, my idea is to make the person to press one key in the mobile to assure that he answered the call and then bridge it to the channel in moh |
| 20:54.29 | Zairus | any idea? |
| 20:55.27 | josephnexus | i think that's a hunt strategy zairus |
| 20:55.30 | josephnexus | unsure of the name of it though |
| 20:56.54 | Zairus | what do you mean with 'hunt'? pick up? |
| 20:58.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:59.29 | citywok | lol, wiki.asterisk.org just died |
| 20:59.51 | citywok | digium, hallllp |
| 21:00.16 | erinspice | Known. :) |
| 21:00.43 | josephnexus | zairus, sorry, ring strategy |
| 21:01.47 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:02.03 | BMJ | citywok: **kpfleming scrambles to address it |
| 21:02.04 | boodu | do you know if misdnv2 works with kernel 2.6.18 ? |
| 21:02.07 | Zairus | ok, thanks josephnexus, i'll take a look |
| 21:02.33 | citywok | BMJ: thanks! :P |
| 21:03.09 | BMJ | citywok: My pleasure! |
| 21:03.28 | devmod | the Playback app doesnt seem to play the video files anymore on ast 1.8 |
| 21:04.55 | pabelanger | ~backtrace |
| 21:04.55 | infobot | it has been said that backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt) |
| 21:06.30 | BMJ | Wiki's back. |
| 21:07.12 | Qwell | citywok: You totally broke it. |
| 21:07.58 | citywok | Qwell, sorry! I was browsing it to see what all is on it :P -- sorry for the high load of 3-4 clicks per minute :P |
| 21:08.22 | Qwell | Just send in a written confession and all will be well. |
| 21:08.23 | citywok | I lied. i'm not sorry. |
| 21:08.52 | citywok | Hmmm. "Dear hostmaster of wiki.asterisk.org, please use /etc/init.d/apache2 restart more often" :P |
| 21:08.59 | Qwell | it's java. |
| 21:09.10 | Qwell | so yeah... |
| 21:09.42 | citywok | lol |
| 21:10.08 | citywok | well in that case it's amazing the site runs as well as it does |
| 21:14.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 21:33.40 | *** part/#asterisk BMJ (~bjohns@cpe-098-026-116-043.nc.res.rr.com) |
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| 21:38.34 | fireman_biff | Hi, if I'm getting red alarms on a PRI, would that be a problem with the pbx or with the provider's equipment? Or is that not enough information to determine the location of the problem? |
| 21:39.03 | *** join/#asterisk [netman] (~netman@59.Red-79-146-230.dynamicIP.rima-tde.net) |
| 21:39.31 | WIMPy | It means you don't have a working connection. For whatever reason. |
| 21:39.38 | citywok | did it work before? has anything changed on your end? |
| 21:39.56 | citywok | your PCI card could have gone bad, or your provider could be having issues. If you provider is good, just call them and they can figure it out. |
| 21:40.02 | fireman_biff | it has been working for months, nothing has changed on my end |
| 21:40.18 | petern_ | could just be a telco fault. they happen sometimes... |
| 21:40.21 | WIMPy | Ah, the famous digger :-) |
| 21:40.42 | citywok | Yes, the guy with the backhoe that decides to chew up a chunk of fibre was always my favorite. |
| 21:40.54 | fireman_biff | unfortunately my provider's first response tends to be "everything's fine on our end"... but I'll give them a call and see what happens |
| 21:40.57 | josephnexus | it's also rather expensive |
| 21:41.10 | fireman_biff | btw, its not a constant red arlams |
| 21:41.17 | Katty | if anyone is interesting in participating in the Asterisk Christmas Card Exchange, please /query me |
| 21:41.18 | fireman_biff | it happens maybe a couple times an hour |
| 21:41.20 | citywok | josephnexus: rolling out to fix the torn up fibre? |
| 21:41.21 | fireman_biff | and makes call drops |
| 21:41.35 | josephnexus | citywok, torn up fibre is costly to replace |
| 21:41.59 | citywok | yea, especially with all those SLA's that just got broken now as well. lol. |
| 21:42.00 | WIMPy | fireman_biff: For how long? |
| 21:42.25 | fireman_biff | WIMPy: it doesn't stay red, just drops the calls and then you have to redial |
| 21:42.28 | citywok | bouncing can very easily be an issue on the providers end. i have 8 mutlilink t1's and had a pair of them bouncing last month, bad interface card. |
| 21:42.50 | WIMPy | fireman_biff: Talk in seconds. |
| 21:44.32 | p3nguin_ | Anyone know a good chipset for a PCI gigabit Ethernet card that's going to have the driver in almost every Linux kernel? I was looking at an RTL8169SC, but it seems that it's not a real popular chipset in some kernels. |
| 21:44.52 | fireman_biff | WIMPy: the time between "alarm detected" and "alarm cleared" is 3 seconds, that's what you're asking about right? |
| 21:44.52 | citywok | The intel stuff works pretty well for me |
| 21:45.00 | WIMPy | inel? |
| 21:45.50 | WIMPy | fireman_biff: Yes. That's really short. I guess you have to work that out together with your telco. |
| 21:46.12 | WIMPy | Howevr, calls are not supposed to be droped that fast. |
| 21:46.38 | fireman_biff | alright, I'll give them a call, thanks |
| 21:47.14 | WIMPy | By default they should survive a L1 failure for 10 seconds. |
| 21:47.59 | citywok | really? i'm pretty sure whenever we dropped T1's the calls were gone pretty fast |
| 21:48.15 | citywok | but then again, when we dropped a T1 to get from a red alarm back to green was like 30 seconds to sync up |
| 21:48.41 | WIMPy | Wow. That's a long time. And >10s. |
| 21:49.19 | citywok | Yea. That was with our Inter-Tel PBX, though i think with Asterisk it was the same. Going through an Adtran DS3 mux |
| 21:49.24 | *** part/#asterisk bougyman (bougyman@pdpc/supporter/gold/bougyman) |
| 21:49.51 | fireman_biff | actually for me the first entry mentioning the alarm happens in the same second that macro-hangupcall gets run |
| 21:49.59 | fireman_biff | so it seems pretty much instant |
| 21:51.24 | WIMPy | T309 is the one. |
| 21:52.03 | WIMPy | Ok. here it says 6s. |
| 21:52.10 | p3nguin_ | The problem I've run into looking for Intel NICs is that there are no regular PCI gig-Ethernet ones out there. Anything that looks promising turns out to be PCI-X. |
| 21:52.43 | WIMPy | PCI is too slow. |
| 21:53.21 | p3nguin_ | It might be slower than PCI-E, but it obviously still works or other brands wouldn't have gigabit Ethernet PCI cards. |
| 21:53.56 | p3nguin_ | I just need one that is going to be well supported. |
| 21:55.04 | p3nguin_ | The box I want to use the card in does not have the rtl8169 driver, so that tells me that it isn't well supported. |
| 21:55.09 | citywok | Hmm. I've used a couple of the PCI-X nic's in a 32bit pci-slot just fine |
| 21:55.23 | titter | Is it possible to set the cdr(userfield) from sip.conf per user? |
| 21:55.34 | p3nguin_ | I'd be taking a risk using the wrong card in my slot. |
| 21:55.42 | citywok | that's what she said |
| 21:56.12 | WIMPy | But beware that PCI-X is usually (always?) 3.3V. |
| 21:56.18 | *** join/#asterisk [cannibalera] (~cannibale@201-24-97-93.fnsce703.dsl.brasiltelecom.net.br) |
| 21:56.33 | WIMPy | And the rtl8169 is very standard as well. Just as tg3. |
| 21:56.48 | *** part/#asterisk [cannibalera] (~cannibale@201-24-97-93.fnsce703.dsl.brasiltelecom.net.br) |
| 21:58.43 | p3nguin_ | Oh, I should be looking for the tg3 module, then? |
| 21:58.56 | p3nguin_ | /lib/modules/2.6.9-78.0.8.EL/kernel/drivers/net/tg3.ko |
| 21:59.01 | p3nguin_ | got that one! |
| 21:59.02 | WIMPy | Only if you have a tg3. |
| 21:59.09 | p3nguin_ | hmm |
| 21:59.15 | p3nguin_ | I don't understand what you meant, then. |
| 22:00.56 | p3nguin_ | I do have r8169.ko in there. I wonder if there is any chance that's the right one. |
| 22:02.03 | WIMPy | has a rtl8101e/rtl8102e working with it. |
| 22:02.16 | WIMPy | But that's not GE. |
| 22:03.06 | p3nguin_ | Lots of people complaining about issues with that r8169.ko module. |
| 22:03.38 | WIMPy | I didn't have any so far. |
| 22:03.40 | p3nguin_ | I guess I could spent a few bucks on the card and try it. The worst that will happen is that it won't work and I'll have a spare NIC for a Windows box. |
| 22:03.50 | p3nguin_ | spend, rather |
| 22:09.47 | *** part/#asterisk fireman_biff (~biff@65.48.133.102) |
| 22:10.26 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 22:10.26 | boodu | WIMPy, the way to misdnv2 is long (kernel recompilation....) |
| 22:11.05 | WIMPy | boodu: It has been there for over two years. |
| 22:12.11 | boodu | i have a 2.6.18 :( |
| 22:13.21 | p3nguin_ | Alright, I bought the Realtek card. We'll see if it's compatible after it arrives. |
| 22:23.14 | WIMPy | is just looking at chan_misdn of 1.8 and I have to say that looks like someone did an excellent job there. But why for a driver model that has been abandoned? |
| 22:24.31 | russellb | WIMPy: customers. :-) |
| 22:24.33 | WIMPy | I actually think I want to try that. But after I get everything smooth with 1.8. |
| 22:29.59 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 22:44.36 | boodu | WIMPy, you say misdn 1.8 works fine with b400E ? |
| 22:45.40 | WIMPy | boodu: I don't know the pci ID of the b400e, but misdn2 definitely supports HFC-4S an HFC-8S. |
| 22:47.35 | boodu | is lost with his b400e |
| 22:48.10 | WIMPy | Can't you find it? |
| 22:50.39 | boodu | WIMPy, may be my problem with misdn can be a jumber or not ? |
| 22:51.05 | WIMPy | What's happening now? |
| 22:51.39 | boodu | it' always the sames problem i can't call more than 1 number |
| 22:51.41 | WIMPy | And does the card have any jumpers except for those to reverse the sockets and add phantom power? |
| 22:51.50 | boodu | now i try to use dahdi |
| 22:52.00 | WIMPy | Ah, that one. |
| 22:52.24 | WIMPy | Well. That actually suggest that the problem might lie somewhere else. |
| 22:52.29 | boodu | i have just connect my card without change any jumper |
| 22:52.48 | WIMPy | What do you have running now? |
| 22:53.01 | boodu | dahdi |
| 22:53.11 | boodu | and dahdi is configurated |
| 22:53.45 | WIMPy | Ok. And it behaves the same as with chan_misdn? |
| 22:53.49 | boodu | i can see my line active |
| 22:54.16 | boodu | it's my first time with dahdi |
| 22:54.33 | boodu | i don't know how use Dial with it :$ |
| 22:55.12 | WIMPy | You define groups there as well. |
| 22:55.34 | WIMPy | But as you are having trouble do dial out, I suggest you just try the schannel drictely. |
| 22:55.55 | WIMPy | Try one by one if you can dial out. |
| 22:56.27 | boodu | good new the incoming seems to be detected because i have : |
| 22:56.31 | boodu | Accepting call from '258838' to '3899' on channel 0/1, span 4 |
| 22:56.32 | boodu | <PROTECTED> |
| 22:56.38 | boodu | :D |
| 22:56.58 | boodu | gloupss i post the number phone |
| 22:57.05 | boodu | ^^ |
| 23:10.25 | Kobaz | how can i get the sip error from a failed call... |
| 23:10.36 | Kobaz | <PROTECTED> |
| 23:16.52 | boodu | WIMPy, with dahdi when i use group i have exactly the same problem that i have with misdn |
| 23:17.18 | *** join/#asterisk ANurmi (~Adam@63.230.70.254) |
| 23:18.59 | citywok | kobaz: sib set debug on? |
| 23:19.05 | citywok | or sip set debug ip 192.168.55.98 |
| 23:19.48 | Kobaz | i know that |
| 23:19.54 | Kobaz | how do i get it via dialplan |
| 23:20.19 | Kobaz | i found a post on the bug tracker about someone trying to do the same thing... the response was use HANGUPCAUSE |
| 23:20.23 | Kobaz | is that still the case? |
| 23:20.34 | Kobaz | the issue was posted in 2007 |
| 23:20.55 | *** join/#asterisk JoseBravo (~Jose@190.144.124.194) |
| 23:21.10 | citywok | Kobaz: this? http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 23:21.13 | JoseBravo | Hello |
| 23:21.19 | WIMPy | Kobaz: I'm pretty sure that's at least true gor 1.6. |
| 23:24.17 | Kobaz | according to the wiki in 1.8 you can get the specific item from the sip header, nice |
| 23:24.23 | Kobaz | using 1.6 though... i'll play with hangup cause |
| 23:25.35 | Kobaz | trying to work around a buggy sip-fxs gateway that randomly returns 503 service unavailable when making outbound calls |
| 23:26.39 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 23:27.55 | WIMPy | That's why I prefer internal stuff. |
| 23:28.17 | Kobaz | heh |
| 23:28.45 | Kobaz | i like external stuff because you never know what kernel update or libpri update, or dahdi update will break everything |
| 23:29.15 | citywok | The other nice part of external is you can have redundant * servers |
| 23:29.43 | Kobaz | yeah |
| 23:30.03 | citywok | thats why i like being pure sip now, just move the IP's from one server to the other. |
| 23:30.14 | citywok | T1's are messy :P |
| 23:30.17 | WIMPy | Don't upgrade :-) But the chances of getting issus fixed is much higher, |
| 23:31.46 | JoseBravo | I have Asterisk 1.6.0 and I changed the codec from my SIP trunk from g711 to g729 and now I have one way audio problem. Any idea? |
| 23:32.14 | Kobaz | if i had t1/pri/q931/q921/etc experience i would write tests up the wazoo for all the asterisk tdm stuff... but instead i just go sip |
| 23:34.10 | bmoraca_work | Kobaz: that could be it's way of saying "the number is busy" |
| 23:34.34 | Kobaz | bmoraca_work: if the number is busy, it passes the audio through of a busy tone |
| 23:34.47 | bmoraca_work | why on earth would it do that? |
| 23:34.58 | Kobaz | 503 unavailable seems to mean that the gateway fscked up and couldn't make the call... you try again 2 seconds later and it works |
| 23:35.15 | Kobaz | bmoraca_work: because it's a grandstream :( |
| 23:35.21 | bmoraca_work | AHAHAHAHAHAHHAHAHA |
| 23:35.24 | Kobaz | heh |
| 23:35.31 | WIMPy | Argh |
| 23:35.49 | bmoraca_work | sorry, i feel your pain |
| 23:36.03 | Kobaz | customer wants cheap... i really don't have much of an option |
| 23:36.18 | bmoraca_work | i went through similar struggles when I configured a largely undocumented Cisco AS5400 appliance as a media gateway |
| 23:36.20 | Kobaz | bmoraca_work: callerid gets screwed up about 30% of the time |
| 23:36.40 | Kobaz | it will come in as callerid: 31711#W#%^@&@@#@#^p^ |
| 23:36.53 | bmoraca_work | nice |
| 23:37.25 | ManxPower | Kobaz, increase or decrease your rxgain by 1db |
| 23:37.55 | ManxPower | maybe more, but not too much |
| 23:38.08 | bmoraca_work | i'm taking a break from VOIP stuff and setting up a miniature VRF network...wooo |
| 23:39.19 | Kobaz | ManxPower: hmm, really? |
| 23:39.42 | ManxPower | Kobaz, I didn't read far enough back, my comment only applies to PSTN |
| 23:40.12 | Kobaz | er i mean.. this is an fxo-sip gateway |
| 23:40.17 | ManxPower | try it |
| 23:40.27 | Kobaz | yeah. just did |
| 23:40.45 | ManxPower | usually the gain needs to be increased but if your calls are loud, you can try decreasing it. don't go too far in either direction or you'll have other audio issues |
| 23:40.51 | Kobaz | it's hard to test... maybe one out of 20 calls fails |
| 23:41.14 | Kobaz | rx gain will affect callerid and dialing out success rates? |
| 23:41.29 | ManxPower | what country are you in Kobaz? |
| 23:41.34 | Kobaz | us |
| 23:42.08 | Kobaz | hmm, that's interesting |
| 23:42.13 | Kobaz | lowering the rx gain made dialing out go faster |
| 23:46.05 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:46.43 | ManxPower | if you can lower your dtmf tone duration, try 50 - 57 ms |
| 23:47.02 | ManxPower | ..er. 50ms to 75ms |
| 23:47.22 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 23:47.47 | ujjain | when I added a firewall, my pap2t can no longer connect to login server, but is possible with x-lite on my pc, same network. |
| 23:47.52 | ujjain | I forwarded ports 5060 and 10k-20k |
| 23:48.05 | boodu | i can use more than 1 channel for call (send) with dahdi or with misdn |
| 23:48.20 | boodu | but i can receive many call |
| 23:48.21 | boodu | :O |
| 23:48.26 | ManxPower | ujjain, UDP. correct? |
| 23:48.31 | ujjain | Yep. |
| 23:48.35 | boodu | my card is the openvox b400e |
| 23:48.46 | boodu | if anyone can help me plz |
| 23:48.47 | ujjain | FreePBX says 2 IP phones online, 6 IP trunks online, 3 ip trunk registrations |
| 23:49.02 | boodu | I don't know what is bad in my configuration |
| 23:49.06 | ujjain | I have not edited any configuration, just added firewall and port-forwarding. |
| 23:49.15 | boodu | i think about bad configuration of jumper |
| 23:49.51 | WIMPy | boodu: Have you tried manually dialling out on explicit channels? |
| 23:49.55 | boodu | yes |
| 23:50.04 | boodu | one is ok with channel 10 |
| 23:50.30 | boodu | and channel 11 on the same time doesn't worked |
| 23:50.31 | WIMPy | How do you get to channel 10 if you only have 6? |
| 23:50.55 | WIMPy | But each channel works on its own? |
| 23:50.56 | boodu | 6 ? |
| 23:51.29 | WIMPy | Didn't you sat you have 3 BRIs? |
| 23:52.23 | ujjain | I can connect via x-lite, but not via iphone and pap2t, same network |
| 23:52.32 | boodu | 4 ports => 1-> channel 1&2 , 2-> channel 4&5, ..., 4,-> channel 10&11 |
| 23:53.15 | boodu | yes 3 bri but one connected to the port 4 |
| 23:54.07 | WIMPy | Hmm. Do the D channels still get numbered in Asterisk? |
| 23:54.52 | boodu | I think |
| 23:55.02 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 23:55.14 | WIMPy | Hmm. I think they do, but the sync channels don't get counted. A little inconsitant. |
| 23:55.59 | WIMPy | I'd suspect that maybe you're simply not allowed to place a 2nd call. |
| 23:56.28 | boodu | I can |
| 23:57.02 | boodu | there are an classic PBX |
| 23:57.16 | boodu | and it's possible to do multi call |
| 23:57.46 | boodu | :O |
| 23:57.55 | WIMPy | What kind of signalling are you using? |
| 23:58.14 | boodu | I don't understand your question :s |
| 23:58.24 | boodu | sip |
| 23:58.26 | boodu | :s |
| 23:58.38 | WIMPy | on the BRIs |
| 23:58.52 | boodu | bri_cpe_ptmp |
| 23:59.16 | WIMPy | what protocol? |
| 23:59.26 | WIMPy | err switchtype |